Release 1.18.0

The HF absorption is applied given the source distance, as relative to the
source's immediate environment, with additional absorption being applied given
the room/reverb environment. This does double up the amount of absorption
compared to the dry path, but it can be assumed the initial reflections travel
a longer distance.

.gitignore

@@ -1 +1,7 @@
 build
+winbuild
+win64build
+include/SLES
+include/sndio.h
+include/sys
+openal-soft.kdev4

.travis.yml

@@ -0,0 +1,70 @@
+language: c
+matrix:
+  include:
+    - os: linux
+      dist: trusty
+    - os: linux
+      dist: trusty
+      env:
+        - BUILD_ANDROID=true
+    - os: osx
+sudo: required
+cache:
+  directories:
+    - $HOME/android-ndk-r14
+install:
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then
+      # Install pulseaudio, portaudio, ALSA, JACK dependencies for
+      # corresponding backends.
+      # Install Qt5 dependency for alsoft-config.
+      sudo apt-get install -qq \
+        libpulse-dev \
+        portaudio19-dev \
+        libasound2-dev \
+        libjack-dev \
+        qtbase5-dev
+    fi
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
+      if [[ ! -d ~/android-ndk-r14 || -z "$(ls -A ~/android-ndk-r14)" ]]; then
+        curl -o ~/android-ndk.zip https://dl.google.com/android/repository/android-ndk-r14-linux-x86_64.zip
+        unzip -q ~/android-ndk.zip -d ~ \
+          'android-ndk-r14/build/cmake/*' \
+          'android-ndk-r14/platforms/android-9/arch-arm/*' \
+          'android-ndk-r14/source.properties' \
+          'android-ndk-r14/sources/cxx-stl/gnu-libstdc++/4.9/libs/armeabi-v7a/*' \
+          'android-ndk-r14/sysroot/*' \
+          'android-ndk-r14/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/*' \
+          'android-ndk-r14/toolchains/llvm/prebuilt/linux-x86_64/*'
+        sed -i -e 's/VERSION 3.6.0/VERSION 3.2/' ~/android-ndk-r14/build/cmake/android.toolchain.cmake
+      fi
+    fi
+script:
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then
+      cmake \
+        -DALSOFT_REQUIRE_ALSA=ON \
+        -DALSOFT_REQUIRE_OSS=ON \
+        -DALSOFT_REQUIRE_PORTAUDIO=ON \
+        -DALSOFT_REQUIRE_PULSEAUDIO=ON \
+        -DALSOFT_REQUIRE_JACK=ON \
+        -DALSOFT_EMBED_HRTF_DATA=YES \
+        .
+    fi
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
+      cmake \
+        -DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r14/build/cmake/android.toolchain.cmake \
+        -DALSOFT_REQUIRE_OPENSL=ON \
+        -DALSOFT_EMBED_HRTF_DATA=YES \
+        .
+    fi
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "osx" ]]; then
+      cmake \
+        -DALSOFT_REQUIRE_COREAUDIO=ON \
+        -DALSOFT_EMBED_HRTF_DATA=YES \
+        .
+    fi
+  - make -j2

Alc/alcConfig.c

@@ -13,14 +13,18 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
 #ifdef _WIN32
+#ifdef __MINGW32__
+#define _WIN32_IE 0x501
+#else
 #define _WIN32_IE 0x400
 #endif
+#endif
 
 #include "config.h"
 
@@ -28,295 +32,619 @@
 #include <stdio.h>
 #include <ctype.h>
 #include <string.h>
-
-#include "alMain.h"
-
 #ifdef _WIN32_IE
+#include <windows.h>
 #include <shlobj.h>
 #endif
 
+#include "alMain.h"
+#include "compat.h"
+#include "bool.h"
+
+
 typedef struct ConfigEntry {
     char *key;
     char *value;
 } ConfigEntry;
 
 typedef struct ConfigBlock {
-    char *name;
     ConfigEntry *entries;
-    size_t entryCount;
+    unsigned int entryCount;
 } ConfigBlock;
+static ConfigBlock cfgBlock;
 
-static ConfigBlock *cfgBlocks;
-static size_t cfgCount;
 
-static char buffer[1024];
+static char *lstrip(char *line)
+{
+    while(isspace(line[0]))
+        line++;
+    return line;
+}
+
+static char *rstrip(char *line)
+{
+    size_t len = strlen(line);
+    while(len > 0 && isspace(line[len-1]))
+        len--;
+    line[len] = 0;
+    return line;
+}
+
+static int readline(FILE *f, char **output, size_t *maxlen)
+{
+    size_t len = 0;
+    int c;
+
+    while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n'))
+        ;
+    if(c == EOF)
+        return 0;
+
+    do {
+        if(len+1 >= *maxlen)
+        {
+            void *temp = NULL;
+            size_t newmax;
+
+            newmax = (*maxlen ? (*maxlen)<<1 : 32);
+            if(newmax > *maxlen)
+                temp = realloc(*output, newmax);
+            if(!temp)
+            {
+                ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen);
+                return 0;
+            }
+
+            *output = temp;
+            *maxlen = newmax;
+        }
+        (*output)[len++] = c;
+        (*output)[len] = '\0';
+    } while((c=fgetc(f)) != EOF && c != '\r' && c != '\n');
+
+    return 1;
+}
+
+
+static char *expdup(const char *str)
+{
+    char *output = NULL;
+    size_t maxlen = 0;
+    size_t len = 0;
+
+    while(*str != '\0')
+    {
+        const char *addstr;
+        size_t addstrlen;
+        size_t i;
+
+        if(str[0] != '$')
+        {
+            const char *next = strchr(str, '$');
+            addstr = str;
+            addstrlen = next ? (size_t)(next-str) : strlen(str);
+
+            str += addstrlen;
+        }
+        else
+        {
+            str++;
+            if(*str == '$')
+            {
+                const char *next = strchr(str+1, '$');
+                addstr = str;
+                addstrlen = next ? (size_t)(next-str) : strlen(str);
+
+                str += addstrlen;
+            }
+            else
+            {
+                bool hasbraces;
+                char envname[1024];
+                size_t k = 0;
+
+                hasbraces = (*str == '{');
+                if(hasbraces) str++;
+
+                while((isalnum(*str) || *str == '_') && k < sizeof(envname)-1)
+                    envname[k++] = *(str++);
+                envname[k++] = '\0';
+
+                if(hasbraces && *str != '}')
+                    continue;
+
+                if(hasbraces) str++;
+                if((addstr=getenv(envname)) == NULL)
+                    continue;
+                addstrlen = strlen(addstr);
+            }
+        }
+        if(addstrlen == 0)
+            continue;
+
+        if(addstrlen >= maxlen-len)
+        {
+            void *temp = NULL;
+            size_t newmax;
+
+            newmax = len+addstrlen+1;
+            if(newmax > maxlen)
+                temp = realloc(output, newmax);
+            if(!temp)
+            {
+                ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, maxlen);
+                return output;
+            }
+
+            output = temp;
+            maxlen = newmax;
+        }
+
+        for(i = 0;i < addstrlen;i++)
+            output[len++] = addstr[i];
+        output[len] = '\0';
+    }
+
+    return output ? output : calloc(1, 1);
+}
+
 
 static void LoadConfigFromFile(FILE *f)
 {
-    ConfigBlock *curBlock = cfgBlocks;
+    char curSection[128] = "";
+    char *buffer = NULL;
+    size_t maxlen = 0;
     ConfigEntry *ent;
 
-    while(fgets(buffer, sizeof(buffer), f))
+    while(readline(f, &buffer, &maxlen))
     {
-        size_t i = 0;
+        char *line, *comment;
+        char key[256] = "";
+        char value[256] = "";
 
-        while(isspace(buffer[i]))
-            i++;
-        if(!buffer[i] || buffer[i] == '#')
-            continue;
+        line = rstrip(lstrip(buffer));
+        if(!line[0]) continue;
 
-        memmove(buffer, buffer+i, strlen(buffer+i)+1);
-
-        if(buffer[0] == '[')
+        if(line[0] == '[')
         {
-            ConfigBlock *nextBlock;
+            char *section = line+1;
+            char *endsection;
 
-            i = 1;
-            while(buffer[i] && buffer[i] != ']')
-                i++;
-
-            if(!buffer[i])
+            endsection = strchr(section, ']');
+            if(!endsection || section == endsection)
             {
-                 AL_PRINT("config parse error: bad line \"%s\"\n", buffer);
-                 continue;
-            }
-            buffer[i] = 0;
-
-            do {
-                i++;
-                if(buffer[i] && !isspace(buffer[i]))
-                {
-                    if(buffer[i] != '#')
-                        AL_PRINT("config warning: extra data after block: \"%s\"\n", buffer+i);
-                    break;
-                }
-            } while(buffer[i]);
-
-            nextBlock = NULL;
-            for(i = 0;i < cfgCount;i++)
-            {
-                if(strcasecmp(cfgBlocks[i].name, buffer+1) == 0)
-                {
-                    nextBlock = cfgBlocks+i;
-//                    AL_PRINT("found block '%s'\n", nextBlock->name);
-                    break;
-                }
-            }
-
-            if(!nextBlock)
-            {
-                nextBlock = realloc(cfgBlocks, (cfgCount+1)*sizeof(ConfigBlock));
-                if(!nextBlock)
-                {
-                     AL_PRINT("config parse error: error reallocating config blocks\n");
-                     continue;
-                }
-                cfgBlocks = nextBlock;
-                nextBlock = cfgBlocks+cfgCount;
-                cfgCount++;
-
-                nextBlock->name = strdup(buffer+1);
-                nextBlock->entries = NULL;
-                nextBlock->entryCount = 0;
-
-//                AL_PRINT("found new block '%s'\n", nextBlock->name);
-            }
-            curBlock = nextBlock;
-            continue;
-        }
-
-        /* Look for the option name */
-        i = 0;
-        while(buffer[i] && buffer[i] != '#' && buffer[i] != '=' &&
-              !isspace(buffer[i]))
-            i++;
-
-        if(!buffer[i] || buffer[i] == '#' || i == 0)
-        {
-            AL_PRINT("config parse error: malformed option line: \"%s\"\n", buffer);
-            continue;
-        }
-
-        /* Seperate the option */
-        if(buffer[i] != '=')
-        {
-            buffer[i++] = 0;
-
-            while(isspace(buffer[i]))
-                i++;
-            if(buffer[i] != '=')
-            {
-                AL_PRINT("config parse error: option without a value: \"%s\"\n", buffer);
+                ERR("config parse error: bad line \"%s\"\n", line);
                 continue;
             }
+            if(endsection[1] != 0)
+            {
+                char *end = endsection+1;
+                while(isspace(*end))
+                    ++end;
+                if(*end != 0 && *end != '#')
+                {
+                    ERR("config parse error: bad line \"%s\"\n", line);
+                    continue;
+                }
+            }
+            *endsection = 0;
+
+            if(strcasecmp(section, "general") == 0)
+                curSection[0] = 0;
+            else
+            {
+                size_t len, p = 0;
+                do {
+                    char *nextp = strchr(section, '%');
+                    if(!nextp)
+                    {
+                        strncpy(curSection+p, section, sizeof(curSection)-1-p);
+                        break;
+                    }
+
+                    len = nextp - section;
+                    if(len > sizeof(curSection)-1-p)
+                        len = sizeof(curSection)-1-p;
+                    strncpy(curSection+p, section, len);
+                    p += len;
+                    section = nextp;
+
+                    if(((section[1] >= '0' && section[1] <= '9') ||
+                        (section[1] >= 'a' && section[1] <= 'f') ||
+                        (section[1] >= 'A' && section[1] <= 'F')) &&
+                       ((section[2] >= '0' && section[2] <= '9') ||
+                        (section[2] >= 'a' && section[2] <= 'f') ||
+                        (section[2] >= 'A' && section[2] <= 'F')))
+                    {
+                        unsigned char b = 0;
+                        if(section[1] >= '0' && section[1] <= '9')
+                            b = (section[1]-'0') << 4;
+                        else if(section[1] >= 'a' && section[1] <= 'f')
+                            b = (section[1]-'a'+0xa) << 4;
+                        else if(section[1] >= 'A' && section[1] <= 'F')
+                            b = (section[1]-'A'+0x0a) << 4;
+                        if(section[2] >= '0' && section[2] <= '9')
+                            b |= (section[2]-'0');
+                        else if(section[2] >= 'a' && section[2] <= 'f')
+                            b |= (section[2]-'a'+0xa);
+                        else if(section[2] >= 'A' && section[2] <= 'F')
+                            b |= (section[2]-'A'+0x0a);
+                        if(p < sizeof(curSection)-1)
+                            curSection[p++] = b;
+                        section += 3;
+                    }
+                    else if(section[1] == '%')
+                    {
+                        if(p < sizeof(curSection)-1)
+                            curSection[p++] = '%';
+                        section += 2;
+                    }
+                    else
+                    {
+                        if(p < sizeof(curSection)-1)
+                            curSection[p++] = '%';
+                        section += 1;
+                    }
+                    if(p < sizeof(curSection)-1)
+                        curSection[p] = 0;
+                } while(p < sizeof(curSection)-1 && *section != 0);
+                curSection[sizeof(curSection)-1] = 0;
+            }
+
+            continue;
+        }
+
+        comment = strchr(line, '#');
+        if(comment) *(comment++) = 0;
+        if(!line[0]) continue;
+
+        if(sscanf(line, "%255[^=] = \"%255[^\"]\"", key, value) == 2 ||
+           sscanf(line, "%255[^=] = '%255[^\']'", key, value) == 2 ||
+           sscanf(line, "%255[^=] = %255[^\n]", key, value) == 2)
+        {
+            /* sscanf doesn't handle '' or "" as empty values, so clip it
+             * manually. */
+            if(strcmp(value, "\"\"") == 0 || strcmp(value, "''") == 0)
+                value[0] = 0;
+        }
+        else if(sscanf(line, "%255[^=] %255[=]", key, value) == 2)
+        {
+            /* Special case for 'key =' */
+            value[0] = 0;
+        }
+        else
+        {
+            ERR("config parse error: malformed option line: \"%s\"\n\n", line);
+            continue;
+        }
+        rstrip(key);
+
+        if(curSection[0] != 0)
+        {
+            size_t len = strlen(curSection);
+            memmove(&key[len+1], key, sizeof(key)-1-len);
+            key[len] = '/';
+            memcpy(key, curSection, len);
         }
-        /* Find the start of the value */
-        buffer[i++] = 0;
-        while(isspace(buffer[i]))
-            i++;
 
         /* Check if we already have this option set */
-        ent = curBlock->entries;
-        while((size_t)(ent-curBlock->entries) < curBlock->entryCount)
+        ent = cfgBlock.entries;
+        while((unsigned int)(ent-cfgBlock.entries) < cfgBlock.entryCount)
         {
-            if(strcasecmp(ent->key, buffer) == 0)
+            if(strcasecmp(ent->key, key) == 0)
                 break;
             ent++;
         }
 
-        if((size_t)(ent-curBlock->entries) >= curBlock->entryCount)
+        if((unsigned int)(ent-cfgBlock.entries) >= cfgBlock.entryCount)
         {
             /* Allocate a new option entry */
-            ent = realloc(curBlock->entries, (curBlock->entryCount+1)*sizeof(ConfigEntry));
+            ent = realloc(cfgBlock.entries, (cfgBlock.entryCount+1)*sizeof(ConfigEntry));
             if(!ent)
             {
-                 AL_PRINT("config parse error: error reallocating config entries\n");
+                 ERR("config parse error: error reallocating config entries\n");
                  continue;
             }
-            curBlock->entries = ent;
-            ent = curBlock->entries + curBlock->entryCount;
-            curBlock->entryCount++;
+            cfgBlock.entries = ent;
+            ent = cfgBlock.entries + cfgBlock.entryCount;
+            cfgBlock.entryCount++;
 
-            ent->key = strdup(buffer);
+            ent->key = strdup(key);
             ent->value = NULL;
         }
 
-        /* Look for the end of the line (Null term, new-line, or #-symbol) and
-           eat up the trailing whitespace */
-        memmove(buffer, buffer+i, strlen(buffer+i)+1);
-
-        i = 0;
-        while(buffer[i] && buffer[i] != '#' && buffer[i] != '\n')
-            i++;
-        do {
-            i--;
-        } while(isspace(buffer[i]));
-        buffer[++i] = 0;
-
         free(ent->value);
-        ent->value = strdup(buffer);
+        ent->value = expdup(value);
 
-//        AL_PRINT("found '%s' = '%s'\n", ent->key, ent->value);
+        TRACE("found '%s' = '%s'\n", ent->key, ent->value);
     }
+
+    free(buffer);
 }
 
+#ifdef _WIN32
 void ReadALConfig(void)
 {
+    WCHAR buffer[PATH_MAX];
+    const WCHAR *str;
+    al_string ppath;
     FILE *f;
 
-    cfgBlocks = calloc(1, sizeof(ConfigBlock));
-    cfgBlocks->name = strdup("general");
-    cfgCount = 1;
-
-#ifdef _WIN32
-    if(SHGetSpecialFolderPathA(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE)
+    if(SHGetSpecialFolderPathW(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE)
     {
-        int p = strlen(buffer);
-        snprintf(buffer+p, sizeof(buffer)-p, "\\alsoft.ini");
-        f = fopen(buffer, "rt");
+        al_string filepath = AL_STRING_INIT_STATIC();
+        alstr_copy_wcstr(&filepath, buffer);
+        alstr_append_cstr(&filepath, "\\alsoft.ini");
+
+        TRACE("Loading config %s...\n", alstr_get_cstr(filepath));
+        f = al_fopen(alstr_get_cstr(filepath), "rt");
+        if(f)
+        {
+            LoadConfigFromFile(f);
+            fclose(f);
+        }
+        alstr_reset(&filepath);
+    }
+
+    ppath = GetProcPath();
+    if(!alstr_empty(ppath))
+    {
+        alstr_append_cstr(&ppath, "\\alsoft.ini");
+        TRACE("Loading config %s...\n", alstr_get_cstr(ppath));
+        f = al_fopen(alstr_get_cstr(ppath), "r");
         if(f)
         {
             LoadConfigFromFile(f);
             fclose(f);
         }
     }
+
+    if((str=_wgetenv(L"ALSOFT_CONF")) != NULL && *str)
+    {
+        al_string filepath = AL_STRING_INIT_STATIC();
+        alstr_copy_wcstr(&filepath, str);
+
+        TRACE("Loading config %s...\n", alstr_get_cstr(filepath));
+        f = al_fopen(alstr_get_cstr(filepath), "rt");
+        if(f)
+        {
+            LoadConfigFromFile(f);
+            fclose(f);
+        }
+        alstr_reset(&filepath);
+    }
+
+    alstr_reset(&ppath);
+}
 #else
-    f = fopen("/etc/openal/alsoft.conf", "r");
+void ReadALConfig(void)
+{
+    char buffer[PATH_MAX];
+    const char *str;
+    al_string ppath;
+    FILE *f;
+
+    str = "/etc/openal/alsoft.conf";
+
+    TRACE("Loading config %s...\n", str);
+    f = al_fopen(str, "r");
     if(f)
     {
         LoadConfigFromFile(f);
         fclose(f);
     }
-    if(getenv("HOME") && *(getenv("HOME")))
+
+    if(!(str=getenv("XDG_CONFIG_DIRS")) || str[0] == 0)
+        str = "/etc/xdg";
+    strncpy(buffer, str, sizeof(buffer)-1);
+    buffer[sizeof(buffer)-1] = 0;
+    /* Go through the list in reverse, since "the order of base directories
+     * denotes their importance; the first directory listed is the most
+     * important". Ergo, we need to load the settings from the later dirs
+     * first so that the settings in the earlier dirs override them.
+     */
+    while(1)
     {
-        snprintf(buffer, sizeof(buffer), "%s/.alsoftrc", getenv("HOME"));
-        f = fopen(buffer, "r");
+        char *next = strrchr(buffer, ':');
+        if(next) *(next++) = 0;
+        else next = buffer;
+
+        if(next[0] != '/')
+            WARN("Ignoring XDG config dir: %s\n", next);
+        else
+        {
+            size_t len = strlen(next);
+            strncpy(next+len, "/alsoft.conf", buffer+sizeof(buffer)-next-len);
+            buffer[sizeof(buffer)-1] = 0;
+
+            TRACE("Loading config %s...\n", next);
+            f = al_fopen(next, "r");
+            if(f)
+            {
+                LoadConfigFromFile(f);
+                fclose(f);
+            }
+        }
+        if(next == buffer)
+            break;
+    }
+
+    if((str=getenv("HOME")) != NULL && *str)
+    {
+        snprintf(buffer, sizeof(buffer), "%s/.alsoftrc", str);
+
+        TRACE("Loading config %s...\n", buffer);
+        f = al_fopen(buffer, "r");
         if(f)
         {
             LoadConfigFromFile(f);
             fclose(f);
         }
     }
-#endif
-    if(getenv("ALSOFT_CONF"))
+
+    if((str=getenv("XDG_CONFIG_HOME")) != NULL && str[0] != 0)
+        snprintf(buffer, sizeof(buffer), "%s/%s", str, "alsoft.conf");
+    else
     {
-        f = fopen(getenv("ALSOFT_CONF"), "r");
+        buffer[0] = 0;
+        if((str=getenv("HOME")) != NULL && str[0] != 0)
+            snprintf(buffer, sizeof(buffer), "%s/.config/%s", str, "alsoft.conf");
+    }
+    if(buffer[0] != 0)
+    {
+        TRACE("Loading config %s...\n", buffer);
+        f = al_fopen(buffer, "r");
         if(f)
         {
             LoadConfigFromFile(f);
             fclose(f);
         }
     }
+
+    ppath = GetProcPath();
+    if(!alstr_empty(ppath))
+    {
+        alstr_append_cstr(&ppath, "/alsoft.conf");
+        TRACE("Loading config %s...\n", alstr_get_cstr(ppath));
+        f = al_fopen(alstr_get_cstr(ppath), "r");
+        if(f)
+        {
+            LoadConfigFromFile(f);
+            fclose(f);
+        }
+    }
+
+    if((str=getenv("ALSOFT_CONF")) != NULL && *str)
+    {
+        TRACE("Loading config %s...\n", str);
+        f = al_fopen(str, "r");
+        if(f)
+        {
+            LoadConfigFromFile(f);
+            fclose(f);
+        }
+    }
+
+    alstr_reset(&ppath);
 }
+#endif
 
 void FreeALConfig(void)
 {
-    size_t i;
+    unsigned int i;
 
-    for(i = 0;i < cfgCount;i++)
+    for(i = 0;i < cfgBlock.entryCount;i++)
     {
-        size_t j;
-        for(j = 0;j < cfgBlocks[i].entryCount;j++)
-        {
-           free(cfgBlocks[i].entries[j].key);
-           free(cfgBlocks[i].entries[j].value);
-        }
-        free(cfgBlocks[i].entries);
-        free(cfgBlocks[i].name);
+        free(cfgBlock.entries[i].key);
+        free(cfgBlock.entries[i].value);
     }
-    free(cfgBlocks);
-    cfgBlocks = NULL;
-    cfgCount = 0;
+    free(cfgBlock.entries);
 }
 
-const char *GetConfigValue(const char *blockName, const char *keyName, const char *def)
+const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def)
 {
-    size_t i, j;
+    unsigned int i;
+    char key[256];
 
-    if(keyName)
+    if(!keyName)
+        return def;
+
+    if(blockName && strcasecmp(blockName, "general") != 0)
     {
-        if(!blockName)
-            blockName = "general";
-
-        for(i = 0;i < cfgCount;i++)
+        if(devName)
+            snprintf(key, sizeof(key), "%s/%s/%s", blockName, devName, keyName);
+        else
+            snprintf(key, sizeof(key), "%s/%s", blockName, keyName);
+    }
+    else
+    {
+        if(devName)
+            snprintf(key, sizeof(key), "%s/%s", devName, keyName);
+        else
         {
-            if(strcasecmp(cfgBlocks[i].name, blockName) != 0)
-                continue;
-
-            for(j = 0;j < cfgBlocks[i].entryCount;j++)
-            {
-                if(strcasecmp(cfgBlocks[i].entries[j].key, keyName) == 0)
-                    return cfgBlocks[i].entries[j].value;
-            }
+            strncpy(key, keyName, sizeof(key)-1);
+            key[sizeof(key)-1] = 0;
         }
     }
 
-    return def;
+    for(i = 0;i < cfgBlock.entryCount;i++)
+    {
+        if(strcmp(cfgBlock.entries[i].key, key) == 0)
+        {
+            TRACE("Found %s = \"%s\"\n", key, cfgBlock.entries[i].value);
+            if(cfgBlock.entries[i].value[0])
+                return cfgBlock.entries[i].value;
+            return def;
+        }
+    }
+
+    if(!devName)
+    {
+        TRACE("Key %s not found\n", key);
+        return def;
+    }
+    return GetConfigValue(NULL, blockName, keyName, def);
 }
 
-int GetConfigValueInt(const char *blockName, const char *keyName, int def)
+int ConfigValueExists(const char *devName, const char *blockName, const char *keyName)
 {
-    const char *val = GetConfigValue(blockName, keyName, "");
-
-    if(!val[0]) return def;
-    return strtol(val, NULL, 0);
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
+    return !!val[0];
 }
 
-float GetConfigValueFloat(const char *blockName, const char *keyName, float def)
+int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret)
 {
-    const char *val = GetConfigValue(blockName, keyName, "");
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
+    if(!val[0]) return 0;
+
+    *ret = val;
+    return 1;
+}
+
+int ConfigValueInt(const char *devName, const char *blockName, const char *keyName, int *ret)
+{
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
+    if(!val[0]) return 0;
+
+    *ret = strtol(val, NULL, 0);
+    return 1;
+}
+
+int ConfigValueUInt(const char *devName, const char *blockName, const char *keyName, unsigned int *ret)
+{
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
+    if(!val[0]) return 0;
+
+    *ret = strtoul(val, NULL, 0);
+    return 1;
+}
+
+int ConfigValueFloat(const char *devName, const char *blockName, const char *keyName, float *ret)
+{
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
+    if(!val[0]) return 0;
 
-    if(!val[0]) return def;
 #ifdef HAVE_STRTOF
-    return strtof(val, NULL);
+    *ret = strtof(val, NULL);
 #else
-    return (float)strtod(val, NULL);
+    *ret = (float)strtod(val, NULL);
 #endif
+    return 1;
 }
 
-int GetConfigValueBool(const char *blockName, const char *keyName, float def)
+int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret)
 {
-    const char *val = GetConfigValue(blockName, keyName, "");
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
+    if(!val[0]) return 0;
+
+    *ret = (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||
+            strcasecmp(val, "on") == 0 || atoi(val) != 0);
+    return 1;
+}
+
+int GetConfigValueBool(const char *devName, const char *blockName, const char *keyName, int def)
+{
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
 
     if(!val[0]) return !!def;
     return (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||

Alc/alcEcho.c

@@ -1,212 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2009 by Chris Robinson.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-// Just a soft maximum. Being higher will cause EchoUpdate to reallocate the
-// sample buffer which may cause an abort if realloc fails
-#define MAX_ECHO_FREQ  192000
-
-typedef struct ALechoState {
-    // Must be first in all effects!
-    ALeffectState state;
-
-    ALfloat *SampleBuffer;
-    ALuint BufferLength;
-
-    // The echo is two tap. The delay is the number of samples from before the
-    // current offset
-    struct {
-        ALuint delay;
-    } Tap[2];
-    ALuint Offset;
-    // The LR gains for the first tap. The second tap uses the reverse
-    ALfloat GainL;
-    ALfloat GainR;
-
-    ALfloat FeedGain;
-
-    FILTER iirFilter;
-    ALfloat history[2];
-} ALechoState;
-
-// Find the next power of 2.  Actually, this will return the input value if
-// it is already a power of 2.
-static ALuint NextPowerOf2(ALuint value)
-{
-    ALuint powerOf2 = 1;
-
-    if(value)
-    {
-        value--;
-        while(value)
-        {
-            value >>= 1;
-            powerOf2 <<= 1;
-        }
-    }
-    return powerOf2;
-}
-
-ALvoid EchoDestroy(ALeffectState *effect)
-{
-    ALechoState *state = (ALechoState*)effect;
-    if(state)
-    {
-        free(state->SampleBuffer);
-        state->SampleBuffer = NULL;
-        free(state);
-    }
-}
-
-ALboolean EchoDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
-{
-    ALechoState *state = (ALechoState*)effect;
-    ALuint maxlen, i;
-
-    // Use the next power of 2 for the buffer length, so the tap offsets can be
-    // wrapped using a mask instead of a modulo
-    maxlen  = (ALuint)(AL_ECHO_MAX_DELAY * Device->Frequency);
-    maxlen += (ALuint)(AL_ECHO_MAX_LRDELAY * Device->Frequency);
-    maxlen  = NextPowerOf2(maxlen+1);
-
-    if(maxlen != state->BufferLength)
-    {
-        void *temp;
-
-        temp = realloc(state->SampleBuffer, maxlen * sizeof(ALfloat));
-        if(!temp)
-        {
-            alSetError(AL_OUT_OF_MEMORY);
-            return AL_FALSE;
-        }
-        state->BufferLength = maxlen;
-    }
-    for(i = 0;i < state->BufferLength;i++)
-        state->SampleBuffer[i] = 0.0f;
-
-    return AL_TRUE;
-}
-
-ALvoid EchoUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffect *Effect)
-{
-    ALechoState *state = (ALechoState*)effect;
-    ALuint frequency = Context->Device->Frequency;
-    ALfloat lrpan, cw, a, g;
-
-    state->Tap[0].delay = (ALuint)(Effect->Echo.Delay * frequency);
-    state->Tap[1].delay = (ALuint)(Effect->Echo.LRDelay * frequency);
-    state->Tap[1].delay += state->Tap[0].delay;
-
-    lrpan = Effect->Echo.Spread*0.5f + 0.5f;
-    state->GainL = aluSqrt(     lrpan);
-    state->GainR = aluSqrt(1.0f-lrpan);
-
-    state->FeedGain = Effect->Echo.Feedback;
-
-    cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / frequency);
-    g = 1.0f - Effect->Echo.Damping;
-    a = 0.0f;
-    if(g < 0.9999f) // 1-epsilon
-        a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
-    state->iirFilter.coeff = a;
-}
-
-ALvoid EchoProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[OUTPUTCHANNELS])
-{
-    ALechoState *state = (ALechoState*)effect;
-    const ALuint mask = state->BufferLength-1;
-    const ALuint tap1 = state->Tap[0].delay;
-    const ALuint tap2 = state->Tap[1].delay;
-    ALuint offset = state->Offset;
-    const ALfloat gain = Slot->Gain;
-    ALfloat samp[2], smp;
-    ALuint i;
-
-    for(i = 0;i < SamplesToDo;i++,offset++)
-    {
-        // Sample first tap
-        smp = state->SampleBuffer[(offset-tap1) & mask];
-        samp[0] = smp * state->GainL;
-        samp[1] = smp * state->GainR;
-        // Sample second tap. Reverse LR panning
-        smp = state->SampleBuffer[(offset-tap2) & mask];
-        samp[0] += smp * state->GainR;
-        samp[1] += smp * state->GainL;
-
-        // Apply damping and feedback gain to the second tap, and mix in the
-        // new sample
-        smp = lpFilter2P(&state->iirFilter, 0, smp+SamplesIn[i]);
-        state->SampleBuffer[offset&mask] = smp * state->FeedGain;
-
-        // Apply slot gain
-        samp[0] *= gain;
-        samp[1] *= gain;
-
-        SamplesOut[i][FRONT_LEFT]  += samp[0];
-        SamplesOut[i][FRONT_RIGHT] += samp[1];
-        SamplesOut[i][SIDE_LEFT]   += samp[0];
-        SamplesOut[i][SIDE_RIGHT]  += samp[1];
-        SamplesOut[i][BACK_LEFT]   += samp[0];
-        SamplesOut[i][BACK_RIGHT]  += samp[1];
-    }
-    state->Offset = offset;
-}
-
-ALeffectState *EchoCreate(void)
-{
-    ALechoState *state;
-
-    state = malloc(sizeof(*state));
-    if(!state)
-    {
-        alSetError(AL_OUT_OF_MEMORY);
-        return NULL;
-    }
-
-    state->state.Destroy = EchoDestroy;
-    state->state.DeviceUpdate = EchoDeviceUpdate;
-    state->state.Update = EchoUpdate;
-    state->state.Process = EchoProcess;
-
-    state->BufferLength = 0;
-    state->SampleBuffer = NULL;
-
-    state->Tap[0].delay = 0;
-    state->Tap[1].delay = 0;
-    state->Offset = 0;
-    state->GainL = 0.0f;
-    state->GainR = 0.0f;
-
-    state->iirFilter.coeff = 0.0f;
-    state->iirFilter.history[0] = 0.0f;
-    state->iirFilter.history[1] = 0.0f;
-
-    return &state->state;
-}

Alc/alcReverb.c

@@ -1,843 +0,0 @@
-/**
- * Reverb for the OpenAL cross platform audio library
- * Copyright (C) 2008-2009 by Christopher Fitzgerald.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alAuxEffectSlot.h"
-#include "alEffect.h"
-#include "alError.h"
-#include "alu.h"
-
-typedef struct DelayLine
-{
-    // The delay lines use sample lengths that are powers of 2 to allow
-    // bitmasking instead of modulus wrapping.
-    ALuint   Mask;
-    ALfloat *Line;
-} DelayLine;
-
-typedef struct ALverbState {
-    // Must be first in all effects!
-    ALeffectState state;
-
-    // All delay lines are allocated as a single buffer to reduce memory
-    // fragmentation and management code.
-    ALfloat  *SampleBuffer;
-    ALuint    TotalLength;
-    // Master effect low-pass filter (2 chained 1-pole filters).
-    FILTER    LpFilter;
-    ALfloat   LpHistory[2];
-    // Initial effect delay and decorrelation.
-    DelayLine Delay;
-    // The tap points for the initial delay.  First tap goes to early
-    // reflections, the last four decorrelate to late reverb.
-    ALuint    Tap[5];
-    struct {
-        // Total gain for early reflections.
-        ALfloat   Gain;
-        // Early reflections are done with 4 delay lines.
-        ALfloat   Coeff[4];
-        DelayLine Delay[4];
-        ALuint    Offset[4];
-        // The gain for each output channel based on 3D panning.
-        ALfloat   PanGain[OUTPUTCHANNELS];
-    } Early;
-    struct {
-        // Total gain for late reverb.
-        ALfloat   Gain;
-        // Attenuation to compensate for modal density and decay rate.
-        ALfloat   DensityGain;
-        // The feed-back and feed-forward all-pass coefficient.
-        ALfloat   ApFeedCoeff;
-        // Mixing matrix coefficient.
-        ALfloat   MixCoeff;
-        // Late reverb has 4 parallel all-pass filters.
-        ALfloat   ApCoeff[4];
-        DelayLine ApDelay[4];
-        ALuint    ApOffset[4];
-        // In addition to 4 cyclical delay lines.
-        ALfloat   Coeff[4];
-        DelayLine Delay[4];
-        ALuint    Offset[4];
-        // The cyclical delay lines are 1-pole low-pass filtered.
-        ALfloat   LpCoeff[4];
-        ALfloat   LpSample[4];
-        // The gain for each output channel based on 3D panning.
-        ALfloat   PanGain[OUTPUTCHANNELS];
-    } Late;
-    // The current read offset for all delay lines.
-    ALuint Offset;
-} ALverbState;
-
-// All delay line lengths are specified in seconds.
-
-// The lengths of the early delay lines.
-static const ALfloat EARLY_LINE_LENGTH[4] =
-{
-    0.0015f, 0.0045f, 0.0135f, 0.0405f
-};
-
-// The lengths of the late all-pass delay lines.
-static const ALfloat ALLPASS_LINE_LENGTH[4] =
-{
-    0.0151f, 0.0167f, 0.0183f, 0.0200f,
-};
-
-// The lengths of the late cyclical delay lines.
-static const ALfloat LATE_LINE_LENGTH[4] =
-{
-    0.0211f, 0.0311f, 0.0461f, 0.0680f
-};
-
-// The late cyclical delay lines have a variable length dependent on the
-// effect's density parameter (inverted for some reason) and this multiplier.
-static const ALfloat LATE_LINE_MULTIPLIER = 4.0f;
-
-// Input into the late reverb is decorrelated between four channels.  Their
-// timings are dependent on a fraction and multiplier.  See VerbUpdate() for
-// the calculations involved.
-static const ALfloat DECO_FRACTION = 1.0f / 32.0f;
-static const ALfloat DECO_MULTIPLIER = 2.0f;
-
-// The maximum length of initial delay for the master delay line (a sum of
-// the maximum early reflection and late reverb delays).
-static const ALfloat MASTER_LINE_LENGTH = 0.3f + 0.1f;
-
-// Find the next power of 2.  Actually, this will return the input value if
-// it is already a power of 2.
-static ALuint NextPowerOf2(ALuint value)
-{
-    ALuint powerOf2 = 1;
-
-    if(value)
-    {
-        value--;
-        while(value)
-        {
-            value >>= 1;
-            powerOf2 <<= 1;
-        }
-    }
-    return powerOf2;
-}
-
-static ALuint CalcLengths(ALuint length[13], ALuint frequency)
-{
-    ALuint samples, totalLength, index;
-
-    // All line lengths are powers of 2, calculated from their lengths, with
-    // an additional sample in case of rounding errors.
-
-    // See VerbUpdate() for an explanation of the additional calculation
-    // added to the master line length.
-    samples = (ALuint)
-              ((MASTER_LINE_LENGTH +
-                (LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER) *
-                 (DECO_FRACTION * ((DECO_MULTIPLIER * DECO_MULTIPLIER *
-                                    DECO_MULTIPLIER) - 1.0f)))) *
-               frequency) + 1;
-    length[0] = NextPowerOf2(samples);
-    totalLength = length[0];
-    for(index = 0;index < 4;index++)
-    {
-        samples = (ALuint)(EARLY_LINE_LENGTH[index] * frequency) + 1;
-        length[1 + index] = NextPowerOf2(samples);
-        totalLength += length[1 + index];
-    }
-    for(index = 0;index < 4;index++)
-    {
-        samples = (ALuint)(ALLPASS_LINE_LENGTH[index] * frequency) + 1;
-        length[5 + index] = NextPowerOf2(samples);
-        totalLength += length[5 + index];
-    }
-    for(index = 0;index < 4;index++)
-    {
-        samples = (ALuint)(LATE_LINE_LENGTH[index] *
-                           (1.0f + LATE_LINE_MULTIPLIER) * frequency) + 1;
-        length[9 + index] = NextPowerOf2(samples);
-        totalLength += length[9 + index];
-    }
-
-    return totalLength;
-}
-
-// Basic delay line input/output routines.
-static __inline ALfloat DelayLineOut(DelayLine *Delay, ALuint offset)
-{
-    return Delay->Line[offset&Delay->Mask];
-}
-
-static __inline ALvoid DelayLineIn(DelayLine *Delay, ALuint offset, ALfloat in)
-{
-    Delay->Line[offset&Delay->Mask] = in;
-}
-
-// Delay line output routine for early reflections.
-static __inline ALfloat EarlyDelayLineOut(ALverbState *State, ALuint index)
-{
-    return State->Early.Coeff[index] *
-           DelayLineOut(&State->Early.Delay[index],
-                        State->Offset - State->Early.Offset[index]);
-}
-
-// Given an input sample, this function produces stereo output for early
-// reflections.
-static __inline ALvoid EarlyReflection(ALverbState *State, ALfloat in, ALfloat *out)
-{
-    ALfloat d[4], v, f[4];
-
-    // Obtain the decayed results of each early delay line.
-    d[0] = EarlyDelayLineOut(State, 0);
-    d[1] = EarlyDelayLineOut(State, 1);
-    d[2] = EarlyDelayLineOut(State, 2);
-    d[3] = EarlyDelayLineOut(State, 3);
-
-    /* The following uses a lossless scattering junction from waveguide
-     * theory.  It actually amounts to a householder mixing matrix, which
-     * will produce a maximally diffuse response, and means this can probably
-     * be considered a simple feedback delay network (FDN).
-     *          N
-     *         ---
-     *         \
-     * v = 2/N /   d_i
-     *         ---
-     *         i=1
-     */
-    v = (d[0] + d[1] + d[2] + d[3]) * 0.5f;
-    // The junction is loaded with the input here.
-    v += in;
-
-    // Calculate the feed values for the delay lines.
-    f[0] = v - d[0];
-    f[1] = v - d[1];
-    f[2] = v - d[2];
-    f[3] = v - d[3];
-
-    // Refeed the delay lines.
-    DelayLineIn(&State->Early.Delay[0], State->Offset, f[0]);
-    DelayLineIn(&State->Early.Delay[1], State->Offset, f[1]);
-    DelayLineIn(&State->Early.Delay[2], State->Offset, f[2]);
-    DelayLineIn(&State->Early.Delay[3], State->Offset, f[3]);
-
-    // Output the results of the junction for all four lines.
-    out[0] = State->Early.Gain * f[0];
-    out[1] = State->Early.Gain * f[1];
-    out[2] = State->Early.Gain * f[2];
-    out[3] = State->Early.Gain * f[3];
-}
-
-// All-pass input/output routine for late reverb.
-static __inline ALfloat LateAllPassInOut(ALverbState *State, ALuint index, ALfloat in)
-{
-    ALfloat out;
-
-    out = State->Late.ApCoeff[index] *
-          DelayLineOut(&State->Late.ApDelay[index],
-                       State->Offset - State->Late.ApOffset[index]);
-    out -= (State->Late.ApFeedCoeff * in);
-    DelayLineIn(&State->Late.ApDelay[index], State->Offset,
-                (State->Late.ApFeedCoeff * out) + in);
-    return out;
-}
-
-// Delay line output routine for late reverb.
-static __inline ALfloat LateDelayLineOut(ALverbState *State, ALuint index)
-{
-    return State->Late.Coeff[index] *
-           DelayLineOut(&State->Late.Delay[index],
-                        State->Offset - State->Late.Offset[index]);
-}
-
-// Low-pass filter input/output routine for late reverb.
-static __inline ALfloat LateLowPassInOut(ALverbState *State, ALuint index, ALfloat in)
-{
-    State->Late.LpSample[index] = in +
-        ((State->Late.LpSample[index] - in) * State->Late.LpCoeff[index]);
-    return State->Late.LpSample[index];
-}
-
-// Given four decorrelated input samples, this function produces stereo
-// output for late reverb.
-static __inline ALvoid LateReverb(ALverbState *State, ALfloat *in, ALfloat *out)
-{
-    ALfloat d[4], f[4];
-
-    // Obtain the decayed results of the cyclical delay lines, and add the
-    // corresponding input channels attenuated by density.  Then pass the
-    // results through the low-pass filters.
-    d[0] = LateLowPassInOut(State, 0, (State->Late.DensityGain * in[0]) +
-                                      LateDelayLineOut(State, 0));
-    d[1] = LateLowPassInOut(State, 1, (State->Late.DensityGain * in[1]) +
-                                      LateDelayLineOut(State, 1));
-    d[2] = LateLowPassInOut(State, 2, (State->Late.DensityGain * in[2]) +
-                                      LateDelayLineOut(State, 2));
-    d[3] = LateLowPassInOut(State, 3, (State->Late.DensityGain * in[3]) +
-                                      LateDelayLineOut(State, 3));
-
-    // To help increase diffusion, run each line through an all-pass filter.
-    // The order of the all-pass filters is selected so that the shortest
-    // all-pass filter will feed the shortest delay line.
-    d[0] = LateAllPassInOut(State, 1, d[0]);
-    d[1] = LateAllPassInOut(State, 3, d[1]);
-    d[2] = LateAllPassInOut(State, 0, d[2]);
-    d[3] = LateAllPassInOut(State, 2, d[3]);
-
-    /* Late reverb is done with a modified feedback delay network (FDN)
-     * topology.  Four input lines are each fed through their own all-pass
-     * filter and then into the mixing matrix.  The four outputs of the
-     * mixing matrix are then cycled back to the inputs.  Each output feeds
-     * a different input to form a circlular feed cycle.
-     *
-     * The mixing matrix used is a 4D skew-symmetric rotation matrix derived
-     * using a single unitary rotational parameter:
-     *
-     *  [  d,  a,  b,  c ]          1 = a^2 + b^2 + c^2 + d^2
-     *  [ -a,  d,  c, -b ]
-     *  [ -b, -c,  d,  a ]
-     *  [ -c,  b, -a,  d ]
-     *
-     * The rotation is constructed from the effect's diffusion parameter,
-     * yielding:  1 = x^2 + 3 y^2; where a, b, and c are the coefficient y
-     * with differing signs, and d is the coefficient x.  The matrix is thus:
-     *
-     *  [  x,  y, -y,  y ]          x = 1 - (0.5 diffusion^3)
-     *  [ -y,  x,  y,  y ]          y = sqrt((1 - x^2) / 3)
-     *  [  y, -y,  x,  y ]
-     *  [ -y, -y, -y,  x ]
-     *
-     * To reduce the number of multiplies, the x coefficient is applied with
-     * the cyclical delay line coefficients.  Thus only the y coefficient is
-     * applied when mixing, and is modified to be:  y / x.
-     */
-    f[0] = d[0] + (State->Late.MixCoeff * ( d[1] - d[2] + d[3]));
-    f[1] = d[1] + (State->Late.MixCoeff * (-d[0] + d[2] + d[3]));
-    f[2] = d[2] + (State->Late.MixCoeff * ( d[0] - d[1] + d[3]));
-    f[3] = d[3] + (State->Late.MixCoeff * (-d[0] - d[1] - d[2]));
-
-    // Output the results of the matrix for all four cyclical delay lines,
-    // attenuated by the late reverb gain (which is attenuated by the 'x'
-    // mix coefficient).
-    out[0] = State->Late.Gain * f[0];
-    out[1] = State->Late.Gain * f[1];
-    out[2] = State->Late.Gain * f[2];
-    out[3] = State->Late.Gain * f[3];
-
-    // The delay lines are fed circularly in the order:
-    // 0 -> 1 -> 3 -> 2 -> 0 ...
-    DelayLineIn(&State->Late.Delay[0], State->Offset, f[2]);
-    DelayLineIn(&State->Late.Delay[1], State->Offset, f[0]);
-    DelayLineIn(&State->Late.Delay[2], State->Offset, f[3]);
-    DelayLineIn(&State->Late.Delay[3], State->Offset, f[1]);
-}
-
-// Process the reverb for a given input sample, resulting in separate four-
-// channel output for both early reflections and late reverb.
-static __inline ALvoid ReverbInOut(ALverbState *State, ALfloat in, ALfloat *early, ALfloat *late)
-{
-    ALfloat taps[4];
-
-    // Low-pass filter the incoming sample.
-    in = lpFilter2P(&State->LpFilter, 0, in);
-
-    // Feed the initial delay line.
-    DelayLineIn(&State->Delay, State->Offset, in);
-
-    // Calculate the early reflection from the first delay tap.
-    in = DelayLineOut(&State->Delay, State->Offset - State->Tap[0]);
-    EarlyReflection(State, in, early);
-
-    // Calculate the late reverb from the last four delay taps.
-    taps[0] = DelayLineOut(&State->Delay, State->Offset - State->Tap[1]);
-    taps[1] = DelayLineOut(&State->Delay, State->Offset - State->Tap[2]);
-    taps[2] = DelayLineOut(&State->Delay, State->Offset - State->Tap[3]);
-    taps[3] = DelayLineOut(&State->Delay, State->Offset - State->Tap[4]);
-    LateReverb(State, taps, late);
-
-    // Step all delays forward one sample.
-    State->Offset++;
-}
-
-// This destroys the reverb state.  It should be called only when the effect
-// slot has a different (or no) effect loaded over the reverb effect.
-ALvoid VerbDestroy(ALeffectState *effect)
-{
-    ALverbState *State = (ALverbState*)effect;
-    if(State)
-    {
-        free(State->SampleBuffer);
-        State->SampleBuffer = NULL;
-        free(State);
-    }
-}
-
-// NOTE:  Temp, remove later.
-static __inline ALint aluCart2LUTpos(ALfloat re, ALfloat im)
-{
-    ALint pos = 0;
-    ALfloat denom = aluFabs(re) + aluFabs(im);
-    if(denom > 0.0f)
-        pos = (ALint)(QUADRANT_NUM*aluFabs(im) / denom + 0.5);
-
-    if(re < 0.0)
-        pos = 2 * QUADRANT_NUM - pos;
-    if(im < 0.0)
-        pos = LUT_NUM - pos;
-    return pos%LUT_NUM;
-}
-
-// This updates the device-dependant reverb state.  This is called on
-// initialization and any time the device parameters (eg. playback frequency,
-// format) have been changed.
-ALboolean VerbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
-{
-    ALverbState *State = (ALverbState*)effect;
-    ALuint length[13], totalLength;
-    ALuint index;
-
-    totalLength = CalcLengths(length, Device->Frequency);
-    if(totalLength != State->TotalLength)
-    {
-        void *temp;
-
-        temp = realloc(State->SampleBuffer, totalLength * sizeof(ALfloat));
-        if(!temp)
-        {
-            alSetError(AL_OUT_OF_MEMORY);
-            return AL_FALSE;
-        }
-        State->TotalLength = totalLength;
-        State->SampleBuffer = temp;
-
-        // All lines share a single sample buffer
-        State->Delay.Mask = length[0] - 1;
-        State->Delay.Line = &State->SampleBuffer[0];
-        totalLength = length[0];
-        for(index = 0;index < 4;index++)
-        {
-            State->Early.Delay[index].Mask = length[1 + index] - 1;
-            State->Early.Delay[index].Line = &State->SampleBuffer[totalLength];
-            totalLength += length[1 + index];
-        }
-        for(index = 0;index < 4;index++)
-        {
-            State->Late.ApDelay[index].Mask = length[5 + index] - 1;
-            State->Late.ApDelay[index].Line = &State->SampleBuffer[totalLength];
-            totalLength += length[5 + index];
-        }
-        for(index = 0;index < 4;index++)
-        {
-            State->Late.Delay[index].Mask = length[9 + index] - 1;
-            State->Late.Delay[index].Line = &State->SampleBuffer[totalLength];
-            totalLength += length[9 + index];
-        }
-    }
-
-    for(index = 0;index < 4;index++)
-    {
-        State->Early.Offset[index] = (ALuint)(EARLY_LINE_LENGTH[index] *
-                                              Device->Frequency);
-        State->Late.ApOffset[index] = (ALuint)(ALLPASS_LINE_LENGTH[index] *
-                                               Device->Frequency);
-    }
-
-    for(index = 0;index < State->TotalLength;index++)
-        State->SampleBuffer[index] = 0.0f;
-
-    return AL_TRUE;
-}
-
-// This updates the reverb state.  This is called any time the reverb effect
-// is loaded into a slot.
-ALvoid VerbUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffect *Effect)
-{
-    ALverbState *State = (ALverbState*)effect;
-    ALuint frequency = Context->Device->Frequency;
-    ALuint index;
-    ALfloat length, mixCoeff, cw, g, coeff;
-    ALfloat hfRatio = Effect->Reverb.DecayHFRatio;
-
-    // Calculate the master low-pass filter (from the master effect HF gain).
-    cw = cos(2.0*M_PI * Effect->Reverb.HFReference / frequency);
-    g = __max(Effect->Reverb.GainHF, 0.0001f);
-    State->LpFilter.coeff = 0.0f;
-    if(g < 0.9999f) // 1-epsilon
-        State->LpFilter.coeff = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
-
-    // Calculate the initial delay taps.
-    length = Effect->Reverb.ReflectionsDelay;
-    State->Tap[0] = (ALuint)(length * frequency);
-
-    length += Effect->Reverb.LateReverbDelay;
-
-    /* The four inputs to the late reverb are decorrelated to smooth the
-     * initial reverb and reduce harsh echos.  The timings are calculated as
-     * multiples of a fraction of the smallest cyclical delay time. This
-     * result is then adjusted so that the first tap occurs immediately (all
-     * taps are reduced by the shortest fraction).
-     *
-     * offset[index] = ((FRACTION MULTIPLIER^index) - 1) delay
-     */
-    for(index = 0;index < 4;index++)
-    {
-        length += LATE_LINE_LENGTH[0] *
-            (1.0f + (Effect->Reverb.Density * LATE_LINE_MULTIPLIER)) *
-            (DECO_FRACTION * (pow(DECO_MULTIPLIER, (ALfloat)index) - 1.0f));
-        State->Tap[1 + index] = (ALuint)(length * frequency);
-    }
-
-    // Calculate the early reflections gain (from the master effect gain, and
-    // reflections gain parameters).
-    State->Early.Gain = Effect->Reverb.Gain * Effect->Reverb.ReflectionsGain;
-
-    // Calculate the gain (coefficient) for each early delay line.
-    for(index = 0;index < 4;index++)
-        State->Early.Coeff[index] = pow(10.0f, EARLY_LINE_LENGTH[index] /
-                                               Effect->Reverb.LateReverbDelay *
-                                               -60.0f / 20.0f);
-
-    // Calculate the first mixing matrix coefficient (x).
-    mixCoeff = 1.0f - (0.5f * pow(Effect->Reverb.Diffusion, 3.0f));
-
-    // Calculate the late reverb gain (from the master effect gain, and late
-    // reverb gain parameters).  Since the output is tapped prior to the
-    // application of the delay line coefficients, this gain needs to be
-    // attenuated by the 'x' mix coefficient from above.
-    State->Late.Gain = Effect->Reverb.Gain * Effect->Reverb.LateReverbGain * mixCoeff;
-
-    /* To compensate for changes in modal density and decay time of the late
-     * reverb signal, the input is attenuated based on the maximal energy of
-     * the outgoing signal.  This is calculated as the ratio between a
-     * reference value and the current approximation of energy for the output
-     * signal.
-     *
-     * Reverb output matches exponential decay of the form Sum(a^n), where a
-     * is the attenuation coefficient, and n is the sample ranging from 0 to
-     * infinity.  The signal energy can thus be approximated using the area
-     * under this curve, calculated as:  1 / (1 - a).
-     *
-     * The reference energy is calculated from a signal at the lowest (effect
-     * at 1.0) density with a decay time of one second.
-     *
-     * The coefficient is calculated as the average length of the cyclical
-     * delay lines.  This produces a better result than calculating the gain
-     * for each line individually (most likely a side effect of diffusion).
-     *
-     * The final result is the square root of the ratio bound to a maximum
-     * value of 1 (no amplification).
-     */
-    length = (LATE_LINE_LENGTH[0] + LATE_LINE_LENGTH[1] +
-              LATE_LINE_LENGTH[2] + LATE_LINE_LENGTH[3]);
-    g = length * (1.0f + LATE_LINE_MULTIPLIER) * 0.25f;
-    g = pow(10.0f, g * -60.0f / 20.0f);
-    g = 1.0f / (1.0f - (g * g));
-    length *= 1.0f + (Effect->Reverb.Density * LATE_LINE_MULTIPLIER) * 0.25f;
-    length = pow(10.0f, length / Effect->Reverb.DecayTime * -60.0f / 20.0f);
-    length = 1.0f / (1.0f - (length * length));
-    State->Late.DensityGain = __min(aluSqrt(g / length), 1.0f);
-
-    // Calculate the all-pass feed-back and feed-forward coefficient.
-    State->Late.ApFeedCoeff = 0.6f * pow(Effect->Reverb.Diffusion, 3.0f);
-
-    // Calculate the mixing matrix coefficient (y / x).
-    g = aluSqrt((1.0f - (mixCoeff * mixCoeff)) / 3.0f);
-    State->Late.MixCoeff = g / mixCoeff;
-
-    for(index = 0;index < 4;index++)
-    {
-        // Calculate the gain (coefficient) for each all-pass line.
-        State->Late.ApCoeff[index] = pow(10.0f, ALLPASS_LINE_LENGTH[index] /
-                                                Effect->Reverb.DecayTime *
-                                                -60.0f / 20.0f);
-    }
-
-    // If the HF limit parameter is flagged, calculate an appropriate limit
-    // based on the air absorption parameter.
-    if(Effect->Reverb.DecayHFLimit && Effect->Reverb.AirAbsorptionGainHF < 1.0f)
-    {
-        ALfloat limitRatio;
-
-        // For each of the cyclical delays, find the attenuation due to air
-        // absorption in dB (converting delay time to meters using the speed
-        // of sound).  Then reversing the decay equation, solve for HF ratio.
-        // The delay length is cancelled out of the equation, so it can be
-        // calculated once for all lines.
-        limitRatio = 1.0f / (log10(Effect->Reverb.AirAbsorptionGainHF) *
-                             SPEEDOFSOUNDMETRESPERSEC *
-                             Effect->Reverb.DecayTime / -60.0f * 20.0f);
-        // Need to limit the result to a minimum of 0.1, just like the HF
-        // ratio parameter.
-        limitRatio = __max(limitRatio, 0.1f);
-
-        // Using the limit calculated above, apply the upper bound to the
-        // HF ratio.
-        hfRatio = __min(hfRatio, limitRatio);
-    }
-
-    // Calculate the low-pass filter frequency.
-    cw = cos(2.0*M_PI * Effect->Reverb.HFReference / frequency);
-
-    for(index = 0;index < 4;index++)
-    {
-        // Calculate the length (in seconds) of each cyclical delay line.
-        length = LATE_LINE_LENGTH[index] * (1.0f + (Effect->Reverb.Density *
-                                                    LATE_LINE_MULTIPLIER));
-        // Calculate the delay offset for the cyclical delay lines.
-        State->Late.Offset[index] = (ALuint)(length * frequency);
-
-        // Calculate the gain (coefficient) for each cyclical line.
-        State->Late.Coeff[index] = pow(10.0f, length / Effect->Reverb.DecayTime *
-                                              -60.0f / 20.0f);
-
-        // Eventually this should boost the high frequencies when the ratio
-        // exceeds 1.
-        coeff = 0.0f;
-        if (hfRatio < 1.0f)
-        {
-            // Calculate the decay equation for each low-pass filter.
-            g = pow(10.0f, length / (Effect->Reverb.DecayTime * hfRatio) *
-                       -60.0f / 20.0f) / State->Late.Coeff[index];
-            g  = __max(g, 0.1f);
-            g *= g;
-
-            // Calculate the gain (coefficient) for each low-pass filter.
-            if(g < 0.9999f) // 1-epsilon
-                coeff = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
-
-            // Very low decay times will produce minimal output, so apply an
-            // upper bound to the coefficient.
-            coeff = __min(coeff, 0.98f);
-        }
-        State->Late.LpCoeff[index] = coeff;
-
-        // Attenuate the cyclical line coefficients by the mixing coefficient
-        // (x).
-        State->Late.Coeff[index] *= mixCoeff;
-    }
-
-    // Calculate the 3D-panning gains for the early reflections and late
-    // reverb (for EAX mode).
-    {
-        ALfloat earlyPan[3] = { Effect->Reverb.ReflectionsPan[0], Effect->Reverb.ReflectionsPan[1], Effect->Reverb.ReflectionsPan[2] };
-        ALfloat latePan[3] = { Effect->Reverb.LateReverbPan[0], Effect->Reverb.LateReverbPan[1], Effect->Reverb.LateReverbPan[2] };
-        ALfloat *speakerGain, dirGain, ambientGain;
-        ALfloat length;
-        ALint pos;
-
-        length = earlyPan[0]*earlyPan[0] + earlyPan[1]*earlyPan[1] + earlyPan[2]*earlyPan[2];
-        if(length > 1.0f)
-        {
-            length = 1.0f / aluSqrt(length);
-            earlyPan[0] *= length;
-            earlyPan[1] *= length;
-            earlyPan[2] *= length;
-        }
-        length = latePan[0]*latePan[0] + latePan[1]*latePan[1] + latePan[2]*latePan[2];
-        if(length > 1.0f)
-        {
-            length = 1.0f / aluSqrt(length);
-            latePan[0] *= length;
-            latePan[1] *= length;
-            latePan[2] *= length;
-        }
-
-        // This code applies directional reverb just like the mixer applies
-        // directional sources.  It diffuses the sound toward all speakers
-        // as the magnitude of the panning vector drops, which is only an
-        // approximation of the expansion of sound across the speakers from
-        // the panning direction.
-        pos = aluCart2LUTpos(earlyPan[2], earlyPan[0]);
-        speakerGain = &Context->PanningLUT[OUTPUTCHANNELS * pos];
-        dirGain = aluSqrt((earlyPan[0] * earlyPan[0]) + (earlyPan[2] * earlyPan[2]));
-        ambientGain = (1.0 - dirGain);
-        for(index = 0;index < OUTPUTCHANNELS;index++)
-             State->Early.PanGain[index] = dirGain * speakerGain[index] + ambientGain;
-
-        pos = aluCart2LUTpos(latePan[2], latePan[0]);
-        speakerGain = &Context->PanningLUT[OUTPUTCHANNELS * pos];
-        dirGain = aluSqrt((latePan[0] * latePan[0]) + (latePan[2] * latePan[2]));
-        ambientGain = (1.0 - dirGain);
-        for(index = 0;index < OUTPUTCHANNELS;index++)
-             State->Late.PanGain[index] = dirGain * speakerGain[index] + ambientGain;
-    }
-}
-
-// This processes the reverb state, given the input samples and an output
-// buffer.
-ALvoid VerbProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[OUTPUTCHANNELS])
-{
-    ALverbState *State = (ALverbState*)effect;
-    ALuint index;
-    ALfloat early[4], late[4], out[4];
-    ALfloat gain = Slot->Gain;
-
-    for(index = 0;index < SamplesToDo;index++)
-    {
-        // Process reverb for this sample.
-        ReverbInOut(State, SamplesIn[index], early, late);
-
-        // Mix early reflections and late reverb.
-        out[0] = (early[0] + late[0]) * gain;
-        out[1] = (early[1] + late[1]) * gain;
-        out[2] = (early[2] + late[2]) * gain;
-        out[3] = (early[3] + late[3]) * gain;
-
-        // Output the results.
-        SamplesOut[index][FRONT_LEFT]   += out[0];
-        SamplesOut[index][FRONT_RIGHT]  += out[1];
-        SamplesOut[index][FRONT_CENTER] += out[3];
-        SamplesOut[index][SIDE_LEFT]    += out[0];
-        SamplesOut[index][SIDE_RIGHT]   += out[1];
-        SamplesOut[index][BACK_LEFT]    += out[0];
-        SamplesOut[index][BACK_RIGHT]   += out[1];
-        SamplesOut[index][BACK_CENTER]  += out[2];
-    }
-}
-
-// This processes the EAX reverb state, given the input samples and an output
-// buffer.
-ALvoid EAXVerbProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[OUTPUTCHANNELS])
-{
-    ALverbState *State = (ALverbState*)effect;
-    ALuint index;
-    ALfloat early[4], late[4];
-    ALfloat gain = Slot->Gain;
-
-    for(index = 0;index < SamplesToDo;index++)
-    {
-        // Process reverb for this sample.
-        ReverbInOut(State, SamplesIn[index], early, late);
-
-        // Unfortunately, while the number and configuration of gains for
-        // panning adjust according to OUTPUTCHANNELS, the output from the
-        // reverb engine is not so scalable.
-        SamplesOut[index][FRONT_LEFT] +=
-           (State->Early.PanGain[FRONT_LEFT]*early[0] +
-            State->Late.PanGain[FRONT_LEFT]*late[0]) * gain;
-        SamplesOut[index][FRONT_RIGHT] +=
-           (State->Early.PanGain[FRONT_RIGHT]*early[1] +
-            State->Late.PanGain[FRONT_RIGHT]*late[1]) * gain;
-        SamplesOut[index][FRONT_CENTER] +=
-           (State->Early.PanGain[FRONT_CENTER]*early[3] +
-            State->Late.PanGain[FRONT_CENTER]*late[3]) * gain;
-        SamplesOut[index][SIDE_LEFT] +=
-           (State->Early.PanGain[SIDE_LEFT]*early[0] +
-            State->Late.PanGain[SIDE_LEFT]*late[0]) * gain;
-        SamplesOut[index][SIDE_RIGHT] +=
-           (State->Early.PanGain[SIDE_RIGHT]*early[1] +
-            State->Late.PanGain[SIDE_RIGHT]*late[1]) * gain;
-        SamplesOut[index][BACK_LEFT] +=
-           (State->Early.PanGain[BACK_LEFT]*early[0] +
-            State->Late.PanGain[BACK_LEFT]*late[0]) * gain;
-        SamplesOut[index][BACK_RIGHT] +=
-           (State->Early.PanGain[BACK_RIGHT]*early[1] +
-            State->Late.PanGain[BACK_RIGHT]*late[1]) * gain;
-        SamplesOut[index][BACK_CENTER] +=
-           (State->Early.PanGain[BACK_CENTER]*early[2] +
-            State->Late.PanGain[BACK_CENTER]*late[2]) * gain;
-    }
-}
-
-// This creates the reverb state.  It should be called only when the reverb
-// effect is loaded into a slot that doesn't already have a reverb effect.
-ALeffectState *VerbCreate(void)
-{
-    ALverbState *State = NULL;
-    ALuint index;
-
-    State = malloc(sizeof(ALverbState));
-    if(!State)
-    {
-        alSetError(AL_OUT_OF_MEMORY);
-        return NULL;
-    }
-
-    State->state.Destroy = VerbDestroy;
-    State->state.DeviceUpdate = VerbDeviceUpdate;
-    State->state.Update = VerbUpdate;
-    State->state.Process = VerbProcess;
-
-    State->TotalLength = 0;
-    State->SampleBuffer = NULL;
-
-    State->LpFilter.coeff = 0.0f;
-    State->LpFilter.history[0] = 0.0f;
-    State->LpFilter.history[1] = 0.0f;
-    State->Delay.Mask = 0;
-    State->Delay.Line = NULL;
-
-    State->Tap[0] = 0;
-    State->Tap[1] = 0;
-    State->Tap[2] = 0;
-    State->Tap[3] = 0;
-    State->Tap[4] = 0;
-
-    State->Early.Gain = 0.0f;
-    for(index = 0;index < 4;index++)
-    {
-        State->Early.Coeff[index] = 0.0f;
-        State->Early.Delay[index].Mask = 0;
-        State->Early.Delay[index].Line = NULL;
-        State->Early.Offset[index] = 0;
-    }
-
-    State->Late.Gain = 0.0f;
-    State->Late.DensityGain = 0.0f;
-    State->Late.ApFeedCoeff = 0.0f;
-    State->Late.MixCoeff = 0.0f;
-
-    for(index = 0;index < 4;index++)
-    {
-        State->Late.ApCoeff[index] = 0.0f;
-        State->Late.ApDelay[index].Mask = 0;
-        State->Late.ApDelay[index].Line = NULL;
-        State->Late.ApOffset[index] = 0;
-
-        State->Late.Coeff[index] = 0.0f;
-        State->Late.Delay[index].Mask = 0;
-        State->Late.Delay[index].Line = NULL;
-        State->Late.Offset[index] = 0;
-
-        State->Late.LpCoeff[index] = 0.0f;
-        State->Late.LpSample[index] = 0.0f;
-    }
-
-    // Panning is applied as an independent gain for each output channel.
-    for(index = 0;index < OUTPUTCHANNELS;index++)
-    {
-        State->Early.PanGain[index] = 0.0f;
-        State->Late.PanGain[index] = 0.0f;
-    }
-
-    State->Offset = 0;
-    return &State->state;
-}
-
-ALeffectState *EAXVerbCreate(void)
-{
-    ALeffectState *State = VerbCreate();
-    if(State) State->Process = EAXVerbProcess;
-    return State;
-}

Alc/alcRing.c

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -24,102 +24,294 @@
 #include <stdlib.h>
 
 #include "alMain.h"
+#include "threads.h"
+#include "almalloc.h"
+#include "compat.h"
 
 
-struct RingBuffer {
-    ALubyte *mem;
+/* NOTE: This lockless ringbuffer implementation is copied from JACK, extended
+ * to include an element size. Consequently, parameters and return values for a
+ * size or count is in 'elements', not bytes. Additionally, it only supports
+ * single-consumer/single-provider operation. */
+struct ll_ringbuffer {
+    ATOMIC(size_t) write_ptr;
+    ATOMIC(size_t) read_ptr;
+    size_t size;
+    size_t size_mask;
+    size_t elem_size;
+    int mlocked;
 
-    ALsizei frame_size;
-    ALsizei length;
-    ALint read_pos;
-    ALint write_pos;
-
-    CRITICAL_SECTION cs;
+    alignas(16) char buf[];
 };
 
-
-RingBuffer *CreateRingBuffer(ALsizei frame_size, ALsizei length)
+/* Create a new ringbuffer to hold at least `sz' elements of `elem_sz' bytes.
+ * The number of elements is rounded up to the next power of two. */
+ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz)
 {
-    RingBuffer *ring = calloc(1, sizeof(*ring));
-    if(ring)
-    {
-        ring->frame_size = frame_size;
-        ring->length = length+1;
-        ring->write_pos = 1;
-        ring->mem = malloc((length+1)*frame_size);
-        if(!ring->mem)
-        {
-            free(ring);
-            ring = NULL;
-        }
+    ll_ringbuffer_t *rb;
+    ALuint power_of_two;
 
-        InitializeCriticalSection(&ring->cs);
-    }
-    return ring;
+    power_of_two = NextPowerOf2(sz);
+    if(power_of_two < sz)
+        return NULL;
+
+    rb = al_malloc(16, sizeof(*rb) + power_of_two*elem_sz);
+    if(!rb) return NULL;
+
+    ATOMIC_INIT(&rb->write_ptr, 0);
+    ATOMIC_INIT(&rb->read_ptr, 0);
+    rb->size = power_of_two;
+    rb->size_mask = rb->size - 1;
+    rb->elem_size = elem_sz;
+    rb->mlocked = 0;
+    return rb;
 }
 
-void DestroyRingBuffer(RingBuffer *ring)
+/* Free all data associated with the ringbuffer `rb'. */
+void ll_ringbuffer_free(ll_ringbuffer_t *rb)
 {
-    if(ring)
+    if(rb)
     {
-        DeleteCriticalSection(&ring->cs);
-        free(ring->mem);
-        free(ring);
+#ifdef USE_MLOCK
+        if(rb->mlocked)
+            munlock(rb, sizeof(*rb) + rb->size*rb->elem_size);
+#endif /* USE_MLOCK */
+        al_free(rb);
     }
 }
 
-ALsizei RingBufferSize(RingBuffer *ring)
+/* Lock the data block of `rb' using the system call 'mlock'. */
+int ll_ringbuffer_mlock(ll_ringbuffer_t *rb)
 {
-    ALsizei s;
-
-    EnterCriticalSection(&ring->cs);
-    s = (ring->write_pos-ring->read_pos-1+ring->length) % ring->length;
-    LeaveCriticalSection(&ring->cs);
-
-    return s;
+#ifdef USE_MLOCK
+    if(!rb->mlocked && mlock(rb, sizeof(*rb) + rb->size*rb->elem_size))
+        return -1;
+#endif /* USE_MLOCK */
+    rb->mlocked = 1;
+    return 0;
 }
 
-void WriteRingBuffer(RingBuffer *ring, const ALubyte *data, ALsizei len)
+/* Reset the read and write pointers to zero. This is not thread safe. */
+void ll_ringbuffer_reset(ll_ringbuffer_t *rb)
 {
-    int remain;
+    ATOMIC_STORE(&rb->write_ptr, 0, almemory_order_release);
+    ATOMIC_STORE(&rb->read_ptr, 0, almemory_order_release);
+    memset(rb->buf, 0, rb->size*rb->elem_size);
+}
 
-    EnterCriticalSection(&ring->cs);
+/* Return the number of elements available for reading. This is the number of
+ * elements in front of the read pointer and behind the write pointer. */
+size_t ll_ringbuffer_read_space(const ll_ringbuffer_t *rb)
+{
+    size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
+    size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
+    return (w-r) & rb->size_mask;
+}
+/* Return the number of elements available for writing. This is the number of
+ * elements in front of the write pointer and behind the read pointer. */
+size_t ll_ringbuffer_write_space(const ll_ringbuffer_t *rb)
+{
+    size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
+    size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
+    return (r-w-1) & rb->size_mask;
+}
 
-    remain = ring->length - ring->write_pos;
-    if((ring->read_pos-ring->write_pos+ring->length)%ring->length < len)
-        ring->read_pos = (ring->write_pos+len) % ring->length;
+/* The copying data reader. Copy at most `cnt' elements from `rb' to `dest'.
+ * Returns the actual number of elements copied. */
+size_t ll_ringbuffer_read(ll_ringbuffer_t *rb, char *dest, size_t cnt)
+{
+    size_t read_ptr;
+    size_t free_cnt;
+    size_t cnt2;
+    size_t to_read;
+    size_t n1, n2;
 
-    if(remain < len)
+    free_cnt = ll_ringbuffer_read_space(rb);
+    if(free_cnt == 0) return 0;
+
+    to_read = (cnt > free_cnt) ? free_cnt : cnt;
+    read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask;
+
+    cnt2 = read_ptr + to_read;
+    if(cnt2 > rb->size)
     {
-        memcpy(ring->mem+(ring->write_pos*ring->frame_size), data, remain*ring->frame_size);
-        memcpy(ring->mem, data+(remain*ring->frame_size), (len-remain)*ring->frame_size);
+        n1 = rb->size - read_ptr;
+        n2 = cnt2 & rb->size_mask;
     }
     else
-        memcpy(ring->mem+(ring->write_pos*ring->frame_size), data, len*ring->frame_size);
+    {
+        n1 = to_read;
+        n2 = 0;
+    }
 
-    ring->write_pos += len;
-    ring->write_pos %= ring->length;
-
-    LeaveCriticalSection(&ring->cs);
+    memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size);
+    read_ptr += n1;
+    if(n2)
+    {
+        memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size],
+               n2*rb->elem_size);
+        read_ptr += n2;
+    }
+    ATOMIC_STORE(&rb->read_ptr, read_ptr, almemory_order_release);
+    return to_read;
 }
 
-void ReadRingBuffer(RingBuffer *ring, ALubyte *data, ALsizei len)
+/* The copying data reader w/o read pointer advance. Copy at most `cnt'
+ * elements from `rb' to `dest'. Returns the actual number of elements copied.
+ */
+size_t ll_ringbuffer_peek(ll_ringbuffer_t *rb, char *dest, size_t cnt)
 {
-    int remain;
+    size_t free_cnt;
+    size_t cnt2;
+    size_t to_read;
+    size_t n1, n2;
+    size_t read_ptr;
 
-    EnterCriticalSection(&ring->cs);
+    free_cnt = ll_ringbuffer_read_space(rb);
+    if(free_cnt == 0) return 0;
 
-    remain = ring->length - ring->read_pos;
-    if(remain < len)
+    to_read = (cnt > free_cnt) ? free_cnt : cnt;
+    read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask;
+
+    cnt2 = read_ptr + to_read;
+    if(cnt2 > rb->size)
     {
-        memcpy(data, ring->mem+(ring->read_pos*ring->frame_size), remain*ring->frame_size);
-        memcpy(data+(remain*ring->frame_size), ring->mem, (len-remain)*ring->frame_size);
+        n1 = rb->size - read_ptr;
+        n2 = cnt2 & rb->size_mask;
     }
     else
-        memcpy(data, ring->mem+(ring->read_pos*ring->frame_size), len*ring->frame_size);
+    {
+        n1 = to_read;
+        n2 = 0;
+    }
 
-    ring->read_pos += len;
-    ring->read_pos %= ring->length;
-
-    LeaveCriticalSection(&ring->cs);
+    memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size);
+    if(n2)
+    {
+        read_ptr += n1;
+        memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size],
+               n2*rb->elem_size);
+    }
+    return to_read;
+}
+
+/* The copying data writer. Copy at most `cnt' elements to `rb' from `src'.
+ * Returns the actual number of elements copied. */
+size_t ll_ringbuffer_write(ll_ringbuffer_t *rb, const char *src, size_t cnt)
+{
+    size_t write_ptr;
+    size_t free_cnt;
+    size_t cnt2;
+    size_t to_write;
+    size_t n1, n2;
+
+    free_cnt = ll_ringbuffer_write_space(rb);
+    if(free_cnt == 0) return 0;
+
+    to_write = (cnt > free_cnt) ? free_cnt : cnt;
+    write_ptr = ATOMIC_LOAD(&rb->write_ptr, almemory_order_relaxed) & rb->size_mask;
+
+    cnt2 = write_ptr + to_write;
+    if(cnt2 > rb->size)
+    {
+        n1 = rb->size - write_ptr;
+        n2 = cnt2 & rb->size_mask;
+    }
+    else
+    {
+        n1 = to_write;
+        n2 = 0;
+    }
+
+    memcpy(&rb->buf[write_ptr*rb->elem_size], src, n1*rb->elem_size);
+    write_ptr += n1;
+    if(n2)
+    {
+        memcpy(&rb->buf[(write_ptr&rb->size_mask)*rb->elem_size], src + n1*rb->elem_size,
+               n2*rb->elem_size);
+        write_ptr += n2;
+    }
+    ATOMIC_STORE(&rb->write_ptr, write_ptr, almemory_order_release);
+    return to_write;
+}
+
+/* Advance the read pointer `cnt' places. */
+void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt)
+{
+    ATOMIC_ADD(&rb->read_ptr, cnt, almemory_order_acq_rel);
+}
+
+/* Advance the write pointer `cnt' places. */
+void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt)
+{
+    ATOMIC_ADD(&rb->write_ptr, cnt, almemory_order_acq_rel);
+}
+
+/* The non-copying data reader. `vec' is an array of two places. Set the values
+ * at `vec' to hold the current readable data at `rb'. If the readable data is
+ * in one segment the second segment has zero length. */
+void ll_ringbuffer_get_read_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t * vec)
+{
+    size_t free_cnt;
+    size_t cnt2;
+    size_t w, r;
+
+    w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
+    r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
+    w &= rb->size_mask;
+    r &= rb->size_mask;
+    free_cnt = (w-r) & rb->size_mask;
+
+    cnt2 = r + free_cnt;
+    if(cnt2 > rb->size)
+    {
+        /* Two part vector: the rest of the buffer after the current write ptr,
+         * plus some from the start of the buffer. */
+        vec[0].buf = (char*)&rb->buf[r*rb->elem_size];
+        vec[0].len = rb->size - r;
+        vec[1].buf = (char*)rb->buf;
+        vec[1].len = cnt2 & rb->size_mask;
+    }
+    else
+    {
+        /* Single part vector: just the rest of the buffer */
+        vec[0].buf = (char*)&rb->buf[r*rb->elem_size];
+        vec[0].len = free_cnt;
+        vec[1].buf = NULL;
+        vec[1].len = 0;
+    }
+}
+
+/* The non-copying data writer. `vec' is an array of two places. Set the values
+ * at `vec' to hold the current writeable data at `rb'. If the writeable data
+ * is in one segment the second segment has zero length. */
+void ll_ringbuffer_get_write_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t *vec)
+{
+    size_t free_cnt;
+    size_t cnt2;
+    size_t w, r;
+
+    w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
+    r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
+    w &= rb->size_mask;
+    r &= rb->size_mask;
+    free_cnt = (r-w-1) & rb->size_mask;
+
+    cnt2 = w + free_cnt;
+    if(cnt2 > rb->size)
+    {
+        /* Two part vector: the rest of the buffer after the current write ptr,
+         * plus some from the start of the buffer. */
+        vec[0].buf = (char*)&rb->buf[w*rb->elem_size];
+        vec[0].len = rb->size - w;
+        vec[1].buf = (char*)rb->buf;
+        vec[1].len = cnt2 & rb->size_mask;
+    }
+    else
+    {
+        vec[0].buf = (char*)&rb->buf[w*rb->elem_size];
+        vec[0].len = free_cnt;
+        vec[1].buf = NULL;
+        vec[1].len = 0;
+    }
 }

Alc/alcThread.c

@@ -1,128 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alThunk.h"
-
-
-#ifdef _WIN32
-
-typedef struct {
-    ALuint (*func)(ALvoid*);
-    ALvoid *ptr;
-    HANDLE thread;
-} ThreadInfo;
-
-static DWORD CALLBACK StarterFunc(void *ptr)
-{
-    ThreadInfo *inf = (ThreadInfo*)ptr;
-    ALint ret;
-
-    ret = inf->func(inf->ptr);
-    ExitThread((DWORD)ret);
-
-    return (DWORD)ret;
-}
-
-ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
-{
-    DWORD dummy;
-    ThreadInfo *inf = malloc(sizeof(ThreadInfo));
-    if(!inf) return 0;
-
-    inf->func = func;
-    inf->ptr = ptr;
-
-    inf->thread = CreateThread(NULL, 0, StarterFunc, inf, 0, &dummy);
-    if(!inf->thread)
-    {
-        free(inf);
-        return NULL;
-    }
-
-    return inf;
-}
-
-ALuint StopThread(ALvoid *thread)
-{
-    ThreadInfo *inf = thread;
-    DWORD ret = 0;
-
-    WaitForSingleObject(inf->thread, INFINITE);
-    GetExitCodeThread(inf->thread, &ret);
-    CloseHandle(inf->thread);
-
-    free(inf);
-
-    return (ALuint)ret;
-}
-
-#else
-
-#include <pthread.h>
-
-typedef struct {
-    ALuint (*func)(ALvoid*);
-    ALvoid *ptr;
-    ALuint ret;
-    pthread_t thread;
-} ThreadInfo;
-
-static void *StarterFunc(void *ptr)
-{
-    ThreadInfo *inf = (ThreadInfo*)ptr;
-    inf->ret = inf->func(inf->ptr);
-    return NULL;
-}
-
-ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
-{
-    ThreadInfo *inf = malloc(sizeof(ThreadInfo));
-    if(!inf) return NULL;
-
-    inf->func = func;
-    inf->ptr = ptr;
-    if(pthread_create(&inf->thread, NULL, StarterFunc, inf) != 0)
-    {
-        free(inf);
-        return NULL;
-    }
-
-    return inf;
-}
-
-ALuint StopThread(ALvoid *thread)
-{
-    ThreadInfo *inf = thread;
-    ALuint ret;
-
-    pthread_join(inf->thread, NULL);
-    ret = inf->ret;
-
-    free(inf);
-
-    return ret;
-}
-
-#endif

Alc/alstring.h

@@ -0,0 +1,49 @@
+#ifndef ALSTRING_H
+#define ALSTRING_H
+
+#include <string.h>
+
+#include "vector.h"
+
+
+typedef char al_string_char_type;
+TYPEDEF_VECTOR(al_string_char_type, al_string)
+TYPEDEF_VECTOR(al_string, vector_al_string)
+
+inline void alstr_reset(al_string *str)
+{ VECTOR_DEINIT(*str); }
+#define AL_STRING_INIT(_x)       do { (_x) = (al_string)NULL; } while(0)
+#define AL_STRING_INIT_STATIC()  ((al_string)NULL)
+#define AL_STRING_DEINIT(_x)     alstr_reset(&(_x))
+
+inline size_t alstr_length(const_al_string str)
+{ return VECTOR_SIZE(str); }
+
+inline ALboolean alstr_empty(const_al_string str)
+{ return alstr_length(str) == 0; }
+
+inline const al_string_char_type *alstr_get_cstr(const_al_string str)
+{ return str ? &VECTOR_FRONT(str) : ""; }
+
+void alstr_clear(al_string *str);
+
+int alstr_cmp(const_al_string str1, const_al_string str2);
+int alstr_cmp_cstr(const_al_string str1, const al_string_char_type *str2);
+
+void alstr_copy(al_string *str, const_al_string from);
+void alstr_copy_cstr(al_string *str, const al_string_char_type *from);
+void alstr_copy_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to);
+
+void alstr_append_char(al_string *str, const al_string_char_type c);
+void alstr_append_cstr(al_string *str, const al_string_char_type *from);
+void alstr_append_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to);
+
+#ifdef _WIN32
+#include <wchar.h>
+/* Windows-only methods to deal with WideChar strings. */
+void alstr_copy_wcstr(al_string *str, const wchar_t *from);
+void alstr_append_wcstr(al_string *str, const wchar_t *from);
+void alstr_append_wrange(al_string *str, const wchar_t *from, const wchar_t *to);
+#endif
+
+#endif /* ALSTRING_H */

Alc/ambdec.c

@@ -0,0 +1,566 @@
+
+#include "config.h"
+
+#include "ambdec.h"
+
+#include <stdio.h>
+#include <string.h>
+#include <ctype.h>
+
+#include "compat.h"
+
+
+static char *lstrip(char *line)
+{
+    while(isspace(line[0]))
+        line++;
+    return line;
+}
+
+static char *rstrip(char *line)
+{
+    size_t len = strlen(line);
+    while(len > 0 && isspace(line[len-1]))
+        len--;
+    line[len] = 0;
+    return line;
+}
+
+static int readline(FILE *f, char **output, size_t *maxlen)
+{
+    size_t len = 0;
+    int c;
+
+    while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n'))
+        ;
+    if(c == EOF)
+        return 0;
+
+    do {
+        if(len+1 >= *maxlen)
+        {
+            void *temp = NULL;
+            size_t newmax;
+
+            newmax = (*maxlen ? (*maxlen)<<1 : 32);
+            if(newmax > *maxlen)
+                temp = realloc(*output, newmax);
+            if(!temp)
+            {
+                ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen);
+                return 0;
+            }
+
+            *output = temp;
+            *maxlen = newmax;
+        }
+        (*output)[len++] = c;
+        (*output)[len] = '\0';
+    } while((c=fgetc(f)) != EOF && c != '\r' && c != '\n');
+
+    return 1;
+}
+
+
+/* Custom strtok_r, since we can't rely on it existing. */
+static char *my_strtok_r(char *str, const char *delim, char **saveptr)
+{
+    /* Sanity check and update internal pointer. */
+    if(!saveptr || !delim) return NULL;
+    if(str) *saveptr = str;
+    str = *saveptr;
+
+    /* Nothing more to do with this string. */
+    if(!str) return NULL;
+
+    /* Find the first non-delimiter character. */
+    while(*str != '\0' && strchr(delim, *str) != NULL)
+        str++;
+    if(*str == '\0')
+    {
+        /* End of string. */
+        *saveptr = NULL;
+        return NULL;
+    }
+
+    /* Find the next delimiter character. */
+    *saveptr = strpbrk(str, delim);
+    if(*saveptr) *((*saveptr)++) = '\0';
+
+    return str;
+}
+
+static char *read_int(ALint *num, const char *line, int base)
+{
+    char *end;
+    *num = strtol(line, &end, base);
+    if(end && *end != '\0')
+        end = lstrip(end);
+    return end;
+}
+
+static char *read_uint(ALuint *num, const char *line, int base)
+{
+    char *end;
+    *num = strtoul(line, &end, base);
+    if(end && *end != '\0')
+        end = lstrip(end);
+    return end;
+}
+
+static char *read_float(ALfloat *num, const char *line)
+{
+    char *end;
+#ifdef HAVE_STRTOF
+    *num = strtof(line, &end);
+#else
+    *num = (ALfloat)strtod(line, &end);
+#endif
+    if(end && *end != '\0')
+        end = lstrip(end);
+    return end;
+}
+
+
+char *read_clipped_line(FILE *f, char **buffer, size_t *maxlen)
+{
+    while(readline(f, buffer, maxlen))
+    {
+        char *line, *comment;
+
+        line = lstrip(*buffer);
+        comment = strchr(line, '#');
+        if(comment) *(comment++) = 0;
+
+        line = rstrip(line);
+        if(line[0]) return line;
+    }
+    return NULL;
+}
+
+static int load_ambdec_speakers(AmbDecConf *conf, FILE *f, char **buffer, size_t *maxlen, char **saveptr)
+{
+    ALsizei cur = 0;
+    while(cur < conf->NumSpeakers)
+    {
+        const char *cmd = my_strtok_r(NULL, " \t", saveptr);
+        if(!cmd)
+        {
+            char *line = read_clipped_line(f, buffer, maxlen);
+            if(!line)
+            {
+                ERR("Unexpected end of file\n");
+                return 0;
+            }
+            cmd = my_strtok_r(line, " \t", saveptr);
+        }
+
+        if(strcmp(cmd, "add_spkr") == 0)
+        {
+            const char *name = my_strtok_r(NULL, " \t", saveptr);
+            const char *dist = my_strtok_r(NULL, " \t", saveptr);
+            const char *az = my_strtok_r(NULL, " \t", saveptr);
+            const char *elev = my_strtok_r(NULL, " \t", saveptr);
+            const char *conn = my_strtok_r(NULL, " \t", saveptr);
+
+            if(!name) WARN("Name not specified for speaker %u\n", cur+1);
+            else alstr_copy_cstr(&conf->Speakers[cur].Name, name);
+            if(!dist) WARN("Distance not specified for speaker %u\n", cur+1);
+            else read_float(&conf->Speakers[cur].Distance, dist);
+            if(!az) WARN("Azimuth not specified for speaker %u\n", cur+1);
+            else read_float(&conf->Speakers[cur].Azimuth, az);
+            if(!elev) WARN("Elevation not specified for speaker %u\n", cur+1);
+            else read_float(&conf->Speakers[cur].Elevation, elev);
+            if(!conn) TRACE("Connection not specified for speaker %u\n", cur+1);
+            else alstr_copy_cstr(&conf->Speakers[cur].Connection, conn);
+
+            cur++;
+        }
+        else
+        {
+            ERR("Unexpected speakers command: %s\n", cmd);
+            return 0;
+        }
+
+        cmd = my_strtok_r(NULL, " \t", saveptr);
+        if(cmd)
+        {
+            ERR("Unexpected junk on line: %s\n", cmd);
+            return 0;
+        }
+    }
+
+    return 1;
+}
+
+static int load_ambdec_matrix(ALfloat *gains, ALfloat (*matrix)[MAX_AMBI_COEFFS], ALsizei maxrow, FILE *f, char **buffer, size_t *maxlen, char **saveptr)
+{
+    int gotgains = 0;
+    ALsizei cur = 0;
+    while(cur < maxrow)
+    {
+        const char *cmd = my_strtok_r(NULL, " \t", saveptr);
+        if(!cmd)
+        {
+            char *line = read_clipped_line(f, buffer, maxlen);
+            if(!line)
+            {
+                ERR("Unexpected end of file\n");
+                return 0;
+            }
+            cmd = my_strtok_r(line, " \t", saveptr);
+        }
+
+        if(strcmp(cmd, "order_gain") == 0)
+        {
+            ALuint curgain = 0;
+            char *line;
+            while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL)
+            {
+                ALfloat value;
+                line = read_float(&value, line);
+                if(line && *line != '\0')
+                {
+                    ERR("Extra junk on gain %u: %s\n", curgain+1, line);
+                    return 0;
+                }
+                if(curgain < MAX_AMBI_ORDER+1)
+                    gains[curgain] = value;
+                curgain++;
+            }
+            while(curgain < MAX_AMBI_ORDER+1)
+                gains[curgain++] = 0.0f;
+            gotgains = 1;
+        }
+        else if(strcmp(cmd, "add_row") == 0)
+        {
+            ALuint curidx = 0;
+            char *line;
+            while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL)
+            {
+                ALfloat value;
+                line = read_float(&value, line);
+                if(line && *line != '\0')
+                {
+                    ERR("Extra junk on matrix element %ux%u: %s\n", cur, curidx, line);
+                    return 0;
+                }
+                if(curidx < MAX_AMBI_COEFFS)
+                    matrix[cur][curidx] = value;
+                curidx++;
+            }
+            while(curidx < MAX_AMBI_COEFFS)
+                matrix[cur][curidx++] = 0.0f;
+            cur++;
+        }
+        else
+        {
+            ERR("Unexpected speakers command: %s\n", cmd);
+            return 0;
+        }
+
+        cmd = my_strtok_r(NULL, " \t", saveptr);
+        if(cmd)
+        {
+            ERR("Unexpected junk on line: %s\n", cmd);
+            return 0;
+        }
+    }
+
+    if(!gotgains)
+    {
+        ERR("Matrix order_gain not specified\n");
+        return 0;
+    }
+
+    return 1;
+}
+
+void ambdec_init(AmbDecConf *conf)
+{
+    ALsizei i;
+
+    memset(conf, 0, sizeof(*conf));
+    AL_STRING_INIT(conf->Description);
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+    {
+        AL_STRING_INIT(conf->Speakers[i].Name);
+        AL_STRING_INIT(conf->Speakers[i].Connection);
+    }
+}
+
+void ambdec_deinit(AmbDecConf *conf)
+{
+    ALsizei i;
+
+    alstr_reset(&conf->Description);
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+    {
+        alstr_reset(&conf->Speakers[i].Name);
+        alstr_reset(&conf->Speakers[i].Connection);
+    }
+    memset(conf, 0, sizeof(*conf));
+}
+
+int ambdec_load(AmbDecConf *conf, const char *fname)
+{
+    char *buffer = NULL;
+    size_t maxlen = 0;
+    char *line;
+    FILE *f;
+
+    f = al_fopen(fname, "r");
+    if(!f)
+    {
+        ERR("Failed to open: %s\n", fname);
+        return 0;
+    }
+
+    while((line=read_clipped_line(f, &buffer, &maxlen)) != NULL)
+    {
+        char *saveptr;
+        char *command;
+
+        command = my_strtok_r(line, "/ \t", &saveptr);
+        if(!command)
+        {
+            ERR("Malformed line: %s\n", line);
+            goto fail;
+        }
+
+        if(strcmp(command, "description") == 0)
+        {
+            char *value = my_strtok_r(NULL, "", &saveptr);
+            alstr_copy_cstr(&conf->Description, lstrip(value));
+        }
+        else if(strcmp(command, "version") == 0)
+        {
+            line = my_strtok_r(NULL, "", &saveptr);
+            line = read_uint(&conf->Version, line, 10);
+            if(line && *line != '\0')
+            {
+                ERR("Extra junk after version: %s\n", line);
+                goto fail;
+            }
+            if(conf->Version != 3)
+            {
+                ERR("Unsupported version: %u\n", conf->Version);
+                goto fail;
+            }
+        }
+        else if(strcmp(command, "dec") == 0)
+        {
+            const char *dec = my_strtok_r(NULL, "/ \t", &saveptr);
+            if(strcmp(dec, "chan_mask") == 0)
+            {
+                line = my_strtok_r(NULL, "", &saveptr);
+                line = read_uint(&conf->ChanMask, line, 16);
+                if(line && *line != '\0')
+                {
+                    ERR("Extra junk after mask: %s\n", line);
+                    goto fail;
+                }
+            }
+            else if(strcmp(dec, "freq_bands") == 0)
+            {
+                line = my_strtok_r(NULL, "", &saveptr);
+                line = read_uint(&conf->FreqBands, line, 10);
+                if(line && *line != '\0')
+                {
+                    ERR("Extra junk after freq_bands: %s\n", line);
+                    goto fail;
+                }
+                if(conf->FreqBands != 1 && conf->FreqBands != 2)
+                {
+                    ERR("Invalid freq_bands value: %u\n", conf->FreqBands);
+                    goto fail;
+                }
+            }
+            else if(strcmp(dec, "speakers") == 0)
+            {
+                line = my_strtok_r(NULL, "", &saveptr);
+                line = read_int(&conf->NumSpeakers, line, 10);
+                if(line && *line != '\0')
+                {
+                    ERR("Extra junk after speakers: %s\n", line);
+                    goto fail;
+                }
+                if(conf->NumSpeakers > MAX_OUTPUT_CHANNELS)
+                {
+                    ERR("Unsupported speaker count: %u\n", conf->NumSpeakers);
+                    goto fail;
+                }
+            }
+            else if(strcmp(dec, "coeff_scale") == 0)
+            {
+                line = my_strtok_r(NULL, " \t", &saveptr);
+                if(strcmp(line, "n3d") == 0)
+                    conf->CoeffScale = ADS_N3D;
+                else if(strcmp(line, "sn3d") == 0)
+                    conf->CoeffScale = ADS_SN3D;
+                else if(strcmp(line, "fuma") == 0)
+                    conf->CoeffScale = ADS_FuMa;
+                else
+                {
+                    ERR("Unsupported coeff scale: %s\n", line);
+                    goto fail;
+                }
+            }
+            else
+            {
+                ERR("Unexpected /dec option: %s\n", dec);
+                goto fail;
+            }
+        }
+        else if(strcmp(command, "opt") == 0)
+        {
+            const char *opt = my_strtok_r(NULL, "/ \t", &saveptr);
+            if(strcmp(opt, "xover_freq") == 0)
+            {
+                line = my_strtok_r(NULL, "", &saveptr);
+                line = read_float(&conf->XOverFreq, line);
+                if(line && *line != '\0')
+                {
+                    ERR("Extra junk after xover_freq: %s\n", line);
+                    goto fail;
+                }
+            }
+            else if(strcmp(opt, "xover_ratio") == 0)
+            {
+                line = my_strtok_r(NULL, "", &saveptr);
+                line = read_float(&conf->XOverRatio, line);
+                if(line && *line != '\0')
+                {
+                    ERR("Extra junk after xover_ratio: %s\n", line);
+                    goto fail;
+                }
+            }
+            else if(strcmp(opt, "input_scale") == 0 || strcmp(opt, "nfeff_comp") == 0 ||
+                    strcmp(opt, "delay_comp") == 0 || strcmp(opt, "level_comp") == 0)
+            {
+                /* Unused */
+                my_strtok_r(NULL, " \t", &saveptr);
+            }
+            else
+            {
+                ERR("Unexpected /opt option: %s\n", opt);
+                goto fail;
+            }
+        }
+        else if(strcmp(command, "speakers") == 0)
+        {
+            const char *value = my_strtok_r(NULL, "/ \t", &saveptr);
+            if(strcmp(value, "{") != 0)
+            {
+                ERR("Expected { after %s command, got %s\n", command, value);
+                goto fail;
+            }
+            if(!load_ambdec_speakers(conf, f, &buffer, &maxlen, &saveptr))
+                goto fail;
+            value = my_strtok_r(NULL, "/ \t", &saveptr);
+            if(!value)
+            {
+                line = read_clipped_line(f, &buffer, &maxlen);
+                if(!line)
+                {
+                    ERR("Unexpected end of file\n");
+                    goto fail;
+                }
+                value = my_strtok_r(line, "/ \t", &saveptr);
+            }
+            if(strcmp(value, "}") != 0)
+            {
+                ERR("Expected } after speaker definitions, got %s\n", value);
+                goto fail;
+            }
+        }
+        else if(strcmp(command, "lfmatrix") == 0 || strcmp(command, "hfmatrix") == 0 ||
+                strcmp(command, "matrix") == 0)
+        {
+            const char *value = my_strtok_r(NULL, "/ \t", &saveptr);
+            if(strcmp(value, "{") != 0)
+            {
+                ERR("Expected { after %s command, got %s\n", command, value);
+                goto fail;
+            }
+            if(conf->FreqBands == 1)
+            {
+                if(strcmp(command, "matrix") != 0)
+                {
+                    ERR("Unexpected \"%s\" type for a single-band decoder\n", command);
+                    goto fail;
+                }
+                if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers,
+                                       f, &buffer, &maxlen, &saveptr))
+                    goto fail;
+            }
+            else
+            {
+                if(strcmp(command, "lfmatrix") == 0)
+                {
+                    if(!load_ambdec_matrix(conf->LFOrderGain, conf->LFMatrix, conf->NumSpeakers,
+                                           f, &buffer, &maxlen, &saveptr))
+                        goto fail;
+                }
+                else if(strcmp(command, "hfmatrix") == 0)
+                {
+                    if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers,
+                                           f, &buffer, &maxlen, &saveptr))
+                        goto fail;
+                }
+                else
+                {
+                    ERR("Unexpected \"%s\" type for a dual-band decoder\n", command);
+                    goto fail;
+                }
+            }
+            value = my_strtok_r(NULL, "/ \t", &saveptr);
+            if(!value)
+            {
+                line = read_clipped_line(f, &buffer, &maxlen);
+                if(!line)
+                {
+                    ERR("Unexpected end of file\n");
+                    goto fail;
+                }
+                value = my_strtok_r(line, "/ \t", &saveptr);
+            }
+            if(strcmp(value, "}") != 0)
+            {
+                ERR("Expected } after matrix definitions, got %s\n", value);
+                goto fail;
+            }
+        }
+        else if(strcmp(command, "end") == 0)
+        {
+            line = my_strtok_r(NULL, "/ \t", &saveptr);
+            if(line)
+            {
+                ERR("Unexpected junk on end: %s\n", line);
+                goto fail;
+            }
+
+            fclose(f);
+            free(buffer);
+            return 1;
+        }
+        else
+        {
+            ERR("Unexpected command: %s\n", command);
+            goto fail;
+        }
+
+        line = my_strtok_r(NULL, "/ \t", &saveptr);
+        if(line)
+        {
+            ERR("Unexpected junk on line: %s\n", line);
+            goto fail;
+        }
+    }
+    ERR("Unexpected end of file\n");
+
+fail:
+    fclose(f);
+    free(buffer);
+    return 0;
+}

Alc/ambdec.h

@@ -0,0 +1,46 @@
+#ifndef AMBDEC_H
+#define AMBDEC_H
+
+#include "alstring.h"
+#include "alMain.h"
+
+/* Helpers to read .ambdec configuration files. */
+
+enum AmbDecScaleType {
+    ADS_N3D,
+    ADS_SN3D,
+    ADS_FuMa,
+};
+typedef struct AmbDecConf {
+    al_string Description;
+    ALuint Version; /* Must be 3 */
+
+    ALuint ChanMask;
+    ALuint FreqBands; /* Must be 1 or 2 */
+    ALsizei NumSpeakers;
+    enum AmbDecScaleType CoeffScale;
+
+    ALfloat XOverFreq;
+    ALfloat XOverRatio;
+
+    struct {
+        al_string Name;
+        ALfloat Distance;
+        ALfloat Azimuth;
+        ALfloat Elevation;
+        al_string Connection;
+    } Speakers[MAX_OUTPUT_CHANNELS];
+
+    /* Unused when FreqBands == 1 */
+    ALfloat LFOrderGain[MAX_AMBI_ORDER+1];
+    ALfloat LFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
+
+    ALfloat HFOrderGain[MAX_AMBI_ORDER+1];
+    ALfloat HFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
+} AmbDecConf;
+
+void ambdec_init(AmbDecConf *conf);
+void ambdec_deinit(AmbDecConf *conf);
+int ambdec_load(AmbDecConf *conf, const char *fname);
+
+#endif /* AMBDEC_H */

Alc/backends/base.c

@@ -0,0 +1,235 @@
+
+#include "config.h"
+
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alu.h"
+
+#include "backends/base.h"
+
+
+extern inline ALuint64 GetDeviceClockTime(ALCdevice *device);
+
+/* Base ALCbackend method implementations. */
+void ALCbackend_Construct(ALCbackend *self, ALCdevice *device)
+{
+    int ret = almtx_init(&self->mMutex, almtx_recursive);
+    assert(ret == althrd_success);
+    self->mDevice = device;
+}
+
+void ALCbackend_Destruct(ALCbackend *self)
+{
+    almtx_destroy(&self->mMutex);
+}
+
+ALCboolean ALCbackend_reset(ALCbackend* UNUSED(self))
+{
+    return ALC_FALSE;
+}
+
+ALCenum ALCbackend_captureSamples(ALCbackend* UNUSED(self), void* UNUSED(buffer), ALCuint UNUSED(samples))
+{
+    return ALC_INVALID_DEVICE;
+}
+
+ALCuint ALCbackend_availableSamples(ALCbackend* UNUSED(self))
+{
+    return 0;
+}
+
+ClockLatency ALCbackend_getClockLatency(ALCbackend *self)
+{
+    ALCdevice *device = self->mDevice;
+    ALuint refcount;
+    ClockLatency ret;
+
+    do {
+        while(((refcount=ATOMIC_LOAD(&device->MixCount, almemory_order_acquire))&1))
+            althrd_yield();
+        ret.ClockTime = GetDeviceClockTime(device);
+        ATOMIC_THREAD_FENCE(almemory_order_acquire);
+    } while(refcount != ATOMIC_LOAD(&device->MixCount, almemory_order_relaxed));
+
+    /* NOTE: The device will generally have about all but one periods filled at
+     * any given time during playback. Without a more accurate measurement from
+     * the output, this is an okay approximation.
+     */
+    ret.Latency = device->UpdateSize * DEVICE_CLOCK_RES / device->Frequency *
+                  maxu(device->NumUpdates-1, 1);
+
+    return ret;
+}
+
+void ALCbackend_lock(ALCbackend *self)
+{
+    int ret = almtx_lock(&self->mMutex);
+    assert(ret == althrd_success);
+}
+
+void ALCbackend_unlock(ALCbackend *self)
+{
+    int ret = almtx_unlock(&self->mMutex);
+    assert(ret == althrd_success);
+}
+
+
+/* Base ALCbackendFactory method implementations. */
+void ALCbackendFactory_deinit(ALCbackendFactory* UNUSED(self))
+{
+}
+
+
+/* Wrappers to use an old-style backend with the new interface. */
+typedef struct PlaybackWrapper {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    const BackendFuncs *Funcs;
+} PlaybackWrapper;
+
+static void PlaybackWrapper_Construct(PlaybackWrapper *self, ALCdevice *device, const BackendFuncs *funcs);
+static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, void, Destruct)
+static ALCenum PlaybackWrapper_open(PlaybackWrapper *self, const ALCchar *name);
+static void PlaybackWrapper_close(PlaybackWrapper *self);
+static ALCboolean PlaybackWrapper_reset(PlaybackWrapper *self);
+static ALCboolean PlaybackWrapper_start(PlaybackWrapper *self);
+static void PlaybackWrapper_stop(PlaybackWrapper *self);
+static DECLARE_FORWARD2(PlaybackWrapper, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, void, lock)
+static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(PlaybackWrapper)
+DEFINE_ALCBACKEND_VTABLE(PlaybackWrapper);
+
+static void PlaybackWrapper_Construct(PlaybackWrapper *self, ALCdevice *device, const BackendFuncs *funcs)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(PlaybackWrapper, ALCbackend, self);
+
+    self->Funcs = funcs;
+}
+
+static ALCenum PlaybackWrapper_open(PlaybackWrapper *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    return self->Funcs->OpenPlayback(device, name);
+}
+
+static void PlaybackWrapper_close(PlaybackWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    self->Funcs->ClosePlayback(device);
+}
+
+static ALCboolean PlaybackWrapper_reset(PlaybackWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    return self->Funcs->ResetPlayback(device);
+}
+
+static ALCboolean PlaybackWrapper_start(PlaybackWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    return self->Funcs->StartPlayback(device);
+}
+
+static void PlaybackWrapper_stop(PlaybackWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    self->Funcs->StopPlayback(device);
+}
+
+
+typedef struct CaptureWrapper {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    const BackendFuncs *Funcs;
+} CaptureWrapper;
+
+static void CaptureWrapper_Construct(CaptureWrapper *self, ALCdevice *device, const BackendFuncs *funcs);
+static DECLARE_FORWARD(CaptureWrapper, ALCbackend, void, Destruct)
+static ALCenum CaptureWrapper_open(CaptureWrapper *self, const ALCchar *name);
+static void CaptureWrapper_close(CaptureWrapper *self);
+static DECLARE_FORWARD(CaptureWrapper, ALCbackend, ALCboolean, reset)
+static ALCboolean CaptureWrapper_start(CaptureWrapper *self);
+static void CaptureWrapper_stop(CaptureWrapper *self);
+static ALCenum CaptureWrapper_captureSamples(CaptureWrapper *self, void *buffer, ALCuint samples);
+static ALCuint CaptureWrapper_availableSamples(CaptureWrapper *self);
+static DECLARE_FORWARD(CaptureWrapper, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(CaptureWrapper, ALCbackend, void, lock)
+static DECLARE_FORWARD(CaptureWrapper, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(CaptureWrapper)
+DEFINE_ALCBACKEND_VTABLE(CaptureWrapper);
+
+static void CaptureWrapper_Construct(CaptureWrapper *self, ALCdevice *device, const BackendFuncs *funcs)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(CaptureWrapper, ALCbackend, self);
+
+    self->Funcs = funcs;
+}
+
+static ALCenum CaptureWrapper_open(CaptureWrapper *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    return self->Funcs->OpenCapture(device, name);
+}
+
+static void CaptureWrapper_close(CaptureWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    self->Funcs->CloseCapture(device);
+}
+
+static ALCboolean CaptureWrapper_start(CaptureWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    self->Funcs->StartCapture(device);
+    return ALC_TRUE;
+}
+
+static void CaptureWrapper_stop(CaptureWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    self->Funcs->StopCapture(device);
+}
+
+static ALCenum CaptureWrapper_captureSamples(CaptureWrapper *self, void *buffer, ALCuint samples)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    return self->Funcs->CaptureSamples(device, buffer, samples);
+}
+
+static ALCuint CaptureWrapper_availableSamples(CaptureWrapper *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    return self->Funcs->AvailableSamples(device);
+}
+
+
+ALCbackend *create_backend_wrapper(ALCdevice *device, const BackendFuncs *funcs, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        PlaybackWrapper *backend;
+
+        NEW_OBJ(backend, PlaybackWrapper)(device, funcs);
+        if(!backend) return NULL;
+
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    if(type == ALCbackend_Capture)
+    {
+        CaptureWrapper *backend;
+
+        NEW_OBJ(backend, CaptureWrapper)(device, funcs);
+        if(!backend) return NULL;
+
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/base.h

@@ -0,0 +1,156 @@
+#ifndef AL_BACKENDS_BASE_H
+#define AL_BACKENDS_BASE_H
+
+#include "alMain.h"
+#include "threads.h"
+
+
+typedef struct ClockLatency {
+    ALint64 ClockTime;
+    ALint64 Latency;
+} ClockLatency;
+
+/* Helper to get the current clock time from the device's ClockBase, and
+ * SamplesDone converted from the sample rate.
+ */
+inline ALuint64 GetDeviceClockTime(ALCdevice *device)
+{
+    return device->ClockBase + (device->SamplesDone * DEVICE_CLOCK_RES /
+                                device->Frequency);
+}
+
+
+struct ALCbackendVtable;
+
+typedef struct ALCbackend {
+    const struct ALCbackendVtable *vtbl;
+
+    ALCdevice *mDevice;
+
+    almtx_t mMutex;
+} ALCbackend;
+
+void ALCbackend_Construct(ALCbackend *self, ALCdevice *device);
+void ALCbackend_Destruct(ALCbackend *self);
+ALCboolean ALCbackend_reset(ALCbackend *self);
+ALCenum ALCbackend_captureSamples(ALCbackend *self, void *buffer, ALCuint samples);
+ALCuint ALCbackend_availableSamples(ALCbackend *self);
+ClockLatency ALCbackend_getClockLatency(ALCbackend *self);
+void ALCbackend_lock(ALCbackend *self);
+void ALCbackend_unlock(ALCbackend *self);
+
+struct ALCbackendVtable {
+    void (*const Destruct)(ALCbackend*);
+
+    ALCenum (*const open)(ALCbackend*, const ALCchar*);
+    void (*const close)(ALCbackend*);
+
+    ALCboolean (*const reset)(ALCbackend*);
+    ALCboolean (*const start)(ALCbackend*);
+    void (*const stop)(ALCbackend*);
+
+    ALCenum (*const captureSamples)(ALCbackend*, void*, ALCuint);
+    ALCuint (*const availableSamples)(ALCbackend*);
+
+    ClockLatency (*const getClockLatency)(ALCbackend*);
+
+    void (*const lock)(ALCbackend*);
+    void (*const unlock)(ALCbackend*);
+
+    void (*const Delete)(void*);
+};
+
+#define DEFINE_ALCBACKEND_VTABLE(T)                                           \
+DECLARE_THUNK(T, ALCbackend, void, Destruct)                                  \
+DECLARE_THUNK1(T, ALCbackend, ALCenum, open, const ALCchar*)                  \
+DECLARE_THUNK(T, ALCbackend, void, close)                                     \
+DECLARE_THUNK(T, ALCbackend, ALCboolean, reset)                               \
+DECLARE_THUNK(T, ALCbackend, ALCboolean, start)                               \
+DECLARE_THUNK(T, ALCbackend, void, stop)                                      \
+DECLARE_THUNK2(T, ALCbackend, ALCenum, captureSamples, void*, ALCuint)        \
+DECLARE_THUNK(T, ALCbackend, ALCuint, availableSamples)                       \
+DECLARE_THUNK(T, ALCbackend, ClockLatency, getClockLatency)                   \
+DECLARE_THUNK(T, ALCbackend, void, lock)                                      \
+DECLARE_THUNK(T, ALCbackend, void, unlock)                                    \
+static void T##_ALCbackend_Delete(void *ptr)                                  \
+{ T##_Delete(STATIC_UPCAST(T, ALCbackend, (ALCbackend*)ptr)); }               \
+                                                                              \
+static const struct ALCbackendVtable T##_ALCbackend_vtable = {                \
+    T##_ALCbackend_Destruct,                                                  \
+                                                                              \
+    T##_ALCbackend_open,                                                      \
+    T##_ALCbackend_close,                                                     \
+    T##_ALCbackend_reset,                                                     \
+    T##_ALCbackend_start,                                                     \
+    T##_ALCbackend_stop,                                                      \
+    T##_ALCbackend_captureSamples,                                            \
+    T##_ALCbackend_availableSamples,                                          \
+    T##_ALCbackend_getClockLatency,                                           \
+    T##_ALCbackend_lock,                                                      \
+    T##_ALCbackend_unlock,                                                    \
+                                                                              \
+    T##_ALCbackend_Delete,                                                    \
+}
+
+
+typedef enum ALCbackend_Type {
+    ALCbackend_Playback,
+    ALCbackend_Capture,
+    ALCbackend_Loopback
+} ALCbackend_Type;
+
+
+struct ALCbackendFactoryVtable;
+
+typedef struct ALCbackendFactory {
+    const struct ALCbackendFactoryVtable *vtbl;
+} ALCbackendFactory;
+
+void ALCbackendFactory_deinit(ALCbackendFactory *self);
+
+struct ALCbackendFactoryVtable {
+    ALCboolean (*const init)(ALCbackendFactory *self);
+    void (*const deinit)(ALCbackendFactory *self);
+
+    ALCboolean (*const querySupport)(ALCbackendFactory *self, ALCbackend_Type type);
+
+    void (*const probe)(ALCbackendFactory *self, enum DevProbe type);
+
+    ALCbackend* (*const createBackend)(ALCbackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+};
+
+#define DEFINE_ALCBACKENDFACTORY_VTABLE(T)                                    \
+DECLARE_THUNK(T, ALCbackendFactory, ALCboolean, init)                         \
+DECLARE_THUNK(T, ALCbackendFactory, void, deinit)                             \
+DECLARE_THUNK1(T, ALCbackendFactory, ALCboolean, querySupport, ALCbackend_Type) \
+DECLARE_THUNK1(T, ALCbackendFactory, void, probe, enum DevProbe)              \
+DECLARE_THUNK2(T, ALCbackendFactory, ALCbackend*, createBackend, ALCdevice*, ALCbackend_Type) \
+                                                                              \
+static const struct ALCbackendFactoryVtable T##_ALCbackendFactory_vtable = {  \
+    T##_ALCbackendFactory_init,                                               \
+    T##_ALCbackendFactory_deinit,                                             \
+    T##_ALCbackendFactory_querySupport,                                       \
+    T##_ALCbackendFactory_probe,                                              \
+    T##_ALCbackendFactory_createBackend,                                      \
+}
+
+
+ALCbackendFactory *ALCpulseBackendFactory_getFactory(void);
+ALCbackendFactory *ALCalsaBackendFactory_getFactory(void);
+ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
+ALCbackendFactory *ALCossBackendFactory_getFactory(void);
+ALCbackendFactory *ALCjackBackendFactory_getFactory(void);
+ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void);
+ALCbackendFactory *ALCsndioBackendFactory_getFactory(void);
+ALCbackendFactory *ALCmmdevBackendFactory_getFactory(void);
+ALCbackendFactory *ALCdsoundBackendFactory_getFactory(void);
+ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void);
+ALCbackendFactory *ALCportBackendFactory_getFactory(void);
+ALCbackendFactory *ALCopenslBackendFactory_getFactory(void);
+ALCbackendFactory *ALCnullBackendFactory_getFactory(void);
+ALCbackendFactory *ALCwaveBackendFactory_getFactory(void);
+ALCbackendFactory *ALCloopbackFactory_getFactory(void);
+
+ALCbackend *create_backend_wrapper(ALCdevice *device, const BackendFuncs *funcs, ALCbackend_Type type);
+
+#endif /* AL_BACKENDS_BASE_H */

Alc/backends/coreaudio.c

@@ -0,0 +1,828 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <alloca.h>
+
+#include "alMain.h"
+#include "alu.h"
+
+#include <CoreServices/CoreServices.h>
+#include <unistd.h>
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+#include "backends/base.h"
+
+
+typedef struct {
+    AudioUnit audioUnit;
+
+    ALuint frameSize;
+    ALdouble sampleRateRatio;              // Ratio of hardware sample rate / requested sample rate
+    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
+
+    AudioConverterRef audioConverter;      // Sample rate converter if needed
+    AudioBufferList *bufferList;           // Buffer for data coming from the input device
+    ALCvoid *resampleBuffer;               // Buffer for returned RingBuffer data when resampling
+
+    ll_ringbuffer_t *ring;
+} ca_data;
+
+static const ALCchar ca_device[] = "CoreAudio Default";
+
+
+static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
+{
+    AudioBufferList *list;
+
+    list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
+    if(list)
+    {
+        list->mNumberBuffers = 1;
+
+        list->mBuffers[0].mNumberChannels = channelCount;
+        list->mBuffers[0].mDataByteSize = byteSize;
+        list->mBuffers[0].mData = malloc(byteSize);
+        if(list->mBuffers[0].mData == NULL)
+        {
+            free(list);
+            list = NULL;
+        }
+    }
+    return list;
+}
+
+static void destroy_buffer_list(AudioBufferList* list)
+{
+    if(list)
+    {
+        UInt32 i;
+        for(i = 0;i < list->mNumberBuffers;i++)
+            free(list->mBuffers[i].mData);
+        free(list);
+    }
+}
+
+
+typedef struct ALCcoreAudioPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    AudioUnit audioUnit;
+
+    ALuint frameSize;
+    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
+} ALCcoreAudioPlayback;
+
+static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device);
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self);
+static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name);
+static void ALCcoreAudioPlayback_close(ALCcoreAudioPlayback *self);
+static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self);
+static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self);
+static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self);
+static DECLARE_FORWARD2(ALCcoreAudioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioPlayback)
+
+DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioPlayback);
+
+
+static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self);
+
+    self->frameSize = 0;
+    memset(&self->format, 0, sizeof(self->format));
+}
+
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
+{
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon,
+  AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp),
+  UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData)
+{
+    ALCcoreAudioPlayback *self = inRefCon;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+
+    ALCdevice_Lock(device);
+    aluMixData(device, ioData->mBuffers[0].mData,
+               ioData->mBuffers[0].mDataByteSize / self->frameSize);
+    ALCdevice_Unlock(device);
+
+    return noErr;
+}
+
+
+static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    AudioComponentDescription desc;
+    AudioComponent comp;
+    OSStatus err;
+
+    if(!name)
+        name = ca_device;
+    else if(strcmp(name, ca_device) != 0)
+        return ALC_INVALID_VALUE;
+
+    /* open the default output unit */
+    desc.componentType = kAudioUnitType_Output;
+    desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+    desc.componentFlags = 0;
+    desc.componentFlagsMask = 0;
+
+    comp = AudioComponentFindNext(NULL, &desc);
+    if(comp == NULL)
+    {
+        ERR("AudioComponentFindNext failed\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioComponentInstanceNew failed\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    /* init and start the default audio unit... */
+    err = AudioUnitInitialize(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioUnitInitialize failed\n");
+        AudioComponentInstanceDispose(self->audioUnit);
+        return ALC_INVALID_VALUE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, name);
+    return ALC_NO_ERROR;
+}
+
+static void ALCcoreAudioPlayback_close(ALCcoreAudioPlayback *self)
+{
+    AudioUnitUninitialize(self->audioUnit);
+    AudioComponentInstanceDispose(self->audioUnit);
+}
+
+static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    AudioStreamBasicDescription streamFormat;
+    AURenderCallbackStruct input;
+    OSStatus err;
+    UInt32 size;
+
+    err = AudioUnitUninitialize(self->audioUnit);
+    if(err != noErr)
+        ERR("-- AudioUnitUninitialize failed.\n");
+
+    /* retrieve default output unit's properties (output side) */
+    size = sizeof(AudioStreamBasicDescription);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
+    if(err != noErr || size != sizeof(AudioStreamBasicDescription))
+    {
+        ERR("AudioUnitGetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+#if 0
+    TRACE("Output streamFormat of default output unit -\n");
+    TRACE("  streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket);
+    TRACE("  streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame);
+    TRACE("  streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel);
+    TRACE("  streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket);
+    TRACE("  streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame);
+    TRACE("  streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate);
+#endif
+
+    /* set default output unit's input side to match output side */
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+    if(device->Frequency != streamFormat.mSampleRate)
+    {
+        device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates *
+                                      streamFormat.mSampleRate /
+                                      device->Frequency);
+        device->Frequency = streamFormat.mSampleRate;
+    }
+
+    /* FIXME: How to tell what channels are what in the output device, and how
+     * to specify what we're giving?  eg, 6.0 vs 5.1 */
+    switch(streamFormat.mChannelsPerFrame)
+    {
+        case 1:
+            device->FmtChans = DevFmtMono;
+            break;
+        case 2:
+            device->FmtChans = DevFmtStereo;
+            break;
+        case 4:
+            device->FmtChans = DevFmtQuad;
+            break;
+        case 6:
+            device->FmtChans = DevFmtX51;
+            break;
+        case 7:
+            device->FmtChans = DevFmtX61;
+            break;
+        case 8:
+            device->FmtChans = DevFmtX71;
+            break;
+        default:
+            ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
+            device->FmtChans = DevFmtStereo;
+            streamFormat.mChannelsPerFrame = 2;
+            break;
+    }
+    SetDefaultWFXChannelOrder(device);
+
+    /* use channel count and sample rate from the default output unit's current
+     * parameters, but reset everything else */
+    streamFormat.mFramesPerPacket = 1;
+    streamFormat.mFormatFlags = 0;
+    switch(device->FmtType)
+    {
+        case DevFmtUByte:
+            device->FmtType = DevFmtByte;
+            /* fall-through */
+        case DevFmtByte:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 8;
+            break;
+        case DevFmtUShort:
+            device->FmtType = DevFmtShort;
+            /* fall-through */
+        case DevFmtShort:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 16;
+            break;
+        case DevFmtUInt:
+            device->FmtType = DevFmtInt;
+            /* fall-through */
+        case DevFmtInt:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 32;
+            break;
+        case DevFmtFloat:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat;
+            streamFormat.mBitsPerChannel = 32;
+            break;
+    }
+    streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame *
+                                  streamFormat.mBitsPerChannel / 8;
+    streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame;
+    streamFormat.mFormatID = kAudioFormatLinearPCM;
+    streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
+                                 kLinearPCMFormatFlagIsPacked;
+
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+    /* setup callback */
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    input.inputProc = ALCcoreAudioPlayback_MixerProc;
+    input.inputProcRefCon = self;
+
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+    /* init the default audio unit... */
+    err = AudioUnitInitialize(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioUnitInitialize failed\n");
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self)
+{
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioOutputUnitStart failed\n");
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self)
+{
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
+    if(err != noErr)
+        ERR("AudioOutputUnitStop failed\n");
+}
+
+
+
+
+typedef struct ALCcoreAudioCapture {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    AudioUnit audioUnit;
+
+    ALuint frameSize;
+    ALdouble sampleRateRatio;              // Ratio of hardware sample rate / requested sample rate
+    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
+
+    AudioConverterRef audioConverter;      // Sample rate converter if needed
+    AudioBufferList *bufferList;           // Buffer for data coming from the input device
+    ALCvoid *resampleBuffer;               // Buffer for returned RingBuffer data when resampling
+
+    ll_ringbuffer_t *ring;
+} ALCcoreAudioCapture;
+
+static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device);
+static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self);
+static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name);
+static void ALCcoreAudioCapture_close(ALCcoreAudioCapture *self);
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ALCboolean, reset)
+static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self);
+static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self);
+static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples);
+static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self);
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioCapture)
+
+DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioCapture);
+
+
+static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self);
+
+}
+
+static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self)
+{
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon,
+  AudioUnitRenderActionFlags* UNUSED(ioActionFlags),
+  const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber),
+  UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData))
+{
+    ALCcoreAudioCapture *self = inRefCon;
+    AudioUnitRenderActionFlags flags = 0;
+    OSStatus err;
+
+    // fill the bufferList with data from the input device
+    err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList);
+    if(err != noErr)
+    {
+        ERR("AudioUnitRender error: %d\n", err);
+        return err;
+    }
+
+    ll_ringbuffer_write(self->ring, self->bufferList->mBuffers[0].mData, inNumberFrames);
+
+    return noErr;
+}
+
+static OSStatus ALCcoreAudioCapture_ConvertCallback(AudioConverterRef UNUSED(inAudioConverter),
+  UInt32 *ioNumberDataPackets, AudioBufferList *ioData,
+  AudioStreamPacketDescription** UNUSED(outDataPacketDescription),
+  void *inUserData)
+{
+    ALCcoreAudioCapture *self = inUserData;
+
+    // Read from the ring buffer and store temporarily in a large buffer
+    ll_ringbuffer_read(self->ring, self->resampleBuffer, *ioNumberDataPackets);
+
+    // Set the input data
+    ioData->mNumberBuffers = 1;
+    ioData->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
+    ioData->mBuffers[0].mData = self->resampleBuffer;
+    ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * self->format.mBytesPerFrame;
+
+    return noErr;
+}
+
+
+static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    AudioStreamBasicDescription requestedFormat;  // The application requested format
+    AudioStreamBasicDescription hardwareFormat;   // The hardware format
+    AudioStreamBasicDescription outputFormat;     // The AudioUnit output format
+    AURenderCallbackStruct input;
+    AudioComponentDescription desc;
+    AudioDeviceID inputDevice;
+    UInt32 outputFrameCount;
+    UInt32 propertySize;
+    AudioObjectPropertyAddress propertyAddress;
+    UInt32 enableIO;
+    AudioComponent comp;
+    OSStatus err;
+
+    if(!name)
+        name = ca_device;
+    else if(strcmp(name, ca_device) != 0)
+        return ALC_INVALID_VALUE;
+
+    desc.componentType = kAudioUnitType_Output;
+    desc.componentSubType = kAudioUnitSubType_HALOutput;
+    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+    desc.componentFlags = 0;
+    desc.componentFlagsMask = 0;
+
+    // Search for component with given description
+    comp = AudioComponentFindNext(NULL, &desc);
+    if(comp == NULL)
+    {
+        ERR("AudioComponentFindNext failed\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    // Open the component
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioComponentInstanceNew failed\n");
+        goto error;
+    }
+
+    // Turn off AudioUnit output
+    enableIO = 0;
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // Turn on AudioUnit input
+    enableIO = 1;
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // Get the default input device
+
+    propertySize = sizeof(AudioDeviceID);
+    propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+    propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
+    propertyAddress.mElement = kAudioObjectPropertyElementMaster;
+
+    err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice);
+    if(err != noErr)
+    {
+        ERR("AudioObjectGetPropertyData failed\n");
+        goto error;
+    }
+
+    if(inputDevice == kAudioDeviceUnknown)
+    {
+        ERR("No input device found\n");
+        goto error;
+    }
+
+    // Track the input device
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // set capture callback
+    input.inputProc = ALCcoreAudioCapture_RecordProc;
+    input.inputProcRefCon = self;
+
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // Initialize the device
+    err = AudioUnitInitialize(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioUnitInitialize failed\n");
+        goto error;
+    }
+
+    // Get the hardware format
+    propertySize = sizeof(AudioStreamBasicDescription);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
+    if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
+    {
+        ERR("AudioUnitGetProperty failed\n");
+        goto error;
+    }
+
+    // Set up the requested format description
+    switch(device->FmtType)
+    {
+        case DevFmtUByte:
+            requestedFormat.mBitsPerChannel = 8;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtShort:
+            requestedFormat.mBitsPerChannel = 16;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtInt:
+            requestedFormat.mBitsPerChannel = 32;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtFloat:
+            requestedFormat.mBitsPerChannel = 32;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtByte:
+        case DevFmtUShort:
+        case DevFmtUInt:
+            ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
+            goto error;
+    }
+
+    switch(device->FmtChans)
+    {
+        case DevFmtMono:
+            requestedFormat.mChannelsPerFrame = 1;
+            break;
+        case DevFmtStereo:
+            requestedFormat.mChannelsPerFrame = 2;
+            break;
+
+        case DevFmtQuad:
+        case DevFmtX51:
+        case DevFmtX51Rear:
+        case DevFmtX61:
+        case DevFmtX71:
+        case DevFmtAmbi3D:
+            ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
+            goto error;
+    }
+
+    requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
+    requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
+    requestedFormat.mSampleRate = device->Frequency;
+    requestedFormat.mFormatID = kAudioFormatLinearPCM;
+    requestedFormat.mReserved = 0;
+    requestedFormat.mFramesPerPacket = 1;
+
+    // save requested format description for later use
+    self->format = requestedFormat;
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    // Use intermediate format for sample rate conversion (outputFormat)
+    // Set sample rate to the same as hardware for resampling later
+    outputFormat = requestedFormat;
+    outputFormat.mSampleRate = hardwareFormat.mSampleRate;
+
+    // Determine sample rate ratio for resampling
+    self->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
+
+    // The output format should be the requested format, but using the hardware sample rate
+    // This is because the AudioUnit will automatically scale other properties, except for sample rate
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // Set the AudioUnit output format frame count
+    outputFrameCount = device->UpdateSize * self->sampleRateRatio;
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed: %d\n", err);
+        goto error;
+    }
+
+    // Set up sample converter
+    err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter);
+    if(err != noErr)
+    {
+        ERR("AudioConverterNew failed: %d\n", err);
+        goto error;
+    }
+
+    // Create a buffer for use in the resample callback
+    self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio);
+
+    // Allocate buffer for the AudioUnit output
+    self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio);
+    if(self->bufferList == NULL)
+        goto error;
+
+    self->ring = ll_ringbuffer_create(
+        device->UpdateSize*self->sampleRateRatio*device->NumUpdates + 1,
+        self->frameSize
+    );
+    if(!self->ring) goto error;
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+
+error:
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+    free(self->resampleBuffer);
+    destroy_buffer_list(self->bufferList);
+
+    if(self->audioConverter)
+        AudioConverterDispose(self->audioConverter);
+    if(self->audioUnit)
+        AudioComponentInstanceDispose(self->audioUnit);
+
+    return ALC_INVALID_VALUE;
+}
+
+
+static void ALCcoreAudioCapture_close(ALCcoreAudioCapture *self)
+{
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+
+    free(self->resampleBuffer);
+
+    destroy_buffer_list(self->bufferList);
+
+    AudioConverterDispose(self->audioConverter);
+    AudioComponentInstanceDispose(self->audioUnit);
+}
+
+static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
+{
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioOutputUnitStart failed\n");
+        return ALC_FALSE;
+    }
+    return ALC_TRUE;
+}
+
+static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
+{
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
+    if(err != noErr)
+        ERR("AudioOutputUnitStop failed\n");
+}
+
+static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
+{
+    AudioBufferList *list;
+    UInt32 frameCount;
+    OSStatus err;
+
+    // If no samples are requested, just return
+    if(samples == 0)
+        return ALC_NO_ERROR;
+
+    // Allocate a temporary AudioBufferList to use as the return resamples data
+    list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
+
+    // Point the resampling buffer to the capture buffer
+    list->mNumberBuffers = 1;
+    list->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
+    list->mBuffers[0].mDataByteSize = samples * self->frameSize;
+    list->mBuffers[0].mData = buffer;
+
+    // Resample into another AudioBufferList
+    frameCount = samples;
+    err = AudioConverterFillComplexBuffer(self->audioConverter,
+        ALCcoreAudioCapture_ConvertCallback, self, &frameCount, list, NULL
+    );
+    if(err != noErr)
+    {
+        ERR("AudioConverterFillComplexBuffer error: %d\n", err);
+        return ALC_INVALID_VALUE;
+    }
+    return ALC_NO_ERROR;
+}
+
+static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self)
+{
+    return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio;
+}
+
+
+typedef struct ALCcoreAudioBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCcoreAudioBackendFactory;
+#define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
+
+static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory *self);
+static DECLARE_FORWARD(ALCcoreAudioBackendFactory, ALCbackendFactory, void, deinit)
+static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory *self, ALCbackend_Type type);
+static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCcoreAudioBackendFactory);
+
+
+ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void)
+{
+    static ALCcoreAudioBackendFactory factory = ALCCOREAUDIOBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory* UNUSED(self))
+{
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback || ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            AppendAllDevicesList(ca_device);
+            break;
+        case CAPTURE_DEVICE_PROBE:
+            AppendCaptureDeviceList(ca_device);
+            break;
+    }
+}
+
+static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCcoreAudioPlayback *backend;
+        NEW_OBJ(backend, ALCcoreAudioPlayback)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+    if(type == ALCbackend_Capture)
+    {
+        ALCcoreAudioCapture *backend;
+        NEW_OBJ(backend, ALCcoreAudioCapture)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/jack.c

@@ -0,0 +1,632 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <memory.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+#include "compat.h"
+
+#include "backends/base.h"
+
+#include <jack/jack.h>
+#include <jack/ringbuffer.h>
+
+
+static const ALCchar jackDevice[] = "JACK Default";
+
+
+#ifdef HAVE_DYNLOAD
+#define JACK_FUNCS(MAGIC)          \
+    MAGIC(jack_client_open);       \
+    MAGIC(jack_client_close);      \
+    MAGIC(jack_client_name_size);  \
+    MAGIC(jack_get_client_name);   \
+    MAGIC(jack_connect);           \
+    MAGIC(jack_activate);          \
+    MAGIC(jack_deactivate);        \
+    MAGIC(jack_port_register);     \
+    MAGIC(jack_port_unregister);   \
+    MAGIC(jack_port_get_buffer);   \
+    MAGIC(jack_port_name);         \
+    MAGIC(jack_get_ports);         \
+    MAGIC(jack_free);              \
+    MAGIC(jack_get_sample_rate);   \
+    MAGIC(jack_set_error_function); \
+    MAGIC(jack_set_process_callback); \
+    MAGIC(jack_set_buffer_size_callback); \
+    MAGIC(jack_set_buffer_size);   \
+    MAGIC(jack_get_buffer_size);
+
+static void *jack_handle;
+#define MAKE_FUNC(f) static __typeof(f) * p##f
+JACK_FUNCS(MAKE_FUNC);
+#undef MAKE_FUNC
+
+#define jack_client_open pjack_client_open
+#define jack_client_close pjack_client_close
+#define jack_client_name_size pjack_client_name_size
+#define jack_get_client_name pjack_get_client_name
+#define jack_connect pjack_connect
+#define jack_activate pjack_activate
+#define jack_deactivate pjack_deactivate
+#define jack_port_register pjack_port_register
+#define jack_port_unregister pjack_port_unregister
+#define jack_port_get_buffer pjack_port_get_buffer
+#define jack_port_name pjack_port_name
+#define jack_get_ports pjack_get_ports
+#define jack_free pjack_free
+#define jack_get_sample_rate pjack_get_sample_rate
+#define jack_set_error_function pjack_set_error_function
+#define jack_set_process_callback pjack_set_process_callback
+#define jack_set_buffer_size_callback pjack_set_buffer_size_callback
+#define jack_set_buffer_size pjack_set_buffer_size
+#define jack_get_buffer_size pjack_get_buffer_size
+#endif
+
+
+static jack_options_t ClientOptions = JackNullOption;
+
+static ALCboolean jack_load(void)
+{
+    ALCboolean error = ALC_FALSE;
+
+#ifdef HAVE_DYNLOAD
+    if(!jack_handle)
+    {
+        al_string missing_funcs = AL_STRING_INIT_STATIC();
+
+#ifdef _WIN32
+#define JACKLIB "libjack.dll"
+#else
+#define JACKLIB "libjack.so.0"
+#endif
+        jack_handle = LoadLib(JACKLIB);
+        if(!jack_handle)
+        {
+            WARN("Failed to load %s\n", JACKLIB);
+            return ALC_FALSE;
+        }
+
+        error = ALC_FALSE;
+#define LOAD_FUNC(f) do {                                                     \
+    p##f = GetSymbol(jack_handle, #f);                                        \
+    if(p##f == NULL) {                                                        \
+        error = ALC_TRUE;                                                     \
+        alstr_append_cstr(&missing_funcs, "\n" #f);                           \
+    }                                                                         \
+} while(0)
+        JACK_FUNCS(LOAD_FUNC);
+#undef LOAD_FUNC
+
+        if(error)
+        {
+            WARN("Missing expected functions:%s\n", alstr_get_cstr(missing_funcs));
+            CloseLib(jack_handle);
+            jack_handle = NULL;
+        }
+        alstr_reset(&missing_funcs);
+    }
+#endif
+
+    return !error;
+}
+
+
+typedef struct ALCjackPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    jack_client_t *Client;
+    jack_port_t *Port[MAX_OUTPUT_CHANNELS];
+
+    ll_ringbuffer_t *Ring;
+    alcnd_t Cond;
+
+    volatile int killNow;
+    althrd_t thread;
+} ALCjackPlayback;
+
+static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg);
+
+static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg);
+static int ALCjackPlayback_mixerProc(void *arg);
+
+static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device);
+static void ALCjackPlayback_Destruct(ALCjackPlayback *self);
+static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name);
+static void ALCjackPlayback_close(ALCjackPlayback *self);
+static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self);
+static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self);
+static void ALCjackPlayback_stop(ALCjackPlayback *self);
+static DECLARE_FORWARD2(ALCjackPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, ALCuint, availableSamples)
+static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self);
+static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCjackPlayback)
+
+DEFINE_ALCBACKEND_VTABLE(ALCjackPlayback);
+
+
+static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device)
+{
+    ALuint i;
+
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCjackPlayback, ALCbackend, self);
+
+    alcnd_init(&self->Cond);
+
+    self->Client = NULL;
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        self->Port[i] = NULL;
+    self->Ring = NULL;
+
+    self->killNow = 1;
+}
+
+static void ALCjackPlayback_Destruct(ALCjackPlayback *self)
+{
+    ALuint i;
+
+    if(self->Client)
+    {
+        for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        {
+            if(self->Port[i])
+                jack_port_unregister(self->Client, self->Port[i]);
+            self->Port[i] = NULL;
+        }
+        jack_client_close(self->Client);
+        self->Client = NULL;
+    }
+
+    alcnd_destroy(&self->Cond);
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg)
+{
+    ALCjackPlayback *self = arg;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    ALuint bufsize;
+
+    ALCjackPlayback_lock(self);
+    device->UpdateSize = numframes;
+    device->NumUpdates = 2;
+
+    bufsize = device->UpdateSize;
+    if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize))
+        bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize);
+    bufsize += device->UpdateSize;
+    device->NumUpdates = bufsize / device->UpdateSize;
+
+    TRACE("%u update size x%u\n", device->UpdateSize, device->NumUpdates);
+
+    ll_ringbuffer_free(self->Ring);
+    self->Ring = ll_ringbuffer_create(bufsize,
+        FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder)
+    );
+    if(!self->Ring)
+    {
+        ERR("Failed to reallocate ringbuffer\n");
+        aluHandleDisconnect(device);
+    }
+    ALCjackPlayback_unlock(self);
+    return 0;
+}
+
+
+static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg)
+{
+    ALCjackPlayback *self = arg;
+    jack_default_audio_sample_t *out[MAX_OUTPUT_CHANNELS];
+    ll_ringbuffer_data_t data[2];
+    jack_nframes_t total = 0;
+    jack_nframes_t todo;
+    ALsizei i, c, numchans;
+
+    ll_ringbuffer_get_read_vector(self->Ring, data);
+
+    for(c = 0;c < MAX_OUTPUT_CHANNELS && self->Port[c];c++)
+        out[c] = jack_port_get_buffer(self->Port[c], numframes);
+    numchans = c;
+
+    todo = minu(numframes, data[0].len);
+    for(c = 0;c < numchans;c++)
+    {
+        const ALfloat *restrict in = ((ALfloat*)data[0].buf) + c;
+        for(i = 0;(jack_nframes_t)i < todo;i++)
+            out[c][i] = in[i*numchans];
+        out[c] += todo;
+    }
+    total += todo;
+
+    todo = minu(numframes-total, data[1].len);
+    if(todo > 0)
+    {
+        for(c = 0;c < numchans;c++)
+        {
+            const ALfloat *restrict in = ((ALfloat*)data[1].buf) + c;
+            for(i = 0;(jack_nframes_t)i < todo;i++)
+                out[c][i] = in[i*numchans];
+            out[c] += todo;
+        }
+        total += todo;
+    }
+
+    ll_ringbuffer_read_advance(self->Ring, total);
+    alcnd_signal(&self->Cond);
+
+    if(numframes > total)
+    {
+        todo = numframes-total;
+        for(c = 0;c < numchans;c++)
+        {
+            for(i = 0;(jack_nframes_t)i < todo;i++)
+                out[c][i] = 0.0f;
+        }
+    }
+
+    return 0;
+}
+
+static int ALCjackPlayback_mixerProc(void *arg)
+{
+    ALCjackPlayback *self = arg;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    ll_ringbuffer_data_t data[2];
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    ALCjackPlayback_lock(self);
+    while(!self->killNow && device->Connected)
+    {
+        ALuint todo, len1, len2;
+
+        /* NOTE: Unfortunately, there is an unavoidable race condition here.
+         * It's possible for the process() method to run, updating the read
+         * pointer and signaling the condition variable, in between the mixer
+         * loop checking the write size and waiting for the condition variable.
+         * This will cause the mixer loop to wait until the *next* process()
+         * invocation, most likely writing silence for it.
+         *
+         * However, this should only happen if the mixer is running behind
+         * anyway (as ideally we'll be asleep in alcnd_wait by the time the
+         * process() method is invoked), so this behavior is not unwarranted.
+         * It's unfortunate since it'll be wasting time sleeping that could be
+         * used to catch up, but there's no way around it without blocking in
+         * the process() method.
+         */
+        if(ll_ringbuffer_write_space(self->Ring) < device->UpdateSize)
+        {
+            alcnd_wait(&self->Cond, &STATIC_CAST(ALCbackend,self)->mMutex);
+            continue;
+        }
+
+        ll_ringbuffer_get_write_vector(self->Ring, data);
+        todo  = data[0].len + data[1].len;
+        todo -= todo%device->UpdateSize;
+
+        len1 = minu(data[0].len, todo);
+        len2 = minu(data[1].len, todo-len1);
+
+        aluMixData(device, data[0].buf, len1);
+        if(len2 > 0)
+            aluMixData(device, data[1].buf, len2);
+        ll_ringbuffer_write_advance(self->Ring, todo);
+    }
+    ALCjackPlayback_unlock(self);
+
+    return 0;
+}
+
+
+static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    const char *client_name = "alsoft";
+    jack_status_t status;
+
+    if(!name)
+        name = jackDevice;
+    else if(strcmp(name, jackDevice) != 0)
+        return ALC_INVALID_VALUE;
+
+    self->Client = jack_client_open(client_name, ClientOptions, &status, NULL);
+    if(self->Client == NULL)
+    {
+        ERR("jack_client_open() failed, status = 0x%02x\n", status);
+        return ALC_INVALID_VALUE;
+    }
+    if((status&JackServerStarted))
+        TRACE("JACK server started\n");
+    if((status&JackNameNotUnique))
+    {
+        client_name = jack_get_client_name(self->Client);
+        TRACE("Client name not unique, got `%s' instead\n", client_name);
+    }
+
+    jack_set_process_callback(self->Client, ALCjackPlayback_process, self);
+    jack_set_buffer_size_callback(self->Client, ALCjackPlayback_bufferSizeNotify, self);
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCjackPlayback_close(ALCjackPlayback *self)
+{
+    ALuint i;
+
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+    {
+        if(self->Port[i])
+            jack_port_unregister(self->Client, self->Port[i]);
+        self->Port[i] = NULL;
+    }
+    jack_client_close(self->Client);
+    self->Client = NULL;
+}
+
+static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ALsizei numchans, i;
+    ALuint bufsize;
+
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+    {
+        if(self->Port[i])
+            jack_port_unregister(self->Client, self->Port[i]);
+        self->Port[i] = NULL;
+    }
+
+    /* Ignore the requested buffer metrics and just keep one JACK-sized buffer
+     * ready for when requested. Note that one period's worth of audio in the
+     * ring buffer will always be left unfilled because one element of the ring
+     * buffer will not be writeable, and we only write in period-sized chunks.
+     */
+    device->Frequency = jack_get_sample_rate(self->Client);
+    device->UpdateSize = jack_get_buffer_size(self->Client);
+    device->NumUpdates = 2;
+
+    bufsize = device->UpdateSize;
+    if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize))
+        bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize);
+    bufsize += device->UpdateSize;
+    device->NumUpdates = bufsize / device->UpdateSize;
+
+    /* Force 32-bit float output. */
+    device->FmtType = DevFmtFloat;
+
+    numchans = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    for(i = 0;i < numchans;i++)
+    {
+        char name[64];
+        snprintf(name, sizeof(name), "channel_%d", i+1);
+        self->Port[i] = jack_port_register(self->Client, name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+        if(self->Port[i] == NULL)
+        {
+            ERR("Not enough JACK ports available for %s output\n", DevFmtChannelsString(device->FmtChans));
+            if(i == 0) return ALC_FALSE;
+            break;
+        }
+    }
+    if(i < numchans)
+    {
+        if(i == 1)
+            device->FmtChans = DevFmtMono;
+        else
+        {
+            for(--i;i >= 2;i--)
+            {
+                jack_port_unregister(self->Client, self->Port[i]);
+                self->Port[i] = NULL;
+            }
+            device->FmtChans = DevFmtStereo;
+        }
+    }
+
+    ll_ringbuffer_free(self->Ring);
+    self->Ring = ll_ringbuffer_create(bufsize,
+        FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder)
+    );
+    if(!self->Ring)
+    {
+        ERR("Failed to allocate ringbuffer\n");
+        return ALC_FALSE;
+    }
+
+    SetDefaultChannelOrder(device);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self)
+{
+    const char **ports;
+    ALsizei i;
+
+    if(jack_activate(self->Client))
+    {
+        ERR("Failed to activate client\n");
+        return ALC_FALSE;
+    }
+
+    ports = jack_get_ports(self->Client, NULL, NULL, JackPortIsPhysical|JackPortIsInput);
+    if(ports == NULL)
+    {
+        ERR("No physical playback ports found\n");
+        jack_deactivate(self->Client);
+        return ALC_FALSE;
+    }
+    for(i = 0;i < MAX_OUTPUT_CHANNELS && self->Port[i];i++)
+    {
+        if(!ports[i])
+        {
+            ERR("No physical playback port for \"%s\"\n", jack_port_name(self->Port[i]));
+            break;
+        }
+        if(jack_connect(self->Client, jack_port_name(self->Port[i]), ports[i]))
+            ERR("Failed to connect output port \"%s\" to \"%s\"\n", jack_port_name(self->Port[i]), ports[i]);
+    }
+    jack_free(ports);
+
+    self->killNow = 0;
+    if(althrd_create(&self->thread, ALCjackPlayback_mixerProc, self) != althrd_success)
+    {
+        jack_deactivate(self->Client);
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCjackPlayback_stop(ALCjackPlayback *self)
+{
+    int res;
+
+    if(self->killNow)
+        return;
+
+    self->killNow = 1;
+    /* Lock the backend to ensure we don't flag the mixer to die and signal the
+     * mixer to wake up in between it checking the flag and going to sleep and
+     * wait for a wakeup (potentially leading to it never waking back up to see
+     * the flag). */
+    ALCjackPlayback_lock(self);
+    ALCjackPlayback_unlock(self);
+    alcnd_signal(&self->Cond);
+    althrd_join(self->thread, &res);
+
+    jack_deactivate(self->Client);
+}
+
+
+static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ClockLatency ret;
+
+    ALCjackPlayback_lock(self);
+    ret.ClockTime = GetDeviceClockTime(device);
+    ret.Latency = ll_ringbuffer_read_space(self->Ring) * DEVICE_CLOCK_RES /
+                  device->Frequency;
+    ALCjackPlayback_unlock(self);
+
+    return ret;
+}
+
+
+static void jack_msg_handler(const char *message)
+{
+    WARN("%s\n", message);
+}
+
+typedef struct ALCjackBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCjackBackendFactory;
+#define ALCJACKBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCjackBackendFactory, ALCbackendFactory) } }
+
+static ALCboolean ALCjackBackendFactory_init(ALCjackBackendFactory* UNUSED(self))
+{
+    jack_client_t *client;
+    jack_status_t status;
+
+    if(!jack_load())
+        return ALC_FALSE;
+
+    if(!GetConfigValueBool(NULL, "jack", "spawn-server", 0))
+        ClientOptions |= JackNoStartServer;
+
+    jack_set_error_function(jack_msg_handler);
+    client = jack_client_open("alsoft", ClientOptions, &status, NULL);
+    jack_set_error_function(NULL);
+    if(client == NULL)
+    {
+        WARN("jack_client_open() failed, 0x%02x\n", status);
+        if((status&JackServerFailed) && !(ClientOptions&JackNoStartServer))
+            ERR("Unable to connect to JACK server\n");
+        return ALC_FALSE;
+    }
+
+    jack_client_close(client);
+    return ALC_TRUE;
+}
+
+static void ALCjackBackendFactory_deinit(ALCjackBackendFactory* UNUSED(self))
+{
+#ifdef HAVE_DYNLOAD
+    if(jack_handle)
+        CloseLib(jack_handle);
+    jack_handle = NULL;
+#endif
+}
+
+static ALCboolean ALCjackBackendFactory_querySupport(ALCjackBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCjackBackendFactory_probe(ALCjackBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            AppendAllDevicesList(jackDevice);
+            break;
+
+        case CAPTURE_DEVICE_PROBE:
+            break;
+    }
+}
+
+static ALCbackend* ALCjackBackendFactory_createBackend(ALCjackBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCjackPlayback *backend;
+        NEW_OBJ(backend, ALCjackPlayback)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}
+
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCjackBackendFactory);
+
+
+ALCbackendFactory *ALCjackBackendFactory_getFactory(void)
+{
+    static ALCjackBackendFactory factory = ALCJACKBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}

Alc/backends/loopback.c

@@ -0,0 +1,133 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2011 by Chris Robinson
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alu.h"
+
+#include "backends/base.h"
+
+
+typedef struct ALCloopback {
+    DERIVE_FROM_TYPE(ALCbackend);
+} ALCloopback;
+
+static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device);
+static DECLARE_FORWARD(ALCloopback, ALCbackend, void, Destruct)
+static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name);
+static void ALCloopback_close(ALCloopback *self);
+static ALCboolean ALCloopback_reset(ALCloopback *self);
+static ALCboolean ALCloopback_start(ALCloopback *self);
+static void ALCloopback_stop(ALCloopback *self);
+static DECLARE_FORWARD2(ALCloopback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCloopback, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCloopback, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCloopback, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCloopback, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCloopback)
+DEFINE_ALCBACKEND_VTABLE(ALCloopback);
+
+
+static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCloopback, ALCbackend, self);
+}
+
+
+static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    alstr_copy_cstr(&device->DeviceName, name);
+    return ALC_NO_ERROR;
+}
+
+static void ALCloopback_close(ALCloopback* UNUSED(self))
+{
+}
+
+static ALCboolean ALCloopback_reset(ALCloopback *self)
+{
+    SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCloopback_start(ALCloopback* UNUSED(self))
+{
+    return ALC_TRUE;
+}
+
+static void ALCloopback_stop(ALCloopback* UNUSED(self))
+{
+}
+
+
+typedef struct ALCloopbackFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCloopbackFactory;
+#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCloopbackFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCloopbackFactory_getFactory(void);
+static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory *self);
+static DECLARE_FORWARD(ALCloopbackFactory, ALCbackendFactory, void, deinit)
+static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory *self, ALCbackend_Type type);
+static void ALCloopbackFactory_probe(ALCloopbackFactory *self, enum DevProbe type);
+static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCloopbackFactory);
+
+
+ALCbackendFactory *ALCloopbackFactory_getFactory(void)
+{
+    static ALCloopbackFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory* UNUSED(self))
+{
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Loopback)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCloopbackFactory_probe(ALCloopbackFactory* UNUSED(self), enum DevProbe UNUSED(type))
+{
+}
+
+static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Loopback)
+    {
+        ALCloopback *backend;
+        NEW_OBJ(backend, ALCloopback)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/null.c

@@ -0,0 +1,225 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2010 by Chris Robinson
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+#ifdef HAVE_WINDOWS_H
+#include <windows.h>
+#endif
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+#include "compat.h"
+
+#include "backends/base.h"
+
+
+typedef struct ALCnullBackend {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    volatile int killNow;
+    althrd_t thread;
+} ALCnullBackend;
+
+static int ALCnullBackend_mixerProc(void *ptr);
+
+static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device);
+static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, Destruct)
+static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name);
+static void ALCnullBackend_close(ALCnullBackend *self);
+static ALCboolean ALCnullBackend_reset(ALCnullBackend *self);
+static ALCboolean ALCnullBackend_start(ALCnullBackend *self);
+static void ALCnullBackend_stop(ALCnullBackend *self);
+static DECLARE_FORWARD2(ALCnullBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCnullBackend)
+
+DEFINE_ALCBACKEND_VTABLE(ALCnullBackend);
+
+
+static const ALCchar nullDevice[] = "No Output";
+
+
+static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCnullBackend, ALCbackend, self);
+}
+
+
+static int ALCnullBackend_mixerProc(void *ptr)
+{
+    ALCnullBackend *self = (ALCnullBackend*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    struct timespec now, start;
+    ALuint64 avail, done;
+    const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 /
+                                 device->Frequency / 2);
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    done = 0;
+    if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
+    {
+        ERR("Failed to get starting time\n");
+        return 1;
+    }
+    while(!self->killNow && device->Connected)
+    {
+        if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC)
+        {
+            ERR("Failed to get current time\n");
+            return 1;
+        }
+
+        avail  = (now.tv_sec - start.tv_sec) * device->Frequency;
+        avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
+        if(avail < done)
+        {
+            /* Oops, time skipped backwards. Reset the number of samples done
+             * with one update available since we (likely) just came back from
+             * sleeping. */
+            done = avail - device->UpdateSize;
+        }
+
+        if(avail-done < device->UpdateSize)
+            al_nssleep(restTime);
+        else while(avail-done >= device->UpdateSize)
+        {
+            ALCnullBackend_lock(self);
+            aluMixData(device, NULL, device->UpdateSize);
+            ALCnullBackend_unlock(self);
+            done += device->UpdateSize;
+        }
+    }
+
+    return 0;
+}
+
+
+static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name)
+{
+    ALCdevice *device;
+
+    if(!name)
+        name = nullDevice;
+    else if(strcmp(name, nullDevice) != 0)
+        return ALC_INVALID_VALUE;
+
+    device = STATIC_CAST(ALCbackend, self)->mDevice;
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCnullBackend_close(ALCnullBackend* UNUSED(self))
+{
+}
+
+static ALCboolean ALCnullBackend_reset(ALCnullBackend *self)
+{
+    SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCnullBackend_start(ALCnullBackend *self)
+{
+    self->killNow = 0;
+    if(althrd_create(&self->thread, ALCnullBackend_mixerProc, self) != althrd_success)
+        return ALC_FALSE;
+    return ALC_TRUE;
+}
+
+static void ALCnullBackend_stop(ALCnullBackend *self)
+{
+    int res;
+
+    if(self->killNow)
+        return;
+
+    self->killNow = 1;
+    althrd_join(self->thread, &res);
+}
+
+
+typedef struct ALCnullBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCnullBackendFactory;
+#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCnullBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCnullBackendFactory_getFactory(void);
+
+static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory *self);
+static DECLARE_FORWARD(ALCnullBackendFactory, ALCbackendFactory, void, deinit)
+static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory *self, ALCbackend_Type type);
+static void ALCnullBackendFactory_probe(ALCnullBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCnullBackendFactory);
+
+
+ALCbackendFactory *ALCnullBackendFactory_getFactory(void)
+{
+    static ALCnullBackendFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory* UNUSED(self))
+{
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCnullBackendFactory_probe(ALCnullBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            AppendAllDevicesList(nullDevice);
+            break;
+        case CAPTURE_DEVICE_PROBE:
+            break;
+    }
+}
+
+static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCnullBackend *backend;
+        NEW_OBJ(backend, ALCnullBackend)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/oss.c

@@ -0,0 +1,860 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <sys/ioctl.h>
+#include <sys/types.h>
+#include <sys/time.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <memory.h>
+#include <unistd.h>
+#include <errno.h>
+#include <math.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+#include "compat.h"
+
+#include "backends/base.h"
+
+#include <sys/soundcard.h>
+
+/*
+ * The OSS documentation talks about SOUND_MIXER_READ, but the header
+ * only contains MIXER_READ. Play safe. Same for WRITE.
+ */
+#ifndef SOUND_MIXER_READ
+#define SOUND_MIXER_READ MIXER_READ
+#endif
+#ifndef SOUND_MIXER_WRITE
+#define SOUND_MIXER_WRITE MIXER_WRITE
+#endif
+
+#if defined(SOUND_VERSION) && (SOUND_VERSION < 0x040000)
+#define ALC_OSS_COMPAT
+#endif
+#ifndef SNDCTL_AUDIOINFO
+#define ALC_OSS_COMPAT
+#endif
+
+/*
+ * FreeBSD strongly discourages the use of specific devices,
+ * such as those returned in oss_audioinfo.devnode
+ */
+#ifdef __FreeBSD__
+#define ALC_OSS_DEVNODE_TRUC
+#endif
+
+struct oss_device {
+    const ALCchar *handle;
+    const char *path;
+    struct oss_device *next;
+};
+
+static struct oss_device oss_playback = {
+    "OSS Default",
+    "/dev/dsp",
+    NULL
+};
+
+static struct oss_device oss_capture = {
+    "OSS Default",
+    "/dev/dsp",
+    NULL
+};
+
+#ifdef ALC_OSS_COMPAT
+
+#define DSP_CAP_OUTPUT 0x00020000
+#define DSP_CAP_INPUT 0x00010000
+static void ALCossListPopulate(struct oss_device *UNUSED(devlist), int UNUSED(type_flag))
+{
+}
+
+#else
+
+#ifndef HAVE_STRNLEN
+static size_t strnlen(const char *str, size_t maxlen)
+{
+    const char *end = memchr(str, 0, maxlen);
+    if(!end) return maxlen;
+    return end - str;
+}
+#endif
+
+static void ALCossListAppend(struct oss_device *list, const char *handle, size_t hlen, const char *path, size_t plen)
+{
+    struct oss_device *next;
+    struct oss_device *last;
+    size_t i;
+
+    /* skip the first item "OSS Default" */
+    last = list;
+    next = list->next;
+#ifdef ALC_OSS_DEVNODE_TRUC
+    for(i = 0;i < plen;i++)
+    {
+        if(path[i] == '.')
+        {
+            if(strncmp(path + i, handle + hlen + i - plen, plen - i) == 0)
+                hlen = hlen + i - plen;
+            plen = i;
+        }
+    }
+#else
+    (void)i;
+#endif
+    if(handle[0] == '\0')
+    {
+        handle = path;
+        hlen = plen;
+    }
+
+    while(next != NULL)
+    {
+        if(strncmp(next->path, path, plen) == 0)
+            return;
+        last = next;
+        next = next->next;
+    }
+
+    next = (struct oss_device*)malloc(sizeof(struct oss_device) + hlen + plen + 2);
+    next->handle = (char*)(next + 1);
+    next->path = next->handle + hlen + 1;
+    next->next = NULL;
+    last->next = next;
+
+    strncpy((char*)next->handle, handle, hlen);
+    ((char*)next->handle)[hlen] = '\0';
+    strncpy((char*)next->path, path, plen);
+    ((char*)next->path)[plen] = '\0';
+
+    TRACE("Got device \"%s\", \"%s\"\n", next->handle, next->path);
+}
+
+static void ALCossListPopulate(struct oss_device *devlist, int type_flag)
+{
+    struct oss_sysinfo si;
+    struct oss_audioinfo ai;
+    int fd, i;
+
+    if((fd=open("/dev/mixer", O_RDONLY)) < 0)
+    {
+        ERR("Could not open /dev/mixer\n");
+        return;
+    }
+    if(ioctl(fd, SNDCTL_SYSINFO, &si) == -1)
+    {
+        ERR("SNDCTL_SYSINFO failed: %s\n", strerror(errno));
+        goto done;
+    }
+    for(i = 0;i < si.numaudios;i++)
+    {
+        const char *handle;
+        size_t len;
+
+        ai.dev = i;
+        if(ioctl(fd, SNDCTL_AUDIOINFO, &ai) == -1)
+        {
+            ERR("SNDCTL_AUDIOINFO (%d) failed: %s\n", i, strerror(errno));
+            continue;
+        }
+        if(ai.devnode[0] == '\0')
+            continue;
+
+        if(ai.handle[0] != '\0')
+        {
+            len = strnlen(ai.handle, sizeof(ai.handle));
+            handle = ai.handle;
+        }
+        else
+        {
+            len = strnlen(ai.name, sizeof(ai.name));
+            handle = ai.name;
+        }
+        if((ai.caps&type_flag))
+            ALCossListAppend(devlist, handle, len, ai.devnode,
+                             strnlen(ai.devnode, sizeof(ai.devnode)));
+    }
+
+done:
+    close(fd);
+}
+
+#endif
+
+static void ALCossListFree(struct oss_device *list)
+{
+    struct oss_device *cur;
+    if(list == NULL)
+        return;
+
+    /* skip the first item "OSS Default" */
+    cur = list->next;
+    list->next = NULL;
+
+    while(cur != NULL)
+    {
+        struct oss_device *next = cur->next;
+        free(cur);
+        cur = next;
+    }
+}
+
+static int log2i(ALCuint x)
+{
+    int y = 0;
+    while (x > 1)
+    {
+        x >>= 1;
+        y++;
+    }
+    return y;
+}
+
+typedef struct ALCplaybackOSS {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    int fd;
+
+    ALubyte *mix_data;
+    int data_size;
+
+    ATOMIC(ALenum) killNow;
+    althrd_t thread;
+} ALCplaybackOSS;
+
+static int ALCplaybackOSS_mixerProc(void *ptr);
+
+static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device);
+static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, Destruct)
+static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name);
+static void ALCplaybackOSS_close(ALCplaybackOSS *self);
+static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self);
+static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self);
+static void ALCplaybackOSS_stop(ALCplaybackOSS *self);
+static DECLARE_FORWARD2(ALCplaybackOSS, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
+static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCplaybackOSS)
+DEFINE_ALCBACKEND_VTABLE(ALCplaybackOSS);
+
+
+static int ALCplaybackOSS_mixerProc(void *ptr)
+{
+    ALCplaybackOSS *self = (ALCplaybackOSS*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    struct timeval timeout;
+    ALubyte *write_ptr;
+    ALint frame_size;
+    ALint to_write;
+    ssize_t wrote;
+    fd_set wfds;
+    int sret;
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    ALCplaybackOSS_lock(self);
+    while(!ATOMIC_LOAD_SEQ(&self->killNow) && device->Connected)
+    {
+        FD_ZERO(&wfds);
+        FD_SET(self->fd, &wfds);
+        timeout.tv_sec = 1;
+        timeout.tv_usec = 0;
+
+        ALCplaybackOSS_unlock(self);
+        sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
+        ALCplaybackOSS_lock(self);
+        if(sret < 0)
+        {
+            if(errno == EINTR)
+                continue;
+            ERR("select failed: %s\n", strerror(errno));
+            aluHandleDisconnect(device);
+            break;
+        }
+        else if(sret == 0)
+        {
+            WARN("select timeout\n");
+            continue;
+        }
+
+        write_ptr = self->mix_data;
+        to_write = self->data_size;
+        aluMixData(device, write_ptr, to_write/frame_size);
+        while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow))
+        {
+            wrote = write(self->fd, write_ptr, to_write);
+            if(wrote < 0)
+            {
+                if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
+                    continue;
+                ERR("write failed: %s\n", strerror(errno));
+                aluHandleDisconnect(device);
+                break;
+            }
+
+            to_write -= wrote;
+            write_ptr += wrote;
+        }
+    }
+    ALCplaybackOSS_unlock(self);
+
+    return 0;
+}
+
+
+static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCplaybackOSS, ALCbackend, self);
+
+    ATOMIC_INIT(&self->killNow, AL_FALSE);
+}
+
+static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name)
+{
+    struct oss_device *dev = &oss_playback;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    if(!name || strcmp(name, dev->handle) == 0)
+        name = dev->handle;
+    else
+    {
+        if(!dev->next)
+        {
+            ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT);
+            dev = &oss_playback;
+        }
+        while(dev != NULL)
+        {
+            if (strcmp(dev->handle, name) == 0)
+                break;
+            dev = dev->next;
+        }
+        if(dev == NULL)
+        {
+            WARN("Could not find \"%s\" in device list\n", name);
+            return ALC_INVALID_VALUE;
+        }
+    }
+
+    self->fd = open(dev->path, O_WRONLY);
+    if(self->fd == -1)
+    {
+        ERR("Could not open %s: %s\n", dev->path, strerror(errno));
+        return ALC_INVALID_VALUE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCplaybackOSS_close(ALCplaybackOSS *self)
+{
+    close(self->fd);
+    self->fd = -1;
+}
+
+static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    int numFragmentsLogSize;
+    int log2FragmentSize;
+    unsigned int periods;
+    audio_buf_info info;
+    ALuint frameSize;
+    int numChannels;
+    int ossFormat;
+    int ossSpeed;
+    char *err;
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            ossFormat = AFMT_S8;
+            break;
+        case DevFmtUByte:
+            ossFormat = AFMT_U8;
+            break;
+        case DevFmtUShort:
+        case DevFmtInt:
+        case DevFmtUInt:
+        case DevFmtFloat:
+            device->FmtType = DevFmtShort;
+            /* fall-through */
+        case DevFmtShort:
+            ossFormat = AFMT_S16_NE;
+            break;
+    }
+
+    periods = device->NumUpdates;
+    numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    ossSpeed = device->Frequency;
+    frameSize = numChannels * BytesFromDevFmt(device->FmtType);
+    /* According to the OSS spec, 16 bytes (log2(16)) is the minimum. */
+    log2FragmentSize = maxi(log2i(device->UpdateSize*frameSize), 4);
+    numFragmentsLogSize = (periods << 16) | log2FragmentSize;
+
+#define CHECKERR(func) if((func) < 0) {                                       \
+    err = #func;                                                              \
+    goto err;                                                                 \
+}
+    /* Don't fail if SETFRAGMENT fails. We can handle just about anything
+     * that's reported back via GETOSPACE */
+    ioctl(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize);
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat));
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels));
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed));
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_GETOSPACE, &info));
+    if(0)
+    {
+    err:
+        ERR("%s failed: %s\n", err, strerror(errno));
+        return ALC_FALSE;
+    }
+#undef CHECKERR
+
+    if((int)ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != numChannels)
+    {
+        ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels);
+        return ALC_FALSE;
+    }
+
+    if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) ||
+         (ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) ||
+         (ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort)))
+    {
+        ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat);
+        return ALC_FALSE;
+    }
+
+    device->Frequency = ossSpeed;
+    device->UpdateSize = info.fragsize / frameSize;
+    device->NumUpdates = info.fragments;
+
+    SetDefaultChannelOrder(device);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
+        device->FmtChans, device->FmtType, device->AmbiOrder
+    );
+    self->mix_data = calloc(1, self->data_size);
+
+    ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
+    if(althrd_create(&self->thread, ALCplaybackOSS_mixerProc, self) != althrd_success)
+    {
+        free(self->mix_data);
+        self->mix_data = NULL;
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCplaybackOSS_stop(ALCplaybackOSS *self)
+{
+    int res;
+
+    if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
+        return;
+    althrd_join(self->thread, &res);
+
+    if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0)
+        ERR("Error resetting device: %s\n", strerror(errno));
+
+    free(self->mix_data);
+    self->mix_data = NULL;
+}
+
+
+typedef struct ALCcaptureOSS {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    int fd;
+
+    ll_ringbuffer_t *ring;
+
+    ATOMIC(ALenum) killNow;
+    althrd_t thread;
+} ALCcaptureOSS;
+
+static int ALCcaptureOSS_recordProc(void *ptr);
+
+static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device);
+static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, Destruct)
+static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name);
+static void ALCcaptureOSS_close(ALCcaptureOSS *self);
+static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ALCboolean, reset)
+static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self);
+static void ALCcaptureOSS_stop(ALCcaptureOSS *self);
+static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples);
+static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self);
+static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCcaptureOSS)
+DEFINE_ALCBACKEND_VTABLE(ALCcaptureOSS);
+
+
+static int ALCcaptureOSS_recordProc(void *ptr)
+{
+    ALCcaptureOSS *self = (ALCcaptureOSS*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    struct timeval timeout;
+    int frame_size;
+    fd_set rfds;
+    ssize_t amt;
+    int sret;
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), RECORD_THREAD_NAME);
+
+    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    while(!ATOMIC_LOAD_SEQ(&self->killNow))
+    {
+        ll_ringbuffer_data_t vec[2];
+
+        FD_ZERO(&rfds);
+        FD_SET(self->fd, &rfds);
+        timeout.tv_sec = 1;
+        timeout.tv_usec = 0;
+
+        sret = select(self->fd+1, &rfds, NULL, NULL, &timeout);
+        if(sret < 0)
+        {
+            if(errno == EINTR)
+                continue;
+            ERR("select failed: %s\n", strerror(errno));
+            aluHandleDisconnect(device);
+            break;
+        }
+        else if(sret == 0)
+        {
+            WARN("select timeout\n");
+            continue;
+        }
+
+        ll_ringbuffer_get_write_vector(self->ring, vec);
+        if(vec[0].len > 0)
+        {
+            amt = read(self->fd, vec[0].buf, vec[0].len*frame_size);
+            if(amt < 0)
+            {
+                ERR("read failed: %s\n", strerror(errno));
+                ALCcaptureOSS_lock(self);
+                aluHandleDisconnect(device);
+                ALCcaptureOSS_unlock(self);
+                break;
+            }
+            ll_ringbuffer_write_advance(self->ring, amt/frame_size);
+        }
+    }
+
+    return 0;
+}
+
+
+static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCcaptureOSS, ALCbackend, self);
+
+    ATOMIC_INIT(&self->killNow, AL_FALSE);
+}
+
+static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    struct oss_device *dev = &oss_capture;
+    int numFragmentsLogSize;
+    int log2FragmentSize;
+    unsigned int periods;
+    audio_buf_info info;
+    ALuint frameSize;
+    int numChannels;
+    int ossFormat;
+    int ossSpeed;
+    char *err;
+
+    if(!name || strcmp(name, dev->handle) == 0)
+        name = dev->handle;
+    else
+    {
+        if(!dev->next)
+        {
+            ALCossListPopulate(&oss_capture, DSP_CAP_INPUT);
+            dev = &oss_capture;
+        }
+        while(dev != NULL)
+        {
+            if (strcmp(dev->handle, name) == 0)
+                break;
+            dev = dev->next;
+        }
+        if(dev == NULL)
+        {
+            WARN("Could not find \"%s\" in device list\n", name);
+            return ALC_INVALID_VALUE;
+        }
+    }
+
+    self->fd = open(dev->path, O_RDONLY);
+    if(self->fd == -1)
+    {
+        ERR("Could not open %s: %s\n", dev->path, strerror(errno));
+        return ALC_INVALID_VALUE;
+    }
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            ossFormat = AFMT_S8;
+            break;
+        case DevFmtUByte:
+            ossFormat = AFMT_U8;
+            break;
+        case DevFmtShort:
+            ossFormat = AFMT_S16_NE;
+            break;
+        case DevFmtUShort:
+        case DevFmtInt:
+        case DevFmtUInt:
+        case DevFmtFloat:
+            ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
+            return ALC_INVALID_VALUE;
+    }
+
+    periods = 4;
+    numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    frameSize = numChannels * BytesFromDevFmt(device->FmtType);
+    ossSpeed = device->Frequency;
+    log2FragmentSize = log2i(device->UpdateSize * device->NumUpdates *
+                             frameSize / periods);
+
+    /* according to the OSS spec, 16 bytes are the minimum */
+    if (log2FragmentSize < 4)
+        log2FragmentSize = 4;
+    numFragmentsLogSize = (periods << 16) | log2FragmentSize;
+
+#define CHECKERR(func) if((func) < 0) {                                       \
+    err = #func;                                                              \
+    goto err;                                                                 \
+}
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize));
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat));
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels));
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed));
+    CHECKERR(ioctl(self->fd, SNDCTL_DSP_GETISPACE, &info));
+    if(0)
+    {
+    err:
+        ERR("%s failed: %s\n", err, strerror(errno));
+        close(self->fd);
+        self->fd = -1;
+        return ALC_INVALID_VALUE;
+    }
+#undef CHECKERR
+
+    if((int)ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != numChannels)
+    {
+        ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels);
+        close(self->fd);
+        self->fd = -1;
+        return ALC_INVALID_VALUE;
+    }
+
+    if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) ||
+         (ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) ||
+         (ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort)))
+    {
+        ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat);
+        close(self->fd);
+        self->fd = -1;
+        return ALC_INVALID_VALUE;
+    }
+
+    self->ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates + 1, frameSize);
+    if(!self->ring)
+    {
+        ERR("Ring buffer create failed\n");
+        close(self->fd);
+        self->fd = -1;
+        return ALC_OUT_OF_MEMORY;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCcaptureOSS_close(ALCcaptureOSS *self)
+{
+    close(self->fd);
+    self->fd = -1;
+
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+}
+
+static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self)
+{
+    ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
+    if(althrd_create(&self->thread, ALCcaptureOSS_recordProc, self) != althrd_success)
+        return ALC_FALSE;
+    return ALC_TRUE;
+}
+
+static void ALCcaptureOSS_stop(ALCcaptureOSS *self)
+{
+    int res;
+
+    if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
+        return;
+
+    althrd_join(self->thread, &res);
+
+    if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0)
+        ERR("Error resetting device: %s\n", strerror(errno));
+}
+
+static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples)
+{
+    ll_ringbuffer_read(self->ring, buffer, samples);
+    return ALC_NO_ERROR;
+}
+
+static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self)
+{
+    return ll_ringbuffer_read_space(self->ring);
+}
+
+
+typedef struct ALCossBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCossBackendFactory;
+#define ALCOSSBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCossBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCossBackendFactory_getFactory(void);
+
+static ALCboolean ALCossBackendFactory_init(ALCossBackendFactory *self);
+static void ALCossBackendFactory_deinit(ALCossBackendFactory *self);
+static ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory *self, ALCbackend_Type type);
+static void ALCossBackendFactory_probe(ALCossBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCossBackendFactory);
+
+
+ALCbackendFactory *ALCossBackendFactory_getFactory(void)
+{
+    static ALCossBackendFactory factory = ALCOSSBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+ALCboolean ALCossBackendFactory_init(ALCossBackendFactory* UNUSED(self))
+{
+    ConfigValueStr(NULL, "oss", "device", &oss_playback.path);
+    ConfigValueStr(NULL, "oss", "capture", &oss_capture.path);
+
+    return ALC_TRUE;
+}
+
+void  ALCossBackendFactory_deinit(ALCossBackendFactory* UNUSED(self))
+{
+    ALCossListFree(&oss_playback);
+    ALCossListFree(&oss_capture);
+}
+
+
+ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback || type == ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+void ALCossBackendFactory_probe(ALCossBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    struct oss_device *cur;
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            ALCossListFree(&oss_playback);
+            ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT);
+            cur = &oss_playback;
+            while(cur != NULL)
+            {
+                AppendAllDevicesList(cur->handle);
+                cur = cur->next;
+            }
+            break;
+
+        case CAPTURE_DEVICE_PROBE:
+            ALCossListFree(&oss_capture);
+            ALCossListPopulate(&oss_capture, DSP_CAP_INPUT);
+            cur = &oss_capture;
+            while(cur != NULL)
+            {
+                AppendCaptureDeviceList(cur->handle);
+                cur = cur->next;
+            }
+            break;
+    }
+}
+
+ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCplaybackOSS *backend;
+        NEW_OBJ(backend, ALCplaybackOSS)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+    if(type == ALCbackend_Capture)
+    {
+        ALCcaptureOSS *backend;
+        NEW_OBJ(backend, ALCcaptureOSS)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/portaudio.c

@@ -0,0 +1,575 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "compat.h"
+
+#include "backends/base.h"
+
+#include <portaudio.h>
+
+
+static const ALCchar pa_device[] = "PortAudio Default";
+
+
+#ifdef HAVE_DYNLOAD
+static void *pa_handle;
+#define MAKE_FUNC(x) static __typeof(x) * p##x
+MAKE_FUNC(Pa_Initialize);
+MAKE_FUNC(Pa_Terminate);
+MAKE_FUNC(Pa_GetErrorText);
+MAKE_FUNC(Pa_StartStream);
+MAKE_FUNC(Pa_StopStream);
+MAKE_FUNC(Pa_OpenStream);
+MAKE_FUNC(Pa_CloseStream);
+MAKE_FUNC(Pa_GetDefaultOutputDevice);
+MAKE_FUNC(Pa_GetDefaultInputDevice);
+MAKE_FUNC(Pa_GetStreamInfo);
+#undef MAKE_FUNC
+
+#define Pa_Initialize                  pPa_Initialize
+#define Pa_Terminate                   pPa_Terminate
+#define Pa_GetErrorText                pPa_GetErrorText
+#define Pa_StartStream                 pPa_StartStream
+#define Pa_StopStream                  pPa_StopStream
+#define Pa_OpenStream                  pPa_OpenStream
+#define Pa_CloseStream                 pPa_CloseStream
+#define Pa_GetDefaultOutputDevice      pPa_GetDefaultOutputDevice
+#define Pa_GetDefaultInputDevice       pPa_GetDefaultInputDevice
+#define Pa_GetStreamInfo               pPa_GetStreamInfo
+#endif
+
+static ALCboolean pa_load(void)
+{
+    PaError err;
+
+#ifdef HAVE_DYNLOAD
+    if(!pa_handle)
+    {
+#ifdef _WIN32
+# define PALIB "portaudio.dll"
+#elif defined(__APPLE__) && defined(__MACH__)
+# define PALIB "libportaudio.2.dylib"
+#elif defined(__OpenBSD__)
+# define PALIB "libportaudio.so"
+#else
+# define PALIB "libportaudio.so.2"
+#endif
+
+        pa_handle = LoadLib(PALIB);
+        if(!pa_handle)
+            return ALC_FALSE;
+
+#define LOAD_FUNC(f) do {                                                     \
+    p##f = GetSymbol(pa_handle, #f);                                          \
+    if(p##f == NULL)                                                          \
+    {                                                                         \
+        CloseLib(pa_handle);                                                  \
+        pa_handle = NULL;                                                     \
+        return ALC_FALSE;                                                     \
+    }                                                                         \
+} while(0)
+        LOAD_FUNC(Pa_Initialize);
+        LOAD_FUNC(Pa_Terminate);
+        LOAD_FUNC(Pa_GetErrorText);
+        LOAD_FUNC(Pa_StartStream);
+        LOAD_FUNC(Pa_StopStream);
+        LOAD_FUNC(Pa_OpenStream);
+        LOAD_FUNC(Pa_CloseStream);
+        LOAD_FUNC(Pa_GetDefaultOutputDevice);
+        LOAD_FUNC(Pa_GetDefaultInputDevice);
+        LOAD_FUNC(Pa_GetStreamInfo);
+#undef LOAD_FUNC
+
+        if((err=Pa_Initialize()) != paNoError)
+        {
+            ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err));
+            CloseLib(pa_handle);
+            pa_handle = NULL;
+            return ALC_FALSE;
+        }
+    }
+#else
+    if((err=Pa_Initialize()) != paNoError)
+    {
+        ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err));
+        return ALC_FALSE;
+    }
+#endif
+    return ALC_TRUE;
+}
+
+
+typedef struct ALCportPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    PaStream *stream;
+    PaStreamParameters params;
+    ALuint update_size;
+} ALCportPlayback;
+
+static int ALCportPlayback_WriteCallback(const void *inputBuffer, void *outputBuffer,
+    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
+    const PaStreamCallbackFlags statusFlags, void *userData);
+
+static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device);
+static void ALCportPlayback_Destruct(ALCportPlayback *self);
+static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name);
+static void ALCportPlayback_close(ALCportPlayback *self);
+static ALCboolean ALCportPlayback_reset(ALCportPlayback *self);
+static ALCboolean ALCportPlayback_start(ALCportPlayback *self);
+static void ALCportPlayback_stop(ALCportPlayback *self);
+static DECLARE_FORWARD2(ALCportPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
+static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCportPlayback)
+
+DEFINE_ALCBACKEND_VTABLE(ALCportPlayback);
+
+
+static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCportPlayback, ALCbackend, self);
+
+    self->stream = NULL;
+}
+
+static void ALCportPlayback_Destruct(ALCportPlayback *self)
+{
+    if(self->stream)
+        Pa_CloseStream(self->stream);
+    self->stream = NULL;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static int ALCportPlayback_WriteCallback(const void *UNUSED(inputBuffer), void *outputBuffer,
+    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo),
+    const PaStreamCallbackFlags UNUSED(statusFlags), void *userData)
+{
+    ALCportPlayback *self = userData;
+
+    ALCportPlayback_lock(self);
+    aluMixData(STATIC_CAST(ALCbackend, self)->mDevice, outputBuffer, framesPerBuffer);
+    ALCportPlayback_unlock(self);
+    return 0;
+}
+
+
+static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    PaError err;
+
+    if(!name)
+        name = pa_device;
+    else if(strcmp(name, pa_device) != 0)
+        return ALC_INVALID_VALUE;
+
+    self->update_size = device->UpdateSize;
+
+    self->params.device = -1;
+    if(!ConfigValueInt(NULL, "port", "device", &self->params.device) ||
+       self->params.device < 0)
+        self->params.device = Pa_GetDefaultOutputDevice();
+    self->params.suggestedLatency = (device->UpdateSize*device->NumUpdates) /
+                                    (float)device->Frequency;
+    self->params.hostApiSpecificStreamInfo = NULL;
+
+    self->params.channelCount = ((device->FmtChans == DevFmtMono) ? 1 : 2);
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            self->params.sampleFormat = paInt8;
+            break;
+        case DevFmtUByte:
+            self->params.sampleFormat = paUInt8;
+            break;
+        case DevFmtUShort:
+            /* fall-through */
+        case DevFmtShort:
+            self->params.sampleFormat = paInt16;
+            break;
+        case DevFmtUInt:
+            /* fall-through */
+        case DevFmtInt:
+            self->params.sampleFormat = paInt32;
+            break;
+        case DevFmtFloat:
+            self->params.sampleFormat = paFloat32;
+            break;
+    }
+
+retry_open:
+    err = Pa_OpenStream(&self->stream, NULL, &self->params,
+        device->Frequency, device->UpdateSize, paNoFlag,
+        ALCportPlayback_WriteCallback, self
+    );
+    if(err != paNoError)
+    {
+        if(self->params.sampleFormat == paFloat32)
+        {
+            self->params.sampleFormat = paInt16;
+            goto retry_open;
+        }
+        ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
+        return ALC_INVALID_VALUE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+
+}
+
+static void ALCportPlayback_close(ALCportPlayback *self)
+{
+    PaError err = Pa_CloseStream(self->stream);
+    if(err != paNoError)
+        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
+    self->stream = NULL;
+}
+
+static ALCboolean ALCportPlayback_reset(ALCportPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    const PaStreamInfo *streamInfo;
+
+    streamInfo = Pa_GetStreamInfo(self->stream);
+    device->Frequency = streamInfo->sampleRate;
+    device->UpdateSize = self->update_size;
+
+    if(self->params.sampleFormat == paInt8)
+        device->FmtType = DevFmtByte;
+    else if(self->params.sampleFormat == paUInt8)
+        device->FmtType = DevFmtUByte;
+    else if(self->params.sampleFormat == paInt16)
+        device->FmtType = DevFmtShort;
+    else if(self->params.sampleFormat == paInt32)
+        device->FmtType = DevFmtInt;
+    else if(self->params.sampleFormat == paFloat32)
+        device->FmtType = DevFmtFloat;
+    else
+    {
+        ERR("Unexpected sample format: 0x%lx\n", self->params.sampleFormat);
+        return ALC_FALSE;
+    }
+
+    if(self->params.channelCount == 2)
+        device->FmtChans = DevFmtStereo;
+    else if(self->params.channelCount == 1)
+        device->FmtChans = DevFmtMono;
+    else
+    {
+        ERR("Unexpected channel count: %u\n", self->params.channelCount);
+        return ALC_FALSE;
+    }
+    SetDefaultChannelOrder(device);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCportPlayback_start(ALCportPlayback *self)
+{
+    PaError err;
+
+    err = Pa_StartStream(self->stream);
+    if(err != paNoError)
+    {
+        ERR("Pa_StartStream() returned an error: %s\n", Pa_GetErrorText(err));
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCportPlayback_stop(ALCportPlayback *self)
+{
+    PaError err = Pa_StopStream(self->stream);
+    if(err != paNoError)
+        ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
+}
+
+
+typedef struct ALCportCapture {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    PaStream *stream;
+    PaStreamParameters params;
+
+    ll_ringbuffer_t *ring;
+} ALCportCapture;
+
+static int ALCportCapture_ReadCallback(const void *inputBuffer, void *outputBuffer,
+    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
+    const PaStreamCallbackFlags statusFlags, void *userData);
+
+static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device);
+static void ALCportCapture_Destruct(ALCportCapture *self);
+static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name);
+static void ALCportCapture_close(ALCportCapture *self);
+static DECLARE_FORWARD(ALCportCapture, ALCbackend, ALCboolean, reset)
+static ALCboolean ALCportCapture_start(ALCportCapture *self);
+static void ALCportCapture_stop(ALCportCapture *self);
+static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples);
+static ALCuint ALCportCapture_availableSamples(ALCportCapture *self);
+static DECLARE_FORWARD(ALCportCapture, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCportCapture)
+
+DEFINE_ALCBACKEND_VTABLE(ALCportCapture);
+
+
+static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCportCapture, ALCbackend, self);
+
+    self->stream = NULL;
+}
+
+static void ALCportCapture_Destruct(ALCportCapture *self)
+{
+    if(self->stream)
+        Pa_CloseStream(self->stream);
+    self->stream = NULL;
+
+    if(self->ring)
+        ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static int ALCportCapture_ReadCallback(const void *inputBuffer, void *UNUSED(outputBuffer),
+    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo),
+    const PaStreamCallbackFlags UNUSED(statusFlags), void *userData)
+{
+    ALCportCapture *self = userData;
+    size_t writable = ll_ringbuffer_write_space(self->ring);
+
+    if(framesPerBuffer > writable)
+        framesPerBuffer = writable;
+    ll_ringbuffer_write(self->ring, inputBuffer, framesPerBuffer);
+    return 0;
+}
+
+
+static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ALuint samples, frame_size;
+    PaError err;
+
+    if(!name)
+        name = pa_device;
+    else if(strcmp(name, pa_device) != 0)
+        return ALC_INVALID_VALUE;
+
+    samples = device->UpdateSize * device->NumUpdates;
+    samples = maxu(samples, 100 * device->Frequency / 1000);
+    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    self->ring = ll_ringbuffer_create(samples, frame_size);
+    if(self->ring == NULL) return ALC_INVALID_VALUE;
+
+    self->params.device = -1;
+    if(!ConfigValueInt(NULL, "port", "capture", &self->params.device) ||
+       self->params.device < 0)
+        self->params.device = Pa_GetDefaultInputDevice();
+    self->params.suggestedLatency = 0.0f;
+    self->params.hostApiSpecificStreamInfo = NULL;
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            self->params.sampleFormat = paInt8;
+            break;
+        case DevFmtUByte:
+            self->params.sampleFormat = paUInt8;
+            break;
+        case DevFmtShort:
+            self->params.sampleFormat = paInt16;
+            break;
+        case DevFmtInt:
+            self->params.sampleFormat = paInt32;
+            break;
+        case DevFmtFloat:
+            self->params.sampleFormat = paFloat32;
+            break;
+        case DevFmtUInt:
+        case DevFmtUShort:
+            ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
+            return ALC_INVALID_VALUE;
+    }
+    self->params.channelCount = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+
+    err = Pa_OpenStream(&self->stream, &self->params, NULL,
+        device->Frequency, paFramesPerBufferUnspecified, paNoFlag,
+        ALCportCapture_ReadCallback, self
+    );
+    if(err != paNoError)
+    {
+        ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
+        return ALC_INVALID_VALUE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCportCapture_close(ALCportCapture *self)
+{
+    PaError err = Pa_CloseStream(self->stream);
+    if(err != paNoError)
+        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
+    self->stream = NULL;
+
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+}
+
+
+static ALCboolean ALCportCapture_start(ALCportCapture *self)
+{
+    PaError err = Pa_StartStream(self->stream);
+    if(err != paNoError)
+    {
+        ERR("Error starting stream: %s\n", Pa_GetErrorText(err));
+        return ALC_FALSE;
+    }
+    return ALC_TRUE;
+}
+
+static void ALCportCapture_stop(ALCportCapture *self)
+{
+    PaError err = Pa_StopStream(self->stream);
+    if(err != paNoError)
+        ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
+}
+
+
+static ALCuint ALCportCapture_availableSamples(ALCportCapture *self)
+{
+    return ll_ringbuffer_read_space(self->ring);
+}
+
+static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples)
+{
+    ll_ringbuffer_read(self->ring, buffer, samples);
+    return ALC_NO_ERROR;
+}
+
+
+typedef struct ALCportBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCportBackendFactory;
+#define ALCPORTBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCportBackendFactory, ALCbackendFactory) } }
+
+static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory *self);
+static void ALCportBackendFactory_deinit(ALCportBackendFactory *self);
+static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory *self, ALCbackend_Type type);
+static void ALCportBackendFactory_probe(ALCportBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCportBackendFactory);
+
+
+static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory* UNUSED(self))
+{
+    if(!pa_load())
+        return ALC_FALSE;
+    return ALC_TRUE;
+}
+
+static void ALCportBackendFactory_deinit(ALCportBackendFactory* UNUSED(self))
+{
+#ifdef HAVE_DYNLOAD
+    if(pa_handle)
+    {
+        Pa_Terminate();
+        CloseLib(pa_handle);
+        pa_handle = NULL;
+    }
+#else
+    Pa_Terminate();
+#endif
+}
+
+static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback || type == ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCportBackendFactory_probe(ALCportBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            AppendAllDevicesList(pa_device);
+            break;
+        case CAPTURE_DEVICE_PROBE:
+            AppendCaptureDeviceList(pa_device);
+            break;
+    }
+}
+
+static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCportPlayback *backend;
+        NEW_OBJ(backend, ALCportPlayback)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+    if(type == ALCbackend_Capture)
+    {
+        ALCportCapture *backend;
+        NEW_OBJ(backend, ALCportCapture)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}
+
+ALCbackendFactory *ALCportBackendFactory_getFactory(void)
+{
+    static ALCportBackendFactory factory = ALCPORTBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}

Alc/backends/qsa.c

@@ -0,0 +1,915 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2011-2013 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <sched.h>
+#include <errno.h>
+#include <memory.h>
+#include <sys/select.h>
+#include <sys/asoundlib.h>
+#include <sys/neutrino.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+
+#include "backends/base.h"
+
+
+typedef struct {
+    snd_pcm_t* pcmHandle;
+    int audio_fd;
+
+    snd_pcm_channel_setup_t  csetup;
+    snd_pcm_channel_params_t cparams;
+
+    ALvoid* buffer;
+    ALsizei size;
+
+    volatile int killNow;
+    althrd_t thread;
+} qsa_data;
+
+typedef struct {
+    ALCchar* name;
+    int card;
+    int dev;
+} DevMap;
+TYPEDEF_VECTOR(DevMap, vector_DevMap)
+
+static vector_DevMap DeviceNameMap;
+static vector_DevMap CaptureNameMap;
+
+static const ALCchar qsaDevice[] = "QSA Default";
+
+static const struct {
+    int32_t format;
+} formatlist[] = {
+    {SND_PCM_SFMT_FLOAT_LE},
+    {SND_PCM_SFMT_S32_LE},
+    {SND_PCM_SFMT_U32_LE},
+    {SND_PCM_SFMT_S16_LE},
+    {SND_PCM_SFMT_U16_LE},
+    {SND_PCM_SFMT_S8},
+    {SND_PCM_SFMT_U8},
+    {0},
+};
+
+static const struct {
+    int32_t rate;
+} ratelist[] = {
+    {192000},
+    {176400},
+    {96000},
+    {88200},
+    {48000},
+    {44100},
+    {32000},
+    {24000},
+    {22050},
+    {16000},
+    {12000},
+    {11025},
+    {8000},
+    {0},
+};
+
+static const struct {
+    int32_t channels;
+} channellist[] = {
+    {8},
+    {7},
+    {6},
+    {4},
+    {2},
+    {1},
+    {0},
+};
+
+static void deviceList(int type, vector_DevMap *devmap)
+{
+    snd_ctl_t* handle;
+    snd_pcm_info_t pcminfo;
+    int max_cards, card, err, dev;
+    DevMap entry;
+    char name[1024];
+    struct snd_ctl_hw_info info;
+
+    max_cards = snd_cards();
+    if(max_cards < 0)
+        return;
+
+    VECTOR_RESIZE(*devmap, 0, max_cards+1);
+
+    entry.name = strdup(qsaDevice);
+    entry.card = 0;
+    entry.dev = 0;
+    VECTOR_PUSH_BACK(*devmap, entry);
+
+    for(card = 0;card < max_cards;card++)
+    {
+        if((err=snd_ctl_open(&handle, card)) < 0)
+            continue;
+
+        if((err=snd_ctl_hw_info(handle, &info)) < 0)
+        {
+            snd_ctl_close(handle);
+            continue;
+        }
+
+        for(dev = 0;dev < (int)info.pcmdevs;dev++)
+        {
+            if((err=snd_ctl_pcm_info(handle, dev, &pcminfo)) < 0)
+                continue;
+
+            if((type==SND_PCM_CHANNEL_PLAYBACK && (pcminfo.flags&SND_PCM_INFO_PLAYBACK)) ||
+               (type==SND_PCM_CHANNEL_CAPTURE && (pcminfo.flags&SND_PCM_INFO_CAPTURE)))
+            {
+                snprintf(name, sizeof(name), "%s [%s] (hw:%d,%d)", info.name, pcminfo.name, card, dev);
+                entry.name = strdup(name);
+                entry.card = card;
+                entry.dev = dev;
+
+                VECTOR_PUSH_BACK(*devmap, entry);
+                TRACE("Got device \"%s\", card %d, dev %d\n", name, card, dev);
+            }
+        }
+        snd_ctl_close(handle);
+    }
+}
+
+
+FORCE_ALIGN static int qsa_proc_playback(void* ptr)
+{
+    ALCdevice* device=(ALCdevice*)ptr;
+    qsa_data* data=(qsa_data*)device->ExtraData;
+    snd_pcm_channel_status_t status;
+    struct sched_param param;
+    struct timeval timeout;
+    char* write_ptr;
+    fd_set wfds;
+    ALint len;
+    int sret;
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    /* Increase default 10 priority to 11 to avoid jerky sound */
+    SchedGet(0, 0, &param);
+    param.sched_priority=param.sched_curpriority+1;
+    SchedSet(0, 0, SCHED_NOCHANGE, &param);
+
+    const ALint frame_size = FrameSizeFromDevFmt(
+        device->FmtChans, device->FmtType, device->AmbiOrder
+    );
+
+    V0(device->Backend,lock)();
+    while(!data->killNow)
+    {
+        FD_ZERO(&wfds);
+        FD_SET(data->audio_fd, &wfds);
+        timeout.tv_sec=2;
+        timeout.tv_usec=0;
+
+        /* Select also works like time slice to OS */
+        V0(device->Backend,unlock)();
+        sret = select(data->audio_fd+1, NULL, &wfds, NULL, &timeout);
+        V0(device->Backend,lock)();
+        if(sret == -1)
+        {
+            ERR("select error: %s\n", strerror(errno));
+            aluHandleDisconnect(device);
+            break;
+        }
+        if(sret == 0)
+        {
+            ERR("select timeout\n");
+            continue;
+        }
+
+        len = data->size;
+        write_ptr = data->buffer;
+        aluMixData(device, write_ptr, len/frame_size);
+        while(len>0 && !data->killNow)
+        {
+            int wrote = snd_pcm_plugin_write(data->pcmHandle, write_ptr, len);
+            if(wrote <= 0)
+            {
+                if(errno==EAGAIN || errno==EWOULDBLOCK)
+                    continue;
+
+                memset(&status, 0, sizeof(status));
+                status.channel = SND_PCM_CHANNEL_PLAYBACK;
+
+                snd_pcm_plugin_status(data->pcmHandle, &status);
+
+                /* we need to reinitialize the sound channel if we've underrun the buffer */
+                if(status.status == SND_PCM_STATUS_UNDERRUN ||
+                   status.status == SND_PCM_STATUS_READY)
+                {
+                    if(snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK) < 0)
+                    {
+                        aluHandleDisconnect(device);
+                        break;
+                    }
+                }
+            }
+            else
+            {
+                write_ptr += wrote;
+                len -= wrote;
+            }
+        }
+    }
+    V0(device->Backend,unlock)();
+
+    return 0;
+}
+
+/************/
+/* Playback */
+/************/
+
+static ALCenum qsa_open_playback(ALCdevice* device, const ALCchar* deviceName)
+{
+    qsa_data *data;
+    int card, dev;
+    int status;
+
+    data = (qsa_data*)calloc(1, sizeof(qsa_data));
+    if(data == NULL)
+        return ALC_OUT_OF_MEMORY;
+
+    if(!deviceName)
+        deviceName = qsaDevice;
+
+    if(strcmp(deviceName, qsaDevice) == 0)
+        status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK);
+    else
+    {
+        const DevMap *iter;
+
+        if(VECTOR_SIZE(DeviceNameMap) == 0)
+            deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap);
+
+#define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0)
+        VECTOR_FIND_IF(iter, const DevMap, DeviceNameMap, MATCH_DEVNAME);
+#undef MATCH_DEVNAME
+        if(iter == VECTOR_END(DeviceNameMap))
+        {
+            free(data);
+            return ALC_INVALID_DEVICE;
+        }
+
+        status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_PLAYBACK);
+    }
+
+    if(status < 0)
+    {
+        free(data);
+        return ALC_INVALID_DEVICE;
+    }
+
+    data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK);
+    if(data->audio_fd < 0)
+    {
+        snd_pcm_close(data->pcmHandle);
+        free(data);
+        return ALC_INVALID_DEVICE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, deviceName);
+    device->ExtraData = data;
+
+    return ALC_NO_ERROR;
+}
+
+static void qsa_close_playback(ALCdevice* device)
+{
+    qsa_data* data=(qsa_data*)device->ExtraData;
+
+    if (data->buffer!=NULL)
+    {
+        free(data->buffer);
+        data->buffer=NULL;
+    }
+
+    snd_pcm_close(data->pcmHandle);
+    free(data);
+
+    device->ExtraData=NULL;
+}
+
+static ALCboolean qsa_reset_playback(ALCdevice* device)
+{
+    qsa_data* data=(qsa_data*)device->ExtraData;
+    int32_t format=-1;
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+             format=SND_PCM_SFMT_S8;
+             break;
+        case DevFmtUByte:
+             format=SND_PCM_SFMT_U8;
+             break;
+        case DevFmtShort:
+             format=SND_PCM_SFMT_S16_LE;
+             break;
+        case DevFmtUShort:
+             format=SND_PCM_SFMT_U16_LE;
+             break;
+        case DevFmtInt:
+             format=SND_PCM_SFMT_S32_LE;
+             break;
+        case DevFmtUInt:
+             format=SND_PCM_SFMT_U32_LE;
+             break;
+        case DevFmtFloat:
+             format=SND_PCM_SFMT_FLOAT_LE;
+             break;
+    }
+
+    /* we actually don't want to block on writes */
+    snd_pcm_nonblock_mode(data->pcmHandle, 1);
+    /* Disable mmap to control data transfer to the audio device */
+    snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP);
+    snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_BUFFER_PARTIAL_BLOCKS);
+
+    // configure a sound channel
+    memset(&data->cparams, 0, sizeof(data->cparams));
+    data->cparams.channel=SND_PCM_CHANNEL_PLAYBACK;
+    data->cparams.mode=SND_PCM_MODE_BLOCK;
+    data->cparams.start_mode=SND_PCM_START_FULL;
+    data->cparams.stop_mode=SND_PCM_STOP_STOP;
+
+    data->cparams.buf.block.frag_size=device->UpdateSize *
+        FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    data->cparams.buf.block.frags_max=device->NumUpdates;
+    data->cparams.buf.block.frags_min=device->NumUpdates;
+
+    data->cparams.format.interleave=1;
+    data->cparams.format.rate=device->Frequency;
+    data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    data->cparams.format.format=format;
+
+    if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
+    {
+        int original_rate=data->cparams.format.rate;
+        int original_voices=data->cparams.format.voices;
+        int original_format=data->cparams.format.format;
+        int it;
+        int jt;
+
+        for (it=0; it<1; it++)
+        {
+            /* Check for second pass */
+            if (it==1)
+            {
+                original_rate=ratelist[0].rate;
+                original_voices=channellist[0].channels;
+                original_format=formatlist[0].format;
+            }
+
+            do {
+                /* At first downgrade sample format */
+                jt=0;
+                do {
+                    if (formatlist[jt].format==data->cparams.format.format)
+                    {
+                        data->cparams.format.format=formatlist[jt+1].format;
+                        break;
+                    }
+                    if (formatlist[jt].format==0)
+                    {
+                        data->cparams.format.format=0;
+                        break;
+                    }
+                    jt++;
+                } while(1);
+
+                if (data->cparams.format.format==0)
+                {
+                    data->cparams.format.format=original_format;
+
+                    /* At secod downgrade sample rate */
+                    jt=0;
+                    do {
+                        if (ratelist[jt].rate==data->cparams.format.rate)
+                        {
+                            data->cparams.format.rate=ratelist[jt+1].rate;
+                            break;
+                        }
+                        if (ratelist[jt].rate==0)
+                        {
+                            data->cparams.format.rate=0;
+                            break;
+                        }
+                        jt++;
+                    } while(1);
+
+                    if (data->cparams.format.rate==0)
+                    {
+                        data->cparams.format.rate=original_rate;
+                        data->cparams.format.format=original_format;
+
+                        /* At third downgrade channels number */
+                        jt=0;
+                        do {
+                            if(channellist[jt].channels==data->cparams.format.voices)
+                            {
+                                data->cparams.format.voices=channellist[jt+1].channels;
+                                break;
+                            }
+                            if (channellist[jt].channels==0)
+                            {
+                                data->cparams.format.voices=0;
+                                break;
+                            }
+                           jt++;
+                        } while(1);
+                    }
+
+                    if (data->cparams.format.voices==0)
+                    {
+                        break;
+                    }
+                }
+
+                data->cparams.buf.block.frag_size=device->UpdateSize*
+                    data->cparams.format.voices*
+                    snd_pcm_format_width(data->cparams.format.format)/8;
+                data->cparams.buf.block.frags_max=device->NumUpdates;
+                data->cparams.buf.block.frags_min=device->NumUpdates;
+                if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
+                {
+                    continue;
+                }
+                else
+                {
+                    break;
+                }
+            } while(1);
+
+            if (data->cparams.format.voices!=0)
+            {
+                break;
+            }
+        }
+
+        if (data->cparams.format.voices==0)
+        {
+            return ALC_FALSE;
+        }
+    }
+
+    if ((snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK))<0)
+    {
+        return ALC_FALSE;
+    }
+
+    memset(&data->csetup, 0, sizeof(data->csetup));
+    data->csetup.channel=SND_PCM_CHANNEL_PLAYBACK;
+    if (snd_pcm_plugin_setup(data->pcmHandle, &data->csetup)<0)
+    {
+        return ALC_FALSE;
+    }
+
+    /* now fill back to the our AL device */
+    device->Frequency=data->cparams.format.rate;
+
+    switch (data->cparams.format.voices)
+    {
+        case 1:
+             device->FmtChans=DevFmtMono;
+             break;
+        case 2:
+             device->FmtChans=DevFmtStereo;
+             break;
+        case 4:
+             device->FmtChans=DevFmtQuad;
+             break;
+        case 6:
+             device->FmtChans=DevFmtX51;
+             break;
+        case 7:
+             device->FmtChans=DevFmtX61;
+             break;
+        case 8:
+             device->FmtChans=DevFmtX71;
+             break;
+        default:
+             device->FmtChans=DevFmtMono;
+             break;
+    }
+
+    switch (data->cparams.format.format)
+    {
+        case SND_PCM_SFMT_S8:
+             device->FmtType=DevFmtByte;
+             break;
+        case SND_PCM_SFMT_U8:
+             device->FmtType=DevFmtUByte;
+             break;
+        case SND_PCM_SFMT_S16_LE:
+             device->FmtType=DevFmtShort;
+             break;
+        case SND_PCM_SFMT_U16_LE:
+             device->FmtType=DevFmtUShort;
+             break;
+        case SND_PCM_SFMT_S32_LE:
+             device->FmtType=DevFmtInt;
+             break;
+        case SND_PCM_SFMT_U32_LE:
+             device->FmtType=DevFmtUInt;
+             break;
+        case SND_PCM_SFMT_FLOAT_LE:
+             device->FmtType=DevFmtFloat;
+             break;
+        default:
+             device->FmtType=DevFmtShort;
+             break;
+    }
+
+    SetDefaultChannelOrder(device);
+
+    device->UpdateSize=data->csetup.buf.block.frag_size/
+        FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    device->NumUpdates=data->csetup.buf.block.frags;
+
+    data->size=data->csetup.buf.block.frag_size;
+    data->buffer=malloc(data->size);
+    if (!data->buffer)
+    {
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static ALCboolean qsa_start_playback(ALCdevice* device)
+{
+    qsa_data *data = (qsa_data*)device->ExtraData;
+
+    data->killNow = 0;
+    if(althrd_create(&data->thread, qsa_proc_playback, device) != althrd_success)
+        return ALC_FALSE;
+
+    return ALC_TRUE;
+}
+
+static void qsa_stop_playback(ALCdevice* device)
+{
+    qsa_data *data = (qsa_data*)device->ExtraData;
+    int res;
+
+    if(data->killNow)
+        return;
+
+    data->killNow = 1;
+    althrd_join(data->thread, &res);
+}
+
+/***********/
+/* Capture */
+/***********/
+
+static ALCenum qsa_open_capture(ALCdevice* device, const ALCchar* deviceName)
+{
+    qsa_data *data;
+    int card, dev;
+    int format=-1;
+    int status;
+
+    data=(qsa_data*)calloc(1, sizeof(qsa_data));
+    if (data==NULL)
+    {
+        return ALC_OUT_OF_MEMORY;
+    }
+
+    if(!deviceName)
+        deviceName = qsaDevice;
+
+    if(strcmp(deviceName, qsaDevice) == 0)
+        status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE);
+    else
+    {
+        const DevMap *iter;
+
+        if(VECTOR_SIZE(CaptureNameMap) == 0)
+            deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap);
+
+#define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0)
+        VECTOR_FIND_IF(iter, const DevMap, CaptureNameMap, MATCH_DEVNAME);
+#undef MATCH_DEVNAME
+        if(iter == VECTOR_END(CaptureNameMap))
+        {
+            free(data);
+            return ALC_INVALID_DEVICE;
+        }
+
+        status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_CAPTURE);
+    }
+
+    if(status < 0)
+    {
+        free(data);
+        return ALC_INVALID_DEVICE;
+    }
+
+    data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE);
+    if(data->audio_fd < 0)
+    {
+        snd_pcm_close(data->pcmHandle);
+        free(data);
+        return ALC_INVALID_DEVICE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, deviceName);
+    device->ExtraData = data;
+
+    switch (device->FmtType)
+    {
+        case DevFmtByte:
+             format=SND_PCM_SFMT_S8;
+             break;
+        case DevFmtUByte:
+             format=SND_PCM_SFMT_U8;
+             break;
+        case DevFmtShort:
+             format=SND_PCM_SFMT_S16_LE;
+             break;
+        case DevFmtUShort:
+             format=SND_PCM_SFMT_U16_LE;
+             break;
+        case DevFmtInt:
+             format=SND_PCM_SFMT_S32_LE;
+             break;
+        case DevFmtUInt:
+             format=SND_PCM_SFMT_U32_LE;
+             break;
+        case DevFmtFloat:
+             format=SND_PCM_SFMT_FLOAT_LE;
+             break;
+    }
+
+    /* we actually don't want to block on reads */
+    snd_pcm_nonblock_mode(data->pcmHandle, 1);
+    /* Disable mmap to control data transfer to the audio device */
+    snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP);
+
+    /* configure a sound channel */
+    memset(&data->cparams, 0, sizeof(data->cparams));
+    data->cparams.mode=SND_PCM_MODE_BLOCK;
+    data->cparams.channel=SND_PCM_CHANNEL_CAPTURE;
+    data->cparams.start_mode=SND_PCM_START_GO;
+    data->cparams.stop_mode=SND_PCM_STOP_STOP;
+
+    data->cparams.buf.block.frag_size=device->UpdateSize*
+        FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    data->cparams.buf.block.frags_max=device->NumUpdates;
+    data->cparams.buf.block.frags_min=device->NumUpdates;
+
+    data->cparams.format.interleave=1;
+    data->cparams.format.rate=device->Frequency;
+    data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    data->cparams.format.format=format;
+
+    if(snd_pcm_plugin_params(data->pcmHandle, &data->cparams) < 0)
+    {
+        snd_pcm_close(data->pcmHandle);
+        free(data);
+        device->ExtraData=NULL;
+
+        return ALC_INVALID_VALUE;
+    }
+
+    return ALC_NO_ERROR;
+}
+
+static void qsa_close_capture(ALCdevice* device)
+{
+    qsa_data* data=(qsa_data*)device->ExtraData;
+
+    if (data->pcmHandle!=NULL)
+        snd_pcm_close(data->pcmHandle);
+
+    free(data);
+    device->ExtraData=NULL;
+}
+
+static void qsa_start_capture(ALCdevice* device)
+{
+    qsa_data* data=(qsa_data*)device->ExtraData;
+    int rstatus;
+
+    if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0)
+    {
+        ERR("capture prepare failed: %s\n", snd_strerror(rstatus));
+        return;
+    }
+
+    memset(&data->csetup, 0, sizeof(data->csetup));
+    data->csetup.channel=SND_PCM_CHANNEL_CAPTURE;
+    if ((rstatus=snd_pcm_plugin_setup(data->pcmHandle, &data->csetup))<0)
+    {
+        ERR("capture setup failed: %s\n", snd_strerror(rstatus));
+        return;
+    }
+
+    snd_pcm_capture_go(data->pcmHandle);
+}
+
+static void qsa_stop_capture(ALCdevice* device)
+{
+    qsa_data* data=(qsa_data*)device->ExtraData;
+
+    snd_pcm_capture_flush(data->pcmHandle);
+}
+
+static ALCuint qsa_available_samples(ALCdevice* device)
+{
+    qsa_data* data=(qsa_data*)device->ExtraData;
+    snd_pcm_channel_status_t status;
+    ALint frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    ALint free_size;
+    int rstatus;
+
+    memset(&status, 0, sizeof (status));
+    status.channel=SND_PCM_CHANNEL_CAPTURE;
+    snd_pcm_plugin_status(data->pcmHandle, &status);
+    if ((status.status==SND_PCM_STATUS_OVERRUN) ||
+        (status.status==SND_PCM_STATUS_READY))
+    {
+        if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0)
+        {
+            ERR("capture prepare failed: %s\n", snd_strerror(rstatus));
+            aluHandleDisconnect(device);
+            return 0;
+        }
+
+        snd_pcm_capture_go(data->pcmHandle);
+        return 0;
+    }
+
+    free_size=data->csetup.buf.block.frag_size*data->csetup.buf.block.frags;
+    free_size-=status.free;
+
+    return free_size/frame_size;
+}
+
+static ALCenum qsa_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
+{
+    qsa_data* data=(qsa_data*)device->ExtraData;
+    char* read_ptr;
+    snd_pcm_channel_status_t status;
+    fd_set rfds;
+    int selectret;
+    struct timeval timeout;
+    int bytes_read;
+    ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    ALint len=samples*frame_size;
+    int rstatus;
+
+    read_ptr=buffer;
+
+    while (len>0)
+    {
+        FD_ZERO(&rfds);
+        FD_SET(data->audio_fd, &rfds);
+        timeout.tv_sec=2;
+        timeout.tv_usec=0;
+
+        /* Select also works like time slice to OS */
+        bytes_read=0;
+        selectret=select(data->audio_fd+1, &rfds, NULL, NULL, &timeout);
+        switch (selectret)
+        {
+            case -1:
+                 aluHandleDisconnect(device);
+                 return ALC_INVALID_DEVICE;
+            case 0:
+                 break;
+            default:
+                 if (FD_ISSET(data->audio_fd, &rfds))
+                 {
+                     bytes_read=snd_pcm_plugin_read(data->pcmHandle, read_ptr, len);
+                     break;
+                 }
+                 break;
+        }
+
+        if (bytes_read<=0)
+        {
+            if ((errno==EAGAIN) || (errno==EWOULDBLOCK))
+            {
+                continue;
+            }
+
+            memset(&status, 0, sizeof (status));
+            status.channel=SND_PCM_CHANNEL_CAPTURE;
+            snd_pcm_plugin_status(data->pcmHandle, &status);
+
+            /* we need to reinitialize the sound channel if we've overrun the buffer */
+            if ((status.status==SND_PCM_STATUS_OVERRUN) ||
+                (status.status==SND_PCM_STATUS_READY))
+            {
+                if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0)
+                {
+                    ERR("capture prepare failed: %s\n", snd_strerror(rstatus));
+                    aluHandleDisconnect(device);
+                    return ALC_INVALID_DEVICE;
+                }
+                snd_pcm_capture_go(data->pcmHandle);
+            }
+        }
+        else
+        {
+            read_ptr+=bytes_read;
+            len-=bytes_read;
+        }
+    }
+
+    return ALC_NO_ERROR;
+}
+
+static const BackendFuncs qsa_funcs= {
+    qsa_open_playback,
+    qsa_close_playback,
+    qsa_reset_playback,
+    qsa_start_playback,
+    qsa_stop_playback,
+    qsa_open_capture,
+    qsa_close_capture,
+    qsa_start_capture,
+    qsa_stop_capture,
+    qsa_capture_samples,
+    qsa_available_samples
+};
+
+ALCboolean alc_qsa_init(BackendFuncs* func_list)
+{
+    *func_list = qsa_funcs;
+    return ALC_TRUE;
+}
+
+void alc_qsa_deinit(void)
+{
+#define FREE_NAME(iter) free((iter)->name)
+    VECTOR_FOR_EACH(DevMap, DeviceNameMap, FREE_NAME);
+    VECTOR_DEINIT(DeviceNameMap);
+
+    VECTOR_FOR_EACH(DevMap, CaptureNameMap, FREE_NAME);
+    VECTOR_DEINIT(CaptureNameMap);
+#undef FREE_NAME
+}
+
+void alc_qsa_probe(enum DevProbe type)
+{
+    switch (type)
+    {
+        case ALL_DEVICE_PROBE:
+#define FREE_NAME(iter) free((iter)->name)
+            VECTOR_FOR_EACH(DevMap, DeviceNameMap, FREE_NAME);
+            VECTOR_RESIZE(DeviceNameMap, 0, 0);
+#undef FREE_NAME
+
+            deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap);
+#define APPEND_DEVICE(iter) AppendAllDevicesList((iter)->name)
+            VECTOR_FOR_EACH(const DevMap, DeviceNameMap, APPEND_DEVICE);
+#undef APPEND_DEVICE
+            break;
+
+        case CAPTURE_DEVICE_PROBE:
+#define FREE_NAME(iter) free((iter)->name)
+            VECTOR_FOR_EACH(DevMap, CaptureNameMap, FREE_NAME);
+            VECTOR_RESIZE(CaptureNameMap, 0, 0);
+#undef FREE_NAME
+
+            deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap);
+#define APPEND_DEVICE(iter) AppendCaptureDeviceList((iter)->name)
+            VECTOR_FOR_EACH(const DevMap, CaptureNameMap, APPEND_DEVICE);
+#undef APPEND_DEVICE
+            break;
+    }
+}

Alc/backends/sndio.c

@@ -0,0 +1,346 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+
+#include "backends/base.h"
+
+#include <sndio.h>
+
+
+
+
+typedef struct ALCsndioBackend {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    struct sio_hdl *sndHandle;
+
+    ALvoid *mix_data;
+    ALsizei data_size;
+
+    volatile int killNow;
+    althrd_t thread;
+} ALCsndioBackend;
+
+static int ALCsndioBackend_mixerProc(void *ptr);
+
+static void ALCsndioBackend_Construct(ALCsndioBackend *self, ALCdevice *device);
+static void ALCsndioBackend_Destruct(ALCsndioBackend *self);
+static ALCenum ALCsndioBackend_open(ALCsndioBackend *self, const ALCchar *name);
+static void ALCsndioBackend_close(ALCsndioBackend *self);
+static ALCboolean ALCsndioBackend_reset(ALCsndioBackend *self);
+static ALCboolean ALCsndioBackend_start(ALCsndioBackend *self);
+static void ALCsndioBackend_stop(ALCsndioBackend *self);
+static DECLARE_FORWARD2(ALCsndioBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCsndioBackend)
+
+DEFINE_ALCBACKEND_VTABLE(ALCsndioBackend);
+
+
+static const ALCchar sndio_device[] = "SndIO Default";
+
+
+static void ALCsndioBackend_Construct(ALCsndioBackend *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCsndioBackend, ALCbackend, self);
+}
+
+static void ALCsndioBackend_Destruct(ALCsndioBackend *self)
+{
+    if(self->sndHandle)
+        sio_close(self->sndHandle);
+    self->sndHandle = NULL;
+
+    al_free(self->mix_data);
+    self->mix_data = NULL;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static int ALCsndioBackend_mixerProc(void *ptr)
+{
+    ALCsndioBackend *self = (ALCsndioBackend*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ALsizei frameSize;
+    size_t wrote;
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    while(!self->killNow && device->Connected)
+    {
+        ALsizei len = self->data_size;
+        ALubyte *WritePtr = self->mix_data;
+
+        ALCsndioBackend_lock(self);
+        aluMixData(device, WritePtr, len/frameSize);
+        ALCsndioBackend_unlock(self);
+        while(len > 0 && !self->killNow)
+        {
+            wrote = sio_write(self->sndHandle, WritePtr, len);
+            if(wrote == 0)
+            {
+                ERR("sio_write failed\n");
+                ALCdevice_Lock(device);
+                aluHandleDisconnect(device);
+                ALCdevice_Unlock(device);
+                break;
+            }
+
+            len -= wrote;
+            WritePtr += wrote;
+        }
+    }
+
+    return 0;
+}
+
+
+static ALCenum ALCsndioBackend_open(ALCsndioBackend *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+
+    if(!name)
+        name = sndio_device;
+    else if(strcmp(name, sndio_device) != 0)
+        return ALC_INVALID_VALUE;
+
+    self->sndHandle = sio_open(NULL, SIO_PLAY, 0);
+    if(self->sndHandle == NULL)
+    {
+        ERR("Could not open device\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCsndioBackend_close(ALCsndioBackend *self)
+{
+    sio_close(self->sndHandle);
+    self->sndHandle = NULL;
+}
+
+static ALCboolean ALCsndioBackend_reset(ALCsndioBackend *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    struct sio_par par;
+
+    sio_initpar(&par);
+
+    par.rate = device->Frequency;
+    par.pchan = ((device->FmtChans != DevFmtMono) ? 2 : 1);
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            par.bits = 8;
+            par.sig = 1;
+            break;
+        case DevFmtUByte:
+            par.bits = 8;
+            par.sig = 0;
+            break;
+        case DevFmtFloat:
+        case DevFmtShort:
+            par.bits = 16;
+            par.sig = 1;
+            break;
+        case DevFmtUShort:
+            par.bits = 16;
+            par.sig = 0;
+            break;
+        case DevFmtInt:
+            par.bits = 32;
+            par.sig = 1;
+            break;
+        case DevFmtUInt:
+            par.bits = 32;
+            par.sig = 0;
+            break;
+    }
+    par.le = SIO_LE_NATIVE;
+
+    par.round = device->UpdateSize;
+    par.appbufsz = device->UpdateSize * (device->NumUpdates-1);
+    if(!par.appbufsz) par.appbufsz = device->UpdateSize;
+
+    if(!sio_setpar(self->sndHandle, &par) || !sio_getpar(self->sndHandle, &par))
+    {
+        ERR("Failed to set device parameters\n");
+        return ALC_FALSE;
+    }
+
+    if(par.bits != par.bps*8)
+    {
+        ERR("Padded samples not supported (%u of %u bits)\n", par.bits, par.bps*8);
+        return ALC_FALSE;
+    }
+
+    device->Frequency = par.rate;
+    device->FmtChans = ((par.pchan==1) ? DevFmtMono : DevFmtStereo);
+
+    if(par.bits == 8 && par.sig == 1)
+        device->FmtType = DevFmtByte;
+    else if(par.bits == 8 && par.sig == 0)
+        device->FmtType = DevFmtUByte;
+    else if(par.bits == 16 && par.sig == 1)
+        device->FmtType = DevFmtShort;
+    else if(par.bits == 16 && par.sig == 0)
+        device->FmtType = DevFmtUShort;
+    else if(par.bits == 32 && par.sig == 1)
+        device->FmtType = DevFmtInt;
+    else if(par.bits == 32 && par.sig == 0)
+        device->FmtType = DevFmtUInt;
+    else
+    {
+        ERR("Unhandled sample format: %s %u-bit\n", (par.sig?"signed":"unsigned"), par.bits);
+        return ALC_FALSE;
+    }
+
+    device->UpdateSize = par.round;
+    device->NumUpdates = (par.bufsz/par.round) + 1;
+
+    SetDefaultChannelOrder(device);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCsndioBackend_start(ALCsndioBackend *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+
+    self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
+        device->FmtChans, device->FmtType, device->AmbiOrder
+    );
+    al_free(self->mix_data);
+    self->mix_data = al_calloc(16, self->data_size);
+
+    if(!sio_start(self->sndHandle))
+    {
+        ERR("Error starting playback\n");
+        return ALC_FALSE;
+    }
+
+    self->killNow = 0;
+    if(althrd_create(&self->thread, ALCsndioBackend_mixerProc, self) != althrd_success)
+    {
+        sio_stop(self->sndHandle);
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCsndioBackend_stop(ALCsndioBackend *self)
+{
+    int res;
+
+    if(self->killNow)
+        return;
+
+    self->killNow = 1;
+    althrd_join(self->thread, &res);
+
+    if(!sio_stop(self->sndHandle))
+        ERR("Error stopping device\n");
+
+    al_free(self->mix_data);
+    self->mix_data = NULL;
+}
+
+
+typedef struct ALCsndioBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCsndioBackendFactory;
+#define ALCSNDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsndioBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCsndioBackendFactory_getFactory(void);
+
+static ALCboolean ALCsndioBackendFactory_init(ALCsndioBackendFactory *self);
+static DECLARE_FORWARD(ALCsndioBackendFactory, ALCbackendFactory, void, deinit)
+static ALCboolean ALCsndioBackendFactory_querySupport(ALCsndioBackendFactory *self, ALCbackend_Type type);
+static void ALCsndioBackendFactory_probe(ALCsndioBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCsndioBackendFactory_createBackend(ALCsndioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsndioBackendFactory);
+
+
+ALCbackendFactory *ALCsndioBackendFactory_getFactory(void)
+{
+    static ALCsndioBackendFactory factory = ALCSNDIOBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCsndioBackendFactory_init(ALCsndioBackendFactory* UNUSED(self))
+{
+    /* No dynamic loading */
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCsndioBackendFactory_querySupport(ALCsndioBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCsndioBackendFactory_probe(ALCsndioBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            AppendAllDevicesList(sndio_device);
+            break;
+        case CAPTURE_DEVICE_PROBE:
+            break;
+    }
+}
+
+static ALCbackend* ALCsndioBackendFactory_createBackend(ALCsndioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCsndioBackend *backend;
+        NEW_OBJ(backend, ALCsndioBackend)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/solaris.c

@@ -0,0 +1,363 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <sys/ioctl.h>
+#include <sys/types.h>
+#include <sys/time.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <memory.h>
+#include <unistd.h>
+#include <errno.h>
+#include <math.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+#include "compat.h"
+
+#include "backends/base.h"
+
+#include <sys/audioio.h>
+
+
+typedef struct ALCsolarisBackend {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    int fd;
+
+    ALubyte *mix_data;
+    int data_size;
+
+    ATOMIC(ALenum) killNow;
+    althrd_t thread;
+} ALCsolarisBackend;
+
+static int ALCsolarisBackend_mixerProc(void *ptr);
+
+static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device);
+static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self);
+static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name);
+static void ALCsolarisBackend_close(ALCsolarisBackend *self);
+static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self);
+static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self);
+static void ALCsolarisBackend_stop(ALCsolarisBackend *self);
+static DECLARE_FORWARD2(ALCsolarisBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCsolarisBackend)
+
+DEFINE_ALCBACKEND_VTABLE(ALCsolarisBackend);
+
+
+static const ALCchar solaris_device[] = "Solaris Default";
+
+static const char *solaris_driver = "/dev/audio";
+
+
+static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCsolarisBackend, ALCbackend, self);
+
+    self->fd = -1;
+    ATOMIC_INIT(&self->killNow, AL_FALSE);
+}
+
+static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self)
+{
+    if(self->fd != -1)
+        close(self->fd);
+    self->fd = -1;
+
+    free(self->mix_data);
+    self->mix_data = NULL;
+    self->data_size = 0;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static int ALCsolarisBackend_mixerProc(void *ptr)
+{
+    ALCsolarisBackend *self = ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    struct timeval timeout;
+    ALubyte *write_ptr;
+    ALint frame_size;
+    ALint to_write;
+    ssize_t wrote;
+    fd_set wfds;
+    int sret;
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    ALCsolarisBackend_lock(self);
+    while(!ATOMIC_LOAD_SEQ(&self->killNow) && device->Connected)
+    {
+        FD_ZERO(&wfds);
+        FD_SET(self->fd, &wfds);
+        timeout.tv_sec = 1;
+        timeout.tv_usec = 0;
+
+        ALCsolarisBackend_unlock(self);
+        sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
+        ALCsolarisBackend_lock(self);
+        if(sret < 0)
+        {
+            if(errno == EINTR)
+                continue;
+            ERR("select failed: %s\n", strerror(errno));
+            aluHandleDisconnect(device);
+            break;
+        }
+        else if(sret == 0)
+        {
+            WARN("select timeout\n");
+            continue;
+        }
+
+        write_ptr = self->mix_data;
+        to_write = self->data_size;
+        aluMixData(device, write_ptr, to_write/frame_size);
+        while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow))
+        {
+            wrote = write(self->fd, write_ptr, to_write);
+            if(wrote < 0)
+            {
+                if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
+                    continue;
+                ERR("write failed: %s\n", strerror(errno));
+                aluHandleDisconnect(device);
+                break;
+            }
+
+            to_write -= wrote;
+            write_ptr += wrote;
+        }
+    }
+    ALCsolarisBackend_unlock(self);
+
+    return 0;
+}
+
+
+static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name)
+{
+    ALCdevice *device;
+
+    if(!name)
+        name = solaris_device;
+    else if(strcmp(name, solaris_device) != 0)
+        return ALC_INVALID_VALUE;
+
+    self->fd = open(solaris_driver, O_WRONLY);
+    if(self->fd == -1)
+    {
+        ERR("Could not open %s: %s\n", solaris_driver, strerror(errno));
+        return ALC_INVALID_VALUE;
+    }
+
+    device = STATIC_CAST(ALCbackend,self)->mDevice;
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCsolarisBackend_close(ALCsolarisBackend *self)
+{
+    close(self->fd);
+    self->fd = -1;
+}
+
+static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    audio_info_t info;
+    ALsizei frameSize;
+    ALsizei numChannels;
+
+    AUDIO_INITINFO(&info);
+
+    info.play.sample_rate = device->Frequency;
+
+    if(device->FmtChans != DevFmtMono)
+        device->FmtChans = DevFmtStereo;
+    numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    info.play.channels = numChannels;
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            info.play.precision = 8;
+            info.play.encoding = AUDIO_ENCODING_LINEAR;
+            break;
+        case DevFmtUByte:
+            info.play.precision = 8;
+            info.play.encoding = AUDIO_ENCODING_LINEAR8;
+            break;
+        case DevFmtUShort:
+        case DevFmtInt:
+        case DevFmtUInt:
+        case DevFmtFloat:
+            device->FmtType = DevFmtShort;
+            /* fall-through */
+        case DevFmtShort:
+            info.play.precision = 16;
+            info.play.encoding = AUDIO_ENCODING_LINEAR;
+            break;
+    }
+
+    frameSize = numChannels * BytesFromDevFmt(device->FmtType);
+    info.play.buffer_size = device->UpdateSize*device->NumUpdates * frameSize;
+
+    if(ioctl(self->fd, AUDIO_SETINFO, &info) < 0)
+    {
+        ERR("ioctl failed: %s\n", strerror(errno));
+        return ALC_FALSE;
+    }
+
+    if(ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)info.play.channels)
+    {
+        ERR("Failed to set %s, got %u channels instead\n", DevFmtChannelsString(device->FmtChans), info.play.channels);
+        return ALC_FALSE;
+    }
+
+    if(!((info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR8 && device->FmtType == DevFmtUByte) ||
+         (info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtByte) ||
+         (info.play.precision == 16 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtShort) ||
+         (info.play.precision == 32 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtInt)))
+    {
+        ERR("Could not set %s samples, got %d (0x%x)\n", DevFmtTypeString(device->FmtType),
+            info.play.precision, info.play.encoding);
+        return ALC_FALSE;
+    }
+
+    device->Frequency = info.play.sample_rate;
+    device->UpdateSize = (info.play.buffer_size/device->NumUpdates) + 1;
+
+    SetDefaultChannelOrder(device);
+
+    free(self->mix_data);
+    self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
+        device->FmtChans, device->FmtType, device->AmbiOrder
+    );
+    self->mix_data = calloc(1, self->data_size);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self)
+{
+    ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
+    if(althrd_create(&self->thread, ALCsolarisBackend_mixerProc, self) != althrd_success)
+        return ALC_FALSE;
+    return ALC_TRUE;
+}
+
+static void ALCsolarisBackend_stop(ALCsolarisBackend *self)
+{
+    int res;
+
+    if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
+        return;
+
+    althrd_join(self->thread, &res);
+
+    if(ioctl(self->fd, AUDIO_DRAIN) < 0)
+        ERR("Error draining device: %s\n", strerror(errno));
+}
+
+
+typedef struct ALCsolarisBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCsolarisBackendFactory;
+#define ALCSOLARISBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsolarisBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void);
+
+static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory *self);
+static DECLARE_FORWARD(ALCsolarisBackendFactory, ALCbackendFactory, void, deinit)
+static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory *self, ALCbackend_Type type);
+static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsolarisBackendFactory);
+
+
+ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void)
+{
+    static ALCsolarisBackendFactory factory = ALCSOLARISBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory* UNUSED(self))
+{
+    ConfigValueStr(NULL, "solaris", "device", &solaris_driver);
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+        {
+#ifdef HAVE_STAT
+            struct stat buf;
+            if(stat(solaris_driver, &buf) == 0)
+#endif
+                AppendAllDevicesList(solaris_device);
+        }
+        break;
+
+        case CAPTURE_DEVICE_PROBE:
+            break;
+    }
+}
+
+ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCsolarisBackend *backend;
+        NEW_OBJ(backend, ALCsolarisBackend)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/wave.c

@@ -0,0 +1,452 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <memory.h>
+#include <errno.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+#include "compat.h"
+
+#include "backends/base.h"
+
+
+static const ALCchar waveDevice[] = "Wave File Writer";
+
+static const ALubyte SUBTYPE_PCM[] = {
+    0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa,
+    0x00, 0x38, 0x9b, 0x71
+};
+static const ALubyte SUBTYPE_FLOAT[] = {
+    0x03, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa,
+    0x00, 0x38, 0x9b, 0x71
+};
+
+static const ALubyte SUBTYPE_BFORMAT_PCM[] = {
+    0x01, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1,
+    0xca, 0x00, 0x00, 0x00
+};
+
+static const ALubyte SUBTYPE_BFORMAT_FLOAT[] = {
+    0x03, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1,
+    0xca, 0x00, 0x00, 0x00
+};
+
+static void fwrite16le(ALushort val, FILE *f)
+{
+    ALubyte data[2] = { val&0xff, (val>>8)&0xff };
+    fwrite(data, 1, 2, f);
+}
+
+static void fwrite32le(ALuint val, FILE *f)
+{
+    ALubyte data[4] = { val&0xff, (val>>8)&0xff, (val>>16)&0xff, (val>>24)&0xff };
+    fwrite(data, 1, 4, f);
+}
+
+
+typedef struct ALCwaveBackend {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    FILE *mFile;
+    long mDataStart;
+
+    ALvoid *mBuffer;
+    ALuint mSize;
+
+    volatile int killNow;
+    althrd_t thread;
+} ALCwaveBackend;
+
+static int ALCwaveBackend_mixerProc(void *ptr);
+
+static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device);
+static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, Destruct)
+static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name);
+static void ALCwaveBackend_close(ALCwaveBackend *self);
+static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self);
+static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self);
+static void ALCwaveBackend_stop(ALCwaveBackend *self);
+static DECLARE_FORWARD2(ALCwaveBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCwaveBackend)
+
+DEFINE_ALCBACKEND_VTABLE(ALCwaveBackend);
+
+
+static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCwaveBackend, ALCbackend, self);
+
+    self->mFile = NULL;
+    self->mDataStart = -1;
+
+    self->mBuffer = NULL;
+    self->mSize = 0;
+
+    self->killNow = 1;
+}
+
+
+static int ALCwaveBackend_mixerProc(void *ptr)
+{
+    ALCwaveBackend *self = (ALCwaveBackend*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    struct timespec now, start;
+    ALint64 avail, done;
+    ALuint frameSize;
+    size_t fs;
+    const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 /
+                                 device->Frequency / 2);
+
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    done = 0;
+    if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
+    {
+        ERR("Failed to get starting time\n");
+        return 1;
+    }
+    while(!self->killNow && device->Connected)
+    {
+        if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC)
+        {
+            ERR("Failed to get current time\n");
+            return 1;
+        }
+
+        avail  = (now.tv_sec - start.tv_sec) * device->Frequency;
+        avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
+        if(avail < done)
+        {
+            /* Oops, time skipped backwards. Reset the number of samples done
+             * with one update available since we (likely) just came back from
+             * sleeping. */
+            done = avail - device->UpdateSize;
+        }
+
+        if(avail-done < device->UpdateSize)
+            al_nssleep(restTime);
+        else while(avail-done >= device->UpdateSize)
+        {
+            ALCwaveBackend_lock(self);
+            aluMixData(device, self->mBuffer, device->UpdateSize);
+            ALCwaveBackend_unlock(self);
+            done += device->UpdateSize;
+
+            if(!IS_LITTLE_ENDIAN)
+            {
+                ALuint bytesize = BytesFromDevFmt(device->FmtType);
+                ALuint i;
+
+                if(bytesize == 2)
+                {
+                    ALushort *samples = self->mBuffer;
+                    ALuint len = self->mSize / 2;
+                    for(i = 0;i < len;i++)
+                    {
+                        ALushort samp = samples[i];
+                        samples[i] = (samp>>8) | (samp<<8);
+                    }
+                }
+                else if(bytesize == 4)
+                {
+                    ALuint *samples = self->mBuffer;
+                    ALuint len = self->mSize / 4;
+                    for(i = 0;i < len;i++)
+                    {
+                        ALuint samp = samples[i];
+                        samples[i] = (samp>>24) | ((samp>>8)&0x0000ff00) |
+                                     ((samp<<8)&0x00ff0000) | (samp<<24);
+                    }
+                }
+            }
+
+            fs = fwrite(self->mBuffer, frameSize, device->UpdateSize, self->mFile);
+            (void)fs;
+            if(ferror(self->mFile))
+            {
+                ERR("Error writing to file\n");
+                ALCdevice_Lock(device);
+                aluHandleDisconnect(device);
+                ALCdevice_Unlock(device);
+                break;
+            }
+        }
+    }
+
+    return 0;
+}
+
+
+static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name)
+{
+    ALCdevice *device;
+    const char *fname;
+
+    fname = GetConfigValue(NULL, "wave", "file", "");
+    if(!fname[0]) return ALC_INVALID_VALUE;
+
+    if(!name)
+        name = waveDevice;
+    else if(strcmp(name, waveDevice) != 0)
+        return ALC_INVALID_VALUE;
+
+    self->mFile = al_fopen(fname, "wb");
+    if(!self->mFile)
+    {
+        ERR("Could not open file '%s': %s\n", fname, strerror(errno));
+        return ALC_INVALID_VALUE;
+    }
+
+    device = STATIC_CAST(ALCbackend, self)->mDevice;
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static void ALCwaveBackend_close(ALCwaveBackend *self)
+{
+    if(self->mFile)
+        fclose(self->mFile);
+    self->mFile = NULL;
+}
+
+static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ALuint channels=0, bits=0, chanmask=0;
+    int isbformat = 0;
+    size_t val;
+
+    fseek(self->mFile, 0, SEEK_SET);
+    clearerr(self->mFile);
+
+    if(GetConfigValueBool(NULL, "wave", "bformat", 0))
+    {
+        device->FmtChans = DevFmtAmbi3D;
+        device->AmbiOrder = 1;
+    }
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            device->FmtType = DevFmtUByte;
+            break;
+        case DevFmtUShort:
+            device->FmtType = DevFmtShort;
+            break;
+        case DevFmtUInt:
+            device->FmtType = DevFmtInt;
+            break;
+        case DevFmtUByte:
+        case DevFmtShort:
+        case DevFmtInt:
+        case DevFmtFloat:
+            break;
+    }
+    switch(device->FmtChans)
+    {
+        case DevFmtMono:   chanmask = 0x04; break;
+        case DevFmtStereo: chanmask = 0x01 | 0x02; break;
+        case DevFmtQuad:   chanmask = 0x01 | 0x02 | 0x10 | 0x20; break;
+        case DevFmtX51: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x200 | 0x400; break;
+        case DevFmtX51Rear: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020; break;
+        case DevFmtX61: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x100 | 0x200 | 0x400; break;
+        case DevFmtX71: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020 | 0x200 | 0x400; break;
+        case DevFmtAmbi3D:
+            /* .amb output requires FuMa */
+            device->AmbiLayout = AmbiLayout_FuMa;
+            device->AmbiScale = AmbiNorm_FuMa;
+            isbformat = 1;
+            chanmask = 0;
+            break;
+    }
+    bits = BytesFromDevFmt(device->FmtType) * 8;
+    channels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+
+    fputs("RIFF", self->mFile);
+    fwrite32le(0xFFFFFFFF, self->mFile); // 'RIFF' header len; filled in at close
+
+    fputs("WAVE", self->mFile);
+
+    fputs("fmt ", self->mFile);
+    fwrite32le(40, self->mFile); // 'fmt ' header len; 40 bytes for EXTENSIBLE
+
+    // 16-bit val, format type id (extensible: 0xFFFE)
+    fwrite16le(0xFFFE, self->mFile);
+    // 16-bit val, channel count
+    fwrite16le(channels, self->mFile);
+    // 32-bit val, frequency
+    fwrite32le(device->Frequency, self->mFile);
+    // 32-bit val, bytes per second
+    fwrite32le(device->Frequency * channels * bits / 8, self->mFile);
+    // 16-bit val, frame size
+    fwrite16le(channels * bits / 8, self->mFile);
+    // 16-bit val, bits per sample
+    fwrite16le(bits, self->mFile);
+    // 16-bit val, extra byte count
+    fwrite16le(22, self->mFile);
+    // 16-bit val, valid bits per sample
+    fwrite16le(bits, self->mFile);
+    // 32-bit val, channel mask
+    fwrite32le(chanmask, self->mFile);
+    // 16 byte GUID, sub-type format
+    val = fwrite(((bits==32) ? (isbformat ? SUBTYPE_BFORMAT_FLOAT : SUBTYPE_FLOAT) :
+                               (isbformat ? SUBTYPE_BFORMAT_PCM : SUBTYPE_PCM)), 1, 16, self->mFile);
+    (void)val;
+
+    fputs("data", self->mFile);
+    fwrite32le(0xFFFFFFFF, self->mFile); // 'data' header len; filled in at close
+
+    if(ferror(self->mFile))
+    {
+        ERR("Error writing header: %s\n", strerror(errno));
+        return ALC_FALSE;
+    }
+    self->mDataStart = ftell(self->mFile);
+
+    SetDefaultWFXChannelOrder(device);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    self->mSize = device->UpdateSize * FrameSizeFromDevFmt(
+        device->FmtChans, device->FmtType, device->AmbiOrder
+    );
+    self->mBuffer = malloc(self->mSize);
+    if(!self->mBuffer)
+    {
+        ERR("Buffer malloc failed\n");
+        return ALC_FALSE;
+    }
+
+    self->killNow = 0;
+    if(althrd_create(&self->thread, ALCwaveBackend_mixerProc, self) != althrd_success)
+    {
+        free(self->mBuffer);
+        self->mBuffer = NULL;
+        self->mSize = 0;
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCwaveBackend_stop(ALCwaveBackend *self)
+{
+    ALuint dataLen;
+    long size;
+    int res;
+
+    if(self->killNow)
+        return;
+
+    self->killNow = 1;
+    althrd_join(self->thread, &res);
+
+    free(self->mBuffer);
+    self->mBuffer = NULL;
+
+    size = ftell(self->mFile);
+    if(size > 0)
+    {
+        dataLen = size - self->mDataStart;
+        if(fseek(self->mFile, self->mDataStart-4, SEEK_SET) == 0)
+            fwrite32le(dataLen, self->mFile); // 'data' header len
+        if(fseek(self->mFile, 4, SEEK_SET) == 0)
+            fwrite32le(size-8, self->mFile); // 'WAVE' header len
+    }
+}
+
+
+typedef struct ALCwaveBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCwaveBackendFactory;
+#define ALCWAVEBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwaveBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCwaveBackendFactory_getFactory(void);
+
+static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory *self);
+static DECLARE_FORWARD(ALCwaveBackendFactory, ALCbackendFactory, void, deinit)
+static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory *self, ALCbackend_Type type);
+static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwaveBackendFactory);
+
+
+ALCbackendFactory *ALCwaveBackendFactory_getFactory(void)
+{
+    static ALCwaveBackendFactory factory = ALCWAVEBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory* UNUSED(self))
+{
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+        return !!ConfigValueExists(NULL, "wave", "file");
+    return ALC_FALSE;
+}
+
+static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            AppendAllDevicesList(waveDevice);
+            break;
+        case CAPTURE_DEVICE_PROBE:
+            break;
+    }
+}
+
+static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCwaveBackend *backend;
+        NEW_OBJ(backend, ALCwaveBackend)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/winmm.c

@@ -0,0 +1,803 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <memory.h>
+
+#include <windows.h>
+#include <mmsystem.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+
+#include "backends/base.h"
+
+#ifndef WAVE_FORMAT_IEEE_FLOAT
+#define WAVE_FORMAT_IEEE_FLOAT  0x0003
+#endif
+
+#define DEVNAME_HEAD "OpenAL Soft on "
+
+
+static vector_al_string PlaybackDevices;
+static vector_al_string CaptureDevices;
+
+static void clear_devlist(vector_al_string *list)
+{
+    VECTOR_FOR_EACH(al_string, *list, alstr_reset);
+    VECTOR_RESIZE(*list, 0, 0);
+}
+
+
+static void ProbePlaybackDevices(void)
+{
+    ALuint numdevs;
+    ALuint i;
+
+    clear_devlist(&PlaybackDevices);
+
+    numdevs = waveOutGetNumDevs();
+    VECTOR_RESIZE(PlaybackDevices, 0, numdevs);
+    for(i = 0;i < numdevs;i++)
+    {
+        WAVEOUTCAPSW WaveCaps;
+        const al_string *iter;
+        al_string dname;
+
+        AL_STRING_INIT(dname);
+        if(waveOutGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR)
+        {
+            ALuint count = 0;
+            while(1)
+            {
+                alstr_copy_cstr(&dname, DEVNAME_HEAD);
+                alstr_append_wcstr(&dname, WaveCaps.szPname);
+                if(count != 0)
+                {
+                    char str[64];
+                    snprintf(str, sizeof(str), " #%d", count+1);
+                    alstr_append_cstr(&dname, str);
+                }
+                count++;
+
+#define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0)
+                VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_ENTRY);
+                if(iter == VECTOR_END(PlaybackDevices)) break;
+#undef MATCH_ENTRY
+            }
+
+            TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i);
+        }
+        VECTOR_PUSH_BACK(PlaybackDevices, dname);
+    }
+}
+
+static void ProbeCaptureDevices(void)
+{
+    ALuint numdevs;
+    ALuint i;
+
+    clear_devlist(&CaptureDevices);
+
+    numdevs = waveInGetNumDevs();
+    VECTOR_RESIZE(CaptureDevices, 0, numdevs);
+    for(i = 0;i < numdevs;i++)
+    {
+        WAVEINCAPSW WaveCaps;
+        const al_string *iter;
+        al_string dname;
+
+        AL_STRING_INIT(dname);
+        if(waveInGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR)
+        {
+            ALuint count = 0;
+            while(1)
+            {
+                alstr_copy_cstr(&dname, DEVNAME_HEAD);
+                alstr_append_wcstr(&dname, WaveCaps.szPname);
+                if(count != 0)
+                {
+                    char str[64];
+                    snprintf(str, sizeof(str), " #%d", count+1);
+                    alstr_append_cstr(&dname, str);
+                }
+                count++;
+
+#define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0)
+                VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_ENTRY);
+                if(iter == VECTOR_END(CaptureDevices)) break;
+#undef MATCH_ENTRY
+            }
+
+            TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i);
+        }
+        VECTOR_PUSH_BACK(CaptureDevices, dname);
+    }
+}
+
+
+typedef struct ALCwinmmPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    RefCount WaveBuffersCommitted;
+    WAVEHDR WaveBuffer[4];
+
+    HWAVEOUT OutHdl;
+
+    WAVEFORMATEX Format;
+
+    volatile ALboolean killNow;
+    althrd_t thread;
+} ALCwinmmPlayback;
+
+static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device);
+static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self);
+
+static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2);
+static int ALCwinmmPlayback_mixerProc(void *arg);
+
+static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *name);
+static void ALCwinmmPlayback_close(ALCwinmmPlayback *self);
+static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self);
+static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self);
+static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self);
+static DECLARE_FORWARD2(ALCwinmmPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
+static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCwinmmPlayback)
+
+DEFINE_ALCBACKEND_VTABLE(ALCwinmmPlayback);
+
+
+static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCwinmmPlayback, ALCbackend, self);
+
+    InitRef(&self->WaveBuffersCommitted, 0);
+    self->OutHdl = NULL;
+
+    self->killNow = AL_TRUE;
+}
+
+static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self)
+{
+    if(self->OutHdl)
+        waveOutClose(self->OutHdl);
+    self->OutHdl = 0;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+/* ALCwinmmPlayback_waveOutProc
+ *
+ * Posts a message to 'ALCwinmmPlayback_mixerProc' everytime a WaveOut Buffer
+ * is completed and returns to the application (for more data)
+ */
+static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2))
+{
+    ALCwinmmPlayback *self = (ALCwinmmPlayback*)instance;
+
+    if(msg != WOM_DONE)
+        return;
+
+    DecrementRef(&self->WaveBuffersCommitted);
+    PostThreadMessage(self->thread, msg, 0, param1);
+}
+
+FORCE_ALIGN static int ALCwinmmPlayback_mixerProc(void *arg)
+{
+    ALCwinmmPlayback *self = arg;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    WAVEHDR *WaveHdr;
+    MSG msg;
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
+
+    while(GetMessage(&msg, NULL, 0, 0))
+    {
+        if(msg.message != WOM_DONE)
+            continue;
+
+        if(self->killNow)
+        {
+            if(ReadRef(&self->WaveBuffersCommitted) == 0)
+                break;
+            continue;
+        }
+
+        WaveHdr = ((WAVEHDR*)msg.lParam);
+        ALCwinmmPlayback_lock(self);
+        aluMixData(device, WaveHdr->lpData, WaveHdr->dwBufferLength /
+                                            self->Format.nBlockAlign);
+        ALCwinmmPlayback_unlock(self);
+
+        // Send buffer back to play more data
+        waveOutWrite(self->OutHdl, WaveHdr, sizeof(WAVEHDR));
+        IncrementRef(&self->WaveBuffersCommitted);
+    }
+
+    return 0;
+}
+
+
+static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *deviceName)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    const al_string *iter;
+    UINT DeviceID;
+    MMRESULT res;
+
+    if(VECTOR_SIZE(PlaybackDevices) == 0)
+        ProbePlaybackDevices();
+
+    // Find the Device ID matching the deviceName if valid
+#define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && \
+                             (!deviceName || alstr_cmp_cstr(*(iter), deviceName) == 0))
+    VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_DEVNAME);
+    if(iter == VECTOR_END(PlaybackDevices))
+        return ALC_INVALID_VALUE;
+#undef MATCH_DEVNAME
+
+    DeviceID = (UINT)(iter - VECTOR_BEGIN(PlaybackDevices));
+
+retry_open:
+    memset(&self->Format, 0, sizeof(WAVEFORMATEX));
+    if(device->FmtType == DevFmtFloat)
+    {
+        self->Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
+        self->Format.wBitsPerSample = 32;
+    }
+    else
+    {
+        self->Format.wFormatTag = WAVE_FORMAT_PCM;
+        if(device->FmtType == DevFmtUByte || device->FmtType == DevFmtByte)
+            self->Format.wBitsPerSample = 8;
+        else
+            self->Format.wBitsPerSample = 16;
+    }
+    self->Format.nChannels = ((device->FmtChans == DevFmtMono) ? 1 : 2);
+    self->Format.nBlockAlign = self->Format.wBitsPerSample *
+                               self->Format.nChannels / 8;
+    self->Format.nSamplesPerSec = device->Frequency;
+    self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec *
+                                   self->Format.nBlockAlign;
+    self->Format.cbSize = 0;
+
+    if((res=waveOutOpen(&self->OutHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmPlayback_waveOutProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
+    {
+        if(device->FmtType == DevFmtFloat)
+        {
+            device->FmtType = DevFmtShort;
+            goto retry_open;
+        }
+        ERR("waveOutOpen failed: %u\n", res);
+        goto failure;
+    }
+
+    alstr_copy(&device->DeviceName, VECTOR_ELEM(PlaybackDevices, DeviceID));
+    return ALC_NO_ERROR;
+
+failure:
+    if(self->OutHdl)
+        waveOutClose(self->OutHdl);
+    self->OutHdl = NULL;
+
+    return ALC_INVALID_VALUE;
+}
+
+static void ALCwinmmPlayback_close(ALCwinmmPlayback* UNUSED(self))
+{ }
+
+static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
+                                  self->Format.nSamplesPerSec /
+                                  device->Frequency);
+    device->UpdateSize = (device->UpdateSize*device->NumUpdates + 3) / 4;
+    device->NumUpdates = 4;
+    device->Frequency = self->Format.nSamplesPerSec;
+
+    if(self->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
+    {
+        if(self->Format.wBitsPerSample == 32)
+            device->FmtType = DevFmtFloat;
+        else
+        {
+            ERR("Unhandled IEEE float sample depth: %d\n", self->Format.wBitsPerSample);
+            return ALC_FALSE;
+        }
+    }
+    else if(self->Format.wFormatTag == WAVE_FORMAT_PCM)
+    {
+        if(self->Format.wBitsPerSample == 16)
+            device->FmtType = DevFmtShort;
+        else if(self->Format.wBitsPerSample == 8)
+            device->FmtType = DevFmtUByte;
+        else
+        {
+            ERR("Unhandled PCM sample depth: %d\n", self->Format.wBitsPerSample);
+            return ALC_FALSE;
+        }
+    }
+    else
+    {
+        ERR("Unhandled format tag: 0x%04x\n", self->Format.wFormatTag);
+        return ALC_FALSE;
+    }
+
+    if(self->Format.nChannels == 2)
+        device->FmtChans = DevFmtStereo;
+    else if(self->Format.nChannels == 1)
+        device->FmtChans = DevFmtMono;
+    else
+    {
+        ERR("Unhandled channel count: %d\n", self->Format.nChannels);
+        return ALC_FALSE;
+    }
+    SetDefaultWFXChannelOrder(device);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ALbyte *BufferData;
+    ALint BufferSize;
+    ALuint i;
+
+    self->killNow = AL_FALSE;
+    if(althrd_create(&self->thread, ALCwinmmPlayback_mixerProc, self) != althrd_success)
+        return ALC_FALSE;
+
+    InitRef(&self->WaveBuffersCommitted, 0);
+
+    // Create 4 Buffers
+    BufferSize  = device->UpdateSize*device->NumUpdates / 4;
+    BufferSize *= FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    BufferData = calloc(4, BufferSize);
+    for(i = 0;i < 4;i++)
+    {
+        memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR));
+        self->WaveBuffer[i].dwBufferLength = BufferSize;
+        self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData :
+                                      (self->WaveBuffer[i-1].lpData +
+                                       self->WaveBuffer[i-1].dwBufferLength));
+        waveOutPrepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
+        waveOutWrite(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
+        IncrementRef(&self->WaveBuffersCommitted);
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self)
+{
+    void *buffer = NULL;
+    int i;
+
+    if(self->killNow)
+        return;
+
+    // Set flag to stop processing headers
+    self->killNow = AL_TRUE;
+    althrd_join(self->thread, &i);
+
+    // Release the wave buffers
+    for(i = 0;i < 4;i++)
+    {
+        waveOutUnprepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
+        if(i == 0) buffer = self->WaveBuffer[i].lpData;
+        self->WaveBuffer[i].lpData = NULL;
+    }
+    free(buffer);
+}
+
+
+
+typedef struct ALCwinmmCapture {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    RefCount WaveBuffersCommitted;
+    WAVEHDR WaveBuffer[4];
+
+    HWAVEIN InHdl;
+
+    ll_ringbuffer_t *Ring;
+
+    WAVEFORMATEX Format;
+
+    volatile ALboolean killNow;
+    althrd_t thread;
+} ALCwinmmCapture;
+
+static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device);
+static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self);
+
+static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2);
+static int ALCwinmmCapture_captureProc(void *arg);
+
+static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name);
+static void ALCwinmmCapture_close(ALCwinmmCapture *self);
+static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ALCboolean, reset)
+static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self);
+static void ALCwinmmCapture_stop(ALCwinmmCapture *self);
+static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples);
+static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self);
+static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCwinmmCapture)
+
+DEFINE_ALCBACKEND_VTABLE(ALCwinmmCapture);
+
+
+static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCwinmmCapture, ALCbackend, self);
+
+    InitRef(&self->WaveBuffersCommitted, 0);
+    self->InHdl = NULL;
+
+    self->killNow = AL_TRUE;
+}
+
+static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self)
+{
+    if(self->InHdl)
+        waveInClose(self->InHdl);
+    self->InHdl = 0;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+/* ALCwinmmCapture_waveInProc
+ *
+ * Posts a message to 'ALCwinmmCapture_captureProc' everytime a WaveIn Buffer
+ * is completed and returns to the application (with more data).
+ */
+static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2))
+{
+    ALCwinmmCapture *self = (ALCwinmmCapture*)instance;
+
+    if(msg != WIM_DATA)
+        return;
+
+    DecrementRef(&self->WaveBuffersCommitted);
+    PostThreadMessage(self->thread, msg, 0, param1);
+}
+
+static int ALCwinmmCapture_captureProc(void *arg)
+{
+    ALCwinmmCapture *self = arg;
+    WAVEHDR *WaveHdr;
+    MSG msg;
+
+    althrd_setname(althrd_current(), RECORD_THREAD_NAME);
+
+    while(GetMessage(&msg, NULL, 0, 0))
+    {
+        if(msg.message != WIM_DATA)
+            continue;
+        /* Don't wait for other buffers to finish before quitting. We're
+         * closing so we don't need them. */
+        if(self->killNow)
+            break;
+
+        WaveHdr = ((WAVEHDR*)msg.lParam);
+        ll_ringbuffer_write(self->Ring, WaveHdr->lpData,
+            WaveHdr->dwBytesRecorded / self->Format.nBlockAlign
+        );
+
+        // Send buffer back to capture more data
+        waveInAddBuffer(self->InHdl, WaveHdr, sizeof(WAVEHDR));
+        IncrementRef(&self->WaveBuffersCommitted);
+    }
+
+    return 0;
+}
+
+
+static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    const al_string *iter;
+    ALbyte *BufferData = NULL;
+    DWORD CapturedDataSize;
+    ALint BufferSize;
+    UINT DeviceID;
+    MMRESULT res;
+    ALuint i;
+
+    if(VECTOR_SIZE(CaptureDevices) == 0)
+        ProbeCaptureDevices();
+
+    // Find the Device ID matching the deviceName if valid
+#define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && (!name || alstr_cmp_cstr(*iter, name) == 0))
+    VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_DEVNAME);
+    if(iter == VECTOR_END(CaptureDevices))
+        return ALC_INVALID_VALUE;
+#undef MATCH_DEVNAME
+
+    DeviceID = (UINT)(iter - VECTOR_BEGIN(CaptureDevices));
+
+    switch(device->FmtChans)
+    {
+        case DevFmtMono:
+        case DevFmtStereo:
+            break;
+
+        case DevFmtQuad:
+        case DevFmtX51:
+        case DevFmtX51Rear:
+        case DevFmtX61:
+        case DevFmtX71:
+        case DevFmtAmbi3D:
+            return ALC_INVALID_ENUM;
+    }
+
+    switch(device->FmtType)
+    {
+        case DevFmtUByte:
+        case DevFmtShort:
+        case DevFmtInt:
+        case DevFmtFloat:
+            break;
+
+        case DevFmtByte:
+        case DevFmtUShort:
+        case DevFmtUInt:
+            return ALC_INVALID_ENUM;
+    }
+
+    memset(&self->Format, 0, sizeof(WAVEFORMATEX));
+    self->Format.wFormatTag = ((device->FmtType == DevFmtFloat) ?
+                               WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM);
+    self->Format.nChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    self->Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8;
+    self->Format.nBlockAlign = self->Format.wBitsPerSample *
+                               self->Format.nChannels / 8;
+    self->Format.nSamplesPerSec = device->Frequency;
+    self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec *
+                                   self->Format.nBlockAlign;
+    self->Format.cbSize = 0;
+
+    if((res=waveInOpen(&self->InHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmCapture_waveInProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
+    {
+        ERR("waveInOpen failed: %u\n", res);
+        goto failure;
+    }
+
+    // Allocate circular memory buffer for the captured audio
+    CapturedDataSize = device->UpdateSize*device->NumUpdates;
+
+    // Make sure circular buffer is at least 100ms in size
+    if(CapturedDataSize < (self->Format.nSamplesPerSec / 10))
+        CapturedDataSize = self->Format.nSamplesPerSec / 10;
+
+    self->Ring = ll_ringbuffer_create(CapturedDataSize+1, self->Format.nBlockAlign);
+    if(!self->Ring) goto failure;
+
+    InitRef(&self->WaveBuffersCommitted, 0);
+
+    // Create 4 Buffers of 50ms each
+    BufferSize = self->Format.nAvgBytesPerSec / 20;
+    BufferSize -= (BufferSize % self->Format.nBlockAlign);
+
+    BufferData = calloc(4, BufferSize);
+    if(!BufferData) goto failure;
+
+    for(i = 0;i < 4;i++)
+    {
+        memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR));
+        self->WaveBuffer[i].dwBufferLength = BufferSize;
+        self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData :
+                                      (self->WaveBuffer[i-1].lpData +
+                                       self->WaveBuffer[i-1].dwBufferLength));
+        self->WaveBuffer[i].dwFlags = 0;
+        self->WaveBuffer[i].dwLoops = 0;
+        waveInPrepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
+        waveInAddBuffer(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
+        IncrementRef(&self->WaveBuffersCommitted);
+    }
+
+    self->killNow = AL_FALSE;
+    if(althrd_create(&self->thread, ALCwinmmCapture_captureProc, self) != althrd_success)
+        goto failure;
+
+    alstr_copy(&device->DeviceName, VECTOR_ELEM(CaptureDevices, DeviceID));
+    return ALC_NO_ERROR;
+
+failure:
+    if(BufferData)
+    {
+        for(i = 0;i < 4;i++)
+            waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
+        free(BufferData);
+    }
+
+    ll_ringbuffer_free(self->Ring);
+    self->Ring = NULL;
+
+    if(self->InHdl)
+        waveInClose(self->InHdl);
+    self->InHdl = NULL;
+
+    return ALC_INVALID_VALUE;
+}
+
+static void ALCwinmmCapture_close(ALCwinmmCapture *self)
+{
+    void *buffer = NULL;
+    int i;
+
+    /* Tell the processing thread to quit and wait for it to do so. */
+    self->killNow = AL_TRUE;
+    PostThreadMessage(self->thread, WM_QUIT, 0, 0);
+
+    althrd_join(self->thread, &i);
+
+    /* Make sure capture is stopped and all pending buffers are flushed. */
+    waveInReset(self->InHdl);
+
+    // Release the wave buffers
+    for(i = 0;i < 4;i++)
+    {
+        waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
+        if(i == 0) buffer = self->WaveBuffer[i].lpData;
+        self->WaveBuffer[i].lpData = NULL;
+    }
+    free(buffer);
+
+    ll_ringbuffer_free(self->Ring);
+    self->Ring = NULL;
+
+    // Close the Wave device
+    waveInClose(self->InHdl);
+    self->InHdl = NULL;
+}
+
+static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self)
+{
+    waveInStart(self->InHdl);
+    return ALC_TRUE;
+}
+
+static void ALCwinmmCapture_stop(ALCwinmmCapture *self)
+{
+    waveInStop(self->InHdl);
+}
+
+static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples)
+{
+    ll_ringbuffer_read(self->Ring, buffer, samples);
+    return ALC_NO_ERROR;
+}
+
+static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self)
+{
+    return ll_ringbuffer_read_space(self->Ring);
+}
+
+
+static inline void AppendAllDevicesList2(const al_string *name)
+{
+    if(!alstr_empty(*name))
+        AppendAllDevicesList(alstr_get_cstr(*name));
+}
+static inline void AppendCaptureDeviceList2(const al_string *name)
+{
+    if(!alstr_empty(*name))
+        AppendCaptureDeviceList(alstr_get_cstr(*name));
+}
+
+typedef struct ALCwinmmBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCwinmmBackendFactory;
+#define ALCWINMMBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwinmmBackendFactory, ALCbackendFactory) } }
+
+static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory *self);
+static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory *self);
+static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory *self, ALCbackend_Type type);
+static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory *self, enum DevProbe type);
+static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwinmmBackendFactory);
+
+
+static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory* UNUSED(self))
+{
+    VECTOR_INIT(PlaybackDevices);
+    VECTOR_INIT(CaptureDevices);
+
+    return ALC_TRUE;
+}
+
+static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory* UNUSED(self))
+{
+    clear_devlist(&PlaybackDevices);
+    VECTOR_DEINIT(PlaybackDevices);
+
+    clear_devlist(&CaptureDevices);
+    VECTOR_DEINIT(CaptureDevices);
+}
+
+static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback || type == ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory* UNUSED(self), enum DevProbe type)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            ProbePlaybackDevices();
+            VECTOR_FOR_EACH(const al_string, PlaybackDevices, AppendAllDevicesList2);
+            break;
+
+        case CAPTURE_DEVICE_PROBE:
+            ProbeCaptureDevices();
+            VECTOR_FOR_EACH(const al_string, CaptureDevices, AppendCaptureDeviceList2);
+            break;
+    }
+}
+
+static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCwinmmPlayback *backend;
+        NEW_OBJ(backend, ALCwinmmPlayback)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+    if(type == ALCbackend_Capture)
+    {
+        ALCwinmmCapture *backend;
+        NEW_OBJ(backend, ALCwinmmCapture)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}
+
+ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void)
+{
+    static ALCwinmmBackendFactory factory = ALCWINMMBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}

Alc/bformatdec.c

@@ -0,0 +1,573 @@
+
+#include "config.h"
+
+#include "bformatdec.h"
+#include "ambdec.h"
+#include "mixer_defs.h"
+#include "alu.h"
+
+#include "bool.h"
+#include "threads.h"
+#include "almalloc.h"
+
+
+void bandsplit_init(BandSplitter *splitter, ALfloat freq_mult)
+{
+    ALfloat w = freq_mult * F_TAU;
+    ALfloat cw = cosf(w);
+    if(cw > FLT_EPSILON)
+        splitter->coeff = (sinf(w) - 1.0f) / cw;
+    else
+        splitter->coeff = cw * -0.5f;
+
+    splitter->lp_z1 = 0.0f;
+    splitter->lp_z2 = 0.0f;
+    splitter->hp_z1 = 0.0f;
+}
+
+void bandsplit_clear(BandSplitter *splitter)
+{
+    splitter->lp_z1 = 0.0f;
+    splitter->lp_z2 = 0.0f;
+    splitter->hp_z1 = 0.0f;
+}
+
+void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout,
+                       const ALfloat *input, ALsizei count)
+{
+    ALfloat coeff, d, x;
+    ALfloat z1, z2;
+    ALsizei i;
+
+    coeff = splitter->coeff*0.5f + 0.5f;
+    z1 = splitter->lp_z1;
+    z2 = splitter->lp_z2;
+    for(i = 0;i < count;i++)
+    {
+        x = input[i];
+
+        d = (x - z1) * coeff;
+        x = z1 + d;
+        z1 = x + d;
+
+        d = (x - z2) * coeff;
+        x = z2 + d;
+        z2 = x + d;
+
+        lpout[i] = x;
+    }
+    splitter->lp_z1 = z1;
+    splitter->lp_z2 = z2;
+
+    coeff = splitter->coeff;
+    z1 = splitter->hp_z1;
+    for(i = 0;i < count;i++)
+    {
+        x = input[i];
+
+        d = x - coeff*z1;
+        x = z1 + coeff*d;
+        z1 = d;
+
+        hpout[i] = x - lpout[i];
+    }
+    splitter->hp_z1 = z1;
+}
+
+
+static const ALfloat UnitScale[MAX_AMBI_COEFFS] = {
+    1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
+    1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f
+};
+static const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS] = {
+    1.000000000f, /* ACN  0 (W), sqrt(1) */
+    1.732050808f, /* ACN  1 (Y), sqrt(3) */
+    1.732050808f, /* ACN  2 (Z), sqrt(3) */
+    1.732050808f, /* ACN  3 (X), sqrt(3) */
+    2.236067978f, /* ACN  4 (V), sqrt(5) */
+    2.236067978f, /* ACN  5 (T), sqrt(5) */
+    2.236067978f, /* ACN  6 (R), sqrt(5) */
+    2.236067978f, /* ACN  7 (S), sqrt(5) */
+    2.236067978f, /* ACN  8 (U), sqrt(5) */
+    2.645751311f, /* ACN  9 (Q), sqrt(7) */
+    2.645751311f, /* ACN 10 (O), sqrt(7) */
+    2.645751311f, /* ACN 11 (M), sqrt(7) */
+    2.645751311f, /* ACN 12 (K), sqrt(7) */
+    2.645751311f, /* ACN 13 (L), sqrt(7) */
+    2.645751311f, /* ACN 14 (N), sqrt(7) */
+    2.645751311f, /* ACN 15 (P), sqrt(7) */
+};
+static const ALfloat FuMa2N3DScale[MAX_AMBI_COEFFS] = {
+    1.414213562f, /* ACN  0 (W), sqrt(2) */
+    1.732050808f, /* ACN  1 (Y), sqrt(3) */
+    1.732050808f, /* ACN  2 (Z), sqrt(3) */
+    1.732050808f, /* ACN  3 (X), sqrt(3) */
+    1.936491673f, /* ACN  4 (V), sqrt(15)/2 */
+    1.936491673f, /* ACN  5 (T), sqrt(15)/2 */
+    2.236067978f, /* ACN  6 (R), sqrt(5) */
+    1.936491673f, /* ACN  7 (S), sqrt(15)/2 */
+    1.936491673f, /* ACN  8 (U), sqrt(15)/2 */
+    2.091650066f, /* ACN  9 (Q), sqrt(35/8) */
+    1.972026594f, /* ACN 10 (O), sqrt(35)/3 */
+    2.231093404f, /* ACN 11 (M), sqrt(224/45) */
+    2.645751311f, /* ACN 12 (K), sqrt(7) */
+    2.231093404f, /* ACN 13 (L), sqrt(224/45) */
+    1.972026594f, /* ACN 14 (N), sqrt(35)/3 */
+    2.091650066f, /* ACN 15 (P), sqrt(35/8) */
+};
+
+
+enum FreqBand {
+    FB_HighFreq,
+    FB_LowFreq,
+    FB_Max
+};
+
+/* These points are in AL coordinates! */
+static const ALfloat Ambi3DPoints[8][3] = {
+    { -0.577350269f,  0.577350269f, -0.577350269f },
+    {  0.577350269f,  0.577350269f, -0.577350269f },
+    { -0.577350269f,  0.577350269f,  0.577350269f },
+    {  0.577350269f,  0.577350269f,  0.577350269f },
+    { -0.577350269f, -0.577350269f, -0.577350269f },
+    {  0.577350269f, -0.577350269f, -0.577350269f },
+    { -0.577350269f, -0.577350269f,  0.577350269f },
+    {  0.577350269f, -0.577350269f,  0.577350269f },
+};
+static const ALfloat Ambi3DDecoder[8][FB_Max][MAX_AMBI_COEFFS] = {
+    { { 0.25f,  0.1443375672f,  0.1443375672f,  0.1443375672f }, { 0.125f,  0.125f,  0.125f,  0.125f } },
+    { { 0.25f, -0.1443375672f,  0.1443375672f,  0.1443375672f }, { 0.125f, -0.125f,  0.125f,  0.125f } },
+    { { 0.25f,  0.1443375672f,  0.1443375672f, -0.1443375672f }, { 0.125f,  0.125f,  0.125f, -0.125f } },
+    { { 0.25f, -0.1443375672f,  0.1443375672f, -0.1443375672f }, { 0.125f, -0.125f,  0.125f, -0.125f } },
+    { { 0.25f,  0.1443375672f, -0.1443375672f,  0.1443375672f }, { 0.125f,  0.125f, -0.125f,  0.125f } },
+    { { 0.25f, -0.1443375672f, -0.1443375672f,  0.1443375672f }, { 0.125f, -0.125f, -0.125f,  0.125f } },
+    { { 0.25f,  0.1443375672f, -0.1443375672f, -0.1443375672f }, { 0.125f,  0.125f, -0.125f, -0.125f } },
+    { { 0.25f, -0.1443375672f, -0.1443375672f, -0.1443375672f }, { 0.125f, -0.125f, -0.125f, -0.125f } },
+};
+
+
+static RowMixerFunc MixMatrixRow = MixRow_C;
+
+
+static alonce_flag bformatdec_inited = AL_ONCE_FLAG_INIT;
+
+static void init_bformatdec(void)
+{
+    MixMatrixRow = SelectRowMixer();
+}
+
+
+/* NOTE: BandSplitter filters are unused with single-band decoding */
+typedef struct BFormatDec {
+    ALboolean Enabled[MAX_OUTPUT_CHANNELS];
+
+    union {
+        alignas(16) ALfloat Dual[MAX_OUTPUT_CHANNELS][FB_Max][MAX_AMBI_COEFFS];
+        alignas(16) ALfloat Single[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
+    } Matrix;
+
+    BandSplitter XOver[MAX_AMBI_COEFFS];
+
+    ALfloat (*Samples)[BUFFERSIZE];
+    /* These two alias into Samples */
+    ALfloat (*SamplesHF)[BUFFERSIZE];
+    ALfloat (*SamplesLF)[BUFFERSIZE];
+
+    alignas(16) ALfloat ChannelMix[BUFFERSIZE];
+
+    struct {
+        BandSplitter XOver;
+        ALfloat Gains[FB_Max];
+    } UpSampler[4];
+
+    ALsizei NumChannels;
+    ALboolean DualBand;
+} BFormatDec;
+
+BFormatDec *bformatdec_alloc()
+{
+    alcall_once(&bformatdec_inited, init_bformatdec);
+    return al_calloc(16, sizeof(BFormatDec));
+}
+
+void bformatdec_free(BFormatDec *dec)
+{
+    if(dec)
+    {
+        al_free(dec->Samples);
+        dec->Samples = NULL;
+        dec->SamplesHF = NULL;
+        dec->SamplesLF = NULL;
+
+        memset(dec, 0, sizeof(*dec));
+        al_free(dec);
+    }
+}
+
+void bformatdec_reset(BFormatDec *dec, const AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS])
+{
+    static const ALsizei map2DTo3D[MAX_AMBI2D_COEFFS] = {
+        0,  1, 3,  4, 8,  9, 15
+    };
+    const ALfloat *coeff_scale = UnitScale;
+    bool periphonic;
+    ALfloat ratio;
+    ALsizei i;
+
+    al_free(dec->Samples);
+    dec->Samples = NULL;
+    dec->SamplesHF = NULL;
+    dec->SamplesLF = NULL;
+
+    dec->NumChannels = chancount;
+    dec->Samples = al_calloc(16, dec->NumChannels*2 * sizeof(dec->Samples[0]));
+    dec->SamplesHF = dec->Samples;
+    dec->SamplesLF = dec->SamplesHF + dec->NumChannels;
+
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        dec->Enabled[i] = AL_FALSE;
+    for(i = 0;i < conf->NumSpeakers;i++)
+        dec->Enabled[chanmap[i]] = AL_TRUE;
+
+    if(conf->CoeffScale == ADS_SN3D)
+        coeff_scale = SN3D2N3DScale;
+    else if(conf->CoeffScale == ADS_FuMa)
+        coeff_scale = FuMa2N3DScale;
+
+    memset(dec->UpSampler, 0, sizeof(dec->UpSampler));
+    ratio = 400.0f / (ALfloat)srate;
+    for(i = 0;i < 4;i++)
+        bandsplit_init(&dec->UpSampler[i].XOver, ratio);
+    if((conf->ChanMask&AMBI_PERIPHONIC_MASK))
+    {
+        periphonic = true;
+
+        dec->UpSampler[0].Gains[FB_HighFreq] = (dec->NumChannels > 9) ? W_SCALE3D_THIRD :
+                                               (dec->NumChannels > 4) ? W_SCALE3D_SECOND : 1.0f;
+        dec->UpSampler[0].Gains[FB_LowFreq] = 1.0f;
+        for(i = 1;i < 4;i++)
+        {
+            dec->UpSampler[i].Gains[FB_HighFreq] = (dec->NumChannels > 9) ? XYZ_SCALE3D_THIRD :
+                                                   (dec->NumChannels > 4) ? XYZ_SCALE3D_SECOND : 1.0f;
+            dec->UpSampler[i].Gains[FB_LowFreq] = 1.0f;
+        }
+    }
+    else
+    {
+        periphonic = false;
+
+        dec->UpSampler[0].Gains[FB_HighFreq] = (dec->NumChannels > 5) ? W_SCALE2D_THIRD :
+                                               (dec->NumChannels > 3) ? W_SCALE2D_SECOND : 1.0f;
+        dec->UpSampler[0].Gains[FB_LowFreq] = 1.0f;
+        for(i = 1;i < 3;i++)
+        {
+            dec->UpSampler[i].Gains[FB_HighFreq] = (dec->NumChannels > 5) ? XYZ_SCALE2D_THIRD :
+                                                   (dec->NumChannels > 3) ? XYZ_SCALE2D_SECOND : 1.0f;
+            dec->UpSampler[i].Gains[FB_LowFreq] = 1.0f;
+        }
+        dec->UpSampler[3].Gains[FB_HighFreq] = 0.0f;
+        dec->UpSampler[3].Gains[FB_LowFreq] = 0.0f;
+    }
+
+    memset(&dec->Matrix, 0, sizeof(dec->Matrix));
+    if(conf->FreqBands == 1)
+    {
+        dec->DualBand = AL_FALSE;
+        for(i = 0;i < conf->NumSpeakers;i++)
+        {
+            ALsizei chan = chanmap[i];
+            ALfloat gain;
+            ALsizei j, k;
+
+            if(!periphonic)
+            {
+                for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
+                {
+                    ALsizei l = map2DTo3D[j];
+                    if(j == 0) gain = conf->HFOrderGain[0];
+                    else if(j == 1) gain = conf->HFOrderGain[1];
+                    else if(j == 3) gain = conf->HFOrderGain[2];
+                    else if(j == 5) gain = conf->HFOrderGain[3];
+                    if((conf->ChanMask&(1<<l)))
+                        dec->Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[l] *
+                                                      gain;
+                }
+            }
+            else
+            {
+                for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
+                {
+                    if(j == 0) gain = conf->HFOrderGain[0];
+                    else if(j == 1) gain = conf->HFOrderGain[1];
+                    else if(j == 4) gain = conf->HFOrderGain[2];
+                    else if(j == 9) gain = conf->HFOrderGain[3];
+                    if((conf->ChanMask&(1<<j)))
+                        dec->Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[j] *
+                                                      gain;
+                }
+            }
+        }
+    }
+    else
+    {
+        dec->DualBand = AL_TRUE;
+
+        ratio = conf->XOverFreq / (ALfloat)srate;
+        for(i = 0;i < MAX_AMBI_COEFFS;i++)
+            bandsplit_init(&dec->XOver[i], ratio);
+
+        ratio = powf(10.0f, conf->XOverRatio / 40.0f);
+        for(i = 0;i < conf->NumSpeakers;i++)
+        {
+            ALsizei chan = chanmap[i];
+            ALfloat gain;
+            ALsizei j, k;
+
+            if(!periphonic)
+            {
+                for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
+                {
+                    ALsizei l = map2DTo3D[j];
+                    if(j == 0) gain = conf->HFOrderGain[0] * ratio;
+                    else if(j == 1) gain = conf->HFOrderGain[1] * ratio;
+                    else if(j == 3) gain = conf->HFOrderGain[2] * ratio;
+                    else if(j == 5) gain = conf->HFOrderGain[3] * ratio;
+                    if((conf->ChanMask&(1<<l)))
+                        dec->Matrix.Dual[chan][FB_HighFreq][j] = conf->HFMatrix[i][k++] /
+                                                                 coeff_scale[l] * gain;
+                }
+                for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
+                {
+                    ALsizei l = map2DTo3D[j];
+                    if(j == 0) gain = conf->LFOrderGain[0] / ratio;
+                    else if(j == 1) gain = conf->LFOrderGain[1] / ratio;
+                    else if(j == 3) gain = conf->LFOrderGain[2] / ratio;
+                    else if(j == 5) gain = conf->LFOrderGain[3] / ratio;
+                    if((conf->ChanMask&(1<<l)))
+                        dec->Matrix.Dual[chan][FB_LowFreq][j] = conf->LFMatrix[i][k++] /
+                                                                coeff_scale[l] * gain;
+                }
+            }
+            else
+            {
+                for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
+                {
+                    if(j == 0) gain = conf->HFOrderGain[0] * ratio;
+                    else if(j == 1) gain = conf->HFOrderGain[1] * ratio;
+                    else if(j == 4) gain = conf->HFOrderGain[2] * ratio;
+                    else if(j == 9) gain = conf->HFOrderGain[3] * ratio;
+                    if((conf->ChanMask&(1<<j)))
+                        dec->Matrix.Dual[chan][FB_HighFreq][j] = conf->HFMatrix[i][k++] /
+                                                                 coeff_scale[j] * gain;
+                }
+                for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
+                {
+                    if(j == 0) gain = conf->LFOrderGain[0] / ratio;
+                    else if(j == 1) gain = conf->LFOrderGain[1] / ratio;
+                    else if(j == 4) gain = conf->LFOrderGain[2] / ratio;
+                    else if(j == 9) gain = conf->LFOrderGain[3] / ratio;
+                    if((conf->ChanMask&(1<<j)))
+                        dec->Matrix.Dual[chan][FB_LowFreq][j] = conf->LFMatrix[i][k++] /
+                                                                coeff_scale[j] * gain;
+                }
+            }
+        }
+    }
+}
+
+
+void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
+{
+    ALsizei chan, i;
+
+    OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
+    if(dec->DualBand)
+    {
+        for(i = 0;i < dec->NumChannels;i++)
+            bandsplit_process(&dec->XOver[i], dec->SamplesHF[i], dec->SamplesLF[i],
+                              InSamples[i], SamplesToDo);
+
+        for(chan = 0;chan < OutChannels;chan++)
+        {
+            if(!dec->Enabled[chan])
+                continue;
+
+            memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
+            MixMatrixRow(dec->ChannelMix, dec->Matrix.Dual[chan][FB_HighFreq],
+                SAFE_CONST(ALfloatBUFFERSIZE*,dec->SamplesHF), dec->NumChannels, 0,
+                SamplesToDo
+            );
+            MixMatrixRow(dec->ChannelMix, dec->Matrix.Dual[chan][FB_LowFreq],
+                SAFE_CONST(ALfloatBUFFERSIZE*,dec->SamplesLF), dec->NumChannels, 0,
+                SamplesToDo
+            );
+
+            for(i = 0;i < SamplesToDo;i++)
+                OutBuffer[chan][i] += dec->ChannelMix[i];
+        }
+    }
+    else
+    {
+        for(chan = 0;chan < OutChannels;chan++)
+        {
+            if(!dec->Enabled[chan])
+                continue;
+
+            memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
+            MixMatrixRow(dec->ChannelMix, dec->Matrix.Single[chan], InSamples,
+                         dec->NumChannels, 0, SamplesToDo);
+
+            for(i = 0;i < SamplesToDo;i++)
+                OutBuffer[chan][i] += dec->ChannelMix[i];
+        }
+    }
+}
+
+
+void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo)
+{
+    ALsizei i;
+
+    /* This up-sampler leverages the differences observed in dual-band second-
+     * and third-order decoder matrices compared to first-order. For the same
+     * output channel configuration, the low-frequency matrix has identical
+     * coefficients in the shared input channels, while the high-frequency
+     * matrix has extra scalars applied to the W channel and X/Y/Z channels.
+     * Mixing the first-order content into the higher-order stream with the
+     * appropriate counter-scales applied to the HF response results in the
+     * subsequent higher-order decode generating the same response as a first-
+     * order decode.
+     */
+    for(i = 0;i < InChannels;i++)
+    {
+        /* First, split the first-order components into low and high frequency
+         * bands.
+         */
+        bandsplit_process(&dec->UpSampler[i].XOver,
+            dec->Samples[FB_HighFreq], dec->Samples[FB_LowFreq],
+            InSamples[i], SamplesToDo
+        );
+
+        /* Now write each band to the output. */
+        MixMatrixRow(OutBuffer[i], dec->UpSampler[i].Gains,
+            SAFE_CONST(ALfloatBUFFERSIZE*,dec->Samples), FB_Max, 0,
+            SamplesToDo
+        );
+    }
+}
+
+
+#define INVALID_UPSAMPLE_INDEX INT_MAX
+
+static ALsizei GetACNIndex(const BFChannelConfig *chans, ALsizei numchans, ALsizei acn)
+{
+    ALsizei i;
+    for(i = 0;i < numchans;i++)
+    {
+        if(chans[i].Index == acn)
+            return i;
+    }
+    return INVALID_UPSAMPLE_INDEX;
+}
+#define GetChannelForACN(b, a) GetACNIndex((b).Ambi.Map, (b).NumChannels, (a))
+
+typedef struct AmbiUpsampler {
+    alignas(16) ALfloat Samples[FB_Max][BUFFERSIZE];
+
+    BandSplitter XOver[4];
+
+    ALfloat Gains[4][MAX_OUTPUT_CHANNELS][FB_Max];
+} AmbiUpsampler;
+
+AmbiUpsampler *ambiup_alloc()
+{
+    alcall_once(&bformatdec_inited, init_bformatdec);
+    return al_calloc(16, sizeof(AmbiUpsampler));
+}
+
+void ambiup_free(struct AmbiUpsampler *ambiup)
+{
+    al_free(ambiup);
+}
+
+void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device)
+{
+    ALfloat ratio;
+    size_t i;
+
+    ratio = 400.0f / (ALfloat)device->Frequency;
+    for(i = 0;i < 4;i++)
+        bandsplit_init(&ambiup->XOver[i], ratio);
+
+    memset(ambiup->Gains, 0, sizeof(ambiup->Gains));
+    if(device->Dry.CoeffCount > 0)
+    {
+        ALfloat encgains[8][MAX_OUTPUT_CHANNELS];
+        ALsizei j;
+        size_t k;
+
+        for(i = 0;i < COUNTOF(Ambi3DPoints);i++)
+        {
+            ALfloat coeffs[MAX_AMBI_COEFFS] = { 0.0f };
+            CalcDirectionCoeffs(Ambi3DPoints[i], 0.0f, coeffs);
+            ComputePanningGains(device->Dry, coeffs, 1.0f, encgains[i]);
+        }
+
+        /* Combine the matrices that do the in->virt and virt->out conversions
+         * so we get a single in->out conversion. NOTE: the Encoder matrix
+         * (encgains) and output are transposed, so the input channels line up
+         * with the rows and the output channels line up with the columns.
+         */
+        for(i = 0;i < 4;i++)
+        {
+            for(j = 0;j < device->Dry.NumChannels;j++)
+            {
+                ALfloat hfgain=0.0f, lfgain=0.0f;
+                for(k = 0;k < COUNTOF(Ambi3DDecoder);k++)
+                {
+                    hfgain += Ambi3DDecoder[k][FB_HighFreq][i]*encgains[k][j];
+                    lfgain += Ambi3DDecoder[k][FB_LowFreq][i]*encgains[k][j];
+                }
+                ambiup->Gains[i][j][FB_HighFreq] = hfgain;
+                ambiup->Gains[i][j][FB_LowFreq] = lfgain;
+            }
+        }
+    }
+    else
+    {
+        /* Assumes full 3D/periphonic on the input and output mixes! */
+        ALfloat w_scale = (device->Dry.NumChannels > 9) ? W_SCALE3D_THIRD :
+                          (device->Dry.NumChannels > 4) ? W_SCALE3D_SECOND : 1.0f;
+        ALfloat xyz_scale = (device->Dry.NumChannels > 9) ? XYZ_SCALE3D_THIRD :
+                            (device->Dry.NumChannels > 4) ? XYZ_SCALE3D_SECOND : 1.0f;
+        for(i = 0;i < 4;i++)
+        {
+            ALsizei index = GetChannelForACN(device->Dry, i);
+            if(index != INVALID_UPSAMPLE_INDEX)
+            {
+                ALfloat scale = device->Dry.Ambi.Map[index].Scale;
+                ambiup->Gains[i][index][FB_HighFreq] = scale * ((i==0) ? w_scale : xyz_scale);
+                ambiup->Gains[i][index][FB_LowFreq] = scale;
+            }
+        }
+    }
+}
+
+void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
+{
+    ALsizei i, j;
+
+    for(i = 0;i < 4;i++)
+    {
+        bandsplit_process(&ambiup->XOver[i],
+            ambiup->Samples[FB_HighFreq], ambiup->Samples[FB_LowFreq],
+            InSamples[i], SamplesToDo
+        );
+
+        for(j = 0;j < OutChannels;j++)
+            MixMatrixRow(OutBuffer[j], ambiup->Gains[i][j],
+                SAFE_CONST(ALfloatBUFFERSIZE*,ambiup->Samples), FB_Max, 0,
+                SamplesToDo
+            );
+    }
+}

Alc/bformatdec.h

@@ -0,0 +1,63 @@
+#ifndef BFORMATDEC_H
+#define BFORMATDEC_H
+
+#include "alMain.h"
+
+
+/* These are the necessary scales for first-order HF responses to play over
+ * higher-order 2D (non-periphonic) decoders.
+ */
+#define W_SCALE2D_SECOND   1.224744871f /* sqrt(1.5) */
+#define XYZ_SCALE2D_SECOND 1.0f
+#define W_SCALE2D_THIRD   1.414213562f /* sqrt(2) */
+#define XYZ_SCALE2D_THIRD 1.082392196f
+
+/* These are the necessary scales for first-order HF responses to play over
+ * higher-order 3D (periphonic) decoders.
+ */
+#define W_SCALE3D_SECOND   1.341640787f /* sqrt(1.8) */
+#define XYZ_SCALE3D_SECOND 1.0f
+#define W_SCALE3D_THIRD   1.695486018f
+#define XYZ_SCALE3D_THIRD 1.136697713f
+
+
+struct AmbDecConf;
+struct BFormatDec;
+struct AmbiUpsampler;
+
+
+struct BFormatDec *bformatdec_alloc();
+void bformatdec_free(struct BFormatDec *dec);
+void bformatdec_reset(struct BFormatDec *dec, const struct AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS]);
+
+/* Decodes the ambisonic input to the given output channels. */
+void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
+
+/* Up-samples a first-order input to the decoder's configuration. */
+void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo);
+
+
+/* Stand-alone first-order upsampler. Kept here because it shares some stuff
+ * with bformatdec.
+ */
+struct AmbiUpsampler *ambiup_alloc();
+void ambiup_free(struct AmbiUpsampler *ambiup);
+void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device);
+
+void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
+
+
+/* Band splitter. Splits a signal into two phase-matching frequency bands. */
+typedef struct BandSplitter {
+    ALfloat coeff;
+    ALfloat lp_z1;
+    ALfloat lp_z2;
+    ALfloat hp_z1;
+} BandSplitter;
+
+void bandsplit_init(BandSplitter *splitter, ALfloat freq_mult);
+void bandsplit_clear(BandSplitter *splitter);
+void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout,
+                       const ALfloat *input, ALsizei count);
+
+#endif /* BFORMATDEC_H */

Alc/bs2b.c

@@ -1,201 +1,187 @@
-/*-
- * Copyright (c) 2005 Boris Mikhaylov
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject to
- * the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#include "config.h"
-
-#include <math.h>
-
-#include "bs2b.h"
-
-#ifndef M_PI
-#define M_PI  3.14159265358979323846
-#endif
-
-/* Single pole IIR filter.
- * O[n] = a0*I[n] + a1*I[n-1] + b1*O[n-1]
- */
-
-/* Lowpass filter */
-#define lo_filter(in, out_1) (bs2b->a0_lo*(in) + bs2b->b1_lo*(out_1))
-
-/* Highboost filter */
-#define hi_filter(in, in_1, out_1) (bs2b->a0_hi*(in) + bs2b->a1_hi*(in_1) + bs2b->b1_hi*(out_1))
-
-/* Set up all data. */
-static void init(struct bs2b *bs2b)
-{
-    double Fc_lo, Fc_hi;
-    double G_lo,  G_hi;
-    double x;
-
-    if ((bs2b->srate > 192000) || (bs2b->srate < 2000))
-        bs2b->srate = BS2B_DEFAULT_SRATE;
-
-    switch(bs2b->level)
-    {
-    case BS2B_LOW_CLEVEL: /* Low crossfeed level */
-        Fc_lo = 360.0;
-        Fc_hi = 501.0;
-        G_lo  = 0.398107170553497;
-        G_hi  = 0.205671765275719;
-        break;
-
-    case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */
-        Fc_lo = 500.0;
-        Fc_hi = 711.0;
-        G_lo  = 0.459726988530872;
-        G_hi  = 0.228208484414988;
-        break;
-
-    case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */
-        Fc_lo = 700.0;
-        Fc_hi = 1021.0;
-        G_lo  = 0.530884444230988;
-        G_hi  = 0.250105790667544;
-        break;
-
-    case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */
-        Fc_lo = 360.0;
-        Fc_hi = 494.0;
-        G_lo  = 0.316227766016838;
-        G_hi  = 0.168236228897329;
-        break;
-
-    case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */
-        Fc_lo = 500.0;
-        Fc_hi = 689.0;
-        G_lo  = 0.354813389233575;
-        G_hi  = 0.187169483835901;
-        break;
-
-    default: /* High easy crossfeed level */
-        bs2b->level = BS2B_HIGH_ECLEVEL;
-
-        Fc_lo = 700.0;
-        Fc_hi = 975.0;
-        G_lo  = 0.398107170553497;
-        G_hi  = 0.205671765275719;
-        break;
-    } /* switch */
-
-    /* $fc = $Fc / $s;
-     * $d  = 1 / 2 / pi / $fc;
-     * $x  = exp(-1 / $d);
-     */
-
-    x           = exp(-2.0 * M_PI * Fc_lo / bs2b->srate);
-    bs2b->b1_lo = x;
-    bs2b->a0_lo = G_lo * (1.0 - x);
-
-    x           = exp(-2.0 * M_PI * Fc_hi / bs2b->srate);
-    bs2b->b1_hi = x;
-    bs2b->a0_hi = 1.0 - G_hi * (1.0 - x);
-    bs2b->a1_hi = -x;
-
-    bs2b->gain  = 1.0 / (1.0 - G_hi + G_lo);
-} /* init */
-
-/* Exported functions.
- * See descriptions in "bs2b.h"
- */
-
-void bs2b_set_level(struct bs2b *bs2b, int level)
-{
-    if(level == bs2b->level)
-        return;
-    bs2b->level = level;
-    init(bs2b);
-} /* bs2b_set_level */
-
-int bs2b_get_level(struct bs2b *bs2b)
-{
-    return bs2b->level;
-} /* bs2b_get_level */
-
-void bs2b_set_srate(struct bs2b *bs2b, int srate)
-{
-    if (srate == bs2b->srate)
-        return;
-    bs2b->srate = srate;
-    init(bs2b);
-} /* bs2b_set_srate */
-
-int bs2b_get_srate(struct bs2b *bs2b)
-{
-    return bs2b->srate;
-} /* bs2b_get_srate */
-
-void bs2b_clear(struct bs2b *bs2b)
-{
-    int loopv = sizeof(bs2b->last_sample);
-
-    while (loopv)
-    {
-        ((char *)&bs2b->last_sample)[--loopv] = 0;
-    }
-} /* bs2b_clear */
-
-int bs2b_is_clear(struct bs2b *bs2b)
-{
-    int loopv = sizeof(bs2b->last_sample);
-
-    while (loopv)
-    {
-        if (((char *)&bs2b->last_sample)[--loopv] != 0)
-            return 0;
-    }
-    return 1;
-} /* bs2b_is_clear */
-
-void bs2b_cross_feed(struct bs2b *bs2b, float *sample)
-{
-    /* Lowpass filter */
-    bs2b->last_sample.lo[0] = lo_filter(sample[0], bs2b->last_sample.lo[0]);
-    bs2b->last_sample.lo[1] = lo_filter(sample[1], bs2b->last_sample.lo[1]);
-
-    /* Highboost filter */
-    bs2b->last_sample.hi[0] = hi_filter(sample[0], bs2b->last_sample.asis[0], bs2b->last_sample.hi[0]);
-    bs2b->last_sample.hi[1] = hi_filter(sample[1], bs2b->last_sample.asis[1], bs2b->last_sample.hi[1]);
-    bs2b->last_sample.asis[0] = sample[0];
-    bs2b->last_sample.asis[1] = sample[1];
-
-    /* Crossfeed */
-    sample[0] = bs2b->last_sample.hi[0] + bs2b->last_sample.lo[1];
-    sample[1] = bs2b->last_sample.hi[1] + bs2b->last_sample.lo[0];
-
-    /* Bass boost cause allpass attenuation */
-    sample[0] *= bs2b->gain;
-    sample[1] *= bs2b->gain;
-
-    /* Clipping of overloaded samples */
-#if 0
-    if (sample[0] > 1.0)
-        sample[0] = 1.0;
-    if (sample[0] < -1.0)
-        sample[0] = -1.0;
-    if (sample[1] > 1.0)
-        sample[1] = 1.0;
-    if (sample[1] < -1.0)
-        sample[1] = -1.0;
-#endif
-} /* bs2b_cross_feed */
+/*-
+ * Copyright (c) 2005 Boris Mikhaylov
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files (the
+ * "Software"), to deal in the Software without restriction, including
+ * without limitation the rights to use, copy, modify, merge, publish,
+ * distribute, sublicense, and/or sell copies of the Software, and to
+ * permit persons to whom the Software is furnished to do so, subject to
+ * the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
+ * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
+ * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
+ * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <string.h>
+
+#include "bs2b.h"
+#include "alu.h"
+
+
+/* Set up all data. */
+static void init(struct bs2b *bs2b)
+{
+    float Fc_lo, Fc_hi;
+    float G_lo,  G_hi;
+    float x, g;
+
+    switch(bs2b->level)
+    {
+    case BS2B_LOW_CLEVEL: /* Low crossfeed level */
+        Fc_lo = 360.0f;
+        Fc_hi = 501.0f;
+        G_lo  = 0.398107170553497f;
+        G_hi  = 0.205671765275719f;
+        break;
+
+    case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */
+        Fc_lo = 500.0f;
+        Fc_hi = 711.0f;
+        G_lo  = 0.459726988530872f;
+        G_hi  = 0.228208484414988f;
+        break;
+
+    case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */
+        Fc_lo = 700.0f;
+        Fc_hi = 1021.0f;
+        G_lo  = 0.530884444230988f;
+        G_hi  = 0.250105790667544f;
+        break;
+
+    case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */
+        Fc_lo = 360.0f;
+        Fc_hi = 494.0f;
+        G_lo  = 0.316227766016838f;
+        G_hi  = 0.168236228897329f;
+        break;
+
+    case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */
+        Fc_lo = 500.0f;
+        Fc_hi = 689.0f;
+        G_lo  = 0.354813389233575f;
+        G_hi  = 0.187169483835901f;
+        break;
+
+    default: /* High easy crossfeed level */
+        bs2b->level = BS2B_HIGH_ECLEVEL;
+
+        Fc_lo = 700.0f;
+        Fc_hi = 975.0f;
+        G_lo  = 0.398107170553497f;
+        G_hi  = 0.205671765275719f;
+        break;
+    } /* switch */
+
+    g = 1.0f / (1.0f - G_hi + G_lo);
+
+    /* $fc = $Fc / $s;
+     * $d  = 1 / 2 / pi / $fc;
+     * $x  = exp(-1 / $d);
+     */
+    x           = expf(-2.0f * F_PI * Fc_lo / bs2b->srate);
+    bs2b->b1_lo = x;
+    bs2b->a0_lo = G_lo * (1.0f - x) * g;
+
+    x           = expf(-2.0f * F_PI * Fc_hi / bs2b->srate);
+    bs2b->b1_hi = x;
+    bs2b->a0_hi = (1.0f - G_hi * (1.0f - x)) * g;
+    bs2b->a1_hi = -x * g;
+} /* init */
+
+
+/* Exported functions.
+ * See descriptions in "bs2b.h"
+ */
+
+void bs2b_set_params(struct bs2b *bs2b, int level, int srate)
+{
+    if(srate <= 0) srate = 1;
+
+    bs2b->level = level;
+    bs2b->srate = srate;
+    init(bs2b);
+} /* bs2b_set_params */
+
+int bs2b_get_level(struct bs2b *bs2b)
+{
+    return bs2b->level;
+} /* bs2b_get_level */
+
+int bs2b_get_srate(struct bs2b *bs2b)
+{
+    return bs2b->srate;
+} /* bs2b_get_srate */
+
+void bs2b_clear(struct bs2b *bs2b)
+{
+    memset(&bs2b->last_sample, 0, sizeof(bs2b->last_sample));
+} /* bs2b_clear */
+
+void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo)
+{
+    float lsamples[128][2];
+    float rsamples[128][2];
+    int base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        int todo = mini(128, SamplesToDo-base);
+        int i;
+
+        /* Process left input */
+        lsamples[0][0] = bs2b->a0_lo*Left[0] +
+                         bs2b->b1_lo*bs2b->last_sample[0].lo;
+        lsamples[0][1] = bs2b->a0_hi*Left[0] +
+                         bs2b->a1_hi*bs2b->last_sample[0].asis +
+                         bs2b->b1_hi*bs2b->last_sample[0].hi;
+        for(i = 1;i < todo;i++)
+        {
+            lsamples[i][0] = bs2b->a0_lo*Left[i] +
+                             bs2b->b1_lo*lsamples[i-1][0];
+            lsamples[i][1] = bs2b->a0_hi*Left[i] +
+                             bs2b->a1_hi*Left[i-1] +
+                             bs2b->b1_hi*lsamples[i-1][1];
+        }
+        bs2b->last_sample[0].asis = Left[i-1];
+        bs2b->last_sample[0].lo = lsamples[i-1][0];
+        bs2b->last_sample[0].hi = lsamples[i-1][1];
+
+        /* Process right input */
+        rsamples[0][0] = bs2b->a0_lo*Right[0] +
+                         bs2b->b1_lo*bs2b->last_sample[1].lo;
+        rsamples[0][1] = bs2b->a0_hi*Right[0] +
+                         bs2b->a1_hi*bs2b->last_sample[1].asis +
+                         bs2b->b1_hi*bs2b->last_sample[1].hi;
+        for(i = 1;i < todo;i++)
+        {
+            rsamples[i][0] = bs2b->a0_lo*Right[i] +
+                             bs2b->b1_lo*rsamples[i-1][0];
+            rsamples[i][1] = bs2b->a0_hi*Right[i] +
+                             bs2b->a1_hi*Right[i-1] +
+                             bs2b->b1_hi*rsamples[i-1][1];
+        }
+        bs2b->last_sample[1].asis = Right[i-1];
+        bs2b->last_sample[1].lo = rsamples[i-1][0];
+        bs2b->last_sample[1].hi = rsamples[i-1][1];
+
+        /* Crossfeed */
+        for(i = 0;i < todo;i++)
+            *(Left++) = lsamples[i][1] + rsamples[i][0];
+        for(i = 0;i < todo;i++)
+            *(Right++) = rsamples[i][1] + lsamples[i][0];
+
+        base += todo;
+    }
+} /* bs2b_cross_feed */

Alc/compat.h

@@ -0,0 +1,57 @@
+#ifndef AL_COMPAT_H
+#define AL_COMPAT_H
+
+#include "alstring.h"
+
+#ifdef _WIN32
+
+#define WIN32_LEAN_AND_MEAN
+#include <windows.h>
+
+WCHAR *strdupW(const WCHAR *str);
+
+/* Opens a file with standard I/O. The filename is expected to be UTF-8. */
+FILE *al_fopen(const char *fname, const char *mode);
+
+#define HAVE_DYNLOAD 1
+
+#else
+
+#define al_fopen fopen
+
+#if defined(HAVE_DLFCN_H) && !defined(IN_IDE_PARSER)
+#define HAVE_DYNLOAD 1
+#endif
+
+#endif
+
+struct FileMapping {
+#ifdef _WIN32
+    HANDLE file;
+    HANDLE fmap;
+#else
+    int fd;
+#endif
+    void *ptr;
+    size_t len;
+};
+struct FileMapping MapFileToMem(const char *fname);
+void UnmapFileMem(const struct FileMapping *mapping);
+
+al_string GetProcPath(void);
+
+#ifdef HAVE_DYNLOAD
+void *LoadLib(const char *name);
+void CloseLib(void *handle);
+void *GetSymbol(void *handle, const char *name);
+#endif
+
+#ifdef __ANDROID__
+#define JCALL(obj, func)  ((*(obj))->func((obj), EXTRACT_VCALL_ARGS
+#define JCALL0(obj, func)  ((*(obj))->func((obj) EXTRACT_VCALL_ARGS
+
+/** Returns a JNIEnv*. */
+void *Android_GetJNIEnv(void);
+#endif
+
+#endif /* AL_COMPAT_H */

Alc/converter.c

@@ -0,0 +1,452 @@
+
+#include "config.h"
+
+#include "converter.h"
+
+#include "mixer_defs.h"
+
+
+SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate)
+{
+    SampleConverter *converter;
+
+    if(numchans <= 0 || srcRate <= 0 || dstRate <= 0)
+        return NULL;
+
+    converter = al_calloc(16, FAM_SIZE(SampleConverter, Chan, numchans));
+    converter->mSrcType = srcType;
+    converter->mDstType = dstType;
+    converter->mNumChannels = numchans;
+    converter->mSrcTypeSize = BytesFromDevFmt(srcType);
+    converter->mDstTypeSize = BytesFromDevFmt(dstType);
+
+    converter->mSrcPrepCount = 0;
+
+    converter->mFracOffset = 0;
+    converter->mIncrement = (ALsizei)clampu64((ALuint64)srcRate*FRACTIONONE/dstRate,
+                                              1, MAX_PITCH*FRACTIONONE);
+    if(converter->mIncrement == FRACTIONONE)
+        converter->mResample = Resample_copy32_C;
+    else
+    {
+        /* TODO: Allow other resamplers. */
+        converter->mResample = SelectResampler(LinearResampler);
+    }
+
+    return converter;
+}
+
+void DestroySampleConverter(SampleConverter **converter)
+{
+    if(converter)
+    {
+        al_free(*converter);
+        *converter = NULL;
+    }
+}
+
+
+static inline ALfloat Sample_ALbyte(ALbyte val)
+{ return val * (1.0f/128.0f); }
+static inline ALfloat Sample_ALubyte(ALubyte val)
+{ return Sample_ALbyte((ALint)val - 128); }
+
+static inline ALfloat Sample_ALshort(ALshort val)
+{ return val * (1.0f/32768.0f); }
+static inline ALfloat Sample_ALushort(ALushort val)
+{ return Sample_ALshort((ALint)val - 32768); }
+
+static inline ALfloat Sample_ALint(ALint val)
+{ return (val>>7) * (1.0f/16777216.0f); }
+static inline ALfloat Sample_ALuint(ALuint val)
+{ return ((ALint)(val>>7) - 16777216) * (1.0f/16777216.0f); }
+
+static inline ALfloat Sample_ALfloat(ALfloat val)
+{ return val; }
+
+#define DECL_TEMPLATE(T)                                                      \
+static inline void Load_##T(ALfloat *restrict dst, const T *restrict src,     \
+                            ALint srcstep, ALsizei samples)                   \
+{                                                                             \
+    ALsizei i;                                                                \
+    for(i = 0;i < samples;i++)                                                \
+        dst[i] = Sample_##T(src[i*srcstep]);                                  \
+}
+
+DECL_TEMPLATE(ALbyte)
+DECL_TEMPLATE(ALubyte)
+DECL_TEMPLATE(ALshort)
+DECL_TEMPLATE(ALushort)
+DECL_TEMPLATE(ALint)
+DECL_TEMPLATE(ALuint)
+DECL_TEMPLATE(ALfloat)
+
+#undef DECL_TEMPLATE
+
+static void LoadSamples(ALfloat *dst, const ALvoid *src, ALint srcstep, enum DevFmtType srctype, ALsizei samples)
+{
+    switch(srctype)
+    {
+        case DevFmtByte:
+            Load_ALbyte(dst, src, srcstep, samples);
+            break;
+        case DevFmtUByte:
+            Load_ALubyte(dst, src, srcstep, samples);
+            break;
+        case DevFmtShort:
+            Load_ALshort(dst, src, srcstep, samples);
+            break;
+        case DevFmtUShort:
+            Load_ALushort(dst, src, srcstep, samples);
+            break;
+        case DevFmtInt:
+            Load_ALint(dst, src, srcstep, samples);
+            break;
+        case DevFmtUInt:
+            Load_ALuint(dst, src, srcstep, samples);
+            break;
+        case DevFmtFloat:
+            Load_ALfloat(dst, src, srcstep, samples);
+            break;
+    }
+}
+
+
+static inline ALbyte ALbyte_Sample(ALfloat val)
+{ return (ALbyte)clampf(val*128.0f, -128.0f, 127.0f); }
+static inline ALubyte ALubyte_Sample(ALfloat val)
+{ return ALbyte_Sample(val)+128; }
+
+static inline ALshort ALshort_Sample(ALfloat val)
+{ return (ALshort)clampf(val*32768.0f, -32768.0f, 32767.0f); }
+static inline ALushort ALushort_Sample(ALfloat val)
+{ return ALshort_Sample(val)+32768; }
+
+static inline ALint ALint_Sample(ALfloat val)
+{ return (ALint)clampf(val*16777216.0f, -16777216.0f, 16777215.0f) << 7; }
+static inline ALuint ALuint_Sample(ALfloat val)
+{ return ALint_Sample(val)+INT_MAX+1; }
+
+static inline ALfloat ALfloat_Sample(ALfloat val)
+{ return val; }
+
+#define DECL_TEMPLATE(T)                                                      \
+static inline void Store_##T(T *restrict dst, const ALfloat *restrict src,    \
+                             ALint dststep, ALsizei samples)                  \
+{                                                                             \
+    ALsizei i;                                                                \
+    for(i = 0;i < samples;i++)                                                \
+        dst[i*dststep] = T##_Sample(src[i]);                                  \
+}
+
+DECL_TEMPLATE(ALbyte)
+DECL_TEMPLATE(ALubyte)
+DECL_TEMPLATE(ALshort)
+DECL_TEMPLATE(ALushort)
+DECL_TEMPLATE(ALint)
+DECL_TEMPLATE(ALuint)
+DECL_TEMPLATE(ALfloat)
+
+#undef DECL_TEMPLATE
+
+static void StoreSamples(ALvoid *dst, const ALfloat *src, ALint dststep, enum DevFmtType dsttype, ALsizei samples)
+{
+    switch(dsttype)
+    {
+        case DevFmtByte:
+            Store_ALbyte(dst, src, dststep, samples);
+            break;
+        case DevFmtUByte:
+            Store_ALubyte(dst, src, dststep, samples);
+            break;
+        case DevFmtShort:
+            Store_ALshort(dst, src, dststep, samples);
+            break;
+        case DevFmtUShort:
+            Store_ALushort(dst, src, dststep, samples);
+            break;
+        case DevFmtInt:
+            Store_ALint(dst, src, dststep, samples);
+            break;
+        case DevFmtUInt:
+            Store_ALuint(dst, src, dststep, samples);
+            break;
+        case DevFmtFloat:
+            Store_ALfloat(dst, src, dststep, samples);
+            break;
+    }
+}
+
+
+ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes)
+{
+    ALint prepcount = converter->mSrcPrepCount;
+    ALsizei increment = converter->mIncrement;
+    ALsizei DataPosFrac = converter->mFracOffset;
+    ALuint64 DataSize64;
+
+    if(prepcount < 0)
+    {
+        /* Negative prepcount means we need to skip that many input samples. */
+        if(-prepcount >= srcframes)
+            return 0;
+        srcframes += prepcount;
+        prepcount = 0;
+    }
+
+    if(prepcount < MAX_POST_SAMPLES+MAX_PRE_SAMPLES &&
+       MAX_POST_SAMPLES+MAX_PRE_SAMPLES-prepcount >= srcframes)
+    {
+        /* Not enough input samples to generate an output sample. */
+        return 0;
+    }
+
+    DataSize64  = prepcount;
+    DataSize64 += srcframes;
+    DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
+    DataSize64 <<= FRACTIONBITS;
+    DataSize64 -= DataPosFrac;
+
+    /* If we have a full prep, we can generate at least one sample. */
+    return (ALsizei)clampu64(DataSize64/increment, 1, INT_MAX);
+}
+
+
+ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes)
+{
+    const ALsizei SrcFrameSize = converter->mNumChannels * converter->mSrcTypeSize;
+    const ALsizei DstFrameSize = converter->mNumChannels * converter->mDstTypeSize;
+    const ALsizei increment = converter->mIncrement;
+    ALsizei pos = 0;
+
+    while(pos < dstframes)
+    {
+        ALfloat *restrict SrcData = ASSUME_ALIGNED(converter->mSrcSamples, 16);
+        ALfloat *restrict DstData = ASSUME_ALIGNED(converter->mDstSamples, 16);
+        ALint prepcount = converter->mSrcPrepCount;
+        ALsizei DataPosFrac = converter->mFracOffset;
+        ALuint64 DataSize64;
+        ALsizei DstSize;
+        ALint toread;
+        ALsizei chan;
+
+        if(prepcount < 0)
+        {
+            /* Negative prepcount means we need to skip that many input samples. */
+            if(-prepcount >= *srcframes)
+            {
+                converter->mSrcPrepCount = prepcount + *srcframes;
+                *srcframes = 0;
+                break;
+            }
+            *src = (const ALbyte*)*src + SrcFrameSize*-prepcount;
+            *srcframes += prepcount;
+            prepcount = 0;
+        }
+        toread = mini(*srcframes, BUFFERSIZE-(MAX_POST_SAMPLES+MAX_PRE_SAMPLES));
+
+        if(prepcount < MAX_POST_SAMPLES+MAX_PRE_SAMPLES &&
+           MAX_POST_SAMPLES+MAX_PRE_SAMPLES-prepcount >= toread)
+        {
+            /* Not enough input samples to generate an output sample. Store
+             * what we're given for later.
+             */
+            for(chan = 0;chan < converter->mNumChannels;chan++)
+                LoadSamples(&converter->Chan[chan].mPrevSamples[prepcount],
+                    (const ALbyte*)src + converter->mSrcTypeSize*chan,
+                    converter->mNumChannels, converter->mSrcType, toread
+                );
+
+            converter->mSrcPrepCount = prepcount + toread;
+            *srcframes = 0;
+            break;
+        }
+
+        DataSize64  = prepcount;
+        DataSize64 += toread;
+        DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
+        DataSize64 <<= FRACTIONBITS;
+        DataSize64 -= DataPosFrac;
+
+        /* If we have a full prep, we can generate at least one sample. */
+        DstSize = (ALsizei)clampu64(DataSize64/increment, 1, BUFFERSIZE);
+        DstSize = mini(DstSize, dstframes-pos);
+
+        for(chan = 0;chan < converter->mNumChannels;chan++)
+        {
+            const ALbyte *SrcSamples = (const ALbyte*)*src + converter->mSrcTypeSize*chan;
+            ALbyte *DstSamples = (ALbyte*)dst + converter->mSrcTypeSize*chan;
+            const ALfloat *ResampledData;
+            ALsizei SrcDataEnd;
+
+            /* Load the previous samples into the source data first, then the
+             * new samples from the input buffer.
+             */
+            memcpy(SrcData, converter->Chan[chan].mPrevSamples,
+                   prepcount*sizeof(ALfloat));
+            LoadSamples(SrcData + prepcount, SrcSamples,
+                converter->mNumChannels, converter->mSrcType, toread
+            );
+
+            /* Store as many prep samples for next time as possible, given the
+             * number of output samples being generated.
+             */
+            SrcDataEnd = (DataPosFrac + increment*DstSize)>>FRACTIONBITS;
+            if(SrcDataEnd >= prepcount+toread)
+                memset(converter->Chan[chan].mPrevSamples, 0,
+                    sizeof(converter->Chan[chan].mPrevSamples));
+            else
+            {
+                size_t len = mini(MAX_PRE_SAMPLES+MAX_POST_SAMPLES, prepcount+toread-SrcDataEnd);
+                memcpy(converter->Chan[chan].mPrevSamples, &SrcData[SrcDataEnd],
+                       len*sizeof(ALfloat));
+                memset(converter->Chan[chan].mPrevSamples+len, 0,
+                       sizeof(converter->Chan[chan].mPrevSamples) - len*sizeof(ALfloat));
+            }
+
+            /* Now resample, and store the result in the output buffer. */
+            ResampledData = converter->mResample(NULL,
+                SrcData+MAX_PRE_SAMPLES, DataPosFrac, increment,
+                DstData, DstSize
+            );
+
+            StoreSamples(DstSamples, ResampledData, converter->mNumChannels,
+                         converter->mDstType, DstSize);
+        }
+
+        /* Update the number of prep samples still available, as well as the
+         * fractional offset.
+         */
+        DataPosFrac += increment*DstSize;
+        converter->mSrcPrepCount = mini(MAX_PRE_SAMPLES+MAX_POST_SAMPLES,
+                                        prepcount+toread-(DataPosFrac>>FRACTIONBITS));
+        converter->mFracOffset = DataPosFrac & FRACTIONMASK;
+
+        /* Update the src and dst pointers in case there's still more to do. */
+        *src = (const ALbyte*)*src + SrcFrameSize*(DataPosFrac>>FRACTIONBITS);
+        *srcframes -= mini(*srcframes, (DataPosFrac>>FRACTIONBITS));
+
+        dst = (ALbyte*)dst + DstFrameSize*DstSize;
+        pos += DstSize;
+    }
+
+    return pos;
+}
+
+
+ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans)
+{
+    ChannelConverter *converter;
+
+    if(srcChans != dstChans && !((srcChans == DevFmtMono && dstChans == DevFmtStereo) ||
+                                 (srcChans == DevFmtStereo && dstChans == DevFmtMono)))
+        return NULL;
+
+    converter = al_calloc(DEF_ALIGN, sizeof(*converter));
+    converter->mSrcType = srcType;
+    converter->mSrcChans = srcChans;
+    converter->mDstChans = dstChans;
+
+    return converter;
+}
+
+void DestroyChannelConverter(ChannelConverter **converter)
+{
+    if(converter)
+    {
+        al_free(*converter);
+        *converter = NULL;
+    }
+}
+
+
+#define DECL_TEMPLATE(T)                                                       \
+static void Mono2Stereo##T(ALfloat *restrict dst, const T *src, ALsizei frames)\
+{                                                                              \
+    ALsizei i;                                                                 \
+    for(i = 0;i < frames;i++)                                                  \
+        dst[i*2 + 1] = dst[i*2 + 0] = Sample_##T(src[i]) * 0.707106781187f;    \
+}                                                                              \
+                                                                               \
+static void Stereo2Mono##T(ALfloat *restrict dst, const T *src, ALsizei frames)\
+{                                                                              \
+    ALsizei i;                                                                 \
+    for(i = 0;i < frames;i++)                                                  \
+        dst[i] = (Sample_##T(src[i*2 + 0])+Sample_##T(src[i*2 + 1])) *         \
+                 0.707106781187f;                                              \
+}
+
+DECL_TEMPLATE(ALbyte)
+DECL_TEMPLATE(ALubyte)
+DECL_TEMPLATE(ALshort)
+DECL_TEMPLATE(ALushort)
+DECL_TEMPLATE(ALint)
+DECL_TEMPLATE(ALuint)
+DECL_TEMPLATE(ALfloat)
+
+#undef DECL_TEMPLATE
+
+void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames)
+{
+    if(converter->mSrcChans == converter->mDstChans)
+    {
+        LoadSamples(dst, src, 1, converter->mSrcType,
+                    frames*ChannelsFromDevFmt(converter->mSrcChans, 0));
+        return;
+    }
+
+    if(converter->mSrcChans == DevFmtStereo && converter->mDstChans == DevFmtMono)
+    {
+        switch(converter->mSrcType)
+        {
+            case DevFmtByte:
+                Stereo2MonoALbyte(dst, src, frames);
+                break;
+            case DevFmtUByte:
+                Stereo2MonoALubyte(dst, src, frames);
+                break;
+            case DevFmtShort:
+                Stereo2MonoALshort(dst, src, frames);
+                break;
+            case DevFmtUShort:
+                Stereo2MonoALushort(dst, src, frames);
+                break;
+            case DevFmtInt:
+                Stereo2MonoALint(dst, src, frames);
+                break;
+            case DevFmtUInt:
+                Stereo2MonoALuint(dst, src, frames);
+                break;
+            case DevFmtFloat:
+                Stereo2MonoALfloat(dst, src, frames);
+                break;
+        }
+    }
+    else /*if(converter->mSrcChans == DevFmtMono && converter->mDstChans == DevFmtStereo)*/
+    {
+        switch(converter->mSrcType)
+        {
+            case DevFmtByte:
+                Mono2StereoALbyte(dst, src, frames);
+                break;
+            case DevFmtUByte:
+                Mono2StereoALubyte(dst, src, frames);
+                break;
+            case DevFmtShort:
+                Mono2StereoALshort(dst, src, frames);
+                break;
+            case DevFmtUShort:
+                Mono2StereoALushort(dst, src, frames);
+                break;
+            case DevFmtInt:
+                Mono2StereoALint(dst, src, frames);
+                break;
+            case DevFmtUInt:
+                Mono2StereoALuint(dst, src, frames);
+                break;
+            case DevFmtFloat:
+                Mono2StereoALfloat(dst, src, frames);
+                break;
+        }
+    }
+}

Alc/converter.h

@@ -0,0 +1,54 @@
+#ifndef CONVERTER_H
+#define CONVERTER_H
+
+#include "alMain.h"
+#include "alu.h"
+
+#ifdef __cpluspluc
+extern "C" {
+#endif
+
+typedef struct SampleConverter {
+    enum DevFmtType mSrcType;
+    enum DevFmtType mDstType;
+    ALsizei mNumChannels;
+    ALsizei mSrcTypeSize;
+    ALsizei mDstTypeSize;
+
+    ALint mSrcPrepCount;
+
+    ALsizei mFracOffset;
+    ALsizei mIncrement;
+    ResamplerFunc mResample;
+
+    alignas(16) ALfloat mSrcSamples[BUFFERSIZE];
+    alignas(16) ALfloat mDstSamples[BUFFERSIZE];
+
+    struct {
+        alignas(16) ALfloat mPrevSamples[MAX_PRE_SAMPLES+MAX_POST_SAMPLES];
+    } Chan[];
+} SampleConverter;
+
+SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate);
+void DestroySampleConverter(SampleConverter **converter);
+
+ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes);
+ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes);
+
+
+typedef struct ChannelConverter {
+    enum DevFmtType mSrcType;
+    enum DevFmtChannels mSrcChans;
+    enum DevFmtChannels mDstChans;
+} ChannelConverter;
+
+ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans);
+void DestroyChannelConverter(ChannelConverter **converter);
+
+void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames);
+
+#ifdef __cpluspluc
+}
+#endif
+
+#endif /* CONVERTER_H */

Alc/dsound.c

@@ -1,572 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#define _WIN32_WINNT 0x0500
-#define INITGUID
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-
-#include <dsound.h>
-#include <mmreg.h>
-#ifndef _WAVEFORMATEXTENSIBLE_
-#include <ks.h>
-#include <ksmedia.h>
-#endif
-
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-
-#ifndef DSSPEAKER_5POINT1
-#define DSSPEAKER_5POINT1       6
-#endif
-#ifndef DSSPEAKER_7POINT1
-#define DSSPEAKER_7POINT1       7
-#endif
-
-DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-
-static void *ds_handle;
-static HRESULT (WINAPI *pDirectSoundCreate)(LPCGUID pcGuidDevice, LPDIRECTSOUND *ppDS, LPUNKNOWN pUnkOuter);
-static HRESULT (WINAPI *pDirectSoundEnumerateA)(LPDSENUMCALLBACKA pDSEnumCallback, LPVOID pContext);
-
-
-typedef struct {
-    // DirectSound Playback Device
-    LPDIRECTSOUND          lpDS;
-    LPDIRECTSOUNDBUFFER    DSpbuffer;
-    LPDIRECTSOUNDBUFFER    DSsbuffer;
-
-    volatile int killNow;
-    ALvoid *thread;
-} DSoundData;
-
-
-typedef struct {
-    ALCchar *name;
-    GUID guid;
-} DevMap;
-
-static const ALCchar dsDevice[] = "DirectSound Software";
-static DevMap *DeviceList;
-static ALuint NumDevices;
-static volatile ALuint load_count;
-
-
-void DSoundLoad(void)
-{
-    if(load_count == 0)
-    {
-#ifdef _WIN32
-        ds_handle = LoadLibraryA("dsound.dll");
-        if(ds_handle == NULL)
-        {
-            AL_PRINT("Failed to load dsound.dll\n");
-            return;
-        }
-
-#define LOAD_FUNC(f) do { \
-    p##f = (void*)GetProcAddress((HMODULE)ds_handle, #f); \
-    if(p##f == NULL) \
-    { \
-        FreeLibrary(ds_handle); \
-        ds_handle = NULL; \
-        AL_PRINT("Could not load %s from dsound.dll\n", #f); \
-        return; \
-    } \
-} while(0)
-#else
-        ds_handle = (void*)0xDEADBEEF;
-#define LOAD_FUNC(f) p##f = f
-#endif
-
-LOAD_FUNC(DirectSoundCreate);
-LOAD_FUNC(DirectSoundEnumerateA);
-#undef LOAD_FUNC
-    }
-    ++load_count;
-}
-
-void DSoundUnload(void)
-{
-    if(load_count == 0 || --load_count > 0)
-        return;
-
-#ifdef _WIN32
-    FreeLibrary(ds_handle);
-#endif
-    ds_handle = NULL;
-}
-
-
-static ALuint DSoundProc(ALvoid *ptr)
-{
-    ALCdevice *pDevice = (ALCdevice*)ptr;
-    DSoundData *pData = (DSoundData*)pDevice->ExtraData;
-    DSBCAPS DSBCaps;
-    DWORD LastCursor = 0;
-    DWORD PlayCursor;
-    VOID *WritePtr1, *WritePtr2;
-    DWORD WriteCnt1,  WriteCnt2;
-    DWORD FrameSize;
-    DWORD FragSize;
-    DWORD avail;
-    HRESULT err;
-
-    memset(&DSBCaps, 0, sizeof(DSBCaps));
-    DSBCaps.dwSize = sizeof(DSBCaps);
-    err = IDirectSoundBuffer_GetCaps(pData->DSsbuffer, &DSBCaps);
-    if(FAILED(err))
-    {
-        AL_PRINT("Failed to get buffer caps: 0x%lx\n", err);
-        aluHandleDisconnect(pDevice);
-        return 1;
-    }
-
-    FrameSize = aluChannelsFromFormat(pDevice->Format) *
-                aluBytesFromFormat(pDevice->Format);
-    FragSize = pDevice->UpdateSize * FrameSize;
-
-    IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &LastCursor, NULL);
-    while(!pData->killNow)
-    {
-        // Get current play and write cursors
-        IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &PlayCursor, NULL);
-        avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes;
-
-        if(avail < FragSize)
-        {
-            Sleep(1);
-            continue;
-        }
-        avail -= avail%FragSize;
-
-        // Lock output buffer
-        WriteCnt1 = 0;
-        WriteCnt2 = 0;
-        err = IDirectSoundBuffer_Lock(pData->DSsbuffer, LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
-
-        // If the buffer is lost, restore it, play and lock
-        if(err == DSERR_BUFFERLOST)
-        {
-            err = IDirectSoundBuffer_Restore(pData->DSsbuffer);
-            if(SUCCEEDED(err))
-                err = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING);
-            if(SUCCEEDED(err))
-                err = IDirectSoundBuffer_Lock(pData->DSsbuffer, LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
-        }
-
-        // Successfully locked the output buffer
-        if(SUCCEEDED(err))
-        {
-            // If we have an active context, mix data directly into output buffer otherwise fill with silence
-            aluMixData(pDevice, WritePtr1, WriteCnt1/FrameSize);
-            aluMixData(pDevice, WritePtr2, WriteCnt2/FrameSize);
-
-            // Unlock output buffer only when successfully locked
-            IDirectSoundBuffer_Unlock(pData->DSsbuffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2);
-        }
-        else
-            AL_PRINT("Buffer lock error: %#lx\n", err);
-
-        // Update old write cursor location
-        LastCursor += WriteCnt1+WriteCnt2;
-        LastCursor %= DSBCaps.dwBufferBytes;
-    }
-
-    return 0;
-}
-
-static ALCboolean DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceName)
-{
-    DSoundData *pData = NULL;
-    LPGUID guid = NULL;
-    HRESULT hr;
-
-    if(!deviceName)
-        deviceName = dsDevice;
-    else if(strcmp(deviceName, dsDevice) != 0)
-    {
-        ALuint i;
-        for(i = 0;i < NumDevices;i++)
-        {
-            if(strcmp(deviceName, DeviceList[i].name) == 0)
-            {
-                guid = &DeviceList[i].guid;
-                break;
-            }
-        }
-        if(i == NumDevices)
-            return ALC_FALSE;
-    }
-
-    DSoundLoad();
-    if(ds_handle == NULL)
-        return ALC_FALSE;
-
-    //Initialise requested device
-
-    pData = calloc(1, sizeof(DSoundData));
-    if(!pData)
-    {
-        alcSetError(ALC_OUT_OF_MEMORY);
-        DSoundUnload();
-        return ALC_FALSE;
-    }
-
-    //DirectSound Init code
-    hr = pDirectSoundCreate(guid, &pData->lpDS, NULL);
-    if(SUCCEEDED(hr))
-        hr = IDirectSound_SetCooperativeLevel(pData->lpDS, GetForegroundWindow(), DSSCL_PRIORITY);
-    if(FAILED(hr))
-    {
-        if(pData->lpDS)
-            IDirectSound_Release(pData->lpDS);
-        free(pData);
-        DSoundUnload();
-        return ALC_FALSE;
-    }
-
-    device->szDeviceName = strdup(deviceName);
-    device->ExtraData = pData;
-    return ALC_TRUE;
-}
-
-static void DSoundClosePlayback(ALCdevice *device)
-{
-    DSoundData *pData = device->ExtraData;
-
-    IDirectSound_Release(pData->lpDS);
-    free(pData);
-    device->ExtraData = NULL;
-
-    DSoundUnload();
-}
-
-static ALCboolean DSoundResetPlayback(ALCdevice *device)
-{
-    DSoundData *pData = (DSoundData*)device->ExtraData;
-    DSBUFFERDESC DSBDescription;
-    WAVEFORMATEXTENSIBLE OutputType;
-    DWORD frameSize = 0;
-    ALenum format = 0;
-    DWORD speakers;
-    HRESULT hr;
-
-    memset(&OutputType, 0, sizeof(OutputType));
-
-    hr = IDirectSound_GetSpeakerConfig(pData->lpDS, &speakers);
-    if(SUCCEEDED(hr) && *(GetConfigValue(NULL, "format", "")) != 0)
-    {
-        if(aluChannelsFromFormat(device->Format) == 1)
-            speakers = DSSPEAKER_COMBINED(DSSPEAKER_MONO, 0);
-        else if(aluChannelsFromFormat(device->Format) == 2)
-            speakers = DSSPEAKER_COMBINED(DSSPEAKER_STEREO, 0);
-        else if(aluChannelsFromFormat(device->Format) == 4)
-            speakers = DSSPEAKER_COMBINED(DSSPEAKER_QUAD, 0);
-        else if(aluChannelsFromFormat(device->Format) == 6)
-            speakers = DSSPEAKER_COMBINED(DSSPEAKER_5POINT1, 0);
-        else if(aluChannelsFromFormat(device->Format) == 8)
-            speakers = DSSPEAKER_COMBINED(DSSPEAKER_7POINT1, 0);
-        else
-        {
-            AL_PRINT("Unknown format: 0x%x\n", device->Format);
-            return ALC_FALSE;
-        }
-    }
-    if(SUCCEEDED(hr))
-    {
-        speakers = DSSPEAKER_CONFIG(speakers);
-        if(speakers == DSSPEAKER_MONO)
-        {
-            if(aluBytesFromFormat(device->Format) == 1)
-                format = AL_FORMAT_MONO8;
-            else if(aluBytesFromFormat(device->Format) == 2)
-                format = AL_FORMAT_MONO16;
-            else if(aluBytesFromFormat(device->Format) == 4)
-                format = AL_FORMAT_MONO_FLOAT32;
-        }
-        else if(speakers == DSSPEAKER_STEREO)
-        {
-            if(aluBytesFromFormat(device->Format) == 1)
-                format = AL_FORMAT_STEREO8;
-            else if(aluBytesFromFormat(device->Format) == 2)
-                format = AL_FORMAT_STEREO16;
-            else if(aluBytesFromFormat(device->Format) == 4)
-                format = AL_FORMAT_STEREO_FLOAT32;
-        }
-        else if(speakers == DSSPEAKER_QUAD)
-        {
-            if(aluBytesFromFormat(device->Format) == 1)
-                format = AL_FORMAT_QUAD8;
-            else if(aluBytesFromFormat(device->Format) == 2)
-                format = AL_FORMAT_QUAD16;
-            else if(aluBytesFromFormat(device->Format) == 4)
-                format = AL_FORMAT_QUAD32;
-            OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
-                                       SPEAKER_FRONT_RIGHT |
-                                       SPEAKER_BACK_LEFT |
-                                       SPEAKER_BACK_RIGHT;
-        }
-        else if(speakers == DSSPEAKER_5POINT1)
-        {
-            if(aluBytesFromFormat(device->Format) == 1)
-                format = AL_FORMAT_51CHN8;
-            else if(aluBytesFromFormat(device->Format) == 2)
-                format = AL_FORMAT_51CHN16;
-            else if(aluBytesFromFormat(device->Format) == 4)
-                format = AL_FORMAT_51CHN32;
-            OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
-                                       SPEAKER_FRONT_RIGHT |
-                                       SPEAKER_FRONT_CENTER |
-                                       SPEAKER_LOW_FREQUENCY |
-                                       SPEAKER_BACK_LEFT |
-                                       SPEAKER_BACK_RIGHT;
-        }
-        else if(speakers == DSSPEAKER_7POINT1)
-        {
-            if(aluBytesFromFormat(device->Format) == 1)
-                format = AL_FORMAT_71CHN8;
-            else if(aluBytesFromFormat(device->Format) == 2)
-                format = AL_FORMAT_71CHN16;
-            else if(aluBytesFromFormat(device->Format) == 4)
-                format = AL_FORMAT_71CHN32;
-            OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
-                                       SPEAKER_FRONT_RIGHT |
-                                       SPEAKER_FRONT_CENTER |
-                                       SPEAKER_LOW_FREQUENCY |
-                                       SPEAKER_BACK_LEFT |
-                                       SPEAKER_BACK_RIGHT |
-                                       SPEAKER_SIDE_LEFT |
-                                       SPEAKER_SIDE_RIGHT;
-        }
-        else
-            format = device->Format;
-        frameSize = aluBytesFromFormat(format) * aluChannelsFromFormat(format);
-
-        OutputType.Format.wFormatTag = WAVE_FORMAT_PCM;
-        OutputType.Format.nChannels = aluChannelsFromFormat(format);
-        OutputType.Format.wBitsPerSample = aluBytesFromFormat(format) * 8;
-        OutputType.Format.nBlockAlign = OutputType.Format.nChannels*OutputType.Format.wBitsPerSample/8;
-        OutputType.Format.nSamplesPerSec = device->Frequency;
-        OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec*OutputType.Format.nBlockAlign;
-        OutputType.Format.cbSize = 0;
-    }
-
-    if(OutputType.Format.nChannels > 2 || OutputType.Format.wBitsPerSample > 16)
-    {
-        OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
-        OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample;
-        OutputType.Format.cbSize = 22;
-        if(OutputType.Format.wBitsPerSample == 32)
-            OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
-        else
-            OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
-    }
-    else
-    {
-        if(SUCCEEDED(hr))
-        {
-            memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
-            DSBDescription.dwSize=sizeof(DSBUFFERDESC);
-            DSBDescription.dwFlags=DSBCAPS_PRIMARYBUFFER;
-            hr = IDirectSound_CreateSoundBuffer(pData->lpDS, &DSBDescription, &pData->DSpbuffer, NULL);
-        }
-        if(SUCCEEDED(hr))
-            hr = IDirectSoundBuffer_SetFormat(pData->DSpbuffer,&OutputType.Format);
-    }
-
-    if(SUCCEEDED(hr))
-    {
-        memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
-        DSBDescription.dwSize=sizeof(DSBUFFERDESC);
-        DSBDescription.dwFlags=DSBCAPS_GLOBALFOCUS|DSBCAPS_GETCURRENTPOSITION2;
-        DSBDescription.dwBufferBytes=device->UpdateSize * device->NumUpdates * frameSize;
-        DSBDescription.lpwfxFormat=&OutputType.Format;
-        hr = IDirectSound_CreateSoundBuffer(pData->lpDS, &DSBDescription, &pData->DSsbuffer, NULL);
-    }
-
-    if(SUCCEEDED(hr))
-        hr = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING);
-
-    if(SUCCEEDED(hr))
-    {
-        device->ExtraData = pData;
-        pData->thread = StartThread(DSoundProc, device);
-        if(!pData->thread)
-            hr = E_FAIL;
-    }
-
-    if(FAILED(hr))
-    {
-        if (pData->DSsbuffer)
-            IDirectSoundBuffer_Release(pData->DSsbuffer);
-        pData->DSsbuffer = NULL;
-        if (pData->DSpbuffer)
-            IDirectSoundBuffer_Release(pData->DSpbuffer);
-        pData->DSpbuffer = NULL;
-        return ALC_FALSE;
-    }
-
-    device->Format = format;
-
-    return ALC_TRUE;
-}
-
-static void DSoundStopPlayback(ALCdevice *device)
-{
-    DSoundData *pData = device->ExtraData;
-
-    if(!pData->thread)
-        return;
-
-    pData->killNow = 1;
-    StopThread(pData->thread);
-    pData->thread = NULL;
-
-    IDirectSoundBuffer_Release(pData->DSsbuffer);
-    pData->DSsbuffer = NULL;
-    if (pData->DSpbuffer)
-        IDirectSoundBuffer_Release(pData->DSpbuffer);
-    pData->DSpbuffer = NULL;
-}
-
-
-static ALCboolean DSoundOpenCapture(ALCdevice *pDevice, const ALCchar *deviceName)
-{
-    (void)pDevice;
-    (void)deviceName;
-    return ALC_FALSE;
-}
-
-static void DSoundCloseCapture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void DSoundStartCapture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void DSoundStopCapture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void DSoundCaptureSamples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
-{
-    (void)pDevice;
-    (void)pBuffer;
-    (void)lSamples;
-}
-
-static ALCuint DSoundAvailableSamples(ALCdevice *pDevice)
-{
-    (void)pDevice;
-    return 0;
-}
-
-
-BackendFuncs DSoundFuncs = {
-    DSoundOpenPlayback,
-    DSoundClosePlayback,
-    DSoundResetPlayback,
-    DSoundStopPlayback,
-    DSoundOpenCapture,
-    DSoundCloseCapture,
-    DSoundStartCapture,
-    DSoundStopCapture,
-    DSoundCaptureSamples,
-    DSoundAvailableSamples
-};
-
-static BOOL CALLBACK DSoundEnumDevices(LPGUID guid, LPCSTR desc, LPCSTR drvname, LPVOID data)
-{
-    (void)data;
-    (void)drvname;
-
-    if(guid)
-    {
-        char str[128];
-        void *temp;
-
-        temp = realloc(DeviceList, sizeof(DevMap) * (NumDevices+1));
-        if(temp)
-        {
-            DeviceList = temp;
-
-            snprintf(str, sizeof(str), "DirectSound Software on %s", desc);
-            AppendAllDeviceList(str);
-
-            DeviceList[NumDevices].name = strdup(str);
-            DeviceList[NumDevices].guid = *guid;
-            NumDevices++;
-        }
-    }
-
-    return TRUE;
-}
-
-void alcDSoundInit(BackendFuncs *FuncList)
-{
-    *FuncList = DSoundFuncs;
-}
-
-void alcDSoundDeinit(void)
-{
-    ALuint i;
-
-    for(i = 0;i < NumDevices;++i)
-        free(DeviceList[i].name);
-    free(DeviceList);
-    DeviceList = NULL;
-    NumDevices = 0;
-}
-
-void alcDSoundProbe(int type)
-{
-    DSoundLoad();
-    if(!ds_handle) return;
-
-    if(type == DEVICE_PROBE)
-        AppendDeviceList(dsDevice);
-    else if(type == ALL_DEVICE_PROBE)
-    {
-        HRESULT hr;
-        ALuint i;
-
-        for(i = 0;i < NumDevices;++i)
-            free(DeviceList[i].name);
-        free(DeviceList);
-        DeviceList = NULL;
-        NumDevices = 0;
-
-        hr = pDirectSoundEnumerateA(DSoundEnumDevices, NULL);
-        if(FAILED(hr))
-            AL_PRINT("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr);
-    }
-
-    DSoundUnload();
-}

Alc/effects/chorus.c

@@ -0,0 +1,418 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2013 by Mike Gorchak
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+enum ChorusWaveForm {
+    CWF_Triangle = AL_CHORUS_WAVEFORM_TRIANGLE,
+    CWF_Sinusoid = AL_CHORUS_WAVEFORM_SINUSOID
+};
+
+typedef struct ALchorusState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    ALfloat *SampleBuffer[2];
+    ALsizei BufferLength;
+    ALsizei offset;
+    ALsizei lfo_range;
+    ALfloat lfo_scale;
+    ALint lfo_disp;
+
+    /* Gains for left and right sides */
+    ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
+
+    /* effect parameters */
+    enum ChorusWaveForm waveform;
+    ALint delay;
+    ALfloat depth;
+    ALfloat feedback;
+} ALchorusState;
+
+static ALvoid ALchorusState_Destruct(ALchorusState *state);
+static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device);
+static ALvoid ALchorusState_update(ALchorusState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
+static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALchorusState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALchorusState);
+
+
+static void ALchorusState_Construct(ALchorusState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALchorusState, ALeffectState, state);
+
+    state->BufferLength = 0;
+    state->SampleBuffer[0] = NULL;
+    state->SampleBuffer[1] = NULL;
+    state->offset = 0;
+    state->lfo_range = 1;
+    state->waveform = CWF_Triangle;
+}
+
+static ALvoid ALchorusState_Destruct(ALchorusState *state)
+{
+    al_free(state->SampleBuffer[0]);
+    state->SampleBuffer[0] = NULL;
+    state->SampleBuffer[1] = NULL;
+
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device)
+{
+    ALsizei maxlen;
+    ALsizei it;
+
+    maxlen = fastf2u(AL_CHORUS_MAX_DELAY * 2.0f * Device->Frequency) + 1;
+    maxlen = NextPowerOf2(maxlen);
+
+    if(maxlen != state->BufferLength)
+    {
+        void *temp = al_calloc(16, maxlen * sizeof(ALfloat) * 2);
+        if(!temp) return AL_FALSE;
+
+        al_free(state->SampleBuffer[0]);
+        state->SampleBuffer[0] = temp;
+        state->SampleBuffer[1] = state->SampleBuffer[0] + maxlen;
+
+        state->BufferLength = maxlen;
+    }
+
+    for(it = 0;it < state->BufferLength;it++)
+    {
+        state->SampleBuffer[0][it] = 0.0f;
+        state->SampleBuffer[1][it] = 0.0f;
+    }
+
+    return AL_TRUE;
+}
+
+static ALvoid ALchorusState_update(ALchorusState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
+{
+    ALfloat frequency = (ALfloat)Device->Frequency;
+    ALfloat coeffs[MAX_AMBI_COEFFS];
+    ALfloat rate;
+    ALint phase;
+
+    switch(props->Chorus.Waveform)
+    {
+        case AL_CHORUS_WAVEFORM_TRIANGLE:
+            state->waveform = CWF_Triangle;
+            break;
+        case AL_CHORUS_WAVEFORM_SINUSOID:
+            state->waveform = CWF_Sinusoid;
+            break;
+    }
+    state->feedback = props->Chorus.Feedback;
+    state->delay = fastf2i(props->Chorus.Delay * frequency);
+    /* The LFO depth is scaled to be relative to the sample delay. */
+    state->depth = props->Chorus.Depth * state->delay;
+
+    /* Gains for left and right sides */
+    CalcAngleCoeffs(-F_PI_2, 0.0f, 0.0f, coeffs);
+    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[0]);
+    CalcAngleCoeffs( F_PI_2, 0.0f, 0.0f, coeffs);
+    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[1]);
+
+    phase = props->Chorus.Phase;
+    rate = props->Chorus.Rate;
+    if(!(rate > 0.0f))
+    {
+        state->lfo_scale = 0.0f;
+        state->lfo_range = 1;
+        state->lfo_disp = 0;
+    }
+    else
+    {
+        /* Calculate LFO coefficient */
+        state->lfo_range = fastf2i(frequency/rate + 0.5f);
+        switch(state->waveform)
+        {
+            case CWF_Triangle:
+                state->lfo_scale = 4.0f / state->lfo_range;
+                break;
+            case CWF_Sinusoid:
+                state->lfo_scale = F_TAU / state->lfo_range;
+                break;
+        }
+
+        /* Calculate lfo phase displacement */
+        if(phase >= 0)
+            state->lfo_disp = fastf2i(state->lfo_range * (phase/360.0f));
+        else
+            state->lfo_disp = fastf2i(state->lfo_range * ((360+phase)/360.0f));
+    }
+}
+
+static inline void Triangle(ALint *delay_left, ALint *delay_right, ALuint offset, const ALchorusState *state)
+{
+    ALfloat lfo_value;
+
+    lfo_value = 1.0f - fabsf(2.0f - state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_left = fastf2i(lfo_value) + state->delay;
+
+    offset += state->lfo_disp;
+    lfo_value = 1.0f - fabsf(2.0f - state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_right = fastf2i(lfo_value) + state->delay;
+}
+
+static inline void Sinusoid(ALint *delay_left, ALint *delay_right, ALuint offset, const ALchorusState *state)
+{
+    ALfloat lfo_value;
+
+    lfo_value = sinf(state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_left = fastf2i(lfo_value) + state->delay;
+
+    offset += state->lfo_disp;
+    lfo_value = sinf(state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_right = fastf2i(lfo_value) + state->delay;
+}
+
+#define DECL_TEMPLATE(Func)                                                   \
+static void Process##Func(ALchorusState *state, const ALsizei SamplesToDo,    \
+  const ALfloat *restrict SamplesIn, ALfloat (*restrict out)[2])              \
+{                                                                             \
+    const ALsizei bufmask = state->BufferLength-1;                            \
+    ALfloat *restrict leftbuf = state->SampleBuffer[0];                       \
+    ALfloat *restrict rightbuf = state->SampleBuffer[1];                      \
+    ALsizei offset = state->offset;                                           \
+    const ALfloat feedback = state->feedback;                                 \
+    ALsizei it;                                                               \
+                                                                              \
+    for(it = 0;it < SamplesToDo;it++)                                         \
+    {                                                                         \
+        ALint delay_left, delay_right;                                        \
+        Func(&delay_left, &delay_right, offset, state);                       \
+                                                                              \
+        leftbuf[offset&bufmask] = SamplesIn[it];                              \
+        out[it][0] = leftbuf[(offset-delay_left)&bufmask];                    \
+        leftbuf[offset&bufmask] += out[it][0] * feedback;                     \
+                                                                              \
+        rightbuf[offset&bufmask] = SamplesIn[it];                             \
+        out[it][1] = rightbuf[(offset-delay_right)&bufmask];                  \
+        rightbuf[offset&bufmask] += out[it][1] * feedback;                    \
+                                                                              \
+        offset++;                                                             \
+    }                                                                         \
+    state->offset = offset;                                                   \
+}
+
+DECL_TEMPLATE(Triangle)
+DECL_TEMPLATE(Sinusoid)
+
+#undef DECL_TEMPLATE
+
+static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALsizei it, kt;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALfloat temps[128][2];
+        ALsizei td = mini(128, SamplesToDo-base);
+
+        switch(state->waveform)
+        {
+            case CWF_Triangle:
+                ProcessTriangle(state, td, SamplesIn[0]+base, temps);
+                break;
+            case CWF_Sinusoid:
+                ProcessSinusoid(state, td, SamplesIn[0]+base, temps);
+                break;
+        }
+
+        for(kt = 0;kt < NumChannels;kt++)
+        {
+            ALfloat gain = state->Gain[0][kt];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(it = 0;it < td;it++)
+                    SamplesOut[kt][it+base] += temps[it][0] * gain;
+            }
+
+            gain = state->Gain[1][kt];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(it = 0;it < td;it++)
+                    SamplesOut[kt][it+base] += temps[it][1] * gain;
+            }
+        }
+
+        base += td;
+    }
+}
+
+
+typedef struct ALchorusStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALchorusStateFactory;
+
+static ALeffectState *ALchorusStateFactory_create(ALchorusStateFactory *UNUSED(factory))
+{
+    ALchorusState *state;
+
+    NEW_OBJ0(state, ALchorusState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALchorusStateFactory);
+
+
+ALeffectStateFactory *ALchorusStateFactory_getFactory(void)
+{
+    static ALchorusStateFactory ChorusFactory = { { GET_VTABLE2(ALchorusStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &ChorusFactory);
+}
+
+
+void ALchorus_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_CHORUS_WAVEFORM:
+            if(!(val >= AL_CHORUS_MIN_WAVEFORM && val <= AL_CHORUS_MAX_WAVEFORM))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Chorus.Waveform = val;
+            break;
+
+        case AL_CHORUS_PHASE:
+            if(!(val >= AL_CHORUS_MIN_PHASE && val <= AL_CHORUS_MAX_PHASE))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Chorus.Phase = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALchorus_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALchorus_setParami(effect, context, param, vals[0]);
+}
+void ALchorus_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_CHORUS_RATE:
+            if(!(val >= AL_CHORUS_MIN_RATE && val <= AL_CHORUS_MAX_RATE))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Chorus.Rate = val;
+            break;
+
+        case AL_CHORUS_DEPTH:
+            if(!(val >= AL_CHORUS_MIN_DEPTH && val <= AL_CHORUS_MAX_DEPTH))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Chorus.Depth = val;
+            break;
+
+        case AL_CHORUS_FEEDBACK:
+            if(!(val >= AL_CHORUS_MIN_FEEDBACK && val <= AL_CHORUS_MAX_FEEDBACK))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Chorus.Feedback = val;
+            break;
+
+        case AL_CHORUS_DELAY:
+            if(!(val >= AL_CHORUS_MIN_DELAY && val <= AL_CHORUS_MAX_DELAY))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Chorus.Delay = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALchorus_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALchorus_setParamf(effect, context, param, vals[0]);
+}
+
+void ALchorus_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_CHORUS_WAVEFORM:
+            *val = props->Chorus.Waveform;
+            break;
+
+        case AL_CHORUS_PHASE:
+            *val = props->Chorus.Phase;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALchorus_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALchorus_getParami(effect, context, param, vals);
+}
+void ALchorus_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_CHORUS_RATE:
+            *val = props->Chorus.Rate;
+            break;
+
+        case AL_CHORUS_DEPTH:
+            *val = props->Chorus.Depth;
+            break;
+
+        case AL_CHORUS_FEEDBACK:
+            *val = props->Chorus.Feedback;
+            break;
+
+        case AL_CHORUS_DELAY:
+            *val = props->Chorus.Delay;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALchorus_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALchorus_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALchorus);

Alc/effects/compressor.c

@@ -0,0 +1,254 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2013 by Anis A. Hireche
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include <stdlib.h>
+
+#include "config.h"
+#include "alError.h"
+#include "alMain.h"
+#include "alAuxEffectSlot.h"
+#include "alu.h"
+
+
+typedef struct ALcompressorState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    /* Effect gains for each channel */
+    ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
+
+    /* Effect parameters */
+    ALboolean Enabled;
+    ALfloat AttackRate;
+    ALfloat ReleaseRate;
+    ALfloat GainCtrl;
+} ALcompressorState;
+
+static ALvoid ALcompressorState_Destruct(ALcompressorState *state);
+static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device);
+static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props);
+static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALcompressorState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALcompressorState);
+
+
+static void ALcompressorState_Construct(ALcompressorState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALcompressorState, ALeffectState, state);
+
+    state->Enabled = AL_TRUE;
+    state->AttackRate = 0.0f;
+    state->ReleaseRate = 0.0f;
+    state->GainCtrl = 1.0f;
+}
+
+static ALvoid ALcompressorState_Destruct(ALcompressorState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device)
+{
+    const ALfloat attackTime = device->Frequency * 0.2f; /* 200ms Attack */
+    const ALfloat releaseTime = device->Frequency * 0.4f; /* 400ms Release */
+
+    state->AttackRate = 1.0f / attackTime;
+    state->ReleaseRate = 1.0f / releaseTime;
+
+    return AL_TRUE;
+}
+
+static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    ALuint i;
+
+    state->Enabled = props->Compressor.OnOff;
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
+    for(i = 0;i < 4;i++)
+        ComputeFirstOrderGains(device->FOAOut, IdentityMatrixf.m[i],
+                               slot->Params.Gain, state->Gain[i]);
+}
+
+static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALsizei i, j, k;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALfloat temps[64][4];
+        ALsizei td = mini(64, SamplesToDo-base);
+
+        /* Load samples into the temp buffer first. */
+        for(j = 0;j < 4;j++)
+        {
+            for(i = 0;i < td;i++)
+                temps[i][j] = SamplesIn[j][i+base];
+        }
+
+        if(state->Enabled)
+        {
+            ALfloat gain = state->GainCtrl;
+            ALfloat output, amplitude;
+
+            for(i = 0;i < td;i++)
+            {
+                /* Roughly calculate the maximum amplitude from the 4-channel
+                 * signal, and attack or release the gain control to reach it.
+                 */
+                amplitude = fabsf(temps[i][0]);
+                amplitude = maxf(amplitude + fabsf(temps[i][1]),
+                                 maxf(amplitude + fabsf(temps[i][2]),
+                                      amplitude + fabsf(temps[i][3])));
+                if(amplitude > gain)
+                    gain = minf(gain+state->AttackRate, amplitude);
+                else if(amplitude < gain)
+                    gain = maxf(gain-state->ReleaseRate, amplitude);
+
+                /* Apply the inverse of the gain control to normalize/compress
+                 * the volume. */
+                output = 1.0f / clampf(gain, 0.5f, 2.0f);
+                for(j = 0;j < 4;j++)
+                    temps[i][j] *= output;
+            }
+
+            state->GainCtrl = gain;
+        }
+        else
+        {
+            ALfloat gain = state->GainCtrl;
+            ALfloat output, amplitude;
+
+            for(i = 0;i < td;i++)
+            {
+                /* Same as above, except the amplitude is forced to 1. This
+                 * helps ensure smooth gain changes when the compressor is
+                 * turned on and off.
+                 */
+                amplitude = 1.0f;
+                if(amplitude > gain)
+                    gain = minf(gain+state->AttackRate, amplitude);
+                else if(amplitude < gain)
+                    gain = maxf(gain-state->ReleaseRate, amplitude);
+
+                output = 1.0f / clampf(gain, 0.5f, 2.0f);
+                for(j = 0;j < 4;j++)
+                    temps[i][j] *= output;
+            }
+
+            state->GainCtrl = gain;
+        }
+
+        /* Now mix to the output. */
+        for(j = 0;j < 4;j++)
+        {
+            for(k = 0;k < NumChannels;k++)
+            {
+                ALfloat gain = state->Gain[j][k];
+                if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+                    continue;
+
+                for(i = 0;i < td;i++)
+                    SamplesOut[k][base+i] += gain * temps[i][j];
+            }
+        }
+
+        base += td;
+    }
+}
+
+
+typedef struct ALcompressorStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALcompressorStateFactory;
+
+static ALeffectState *ALcompressorStateFactory_create(ALcompressorStateFactory *UNUSED(factory))
+{
+    ALcompressorState *state;
+
+    NEW_OBJ0(state, ALcompressorState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALcompressorStateFactory);
+
+ALeffectStateFactory *ALcompressorStateFactory_getFactory(void)
+{
+    static ALcompressorStateFactory CompressorFactory = { { GET_VTABLE2(ALcompressorStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &CompressorFactory);
+}
+
+
+void ALcompressor_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_COMPRESSOR_ONOFF:
+            if(!(val >= AL_COMPRESSOR_MIN_ONOFF && val <= AL_COMPRESSOR_MAX_ONOFF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Compressor.OnOff = val;
+            break;
+
+    default:
+        SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALcompressor_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALcompressor_setParami(effect, context, param, vals[0]);
+}
+void ALcompressor_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALfloat UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALcompressor_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALcompressor_setParamf(effect, context, param, vals[0]);
+}
+
+void ALcompressor_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{ 
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_COMPRESSOR_ONOFF:
+            *val = props->Compressor.OnOff;
+            break;
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALcompressor_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALcompressor_getParami(effect, context, param, vals);
+}
+void ALcompressor_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALfloat *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALcompressor_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALcompressor_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALcompressor);

Alc/effects/dedicated.c

@@ -0,0 +1,203 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2011 by Chris Robinson.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+typedef struct ALdedicatedState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    ALfloat gains[MAX_OUTPUT_CHANNELS];
+} ALdedicatedState;
+
+static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state);
+static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *state, ALCdevice *device);
+static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCdevice *device, const ALeffectslot *Slot, const ALeffectProps *props);
+static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALdedicatedState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALdedicatedState);
+
+
+static void ALdedicatedState_Construct(ALdedicatedState *state)
+{
+    ALsizei s;
+
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALdedicatedState, ALeffectState, state);
+
+    for(s = 0;s < MAX_OUTPUT_CHANNELS;s++)
+        state->gains[s] = 0.0f;
+}
+
+static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *UNUSED(state), ALCdevice *UNUSED(device))
+{
+    return AL_TRUE;
+}
+
+static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCdevice *device, const ALeffectslot *Slot, const ALeffectProps *props)
+{
+    ALfloat Gain;
+    ALuint i;
+
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        state->gains[i] = 0.0f;
+
+    Gain = Slot->Params.Gain * props->Dedicated.Gain;
+    if(Slot->Params.EffectType == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
+    {
+        int idx;
+        if((idx=GetChannelIdxByName(device->RealOut, LFE)) != -1)
+        {
+            STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer;
+            STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels;
+            state->gains[idx] = Gain;
+        }
+    }
+    else if(Slot->Params.EffectType == AL_EFFECT_DEDICATED_DIALOGUE)
+    {
+        int idx;
+        /* Dialog goes to the front-center speaker if it exists, otherwise it
+         * plays from the front-center location. */
+        if((idx=GetChannelIdxByName(device->RealOut, FrontCenter)) != -1)
+        {
+            STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer;
+            STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels;
+            state->gains[idx] = Gain;
+        }
+        else
+        {
+            ALfloat coeffs[MAX_AMBI_COEFFS];
+            CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
+
+            STATIC_CAST(ALeffectState,state)->OutBuffer = device->Dry.Buffer;
+            STATIC_CAST(ALeffectState,state)->OutChannels = device->Dry.NumChannels;
+            ComputePanningGains(device->Dry, coeffs, Gain, state->gains);
+        }
+    }
+}
+
+static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALsizei i, c;
+
+    SamplesIn = ASSUME_ALIGNED(SamplesIn, 16);
+    SamplesOut = ASSUME_ALIGNED(SamplesOut, 16);
+    for(c = 0;c < NumChannels;c++)
+    {
+        const ALfloat gain = state->gains[c];
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+
+        for(i = 0;i < SamplesToDo;i++)
+            SamplesOut[c][i] += SamplesIn[0][i] * gain;
+    }
+}
+
+
+typedef struct ALdedicatedStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALdedicatedStateFactory;
+
+ALeffectState *ALdedicatedStateFactory_create(ALdedicatedStateFactory *UNUSED(factory))
+{
+    ALdedicatedState *state;
+
+    NEW_OBJ0(state, ALdedicatedState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALdedicatedStateFactory);
+
+
+ALeffectStateFactory *ALdedicatedStateFactory_getFactory(void)
+{
+    static ALdedicatedStateFactory DedicatedFactory = { { GET_VTABLE2(ALdedicatedStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &DedicatedFactory);
+}
+
+
+void ALdedicated_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALdedicated_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALdedicated_setParami(effect, context, param, vals[0]);
+}
+void ALdedicated_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_DEDICATED_GAIN:
+            if(!(val >= 0.0f && isfinite(val)))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Dedicated.Gain = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALdedicated_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALdedicated_setParamf(effect, context, param, vals[0]);
+}
+
+void ALdedicated_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALdedicated_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALdedicated_getParami(effect, context, param, vals);
+}
+void ALdedicated_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_DEDICATED_GAIN:
+            *val = props->Dedicated.Gain;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALdedicated_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALdedicated_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALdedicated);

Alc/effects/distortion.c

@@ -0,0 +1,297 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2013 by Mike Gorchak
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+typedef struct ALdistortionState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    /* Effect gains for each channel */
+    ALfloat Gain[MAX_OUTPUT_CHANNELS];
+
+    /* Effect parameters */
+    ALfilterState lowpass;
+    ALfilterState bandpass;
+    ALfloat attenuation;
+    ALfloat edge_coeff;
+} ALdistortionState;
+
+static ALvoid ALdistortionState_Destruct(ALdistortionState *state);
+static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *device);
+static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
+static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALdistortionState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALdistortionState);
+
+
+static void ALdistortionState_Construct(ALdistortionState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALdistortionState, ALeffectState, state);
+
+    ALfilterState_clear(&state->lowpass);
+    ALfilterState_clear(&state->bandpass);
+}
+
+static ALvoid ALdistortionState_Destruct(ALdistortionState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *UNUSED(state), ALCdevice *UNUSED(device))
+{
+    return AL_TRUE;
+}
+
+static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
+{
+    ALfloat frequency = (ALfloat)Device->Frequency;
+    ALfloat bandwidth;
+    ALfloat cutoff;
+    ALfloat edge;
+
+    /* Store distorted signal attenuation settings. */
+    state->attenuation = props->Distortion.Gain;
+
+    /* Store waveshaper edge settings. */
+    edge = sinf(props->Distortion.Edge * (F_PI_2));
+    edge = minf(edge, 0.99f);
+    state->edge_coeff = 2.0f * edge / (1.0f-edge);
+
+    cutoff = props->Distortion.LowpassCutoff;
+    /* Bandwidth value is constant in octaves. */
+    bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
+    /* Multiply sampling frequency by the amount of oversampling done during
+     * processing.
+     */
+    ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f,
+        cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
+    );
+
+    cutoff = props->Distortion.EQCenter;
+    /* Convert bandwidth in Hz to octaves. */
+    bandwidth = props->Distortion.EQBandwidth / (cutoff * 0.67f);
+    ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f,
+        cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
+    );
+
+    ComputeAmbientGains(Device->Dry, Slot->Params.Gain, state->Gain);
+}
+
+static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    const ALfloat fc = state->edge_coeff;
+    ALsizei it, kt;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        float buffer[2][64 * 4];
+        ALsizei td = mini(64, SamplesToDo-base);
+
+        /* Perform 4x oversampling to avoid aliasing. Oversampling greatly
+         * improves distortion quality and allows to implement lowpass and
+         * bandpass filters using high frequencies, at which classic IIR
+         * filters became unstable.
+         */
+
+        /* Fill oversample buffer using zero stuffing. */
+        for(it = 0;it < td;it++)
+        {
+            /* Multiply the sample by the amount of oversampling to maintain
+             * the signal's power.
+             */
+            buffer[0][it*4 + 0] = SamplesIn[0][it+base] * 4.0f;
+            buffer[0][it*4 + 1] = 0.0f;
+            buffer[0][it*4 + 2] = 0.0f;
+            buffer[0][it*4 + 3] = 0.0f;
+        }
+
+        /* First step, do lowpass filtering of original signal. Additionally
+         * perform buffer interpolation and lowpass cutoff for oversampling
+         * (which is fortunately first step of distortion). So combine three
+         * operations into the one.
+         */
+        ALfilterState_process(&state->lowpass, buffer[1], buffer[0], td*4);
+
+        /* Second step, do distortion using waveshaper function to emulate
+         * signal processing during tube overdriving. Three steps of
+         * waveshaping are intended to modify waveform without boost/clipping/
+         * attenuation process.
+         */
+        for(it = 0;it < td*4;it++)
+        {
+            ALfloat smp = buffer[1][it];
+
+            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
+            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f;
+            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
+
+            buffer[0][it] = smp;
+        }
+
+        /* Third step, do bandpass filtering of distorted signal. */
+        ALfilterState_process(&state->bandpass, buffer[1], buffer[0], td*4);
+
+        for(kt = 0;kt < NumChannels;kt++)
+        {
+            /* Fourth step, final, do attenuation and perform decimation,
+             * store only one sample out of 4.
+             */
+            ALfloat gain = state->Gain[kt] * state->attenuation;
+            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+                continue;
+
+            for(it = 0;it < td;it++)
+                SamplesOut[kt][base+it] += gain * buffer[1][it*4];
+        }
+
+        base += td;
+    }
+}
+
+
+typedef struct ALdistortionStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALdistortionStateFactory;
+
+static ALeffectState *ALdistortionStateFactory_create(ALdistortionStateFactory *UNUSED(factory))
+{
+    ALdistortionState *state;
+
+    NEW_OBJ0(state, ALdistortionState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALdistortionStateFactory);
+
+
+ALeffectStateFactory *ALdistortionStateFactory_getFactory(void)
+{
+    static ALdistortionStateFactory DistortionFactory = { { GET_VTABLE2(ALdistortionStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &DistortionFactory);
+}
+
+
+void ALdistortion_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALdistortion_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALdistortion_setParami(effect, context, param, vals[0]);
+}
+void ALdistortion_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_DISTORTION_EDGE:
+            if(!(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Distortion.Edge = val;
+            break;
+
+        case AL_DISTORTION_GAIN:
+            if(!(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Distortion.Gain = val;
+            break;
+
+        case AL_DISTORTION_LOWPASS_CUTOFF:
+            if(!(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Distortion.LowpassCutoff = val;
+            break;
+
+        case AL_DISTORTION_EQCENTER:
+            if(!(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Distortion.EQCenter = val;
+            break;
+
+        case AL_DISTORTION_EQBANDWIDTH:
+            if(!(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Distortion.EQBandwidth = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALdistortion_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALdistortion_setParamf(effect, context, param, vals[0]);
+}
+
+void ALdistortion_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALdistortion_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALdistortion_getParami(effect, context, param, vals);
+}
+void ALdistortion_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_DISTORTION_EDGE:
+            *val = props->Distortion.Edge;
+            break;
+
+        case AL_DISTORTION_GAIN:
+            *val = props->Distortion.Gain;
+            break;
+
+        case AL_DISTORTION_LOWPASS_CUTOFF:
+            *val = props->Distortion.LowpassCutoff;
+            break;
+
+        case AL_DISTORTION_EQCENTER:
+            *val = props->Distortion.EQCenter;
+            break;
+
+        case AL_DISTORTION_EQBANDWIDTH:
+            *val = props->Distortion.EQBandwidth;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALdistortion_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALdistortion_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALdistortion);

Alc/effects/echo.c

@@ -0,0 +1,326 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2009 by Chris Robinson.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+typedef struct ALechoState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    ALfloat *SampleBuffer;
+    ALsizei BufferLength;
+
+    // The echo is two tap. The delay is the number of samples from before the
+    // current offset
+    struct {
+        ALsizei delay;
+    } Tap[2];
+    ALsizei Offset;
+    /* The panning gains for the two taps */
+    ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
+
+    ALfloat FeedGain;
+
+    ALfilterState Filter;
+} ALechoState;
+
+static ALvoid ALechoState_Destruct(ALechoState *state);
+static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device);
+static ALvoid ALechoState_update(ALechoState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
+static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALechoState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALechoState);
+
+
+static void ALechoState_Construct(ALechoState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALechoState, ALeffectState, state);
+
+    state->BufferLength = 0;
+    state->SampleBuffer = NULL;
+
+    state->Tap[0].delay = 0;
+    state->Tap[1].delay = 0;
+    state->Offset = 0;
+
+    ALfilterState_clear(&state->Filter);
+}
+
+static ALvoid ALechoState_Destruct(ALechoState *state)
+{
+    al_free(state->SampleBuffer);
+    state->SampleBuffer = NULL;
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device)
+{
+    ALsizei maxlen, i;
+
+    // Use the next power of 2 for the buffer length, so the tap offsets can be
+    // wrapped using a mask instead of a modulo
+    maxlen  = fastf2i(AL_ECHO_MAX_DELAY * Device->Frequency) + 1;
+    maxlen += fastf2i(AL_ECHO_MAX_LRDELAY * Device->Frequency) + 1;
+    maxlen  = NextPowerOf2(maxlen);
+
+    if(maxlen != state->BufferLength)
+    {
+        void *temp = al_calloc(16, maxlen * sizeof(ALfloat));
+        if(!temp) return AL_FALSE;
+
+        al_free(state->SampleBuffer);
+        state->SampleBuffer = temp;
+        state->BufferLength = maxlen;
+    }
+    for(i = 0;i < state->BufferLength;i++)
+        state->SampleBuffer[i] = 0.0f;
+
+    return AL_TRUE;
+}
+
+static ALvoid ALechoState_update(ALechoState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
+{
+    ALuint frequency = Device->Frequency;
+    ALfloat coeffs[MAX_AMBI_COEFFS];
+    ALfloat gain, lrpan, spread;
+
+    state->Tap[0].delay = fastf2i(props->Echo.Delay * frequency) + 1;
+    state->Tap[1].delay = fastf2i(props->Echo.LRDelay * frequency);
+    state->Tap[1].delay += state->Tap[0].delay;
+
+    spread = props->Echo.Spread;
+    if(spread < 0.0f) lrpan = -1.0f;
+    else lrpan = 1.0f;
+    /* Convert echo spread (where 0 = omni, +/-1 = directional) to coverage
+     * spread (where 0 = point, tau = omni).
+     */
+    spread = asinf(1.0f - fabsf(spread))*4.0f;
+
+    state->FeedGain = props->Echo.Feedback;
+
+    gain = maxf(1.0f - props->Echo.Damping, 0.0625f); /* Limit -24dB */
+    ALfilterState_setParams(&state->Filter, ALfilterType_HighShelf,
+                            gain, LOWPASSFREQREF/frequency,
+                            calc_rcpQ_from_slope(gain, 1.0f));
+
+    gain = Slot->Params.Gain;
+
+    /* First tap panning */
+    CalcAngleCoeffs(-F_PI_2*lrpan, 0.0f, spread, coeffs);
+    ComputePanningGains(Device->Dry, coeffs, gain, state->Gain[0]);
+
+    /* Second tap panning */
+    CalcAngleCoeffs( F_PI_2*lrpan, 0.0f, spread, coeffs);
+    ComputePanningGains(Device->Dry, coeffs, gain, state->Gain[1]);
+}
+
+static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    const ALsizei mask = state->BufferLength-1;
+    const ALsizei tap1 = state->Tap[0].delay;
+    const ALsizei tap2 = state->Tap[1].delay;
+    ALsizei offset = state->Offset;
+    ALfloat x[2], y[2], in, out;
+    ALsizei base, k;
+    ALsizei i;
+
+    x[0] = state->Filter.x[0];
+    x[1] = state->Filter.x[1];
+    y[0] = state->Filter.y[0];
+    y[1] = state->Filter.y[1];
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALfloat temps[128][2];
+        ALsizei td = mini(128, SamplesToDo-base);
+
+        for(i = 0;i < td;i++)
+        {
+            /* First tap */
+            temps[i][0] = state->SampleBuffer[(offset-tap1) & mask];
+            /* Second tap */
+            temps[i][1] = state->SampleBuffer[(offset-tap2) & mask];
+
+            // Apply damping and feedback gain to the second tap, and mix in the
+            // new sample
+            in = temps[i][1] + SamplesIn[0][i+base];
+            out = in*state->Filter.b0 +
+                  x[0]*state->Filter.b1 + x[1]*state->Filter.b2 -
+                  y[0]*state->Filter.a1 - y[1]*state->Filter.a2;
+            x[1] = x[0]; x[0] = in;
+            y[1] = y[0]; y[0] = out;
+
+            state->SampleBuffer[offset&mask] = out * state->FeedGain;
+            offset++;
+        }
+
+        for(k = 0;k < NumChannels;k++)
+        {
+            ALfloat gain = state->Gain[0][k];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(i = 0;i < td;i++)
+                    SamplesOut[k][i+base] += temps[i][0] * gain;
+            }
+
+            gain = state->Gain[1][k];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(i = 0;i < td;i++)
+                    SamplesOut[k][i+base] += temps[i][1] * gain;
+            }
+        }
+
+        base += td;
+    }
+    state->Filter.x[0] = x[0];
+    state->Filter.x[1] = x[1];
+    state->Filter.y[0] = y[0];
+    state->Filter.y[1] = y[1];
+
+    state->Offset = offset;
+}
+
+
+typedef struct ALechoStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALechoStateFactory;
+
+ALeffectState *ALechoStateFactory_create(ALechoStateFactory *UNUSED(factory))
+{
+    ALechoState *state;
+
+    NEW_OBJ0(state, ALechoState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALechoStateFactory);
+
+ALeffectStateFactory *ALechoStateFactory_getFactory(void)
+{
+    static ALechoStateFactory EchoFactory = { { GET_VTABLE2(ALechoStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &EchoFactory);
+}
+
+
+void ALecho_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALecho_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALecho_setParami(effect, context, param, vals[0]);
+}
+void ALecho_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_ECHO_DELAY:
+            if(!(val >= AL_ECHO_MIN_DELAY && val <= AL_ECHO_MAX_DELAY))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Echo.Delay = val;
+            break;
+
+        case AL_ECHO_LRDELAY:
+            if(!(val >= AL_ECHO_MIN_LRDELAY && val <= AL_ECHO_MAX_LRDELAY))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Echo.LRDelay = val;
+            break;
+
+        case AL_ECHO_DAMPING:
+            if(!(val >= AL_ECHO_MIN_DAMPING && val <= AL_ECHO_MAX_DAMPING))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Echo.Damping = val;
+            break;
+
+        case AL_ECHO_FEEDBACK:
+            if(!(val >= AL_ECHO_MIN_FEEDBACK && val <= AL_ECHO_MAX_FEEDBACK))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Echo.Feedback = val;
+            break;
+
+        case AL_ECHO_SPREAD:
+            if(!(val >= AL_ECHO_MIN_SPREAD && val <= AL_ECHO_MAX_SPREAD))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Echo.Spread = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALecho_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALecho_setParamf(effect, context, param, vals[0]);
+}
+
+void ALecho_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALecho_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALecho_getParami(effect, context, param, vals);
+}
+void ALecho_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_ECHO_DELAY:
+            *val = props->Echo.Delay;
+            break;
+
+        case AL_ECHO_LRDELAY:
+            *val = props->Echo.LRDelay;
+            break;
+
+        case AL_ECHO_DAMPING:
+            *val = props->Echo.Damping;
+            break;
+
+        case AL_ECHO_FEEDBACK:
+            *val = props->Echo.Feedback;
+            break;
+
+        case AL_ECHO_SPREAD:
+            *val = props->Echo.Spread;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALecho_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALecho_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALecho);

Alc/effects/equalizer.c

@@ -0,0 +1,380 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2013 by Mike Gorchak
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+/*  The document  "Effects Extension Guide.pdf"  says that low and high  *
+ *  frequencies are cutoff frequencies. This is not fully correct, they  *
+ *  are corner frequencies for low and high shelf filters. If they were  *
+ *  just cutoff frequencies, there would be no need in cutoff frequency  *
+ *  gains, which are present.  Documentation for  "Creative Proteus X2"  *
+ *  software describes  4-band equalizer functionality in a much better  *
+ *  way.  This equalizer seems  to be a predecessor  of  OpenAL  4-band  *
+ *  equalizer.  With low and high  shelf filters  we are able to cutoff  *
+ *  frequencies below and/or above corner frequencies using attenuation  *
+ *  gains (below 1.0) and amplify all low and/or high frequencies using  *
+ *  gains above 1.0.                                                     *
+ *                                                                       *
+ *     Low-shelf       Low Mid Band      High Mid Band     High-shelf    *
+ *      corner            center             center          corner      *
+ *     frequency        frequency          frequency       frequency     *
+ *    50Hz..800Hz     200Hz..3000Hz      1000Hz..8000Hz  4000Hz..16000Hz *
+ *                                                                       *
+ *          |               |                  |               |         *
+ *          |               |                  |               |         *
+ *   B -----+            /--+--\            /--+--\            +-----    *
+ *   O      |\          |   |   |          |   |   |          /|         *
+ *   O      | \        -    |    -        -    |    -        / |         *
+ *   S +    |  \      |     |     |      |     |     |      /  |         *
+ *   T      |   |    |      |      |    |      |      |    |   |         *
+ * ---------+---------------+------------------+---------------+-------- *
+ *   C      |   |    |      |      |    |      |      |    |   |         *
+ *   U -    |  /      |     |     |      |     |     |      \  |         *
+ *   T      | /        -    |    -        -    |    -        \ |         *
+ *   O      |/          |   |   |          |   |   |          \|         *
+ *   F -----+            \--+--/            \--+--/            +-----    *
+ *   F      |               |                  |               |         *
+ *          |               |                  |               |         *
+ *                                                                       *
+ * Gains vary from 0.126 up to 7.943, which means from -18dB attenuation *
+ * up to +18dB amplification. Band width varies from 0.01 up to 1.0 in   *
+ * octaves for two mid bands.                                            *
+ *                                                                       *
+ * Implementation is based on the "Cookbook formulae for audio EQ biquad *
+ * filter coefficients" by Robert Bristow-Johnson                        *
+ * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt                   */
+
+
+/* The maximum number of sample frames per update. */
+#define MAX_UPDATE_SAMPLES 256
+
+typedef struct ALequalizerState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    /* Effect gains for each channel */
+    ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
+
+    /* Effect parameters */
+    ALfilterState filter[4][MAX_EFFECT_CHANNELS];
+
+    ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES];
+} ALequalizerState;
+
+static ALvoid ALequalizerState_Destruct(ALequalizerState *state);
+static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *device);
+static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props);
+static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALequalizerState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
+
+
+static void ALequalizerState_Construct(ALequalizerState *state)
+{
+    int it, ft;
+
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALequalizerState, ALeffectState, state);
+
+    /* Initialize sample history only on filter creation to avoid */
+    /* sound clicks if filter settings were changed in runtime.   */
+    for(it = 0; it < 4; it++)
+    {
+        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
+            ALfilterState_clear(&state->filter[it][ft]);
+    }
+}
+
+static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), ALCdevice *UNUSED(device))
+{
+    return AL_TRUE;
+}
+
+static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    ALfloat frequency = (ALfloat)device->Frequency;
+    ALfloat gain, freq_mult;
+    ALuint i;
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+        ComputeFirstOrderGains(device->FOAOut, IdentityMatrixf.m[i],
+                               slot->Params.Gain, state->Gain[i]);
+
+    /* Calculate coefficients for the each type of filter. Note that the shelf
+     * filters' gain is for the reference frequency, which is the centerpoint
+     * of the transition band.
+     */
+    gain = maxf(sqrtf(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */
+    freq_mult = props->Equalizer.LowCutoff/frequency;
+    ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf,
+        gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
+    );
+    /* Copy the filter coefficients for the other input channels. */
+    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
+        ALfilterState_copyParams(&state->filter[0][i], &state->filter[0][0]);
+
+    gain = maxf(props->Equalizer.Mid1Gain, 0.0625f);
+    freq_mult = props->Equalizer.Mid1Center/frequency;
+    ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking,
+        gain, freq_mult, calc_rcpQ_from_bandwidth(
+            freq_mult, props->Equalizer.Mid1Width
+        )
+    );
+    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
+        ALfilterState_copyParams(&state->filter[1][i], &state->filter[1][0]);
+
+    gain = maxf(props->Equalizer.Mid2Gain, 0.0625f);
+    freq_mult = props->Equalizer.Mid2Center/frequency;
+    ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking,
+        gain, freq_mult, calc_rcpQ_from_bandwidth(
+            freq_mult, props->Equalizer.Mid2Width
+        )
+    );
+    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
+        ALfilterState_copyParams(&state->filter[2][i], &state->filter[2][0]);
+
+    gain = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f);
+    freq_mult = props->Equalizer.HighCutoff/frequency;
+    ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf,
+        gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
+    );
+    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
+        ALfilterState_copyParams(&state->filter[3][i], &state->filter[3][0]);
+}
+
+static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer;
+    ALsizei it, kt, ft;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base);
+
+        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
+            ALfilterState_process(&state->filter[0][ft], Samples[0][ft], &SamplesIn[ft][base], td);
+        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
+            ALfilterState_process(&state->filter[1][ft], Samples[1][ft], Samples[0][ft], td);
+        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
+            ALfilterState_process(&state->filter[2][ft], Samples[2][ft], Samples[1][ft], td);
+        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
+            ALfilterState_process(&state->filter[3][ft], Samples[3][ft], Samples[2][ft], td);
+
+        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
+        {
+            for(kt = 0;kt < NumChannels;kt++)
+            {
+                ALfloat gain = state->Gain[ft][kt];
+                if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+                    continue;
+
+                for(it = 0;it < td;it++)
+                    SamplesOut[kt][base+it] += gain * Samples[3][ft][it];
+            }
+        }
+
+        base += td;
+    }
+}
+
+
+typedef struct ALequalizerStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALequalizerStateFactory;
+
+ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory))
+{
+    ALequalizerState *state;
+
+    NEW_OBJ0(state, ALequalizerState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALequalizerStateFactory);
+
+ALeffectStateFactory *ALequalizerStateFactory_getFactory(void)
+{
+    static ALequalizerStateFactory EqualizerFactory = { { GET_VTABLE2(ALequalizerStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
+}
+
+
+void ALequalizer_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALequalizer_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALequalizer_setParami(effect, context, param, vals[0]);
+}
+void ALequalizer_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_EQUALIZER_LOW_GAIN:
+            if(!(val >= AL_EQUALIZER_MIN_LOW_GAIN && val <= AL_EQUALIZER_MAX_LOW_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.LowGain = val;
+            break;
+
+        case AL_EQUALIZER_LOW_CUTOFF:
+            if(!(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.LowCutoff = val;
+            break;
+
+        case AL_EQUALIZER_MID1_GAIN:
+            if(!(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.Mid1Gain = val;
+            break;
+
+        case AL_EQUALIZER_MID1_CENTER:
+            if(!(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.Mid1Center = val;
+            break;
+
+        case AL_EQUALIZER_MID1_WIDTH:
+            if(!(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.Mid1Width = val;
+            break;
+
+        case AL_EQUALIZER_MID2_GAIN:
+            if(!(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.Mid2Gain = val;
+            break;
+
+        case AL_EQUALIZER_MID2_CENTER:
+            if(!(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.Mid2Center = val;
+            break;
+
+        case AL_EQUALIZER_MID2_WIDTH:
+            if(!(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.Mid2Width = val;
+            break;
+
+        case AL_EQUALIZER_HIGH_GAIN:
+            if(!(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.HighGain = val;
+            break;
+
+        case AL_EQUALIZER_HIGH_CUTOFF:
+            if(!(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Equalizer.HighCutoff = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALequalizer_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALequalizer_setParamf(effect, context, param, vals[0]);
+}
+
+void ALequalizer_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+void ALequalizer_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALequalizer_getParami(effect, context, param, vals);
+}
+void ALequalizer_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_EQUALIZER_LOW_GAIN:
+            *val = props->Equalizer.LowGain;
+            break;
+
+        case AL_EQUALIZER_LOW_CUTOFF:
+            *val = props->Equalizer.LowCutoff;
+            break;
+
+        case AL_EQUALIZER_MID1_GAIN:
+            *val = props->Equalizer.Mid1Gain;
+            break;
+
+        case AL_EQUALIZER_MID1_CENTER:
+            *val = props->Equalizer.Mid1Center;
+            break;
+
+        case AL_EQUALIZER_MID1_WIDTH:
+            *val = props->Equalizer.Mid1Width;
+            break;
+
+        case AL_EQUALIZER_MID2_GAIN:
+            *val = props->Equalizer.Mid2Gain;
+            break;
+
+        case AL_EQUALIZER_MID2_CENTER:
+            *val = props->Equalizer.Mid2Center;
+            break;
+
+        case AL_EQUALIZER_MID2_WIDTH:
+            *val = props->Equalizer.Mid2Width;
+            break;
+
+        case AL_EQUALIZER_HIGH_GAIN:
+            *val = props->Equalizer.HighGain;
+            break;
+
+        case AL_EQUALIZER_HIGH_CUTOFF:
+            *val = props->Equalizer.HighCutoff;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALequalizer_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALequalizer_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALequalizer);

Alc/effects/flanger.c

@@ -0,0 +1,417 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2013 by Mike Gorchak
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+enum FlangerWaveForm {
+    FWF_Triangle = AL_FLANGER_WAVEFORM_TRIANGLE,
+    FWF_Sinusoid = AL_FLANGER_WAVEFORM_SINUSOID
+};
+
+typedef struct ALflangerState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    ALfloat *SampleBuffer[2];
+    ALsizei BufferLength;
+    ALsizei offset;
+    ALsizei lfo_range;
+    ALfloat lfo_scale;
+    ALint lfo_disp;
+
+    /* Gains for left and right sides */
+    ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
+
+    /* effect parameters */
+    enum FlangerWaveForm waveform;
+    ALint delay;
+    ALfloat depth;
+    ALfloat feedback;
+} ALflangerState;
+
+static ALvoid ALflangerState_Destruct(ALflangerState *state);
+static ALboolean ALflangerState_deviceUpdate(ALflangerState *state, ALCdevice *Device);
+static ALvoid ALflangerState_update(ALflangerState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
+static ALvoid ALflangerState_process(ALflangerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALflangerState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALflangerState);
+
+
+static void ALflangerState_Construct(ALflangerState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALflangerState, ALeffectState, state);
+
+    state->BufferLength = 0;
+    state->SampleBuffer[0] = NULL;
+    state->SampleBuffer[1] = NULL;
+    state->offset = 0;
+    state->lfo_range = 1;
+    state->waveform = FWF_Triangle;
+}
+
+static ALvoid ALflangerState_Destruct(ALflangerState *state)
+{
+    al_free(state->SampleBuffer[0]);
+    state->SampleBuffer[0] = NULL;
+    state->SampleBuffer[1] = NULL;
+
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALflangerState_deviceUpdate(ALflangerState *state, ALCdevice *Device)
+{
+    ALsizei maxlen;
+    ALsizei it;
+
+    maxlen = fastf2i(AL_FLANGER_MAX_DELAY * 2.0f * Device->Frequency) + 1;
+    maxlen = NextPowerOf2(maxlen);
+
+    if(maxlen != state->BufferLength)
+    {
+        void *temp = al_calloc(16, maxlen * sizeof(ALfloat) * 2);
+        if(!temp) return AL_FALSE;
+
+        al_free(state->SampleBuffer[0]);
+        state->SampleBuffer[0] = temp;
+        state->SampleBuffer[1] = state->SampleBuffer[0] + maxlen;
+
+        state->BufferLength = maxlen;
+    }
+
+    for(it = 0;it < state->BufferLength;it++)
+    {
+        state->SampleBuffer[0][it] = 0.0f;
+        state->SampleBuffer[1][it] = 0.0f;
+    }
+
+    return AL_TRUE;
+}
+
+static ALvoid ALflangerState_update(ALflangerState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
+{
+    ALfloat frequency = (ALfloat)Device->Frequency;
+    ALfloat coeffs[MAX_AMBI_COEFFS];
+    ALfloat rate;
+    ALint phase;
+
+    switch(props->Flanger.Waveform)
+    {
+        case AL_FLANGER_WAVEFORM_TRIANGLE:
+            state->waveform = FWF_Triangle;
+            break;
+        case AL_FLANGER_WAVEFORM_SINUSOID:
+            state->waveform = FWF_Sinusoid;
+            break;
+    }
+    state->feedback = props->Flanger.Feedback;
+    state->delay = fastf2i(props->Flanger.Delay * frequency);
+    /* The LFO depth is scaled to be relative to the sample delay. */
+    state->depth = props->Flanger.Depth * state->delay;
+
+    /* Gains for left and right sides */
+    CalcAngleCoeffs(-F_PI_2, 0.0f, 0.0f, coeffs);
+    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[0]);
+    CalcAngleCoeffs( F_PI_2, 0.0f, 0.0f, coeffs);
+    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[1]);
+
+    phase = props->Flanger.Phase;
+    rate = props->Flanger.Rate;
+    if(!(rate > 0.0f))
+    {
+        state->lfo_scale = 0.0f;
+        state->lfo_range = 1;
+        state->lfo_disp = 0;
+    }
+    else
+    {
+        /* Calculate LFO coefficient */
+        state->lfo_range = fastf2i(frequency/rate + 0.5f);
+        switch(state->waveform)
+        {
+            case FWF_Triangle:
+                state->lfo_scale = 4.0f / state->lfo_range;
+                break;
+            case FWF_Sinusoid:
+                state->lfo_scale = F_TAU / state->lfo_range;
+                break;
+        }
+
+        /* Calculate lfo phase displacement */
+        if(phase >= 0)
+            state->lfo_disp = fastf2i(state->lfo_range * (phase/360.0f));
+        else
+            state->lfo_disp = fastf2i(state->lfo_range * ((360+phase)/360.0f));
+    }
+}
+
+static inline void Triangle(ALint *delay_left, ALint *delay_right, ALuint offset, const ALflangerState *state)
+{
+    ALfloat lfo_value;
+
+    lfo_value = 1.0f - fabsf(2.0f - state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_left = fastf2i(lfo_value) + state->delay;
+
+    offset += state->lfo_disp;
+    lfo_value = 1.0f - fabsf(2.0f - state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_right = fastf2i(lfo_value) + state->delay;
+}
+
+static inline void Sinusoid(ALint *delay_left, ALint *delay_right, ALuint offset, const ALflangerState *state)
+{
+    ALfloat lfo_value;
+
+    lfo_value = sinf(state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_left = fastf2i(lfo_value) + state->delay;
+
+    offset += state->lfo_disp;
+    lfo_value = sinf(state->lfo_scale*(offset%state->lfo_range));
+    lfo_value *= state->depth;
+    *delay_right = fastf2i(lfo_value) + state->delay;
+}
+
+#define DECL_TEMPLATE(Func)                                                   \
+static void Process##Func(ALflangerState *state, const ALsizei SamplesToDo,   \
+  const ALfloat *restrict SamplesIn, ALfloat (*restrict out)[2])              \
+{                                                                             \
+    const ALsizei bufmask = state->BufferLength-1;                            \
+    ALfloat *restrict leftbuf = state->SampleBuffer[0];                       \
+    ALfloat *restrict rightbuf = state->SampleBuffer[1];                      \
+    ALsizei offset = state->offset;                                           \
+    const ALfloat feedback = state->feedback;                                 \
+    ALsizei it;                                                               \
+                                                                              \
+    for(it = 0;it < SamplesToDo;it++)                                         \
+    {                                                                         \
+        ALint delay_left, delay_right;                                        \
+        Func(&delay_left, &delay_right, offset, state);                       \
+                                                                              \
+        leftbuf[offset&bufmask] = SamplesIn[it];                              \
+        out[it][0] = leftbuf[(offset-delay_left)&bufmask];                    \
+        leftbuf[offset&bufmask] += out[it][0] * feedback;                     \
+                                                                              \
+        rightbuf[offset&bufmask] = SamplesIn[it];                             \
+        out[it][1] = rightbuf[(offset-delay_right)&bufmask];                  \
+        rightbuf[offset&bufmask] += out[it][1] * feedback;                    \
+                                                                              \
+        offset++;                                                             \
+    }                                                                         \
+    state->offset = offset;                                                   \
+}
+
+DECL_TEMPLATE(Triangle)
+DECL_TEMPLATE(Sinusoid)
+
+#undef DECL_TEMPLATE
+
+static ALvoid ALflangerState_process(ALflangerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALsizei it, kt;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALfloat temps[128][2];
+        ALsizei td = mini(128, SamplesToDo-base);
+
+        switch(state->waveform)
+        {
+            case FWF_Triangle:
+                ProcessTriangle(state, td, SamplesIn[0]+base, temps);
+                break;
+            case FWF_Sinusoid:
+                ProcessSinusoid(state, td, SamplesIn[0]+base, temps);
+                break;
+        }
+
+        for(kt = 0;kt < NumChannels;kt++)
+        {
+            ALfloat gain = state->Gain[0][kt];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(it = 0;it < td;it++)
+                    SamplesOut[kt][it+base] += temps[it][0] * gain;
+            }
+
+            gain = state->Gain[1][kt];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(it = 0;it < td;it++)
+                    SamplesOut[kt][it+base] += temps[it][1] * gain;
+            }
+        }
+
+        base += td;
+    }
+}
+
+
+typedef struct ALflangerStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALflangerStateFactory;
+
+ALeffectState *ALflangerStateFactory_create(ALflangerStateFactory *UNUSED(factory))
+{
+    ALflangerState *state;
+
+    NEW_OBJ0(state, ALflangerState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALflangerStateFactory);
+
+ALeffectStateFactory *ALflangerStateFactory_getFactory(void)
+{
+    static ALflangerStateFactory FlangerFactory = { { GET_VTABLE2(ALflangerStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &FlangerFactory);
+}
+
+
+void ALflanger_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FLANGER_WAVEFORM:
+            if(!(val >= AL_FLANGER_MIN_WAVEFORM && val <= AL_FLANGER_MAX_WAVEFORM))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Flanger.Waveform = val;
+            break;
+
+        case AL_FLANGER_PHASE:
+            if(!(val >= AL_FLANGER_MIN_PHASE && val <= AL_FLANGER_MAX_PHASE))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Flanger.Phase = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALflanger_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALflanger_setParami(effect, context, param, vals[0]);
+}
+void ALflanger_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FLANGER_RATE:
+            if(!(val >= AL_FLANGER_MIN_RATE && val <= AL_FLANGER_MAX_RATE))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Flanger.Rate = val;
+            break;
+
+        case AL_FLANGER_DEPTH:
+            if(!(val >= AL_FLANGER_MIN_DEPTH && val <= AL_FLANGER_MAX_DEPTH))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Flanger.Depth = val;
+            break;
+
+        case AL_FLANGER_FEEDBACK:
+            if(!(val >= AL_FLANGER_MIN_FEEDBACK && val <= AL_FLANGER_MAX_FEEDBACK))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Flanger.Feedback = val;
+            break;
+
+        case AL_FLANGER_DELAY:
+            if(!(val >= AL_FLANGER_MIN_DELAY && val <= AL_FLANGER_MAX_DELAY))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Flanger.Delay = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALflanger_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALflanger_setParamf(effect, context, param, vals[0]);
+}
+
+void ALflanger_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FLANGER_WAVEFORM:
+            *val = props->Flanger.Waveform;
+            break;
+
+        case AL_FLANGER_PHASE:
+            *val = props->Flanger.Phase;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALflanger_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALflanger_getParami(effect, context, param, vals);
+}
+void ALflanger_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FLANGER_RATE:
+            *val = props->Flanger.Rate;
+            break;
+
+        case AL_FLANGER_DEPTH:
+            *val = props->Flanger.Depth;
+            break;
+
+        case AL_FLANGER_FEEDBACK:
+            *val = props->Flanger.Feedback;
+            break;
+
+        case AL_FLANGER_DELAY:
+            *val = props->Flanger.Delay;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALflanger_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALflanger_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALflanger);

Alc/effects/modulator.c

@@ -0,0 +1,311 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2009 by Chris Robinson.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+typedef struct ALmodulatorState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    void (*Process)(ALfloat*, const ALfloat*, ALsizei, const ALsizei, ALsizei);
+
+    ALsizei index;
+    ALsizei step;
+
+    ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
+
+    ALfilterState Filter[MAX_EFFECT_CHANNELS];
+} ALmodulatorState;
+
+static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state);
+static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *device);
+static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
+static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALmodulatorState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALmodulatorState);
+
+
+#define WAVEFORM_FRACBITS  24
+#define WAVEFORM_FRACONE   (1<<WAVEFORM_FRACBITS)
+#define WAVEFORM_FRACMASK  (WAVEFORM_FRACONE-1)
+
+static inline ALfloat Sin(ALsizei index)
+{
+    return sinf(index*(F_TAU/WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f;
+}
+
+static inline ALfloat Saw(ALsizei index)
+{
+    return (ALfloat)index / WAVEFORM_FRACONE;
+}
+
+static inline ALfloat Square(ALsizei index)
+{
+    return (ALfloat)((index >> (WAVEFORM_FRACBITS - 1)) & 1);
+}
+
+#define DECL_TEMPLATE(func)                                                   \
+static void Modulate##func(ALfloat *restrict dst, const ALfloat *restrict src,\
+                           ALsizei index, const ALsizei step, ALsizei todo)   \
+{                                                                             \
+    ALsizei i;                                                                \
+    for(i = 0;i < todo;i++)                                                   \
+    {                                                                         \
+        index += step;                                                        \
+        index &= WAVEFORM_FRACMASK;                                           \
+        dst[i] = src[i] * func(index);                                        \
+    }                                                                         \
+}
+
+DECL_TEMPLATE(Sin)
+DECL_TEMPLATE(Saw)
+DECL_TEMPLATE(Square)
+
+#undef DECL_TEMPLATE
+
+
+static void ALmodulatorState_Construct(ALmodulatorState *state)
+{
+    ALuint i;
+
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALmodulatorState, ALeffectState, state);
+
+    state->index = 0;
+    state->step = 1;
+
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+        ALfilterState_clear(&state->Filter[i]);
+}
+
+static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *UNUSED(state), ALCdevice *UNUSED(device))
+{
+    return AL_TRUE;
+}
+
+static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
+{
+    ALfloat cw, a;
+    ALsizei i;
+
+    if(props->Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
+        state->Process = ModulateSin;
+    else if(props->Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
+        state->Process = ModulateSaw;
+    else /*if(Slot->Params.EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)*/
+        state->Process = ModulateSquare;
+
+    state->step = fastf2u(props->Modulator.Frequency*WAVEFORM_FRACONE /
+                          Device->Frequency);
+    if(state->step == 0) state->step = 1;
+
+    /* Custom filter coeffs, which match the old version instead of a low-shelf. */
+    cw = cosf(F_TAU * props->Modulator.HighPassCutoff / Device->Frequency);
+    a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
+
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+    {
+        state->Filter[i].b0 = a;
+        state->Filter[i].b1 = -a;
+        state->Filter[i].b2 = 0.0f;
+        state->Filter[i].a1 = -a;
+        state->Filter[i].a2 = 0.0f;
+    }
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = Device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = Device->FOAOut.NumChannels;
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+        ComputeFirstOrderGains(Device->FOAOut, IdentityMatrixf.m[i],
+                               Slot->Params.Gain, state->Gain[i]);
+}
+
+static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    const ALsizei step = state->step;
+    ALsizei index = state->index;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALfloat temps[2][128];
+        ALsizei td = mini(128, SamplesToDo-base);
+        ALsizei i, j, k;
+
+        for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
+        {
+            ALfilterState_process(&state->Filter[j], temps[0], &SamplesIn[j][base], td);
+            state->Process(temps[1], temps[0], index, step, td);
+
+            for(k = 0;k < NumChannels;k++)
+            {
+                ALfloat gain = state->Gain[j][k];
+                if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+                    continue;
+
+                for(i = 0;i < td;i++)
+                    SamplesOut[k][base+i] += gain * temps[1][i];
+            }
+        }
+
+        for(i = 0;i < td;i++)
+        {
+            index += step;
+            index &= WAVEFORM_FRACMASK;
+        }
+        base += td;
+    }
+    state->index = index;
+}
+
+
+typedef struct ALmodulatorStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALmodulatorStateFactory;
+
+static ALeffectState *ALmodulatorStateFactory_create(ALmodulatorStateFactory *UNUSED(factory))
+{
+    ALmodulatorState *state;
+
+    NEW_OBJ0(state, ALmodulatorState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALmodulatorStateFactory);
+
+ALeffectStateFactory *ALmodulatorStateFactory_getFactory(void)
+{
+    static ALmodulatorStateFactory ModulatorFactory = { { GET_VTABLE2(ALmodulatorStateFactory, ALeffectStateFactory) } };
+
+    return STATIC_CAST(ALeffectStateFactory, &ModulatorFactory);
+}
+
+
+void ALmodulator_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_RING_MODULATOR_FREQUENCY:
+            if(!(val >= AL_RING_MODULATOR_MIN_FREQUENCY && val <= AL_RING_MODULATOR_MAX_FREQUENCY))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Modulator.Frequency = val;
+            break;
+
+        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
+            if(!(val >= AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF && val <= AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Modulator.HighPassCutoff = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALmodulator_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALmodulator_setParamf(effect, context, param, vals[0]);
+}
+void ALmodulator_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_RING_MODULATOR_FREQUENCY:
+        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
+            ALmodulator_setParamf(effect, context, param, (ALfloat)val);
+            break;
+
+        case AL_RING_MODULATOR_WAVEFORM:
+            if(!(val >= AL_RING_MODULATOR_MIN_WAVEFORM && val <= AL_RING_MODULATOR_MAX_WAVEFORM))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            props->Modulator.Waveform = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALmodulator_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALmodulator_setParami(effect, context, param, vals[0]);
+}
+
+void ALmodulator_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_RING_MODULATOR_FREQUENCY:
+            *val = (ALint)props->Modulator.Frequency;
+            break;
+        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
+            *val = (ALint)props->Modulator.HighPassCutoff;
+            break;
+        case AL_RING_MODULATOR_WAVEFORM:
+            *val = props->Modulator.Waveform;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALmodulator_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALmodulator_getParami(effect, context, param, vals);
+}
+void ALmodulator_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_RING_MODULATOR_FREQUENCY:
+            *val = props->Modulator.Frequency;
+            break;
+        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
+            *val = props->Modulator.HighPassCutoff;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALmodulator_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALmodulator_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALmodulator);

Alc/effects/null.c

@@ -0,0 +1,179 @@
+#include "config.h"
+
+#include <stdlib.h>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "alMain.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+
+
+typedef struct ALnullState {
+    DERIVE_FROM_TYPE(ALeffectState);
+} ALnullState;
+
+/* Forward-declare "virtual" functions to define the vtable with. */
+static ALvoid ALnullState_Destruct(ALnullState *state);
+static ALboolean ALnullState_deviceUpdate(ALnullState *state, ALCdevice *device);
+static ALvoid ALnullState_update(ALnullState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props);
+static ALvoid ALnullState_process(ALnullState *state, ALsizei samplesToDo, const ALfloatBUFFERSIZE*restrict samplesIn, ALfloatBUFFERSIZE*restrict samplesOut, ALsizei NumChannels);
+static void *ALnullState_New(size_t size);
+static void ALnullState_Delete(void *ptr);
+
+/* Define the ALeffectState vtable for this type. */
+DEFINE_ALEFFECTSTATE_VTABLE(ALnullState);
+
+
+/* This constructs the effect state. It's called when the object is first
+ * created. Make sure to call the parent Construct function first, and set the
+ * vtable!
+ */
+static void ALnullState_Construct(ALnullState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALnullState, ALeffectState, state);
+}
+
+/* This destructs (not free!) the effect state. It's called only when the
+ * effect slot is no longer used. Make sure to call the parent Destruct
+ * function before returning!
+ */
+static ALvoid ALnullState_Destruct(ALnullState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+/* This updates the device-dependant effect state. This is called on
+ * initialization and any time the device parameters (eg. playback frequency,
+ * format) have been changed.
+ */
+static ALboolean ALnullState_deviceUpdate(ALnullState* UNUSED(state), ALCdevice* UNUSED(device))
+{
+    return AL_TRUE;
+}
+
+/* This updates the effect state. This is called any time the effect is
+ * (re)loaded into a slot.
+ */
+static ALvoid ALnullState_update(ALnullState* UNUSED(state), const ALCdevice* UNUSED(device), const ALeffectslot* UNUSED(slot), const ALeffectProps* UNUSED(props))
+{
+}
+
+/* This processes the effect state, for the given number of samples from the
+ * input to the output buffer. The result should be added to the output buffer,
+ * not replace it.
+ */
+static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALsizei UNUSED(samplesToDo), const ALfloatBUFFERSIZE*restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALsizei UNUSED(NumChannels))
+{
+}
+
+/* This allocates memory to store the object, before it gets constructed.
+ * DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method.
+ */
+static void *ALnullState_New(size_t size)
+{
+    return al_malloc(16, size);
+}
+
+/* This frees the memory used by the object, after it has been destructed.
+ * DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method.
+ */
+static void ALnullState_Delete(void *ptr)
+{
+    al_free(ptr);
+}
+
+
+typedef struct ALnullStateFactory {
+    DERIVE_FROM_TYPE(ALeffectStateFactory);
+} ALnullStateFactory;
+
+/* Creates ALeffectState objects of the appropriate type. */
+ALeffectState *ALnullStateFactory_create(ALnullStateFactory *UNUSED(factory))
+{
+    ALnullState *state;
+
+    NEW_OBJ0(state, ALnullState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+/* Define the ALeffectStateFactory vtable for this type. */
+DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALnullStateFactory);
+
+ALeffectStateFactory *ALnullStateFactory_getFactory(void)
+{
+    static ALnullStateFactory NullFactory = { { GET_VTABLE2(ALnullStateFactory, ALeffectStateFactory) } };
+    return STATIC_CAST(ALeffectStateFactory, &NullFactory);
+}
+
+
+void ALnull_setParami(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALnull_setParamiv(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, const ALint* UNUSED(vals))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALnull_setParamf(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALnull_setParamfv(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat* UNUSED(vals))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+
+void ALnull_getParami(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(val))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALnull_getParamiv(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(vals))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALnull_getParamf(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(val))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+void ALnull_getParamfv(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(vals))
+{
+    switch(param)
+    {
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+
+DEFINE_ALEFFECT_VTABLE(ALnull);

Alc/hrtf.h

@@ -0,0 +1,56 @@
+#ifndef ALC_HRTF_H
+#define ALC_HRTF_H
+
+#include "AL/al.h"
+#include "AL/alc.h"
+
+#include "alMain.h"
+#include "alstring.h"
+#include "atomic.h"
+
+
+#define HRTFDELAY_BITS    (20)
+#define HRTFDELAY_FRACONE (1<<HRTFDELAY_BITS)
+#define HRTFDELAY_MASK    (HRTFDELAY_FRACONE-1)
+
+/* The maximum number of virtual speakers used to generate HRTF coefficients
+ * for decoding B-Format.
+ */
+#define HRTF_AMBI_MAX_CHANNELS 16
+
+
+struct HrtfEntry;
+
+struct Hrtf {
+    RefCount ref;
+
+    ALuint sampleRate;
+    ALsizei irSize;
+    ALubyte evCount;
+
+    const ALubyte *azCount;
+    const ALushort *evOffset;
+    const ALfloat (*coeffs)[2];
+    const ALubyte (*delays)[2];
+};
+
+
+void FreeHrtfs(void);
+
+vector_EnumeratedHrtf EnumerateHrtf(const_al_string devname);
+void FreeHrtfList(vector_EnumeratedHrtf *list);
+struct Hrtf *GetLoadedHrtf(struct HrtfEntry *entry);
+void Hrtf_IncRef(struct Hrtf *hrtf);
+void Hrtf_DecRef(struct Hrtf *hrtf);
+
+void GetHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat spread, ALfloat (*coeffs)[2], ALsizei *delays);
+
+/**
+ * Produces HRTF filter coefficients for decoding B-Format, given a set of
+ * virtual speaker positions and HF/LF matrices for decoding to them. The
+ * returned coefficients are ordered and scaled according to the matrices.
+ * Returns the maximum impulse-response length of the generated coefficients.
+ */
+ALsizei BuildBFormatHrtf(const struct Hrtf *Hrtf, DirectHrtfState *state, ALsizei NumChannels, const ALfloat (*restrict AmbiPoints)[2], const ALfloat (*restrict AmbiMatrix)[2][MAX_AMBI_COEFFS], ALsizei AmbiCount);
+
+#endif /* ALC_HRTF_H */

Alc/mastering.c

@@ -0,0 +1,255 @@
+#include "config.h"
+
+#include <math.h>
+
+#include "alu.h"
+#include "almalloc.h"
+
+#define RMS_WINDOW_SIZE (1<<7)
+#define RMS_WINDOW_MASK (RMS_WINDOW_SIZE-1)
+#define RMS_VALUE_MAX  (1<<24)
+
+#define LOOKAHEAD_SIZE (1<<13)
+#define LOOKAHEAD_MASK (LOOKAHEAD_SIZE-1)
+
+static_assert(RMS_VALUE_MAX < (UINT_MAX / RMS_WINDOW_SIZE), "RMS_VALUE_MAX is too big");
+
+typedef struct Compressor {
+    ALfloat PreGain;
+    ALfloat PostGain;
+    ALboolean SummedLink;
+    ALfloat AttackMin;
+    ALfloat AttackMax;
+    ALfloat ReleaseMin;
+    ALfloat ReleaseMax;
+    ALfloat Ratio;
+    ALfloat Threshold;
+    ALfloat Knee;
+    ALuint SampleRate;
+
+    ALuint RmsSum;
+    ALuint *RmsWindow;
+    ALsizei RmsIndex;
+    ALfloat Envelope[BUFFERSIZE];
+    ALfloat EnvLast;
+} Compressor;
+
+/* Multichannel compression is linked via one of two modes:
+ *
+ *   Summed - Absolute sum of all channels.
+ *   Maxed  - Absolute maximum of any channel.
+ */
+static void SumChannels(Compressor *Comp, const ALsizei NumChans, const ALsizei SamplesToDo,
+                        ALfloat (*restrict OutBuffer)[BUFFERSIZE])
+{
+    ALsizei c, i;
+
+    for(i = 0;i < SamplesToDo;i++)
+        Comp->Envelope[i] = 0.0f;
+
+    for(c = 0;c < NumChans;c++)
+    {
+        for(i = 0;i < SamplesToDo;i++)
+            Comp->Envelope[i] += OutBuffer[c][i];
+    }
+
+    for(i = 0;i < SamplesToDo;i++)
+        Comp->Envelope[i] = fabsf(Comp->Envelope[i]);
+}
+
+static void MaxChannels(Compressor *Comp, const ALsizei NumChans, const ALsizei SamplesToDo,
+                        ALfloat (*restrict OutBuffer)[BUFFERSIZE])
+{
+    ALsizei c, i;
+
+    for(i = 0;i < SamplesToDo;i++)
+        Comp->Envelope[i] = 0.0f;
+
+    for(c = 0;c < NumChans;c++)
+    {
+        for(i = 0;i < SamplesToDo;i++)
+            Comp->Envelope[i] = maxf(Comp->Envelope[i], fabsf(OutBuffer[c][i]));
+    }
+}
+
+/* Envelope detection/sensing can be done via:
+ *
+ *   RMS  - Rectangular windowed root mean square of linking stage.
+ *   Peak - Implicit output from linking stage.
+ */
+static void RmsDetection(Compressor *Comp, const ALsizei SamplesToDo)
+{
+    ALuint sum = Comp->RmsSum;
+    ALuint *window = Comp->RmsWindow;
+    ALsizei index = Comp->RmsIndex;
+    ALsizei i;
+
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        ALfloat sig = Comp->Envelope[i];
+
+        sum -= window[index];
+        window[index] = fastf2u(minf(sig * sig * 65536.0f, RMS_VALUE_MAX));
+        sum += window[index];
+        index = (index + 1) & RMS_WINDOW_MASK;
+
+        Comp->Envelope[i] = sqrtf(sum / 65536.0f / RMS_WINDOW_SIZE);
+    }
+
+    Comp->RmsSum = sum;
+    Comp->RmsIndex = index;
+}
+
+/* This isn't a very sophisticated envelope follower, but it gets the job
+ * done.  First, it operates at logarithmic scales to keep transitions
+ * appropriate for human hearing.  Second, it can apply adaptive (automated)
+ * attack/release adjustments based on the signal.
+ */
+static void FollowEnvelope(Compressor *Comp, const ALsizei SamplesToDo)
+{
+    ALfloat attackMin = Comp->AttackMin;
+    ALfloat attackMax = Comp->AttackMax;
+    ALfloat releaseMin = Comp->ReleaseMin;
+    ALfloat releaseMax = Comp->ReleaseMax;
+    ALfloat last = Comp->EnvLast;
+    ALsizei i;
+
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        ALfloat env = maxf(-6.0f, log10f(Comp->Envelope[i]));
+        ALfloat slope = minf(1.0f, fabsf(env - last) / 4.5f);
+
+        if(env > last)
+            last = minf(env, last + lerp(attackMin, attackMax, 1.0f - (slope * slope)));
+        else
+            last = maxf(env, last + lerp(releaseMin, releaseMax, 1.0f - (slope * slope)));
+
+        Comp->Envelope[i] = last;
+    }
+
+    Comp->EnvLast = last;
+}
+
+/* The envelope is converted to control gain with an optional soft knee. */
+static void EnvelopeGain(Compressor *Comp, const ALsizei SamplesToDo, const ALfloat Slope)
+{
+    const ALfloat threshold = Comp->Threshold;
+    const ALfloat knee = Comp->Knee;
+    ALsizei i;
+
+    if(!(knee > 0.0f))
+    {
+        for(i = 0;i < SamplesToDo;i++)
+        {
+            ALfloat gain = Slope * (threshold - Comp->Envelope[i]);
+            Comp->Envelope[i] = powf(10.0f, minf(0.0f, gain));
+        }
+    }
+    else
+    {
+        const ALfloat lower = threshold - (0.5f * knee);
+        const ALfloat upper = threshold + (0.5f * knee);
+        const ALfloat m = 0.5f * Slope / knee;
+
+        for(i = 0;i < SamplesToDo;i++)
+        {
+            ALfloat env = Comp->Envelope[i];
+            ALfloat gain;
+
+            if(env > lower && env < upper)
+                gain = m * (env - lower) * (lower - env);
+            else
+                gain = Slope * (threshold - env);
+
+            Comp->Envelope[i] = powf(10.0f, minf(0.0f, gain));
+        }
+    }
+}
+
+
+Compressor *CompressorInit(const ALfloat PreGainDb, const ALfloat PostGainDb,
+                           const ALboolean SummedLink, const ALboolean RmsSensing,
+                           const ALfloat AttackTimeMin, const ALfloat AttackTimeMax,
+                           const ALfloat ReleaseTimeMin, const ALfloat ReleaseTimeMax,
+                           const ALfloat Ratio, const ALfloat ThresholdDb,
+                           const ALfloat KneeDb, const ALuint SampleRate)
+{
+    Compressor *Comp;
+    size_t size;
+    ALsizei i;
+
+    size = sizeof(*Comp);
+    if(RmsSensing)
+        size += sizeof(Comp->RmsWindow[0]) * RMS_WINDOW_SIZE;
+    Comp = al_calloc(16, size);
+
+    Comp->PreGain = powf(10.0f, PreGainDb / 20.0f);
+    Comp->PostGain = powf(10.0f, PostGainDb / 20.0f);
+    Comp->SummedLink = SummedLink;
+    Comp->AttackMin = 1.0f / maxf(0.000001f, AttackTimeMin * SampleRate * logf(10.0f));
+    Comp->AttackMax = 1.0f / maxf(0.000001f, AttackTimeMax * SampleRate * logf(10.0f));
+    Comp->ReleaseMin = -1.0f / maxf(0.000001f, ReleaseTimeMin * SampleRate * logf(10.0f));
+    Comp->ReleaseMax = -1.0f / maxf(0.000001f, ReleaseTimeMax * SampleRate * logf(10.0f));
+    Comp->Ratio = Ratio;
+    Comp->Threshold = ThresholdDb / 20.0f;
+    Comp->Knee = maxf(0.0f, KneeDb / 20.0f);
+    Comp->SampleRate = SampleRate;
+
+    Comp->RmsSum = 0;
+    if(RmsSensing)
+        Comp->RmsWindow = (ALuint*)(Comp+1);
+    else
+        Comp->RmsWindow = NULL;
+    Comp->RmsIndex = 0;
+
+    for(i = 0;i < BUFFERSIZE;i++)
+        Comp->Envelope[i] = 0.0f;
+    Comp->EnvLast = -6.0f;
+
+    return Comp;
+}
+
+ALuint GetCompressorSampleRate(const Compressor *Comp)
+{
+    return Comp->SampleRate;
+}
+
+void ApplyCompression(Compressor *Comp, const ALsizei NumChans, const ALsizei SamplesToDo,
+                      ALfloat (*restrict OutBuffer)[BUFFERSIZE])
+{
+    ALsizei c, i;
+
+    if(Comp->PreGain != 1.0f)
+    {
+        for(c = 0;c < NumChans;c++)
+        {
+            for(i = 0;i < SamplesToDo;i++)
+                OutBuffer[c][i] *= Comp->PreGain;
+        }
+    }
+
+    if(Comp->SummedLink)
+        SumChannels(Comp, NumChans, SamplesToDo, OutBuffer);
+    else
+        MaxChannels(Comp, NumChans, SamplesToDo, OutBuffer);
+
+    if(Comp->RmsWindow)
+        RmsDetection(Comp, SamplesToDo);
+    FollowEnvelope(Comp, SamplesToDo);
+
+    if(Comp->Ratio > 0.0f)
+        EnvelopeGain(Comp, SamplesToDo, 1.0f - (1.0f / Comp->Ratio));
+    else
+        EnvelopeGain(Comp, SamplesToDo, 1.0f);
+
+    if(Comp->PostGain != 1.0f)
+    {
+        for(i = 0;i < SamplesToDo;i++)
+            Comp->Envelope[i] *= Comp->PostGain;
+    }
+    for(c = 0;c < NumChans;c++)
+    {
+        for(i = 0;i < SamplesToDo;i++)
+            OutBuffer[c][i] *= Comp->Envelope[i];
+    }
+}

Alc/mixer.c

@@ -0,0 +1,673 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <ctype.h>
+#include <assert.h>
+
+#include "alMain.h"
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "alSource.h"
+#include "alBuffer.h"
+#include "alListener.h"
+#include "alAuxEffectSlot.h"
+#include "alu.h"
+
+#include "mixer_defs.h"
+
+
+static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
+              "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
+
+extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALint *restrict pos_arr, ALsizei size);
+
+
+/* BSinc requires up to 11 extra samples before the current position, and 12 after. */
+static_assert(MAX_PRE_SAMPLES >= 11, "MAX_PRE_SAMPLES must be at least 11!");
+static_assert(MAX_POST_SAMPLES >= 12, "MAX_POST_SAMPLES must be at least 12!");
+
+
+enum Resampler ResamplerDefault = LinearResampler;
+
+static MixerFunc MixSamples = Mix_C;
+static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
+HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
+
+MixerFunc SelectMixer(void)
+{
+#ifdef HAVE_NEON
+    if((CPUCapFlags&CPU_CAP_NEON))
+        return Mix_Neon;
+#endif
+#ifdef HAVE_SSE
+    if((CPUCapFlags&CPU_CAP_SSE))
+        return Mix_SSE;
+#endif
+    return Mix_C;
+}
+
+RowMixerFunc SelectRowMixer(void)
+{
+#ifdef HAVE_NEON
+    if((CPUCapFlags&CPU_CAP_NEON))
+        return MixRow_Neon;
+#endif
+#ifdef HAVE_SSE
+    if((CPUCapFlags&CPU_CAP_SSE))
+        return MixRow_SSE;
+#endif
+    return MixRow_C;
+}
+
+static inline HrtfMixerFunc SelectHrtfMixer(void)
+{
+#ifdef HAVE_NEON
+    if((CPUCapFlags&CPU_CAP_NEON))
+        return MixHrtf_Neon;
+#endif
+#ifdef HAVE_SSE
+    if((CPUCapFlags&CPU_CAP_SSE))
+        return MixHrtf_SSE;
+#endif
+    return MixHrtf_C;
+}
+
+static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
+{
+#ifdef HAVE_NEON
+    if((CPUCapFlags&CPU_CAP_NEON))
+        return MixHrtfBlend_Neon;
+#endif
+#ifdef HAVE_SSE
+    if((CPUCapFlags&CPU_CAP_SSE))
+        return MixHrtfBlend_SSE;
+#endif
+    return MixHrtfBlend_C;
+}
+
+ResamplerFunc SelectResampler(enum Resampler resampler)
+{
+    switch(resampler)
+    {
+        case PointResampler:
+            return Resample_point32_C;
+        case LinearResampler:
+#ifdef HAVE_NEON
+            if((CPUCapFlags&CPU_CAP_NEON))
+                return Resample_lerp32_Neon;
+#endif
+#ifdef HAVE_SSE4_1
+            if((CPUCapFlags&CPU_CAP_SSE4_1))
+                return Resample_lerp32_SSE41;
+#endif
+#ifdef HAVE_SSE2
+            if((CPUCapFlags&CPU_CAP_SSE2))
+                return Resample_lerp32_SSE2;
+#endif
+            return Resample_lerp32_C;
+        case FIR4Resampler:
+#ifdef HAVE_NEON
+            if((CPUCapFlags&CPU_CAP_NEON))
+                return Resample_fir4_32_Neon;
+#endif
+#ifdef HAVE_SSE4_1
+            if((CPUCapFlags&CPU_CAP_SSE4_1))
+                return Resample_fir4_32_SSE41;
+#endif
+#ifdef HAVE_SSE3
+            if((CPUCapFlags&CPU_CAP_SSE3))
+                return Resample_fir4_32_SSE3;
+#endif
+            return Resample_fir4_32_C;
+        case BSincResampler:
+#ifdef HAVE_NEON
+            if((CPUCapFlags&CPU_CAP_NEON))
+                return Resample_bsinc32_Neon;
+#endif
+#ifdef HAVE_SSE
+            if((CPUCapFlags&CPU_CAP_SSE))
+                return Resample_bsinc32_SSE;
+#endif
+            return Resample_bsinc32_C;
+    }
+
+    return Resample_point32_C;
+}
+
+
+void aluInitMixer(void)
+{
+    const char *str;
+
+    if(ConfigValueStr(NULL, NULL, "resampler", &str))
+    {
+        if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
+            ResamplerDefault = PointResampler;
+        else if(strcasecmp(str, "linear") == 0)
+            ResamplerDefault = LinearResampler;
+        else if(strcasecmp(str, "sinc4") == 0)
+            ResamplerDefault = FIR4Resampler;
+        else if(strcasecmp(str, "bsinc") == 0)
+            ResamplerDefault = BSincResampler;
+        else if(strcasecmp(str, "cubic") == 0 || strcasecmp(str, "sinc8") == 0)
+        {
+            WARN("Resampler option \"%s\" is deprecated, using sinc4\n", str);
+            ResamplerDefault = FIR4Resampler;
+        }
+        else
+        {
+            char *end;
+            long n = strtol(str, &end, 0);
+            if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
+                ResamplerDefault = n;
+            else
+                WARN("Invalid resampler: %s\n", str);
+        }
+    }
+
+    MixHrtfBlendSamples = SelectHrtfBlendMixer();
+    MixHrtfSamples = SelectHrtfMixer();
+    MixSamples = SelectMixer();
+}
+
+
+static inline ALfloat Sample_ALbyte(ALbyte val)
+{ return val * (1.0f/128.0f); }
+
+static inline ALfloat Sample_ALshort(ALshort val)
+{ return val * (1.0f/32768.0f); }
+
+static inline ALfloat Sample_ALfloat(ALfloat val)
+{ return val; }
+
+#define DECL_TEMPLATE(T)                                                      \
+static inline void Load_##T(ALfloat *dst, const T *src, ALint srcstep, ALsizei samples)\
+{                                                                             \
+    ALsizei i;                                                                \
+    for(i = 0;i < samples;i++)                                                \
+        dst[i] = Sample_##T(src[i*srcstep]);                                  \
+}
+
+DECL_TEMPLATE(ALbyte)
+DECL_TEMPLATE(ALshort)
+DECL_TEMPLATE(ALfloat)
+
+#undef DECL_TEMPLATE
+
+static void LoadSamples(ALfloat *dst, const ALvoid *src, ALint srcstep, enum FmtType srctype, ALsizei samples)
+{
+    switch(srctype)
+    {
+        case FmtByte:
+            Load_ALbyte(dst, src, srcstep, samples);
+            break;
+        case FmtShort:
+            Load_ALshort(dst, src, srcstep, samples);
+            break;
+        case FmtFloat:
+            Load_ALfloat(dst, src, srcstep, samples);
+            break;
+    }
+}
+
+static inline void SilenceSamples(ALfloat *dst, ALsizei samples)
+{
+    ALsizei i;
+    for(i = 0;i < samples;i++)
+        dst[i] = 0.0f;
+}
+
+
+static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
+                                ALfloat *restrict dst, const ALfloat *restrict src,
+                                ALsizei numsamples, enum ActiveFilters type)
+{
+    ALsizei i;
+    switch(type)
+    {
+        case AF_None:
+            ALfilterState_processPassthru(lpfilter, src, numsamples);
+            ALfilterState_processPassthru(hpfilter, src, numsamples);
+            break;
+
+        case AF_LowPass:
+            ALfilterState_process(lpfilter, dst, src, numsamples);
+            ALfilterState_processPassthru(hpfilter, dst, numsamples);
+            return dst;
+        case AF_HighPass:
+            ALfilterState_processPassthru(lpfilter, src, numsamples);
+            ALfilterState_process(hpfilter, dst, src, numsamples);
+            return dst;
+
+        case AF_BandPass:
+            for(i = 0;i < numsamples;)
+            {
+                ALfloat temp[256];
+                ALsizei todo = mini(256, numsamples-i);
+
+                ALfilterState_process(lpfilter, temp, src+i, todo);
+                ALfilterState_process(hpfilter, dst+i, temp, todo);
+                i += todo;
+            }
+            return dst;
+    }
+    return src;
+}
+
+
+ALboolean MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo)
+{
+    ALbufferlistitem *BufferListItem;
+    ALbufferlistitem *BufferLoopItem;
+    ALsizei NumChannels, SampleSize;
+    ResamplerFunc Resample;
+    ALsizei DataPosInt;
+    ALsizei DataPosFrac;
+    ALint64 DataSize64;
+    ALint increment;
+    ALsizei Counter;
+    ALsizei OutPos;
+    ALsizei IrSize;
+    bool isplaying;
+    bool firstpass;
+    ALsizei chan;
+    ALsizei send;
+
+    /* Get source info */
+    isplaying      = true; /* Will only be called while playing. */
+    DataPosInt     = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
+    DataPosFrac    = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
+    BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
+    BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
+    NumChannels    = voice->NumChannels;
+    SampleSize     = voice->SampleSize;
+    increment      = voice->Step;
+
+    IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
+
+    Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
+                Resample_copy32_C : voice->Resampler);
+
+    Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
+    firstpass = true;
+    OutPos = 0;
+
+    do {
+        ALsizei SrcBufferSize, DstBufferSize;
+
+        /* Figure out how many buffer samples will be needed */
+        DataSize64  = SamplesToDo-OutPos;
+        DataSize64 *= increment;
+        DataSize64 += DataPosFrac+FRACTIONMASK;
+        DataSize64 >>= FRACTIONBITS;
+        DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
+
+        SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
+
+        /* Figure out how many samples we can actually mix from this. */
+        DataSize64  = SrcBufferSize;
+        DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
+        DataSize64 <<= FRACTIONBITS;
+        DataSize64 -= DataPosFrac;
+
+        DstBufferSize = (ALsizei)((DataSize64+(increment-1)) / increment);
+        DstBufferSize = mini(DstBufferSize, (SamplesToDo-OutPos));
+
+        /* Some mixers like having a multiple of 4, so try to give that unless
+         * this is the last update. */
+        if(OutPos+DstBufferSize < SamplesToDo)
+            DstBufferSize &= ~3;
+
+        for(chan = 0;chan < NumChannels;chan++)
+        {
+            const ALfloat *ResampledData;
+            ALfloat *SrcData = Device->SourceData;
+            ALsizei SrcDataSize;
+
+            /* Load the previous samples into the source data first. */
+            memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
+            SrcDataSize = MAX_PRE_SAMPLES;
+
+            if(Source->SourceType == AL_STATIC)
+            {
+                const ALbuffer *ALBuffer = BufferListItem->buffer;
+                const ALubyte *Data = ALBuffer->data;
+                ALsizei DataSize;
+
+                /* Offset buffer data to current channel */
+                Data += chan*SampleSize;
+
+                /* If current pos is beyond the loop range, do not loop */
+                if(!BufferLoopItem || DataPosInt >= ALBuffer->LoopEnd)
+                {
+                    BufferLoopItem = NULL;
+
+                    /* Load what's left to play from the source buffer, and
+                     * clear the rest of the temp buffer */
+                    DataSize = minu(SrcBufferSize - SrcDataSize,
+                                    ALBuffer->SampleLen - DataPosInt);
+
+                    LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
+                                NumChannels, ALBuffer->FmtType, DataSize);
+                    SrcDataSize += DataSize;
+
+                    SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
+                    SrcDataSize += SrcBufferSize - SrcDataSize;
+                }
+                else
+                {
+                    ALsizei LoopStart = ALBuffer->LoopStart;
+                    ALsizei LoopEnd   = ALBuffer->LoopEnd;
+
+                    /* Load what's left of this loop iteration, then load
+                     * repeats of the loop section */
+                    DataSize = minu(SrcBufferSize - SrcDataSize, LoopEnd - DataPosInt);
+
+                    LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
+                                NumChannels, ALBuffer->FmtType, DataSize);
+                    SrcDataSize += DataSize;
+
+                    DataSize = LoopEnd-LoopStart;
+                    while(SrcBufferSize > SrcDataSize)
+                    {
+                        DataSize = mini(SrcBufferSize - SrcDataSize, DataSize);
+
+                        LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize],
+                                    NumChannels, ALBuffer->FmtType, DataSize);
+                        SrcDataSize += DataSize;
+                    }
+                }
+            }
+            else
+            {
+                /* Crawl the buffer queue to fill in the temp buffer */
+                ALbufferlistitem *tmpiter = BufferListItem;
+                ALsizei pos = DataPosInt;
+
+                while(tmpiter && SrcBufferSize > SrcDataSize)
+                {
+                    const ALbuffer *ALBuffer;
+                    if((ALBuffer=tmpiter->buffer) != NULL)
+                    {
+                        const ALubyte *Data = ALBuffer->data;
+                        ALsizei DataSize = ALBuffer->SampleLen;
+
+                        /* Skip the data already played */
+                        if(DataSize <= pos)
+                            pos -= DataSize;
+                        else
+                        {
+                            Data += (pos*NumChannels + chan)*SampleSize;
+                            DataSize -= pos;
+                            pos -= pos;
+
+                            DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
+                            LoadSamples(&SrcData[SrcDataSize], Data, NumChannels,
+                                        ALBuffer->FmtType, DataSize);
+                            SrcDataSize += DataSize;
+                        }
+                    }
+                    tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
+                    if(!tmpiter && BufferLoopItem)
+                        tmpiter = BufferLoopItem;
+                    else if(!tmpiter)
+                    {
+                        SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
+                        SrcDataSize += SrcBufferSize - SrcDataSize;
+                    }
+                }
+            }
+
+            /* Store the last source samples used for next time. */
+            memcpy(voice->PrevSamples[chan],
+                &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
+                MAX_PRE_SAMPLES*sizeof(ALfloat)
+            );
+
+            /* Now resample, then filter and mix to the appropriate outputs. */
+            ResampledData = Resample(&voice->ResampleState,
+                &SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment,
+                Device->ResampledData, DstBufferSize
+            );
+            {
+                DirectParams *parms = &voice->Direct.Params[chan];
+                const ALfloat *samples;
+
+                samples = DoFilters(
+                    &parms->LowPass, &parms->HighPass, Device->FilteredData,
+                    ResampledData, DstBufferSize, voice->Direct.FilterType
+                );
+                if(!(voice->Flags&VOICE_HAS_HRTF))
+                {
+                    if(!Counter)
+                        memcpy(parms->Gains.Current, parms->Gains.Target,
+                               sizeof(parms->Gains.Current));
+                    if(!(voice->Flags&VOICE_HAS_NFC))
+                        MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
+                            parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
+                            DstBufferSize
+                        );
+                    else
+                    {
+                        static void (*const NfcUpdate[MAX_AMBI_ORDER])(
+                            NfcFilter*,float*,const float*,const int
+                        ) = {
+                            NfcFilterUpdate1, NfcFilterUpdate2, NfcFilterUpdate3
+                        };
+                        ALfloat *nfcsamples = Device->NFCtrlData;
+                        ALsizei ord, chanoffset = 0;
+
+                        MixSamples(samples,
+                            voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
+                            parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
+                            DstBufferSize
+                        );
+                        chanoffset += voice->Direct.ChannelsPerOrder[0];
+                        for(ord = 1;ord < MAX_AMBI_ORDER+1;ord++)
+                        {
+                            if(voice->Direct.ChannelsPerOrder[ord] <= 0)
+                                break;
+                            NfcUpdate[ord-1](&parms->NFCtrlFilter[ord-1], nfcsamples, samples,
+                                             DstBufferSize);
+                            MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[ord],
+                                voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset,
+                                parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize
+                            );
+                            chanoffset += voice->Direct.ChannelsPerOrder[ord];
+                        }
+                    }
+                }
+                else
+                {
+                    MixHrtfParams hrtfparams;
+                    ALsizei fademix = 0;
+                    int lidx, ridx;
+
+                    lidx = GetChannelIdxByName(Device->RealOut, FrontLeft);
+                    ridx = GetChannelIdxByName(Device->RealOut, FrontRight);
+                    assert(lidx != -1 && ridx != -1);
+
+                    if(!Counter)
+                    {
+                        /* No fading, just overwrite the old HRTF params. */
+                        parms->Hrtf.Old = parms->Hrtf.Target;
+                    }
+                    else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
+                    {
+                        /* The old HRTF params are silent, so overwrite the old
+                         * coefficients with the new, and reset the old gain to
+                         * 0. The future mix will then fade from silence.
+                         */
+                        parms->Hrtf.Old = parms->Hrtf.Target;
+                        parms->Hrtf.Old.Gain = 0.0f;
+                    }
+                    else if(firstpass)
+                    {
+                        ALfloat gain;
+
+                        /* Fade between the coefficients over 128 samples. */
+                        fademix = mini(DstBufferSize, 128);
+
+                        /* The new coefficients need to fade in completely
+                         * since they're replacing the old ones. To keep the
+                         * gain fading consistent, interpolate between the old
+                         * and new target gains given how much of the fade time
+                         * this mix handles.
+                         */
+                        gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
+                                    minf(1.0f, (ALfloat)fademix/Counter));
+                        hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
+                        hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
+                        hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
+                        hrtfparams.Gain = 0.0f;
+                        hrtfparams.GainStep = gain / (ALfloat)fademix;
+
+                        MixHrtfBlendSamples(
+                            voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
+                            samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
+                            &hrtfparams, &parms->Hrtf.State, fademix
+                        );
+                        /* Update the old parameters with the result. */
+                        parms->Hrtf.Old = parms->Hrtf.Target;
+                        if(fademix < Counter)
+                            parms->Hrtf.Old.Gain = hrtfparams.Gain;
+                    }
+
+                    if(fademix < DstBufferSize)
+                    {
+                        ALsizei todo = DstBufferSize - fademix;
+                        ALfloat gain = parms->Hrtf.Target.Gain;
+
+                        /* Interpolate the target gain if the gain fading lasts
+                         * longer than this mix.
+                         */
+                        if(Counter > DstBufferSize)
+                            gain = lerp(parms->Hrtf.Old.Gain, gain,
+                                        (ALfloat)todo/(Counter-fademix));
+
+                        hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
+                        hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
+                        hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
+                        hrtfparams.Gain = parms->Hrtf.Old.Gain;
+                        hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
+                        MixHrtfSamples(
+                            voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
+                            samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
+                            &hrtfparams, &parms->Hrtf.State, todo
+                        );
+                        /* Store the interpolated gain or the final target gain
+                         * depending if the fade is done.
+                         */
+                        if(DstBufferSize < Counter)
+                            parms->Hrtf.Old.Gain = gain;
+                        else
+                            parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
+                    }
+                }
+            }
+
+            for(send = 0;send < Device->NumAuxSends;send++)
+            {
+                SendParams *parms = &voice->Send[send].Params[chan];
+                const ALfloat *samples;
+
+                if(!voice->Send[send].Buffer)
+                    continue;
+
+                samples = DoFilters(
+                    &parms->LowPass, &parms->HighPass, Device->FilteredData,
+                    ResampledData, DstBufferSize, voice->Send[send].FilterType
+                );
+
+                if(!Counter)
+                    memcpy(parms->Gains.Current, parms->Gains.Target,
+                           sizeof(parms->Gains.Current));
+                MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
+                    parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
+                );
+            }
+        }
+        /* Update positions */
+        DataPosFrac += increment*DstBufferSize;
+        DataPosInt  += DataPosFrac>>FRACTIONBITS;
+        DataPosFrac &= FRACTIONMASK;
+
+        OutPos += DstBufferSize;
+        voice->Offset += DstBufferSize;
+        Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
+        firstpass = false;
+
+        /* Handle looping sources */
+        while(1)
+        {
+            const ALbuffer *ALBuffer;
+            ALsizei DataSize = 0;
+            ALsizei LoopStart = 0;
+            ALsizei LoopEnd = 0;
+
+            if((ALBuffer=BufferListItem->buffer) != NULL)
+            {
+                DataSize = ALBuffer->SampleLen;
+                LoopStart = ALBuffer->LoopStart;
+                LoopEnd = ALBuffer->LoopEnd;
+                if(LoopEnd > DataPosInt)
+                    break;
+            }
+
+            if(BufferLoopItem && Source->SourceType == AL_STATIC)
+            {
+                assert(LoopEnd > LoopStart);
+                DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
+                break;
+            }
+
+            if(DataSize > DataPosInt)
+                break;
+
+            BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
+            if(!BufferListItem)
+            {
+                BufferListItem = BufferLoopItem;
+                if(!BufferListItem)
+                {
+                    isplaying = false;
+                    DataPosInt = 0;
+                    DataPosFrac = 0;
+                    break;
+                }
+            }
+
+            DataPosInt -= DataSize;
+        }
+    } while(isplaying && OutPos < SamplesToDo);
+
+    voice->Flags |= VOICE_IS_FADING;
+
+    /* Update source info */
+    ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
+    ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
+    ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
+    return isplaying;
+}

Alc/mixer_c.c

@@ -0,0 +1,208 @@
+#include "config.h"
+
+#include <assert.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "alSource.h"
+#include "alAuxEffectSlot.h"
+
+
+static inline ALfloat point32(const ALfloat *restrict vals, ALsizei UNUSED(frac))
+{ return vals[0]; }
+static inline ALfloat lerp32(const ALfloat *restrict vals, ALsizei frac)
+{ return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); }
+static inline ALfloat fir4_32(const ALfloat *restrict vals, ALsizei frac)
+{ return resample_fir4(vals[-1], vals[0], vals[1], vals[2], frac); }
+
+
+const ALfloat *Resample_copy32_C(const InterpState* UNUSED(state),
+  const ALfloat *restrict src, ALsizei UNUSED(frac), ALint UNUSED(increment),
+  ALfloat *restrict dst, ALsizei numsamples)
+{
+#if defined(HAVE_SSE) || defined(HAVE_NEON)
+    /* Avoid copying the source data if it's aligned like the destination. */
+    if((((intptr_t)src)&15) == (((intptr_t)dst)&15))
+        return src;
+#endif
+    memcpy(dst, src, numsamples*sizeof(ALfloat));
+    return dst;
+}
+
+#define DECL_TEMPLATE(Sampler)                                                \
+const ALfloat *Resample_##Sampler##_C(const InterpState* UNUSED(state),       \
+  const ALfloat *restrict src, ALsizei frac, ALint increment,                 \
+  ALfloat *restrict dst, ALsizei numsamples)                                  \
+{                                                                             \
+    ALsizei i;                                                                \
+    for(i = 0;i < numsamples;i++)                                             \
+    {                                                                         \
+        dst[i] = Sampler(src, frac);                                          \
+                                                                              \
+        frac += increment;                                                    \
+        src  += frac>>FRACTIONBITS;                                           \
+        frac &= FRACTIONMASK;                                                 \
+    }                                                                         \
+    return dst;                                                               \
+}
+
+DECL_TEMPLATE(point32)
+DECL_TEMPLATE(lerp32)
+DECL_TEMPLATE(fir4_32)
+
+#undef DECL_TEMPLATE
+
+const ALfloat *Resample_bsinc32_C(const InterpState *state, const ALfloat *restrict src,
+                                  ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                  ALsizei dstlen)
+{
+    const ALfloat *fil, *scd, *phd, *spd;
+    const ALfloat sf = state->bsinc.sf;
+    const ALsizei m = state->bsinc.m;
+    ALsizei j_f, pi, i;
+    ALfloat pf, r;
+
+    src += state->bsinc.l;
+    for(i = 0;i < dstlen;i++)
+    {
+        // Calculate the phase index and factor.
+#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
+        pi = frac >> FRAC_PHASE_BITDIFF;
+        pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
+#undef FRAC_PHASE_BITDIFF
+
+        fil = ASSUME_ALIGNED(state->bsinc.coeffs[pi].filter, 16);
+        scd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].scDelta, 16);
+        phd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].phDelta, 16);
+        spd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].spDelta, 16);
+
+        // Apply the scale and phase interpolated filter.
+        r = 0.0f;
+        for(j_f = 0;j_f < m;j_f++)
+            r += (fil[j_f] + sf*scd[j_f] + pf*(phd[j_f] + sf*spd[j_f])) * src[j_f];
+        dst[i] = r;
+
+        frac += increment;
+        src  += frac>>FRACTIONBITS;
+        frac &= FRACTIONMASK;
+    }
+    return dst;
+}
+
+
+void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples)
+{
+    ALsizei i;
+    if(numsamples > 1)
+    {
+        dst[0] = filter->b0 * src[0] +
+                 filter->b1 * filter->x[0] +
+                 filter->b2 * filter->x[1] -
+                 filter->a1 * filter->y[0] -
+                 filter->a2 * filter->y[1];
+        dst[1] = filter->b0 * src[1] +
+                 filter->b1 * src[0] +
+                 filter->b2 * filter->x[0] -
+                 filter->a1 * dst[0] -
+                 filter->a2 * filter->y[0];
+        for(i = 2;i < numsamples;i++)
+            dst[i] = filter->b0 * src[i] +
+                     filter->b1 * src[i-1] +
+                     filter->b2 * src[i-2] -
+                     filter->a1 * dst[i-1] -
+                     filter->a2 * dst[i-2];
+        filter->x[0] = src[i-1];
+        filter->x[1] = src[i-2];
+        filter->y[0] = dst[i-1];
+        filter->y[1] = dst[i-2];
+    }
+    else if(numsamples == 1)
+    {
+        dst[0] = filter->b0 * src[0] +
+                 filter->b1 * filter->x[0] +
+                 filter->b2 * filter->x[1] -
+                 filter->a1 * filter->y[0] -
+                 filter->a2 * filter->y[1];
+        filter->x[1] = filter->x[0];
+        filter->x[0] = src[0];
+        filter->y[1] = filter->y[0];
+        filter->y[0] = dst[0];
+    }
+}
+
+
+static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
+                               const ALsizei IrSize,
+                               const ALfloat (*restrict Coeffs)[2],
+                               ALfloat left, ALfloat right)
+{
+    ALsizei c;
+    for(c = 0;c < IrSize;c++)
+    {
+        const ALsizei off = (Offset+c)&HRIR_MASK;
+        Values[off][0] += Coeffs[c][0] * left;
+        Values[off][1] += Coeffs[c][1] * right;
+    }
+}
+
+#define MixHrtf MixHrtf_C
+#define MixHrtfBlend MixHrtfBlend_C
+#define MixDirectHrtf MixDirectHrtf_C
+#include "mixer_inc.c"
+#undef MixHrtf
+
+
+void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+           ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+           ALsizei BufferSize)
+{
+    ALfloat gain, delta, step;
+    ALsizei c;
+
+    delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
+
+    for(c = 0;c < OutChans;c++)
+    {
+        ALsizei pos = 0;
+        gain = CurrentGains[c];
+        step = (TargetGains[c] - gain) * delta;
+        if(fabsf(step) > FLT_EPSILON)
+        {
+            ALsizei minsize = mini(BufferSize, Counter);
+            for(;pos < minsize;pos++)
+            {
+                OutBuffer[c][OutPos+pos] += data[pos]*gain;
+                gain += step;
+            }
+            if(pos == Counter)
+                gain = TargetGains[c];
+            CurrentGains[c] = gain;
+        }
+
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+        for(;pos < BufferSize;pos++)
+            OutBuffer[c][OutPos+pos] += data[pos]*gain;
+    }
+}
+
+/* Basically the inverse of the above. Rather than one input going to multiple
+ * outputs (each with its own gain), it's multiple inputs (each with its own
+ * gain) going to one output. This applies one row (vs one column) of a matrix
+ * transform. And as the matrices are more or less static once set up, no
+ * stepping is necessary.
+ */
+void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
+{
+    ALsizei c, i;
+
+    for(c = 0;c < InChans;c++)
+    {
+        ALfloat gain = Gains[c];
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+
+        for(i = 0;i < BufferSize;i++)
+            OutBuffer[i] += data[c][InPos+i] * gain;
+    }
+}

Alc/mixer_defs.h

@@ -0,0 +1,129 @@
+#ifndef MIXER_DEFS_H
+#define MIXER_DEFS_H
+
+#include "AL/alc.h"
+#include "AL/al.h"
+#include "alMain.h"
+#include "alu.h"
+
+struct MixGains;
+
+struct MixHrtfParams;
+struct HrtfState;
+
+/* C resamplers */
+const ALfloat *Resample_copy32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_point32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_lerp32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_fir4_32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_bsinc32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+
+
+/* C mixers */
+void MixHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+               const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+               const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
+               struct HrtfState *hrtfstate, ALsizei BufferSize);
+void MixHrtfBlend_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                    const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                    const ALsizei IrSize, const HrtfParams *oldparams,
+                    MixHrtfParams *newparams, HrtfState *hrtfstate,
+                    ALsizei BufferSize);
+void MixDirectHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                     const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
+                     const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
+                     ALsizei BufferSize);
+void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+           ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+           ALsizei BufferSize);
+void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains,
+              const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
+              ALsizei InPos, ALsizei BufferSize);
+
+/* SSE mixers */
+void MixHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                 const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                 const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
+                 struct HrtfState *hrtfstate, ALsizei BufferSize);
+void MixHrtfBlend_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                      const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                      const ALsizei IrSize, const HrtfParams *oldparams,
+                      MixHrtfParams *newparams, HrtfState *hrtfstate,
+                      ALsizei BufferSize);
+void MixDirectHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                       const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
+                       const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
+                       ALsizei BufferSize);
+void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+             ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+             ALsizei BufferSize);
+void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains,
+                const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
+                ALsizei InPos, ALsizei BufferSize);
+
+/* SSE resamplers */
+inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALint *restrict pos_arr, ALsizei size)
+{
+    ALsizei i;
+
+    pos_arr[0] = 0;
+    frac_arr[0] = frac;
+    for(i = 1;i < size;i++)
+    {
+        ALint frac_tmp = frac_arr[i-1] + increment;
+        pos_arr[i] = pos_arr[i-1] + (frac_tmp>>FRACTIONBITS);
+        frac_arr[i] = frac_tmp&FRACTIONMASK;
+    }
+}
+
+const ALfloat *Resample_lerp32_SSE2(const InterpState *state, const ALfloat *restrict src,
+                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                    ALsizei numsamples);
+const ALfloat *Resample_lerp32_SSE41(const InterpState *state, const ALfloat *restrict src,
+                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                     ALsizei numsamples);
+
+const ALfloat *Resample_fir4_32_SSE3(const InterpState *state, const ALfloat *restrict src,
+                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                     ALsizei numsamples);
+const ALfloat *Resample_fir4_32_SSE41(const InterpState *state, const ALfloat *restrict src,
+                                      ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                      ALsizei numsamples);
+
+const ALfloat *Resample_bsinc32_SSE(const InterpState *state, const ALfloat *restrict src,
+                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                    ALsizei dstlen);
+
+/* Neon mixers */
+void MixHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                  const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                  const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
+                  struct HrtfState *hrtfstate, ALsizei BufferSize);
+void MixHrtfBlend_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                       const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                       const ALsizei IrSize, const HrtfParams *oldparams,
+                       MixHrtfParams *newparams, HrtfState *hrtfstate,
+                       ALsizei BufferSize);
+void MixDirectHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                        const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
+                        const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
+                        ALsizei BufferSize);
+void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+              ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+              ALsizei BufferSize);
+void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains,
+                 const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
+                 ALsizei InPos, ALsizei BufferSize);
+
+/* Neon resamplers */
+const ALfloat *Resample_lerp32_Neon(const InterpState *state, const ALfloat *restrict src,
+                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                    ALsizei numsamples);
+const ALfloat *Resample_fir4_32_Neon(const InterpState *state, const ALfloat *restrict src,
+                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                     ALsizei numsamples);
+const ALfloat *Resample_bsinc32_Neon(const InterpState *state, const ALfloat *restrict src,
+                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                     ALsizei dstlen);
+
+#endif /* MIXER_DEFS_H */

Alc/mixer_inc.c

@@ -0,0 +1,114 @@
+#include "config.h"
+
+#include "alMain.h"
+#include "alSource.h"
+
+#include "hrtf.h"
+#include "mixer_defs.h"
+#include "align.h"
+#include "alu.h"
+
+
+static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
+                               const ALsizei irSize,
+                               const ALfloat (*restrict Coeffs)[2],
+                               ALfloat left, ALfloat right);
+
+
+void MixHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+             const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+             const ALsizei IrSize, MixHrtfParams *hrtfparams, HrtfState *hrtfstate,
+             ALsizei BufferSize)
+{
+    const ALfloat (*Coeffs)[2] = ASSUME_ALIGNED(hrtfparams->Coeffs, 16);
+    const ALsizei Delay[2] = { hrtfparams->Delay[0], hrtfparams->Delay[1] };
+    ALfloat gainstep = hrtfparams->GainStep;
+    ALfloat gain = hrtfparams->Gain;
+    ALfloat left, right;
+    ALsizei i;
+
+    LeftOut  += OutPos;
+    RightOut += OutPos;
+    for(i = 0;i < BufferSize;i++)
+    {
+        hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++);
+        left = hrtfstate->History[(Offset-Delay[0])&HRTF_HISTORY_MASK]*gain;
+        right = hrtfstate->History[(Offset-Delay[1])&HRTF_HISTORY_MASK]*gain;
+
+        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f;
+        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f;
+
+        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, Coeffs, left, right);
+        *(LeftOut++)  += hrtfstate->Values[Offset&HRIR_MASK][0];
+        *(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1];
+
+        gain += gainstep;
+        Offset++;
+    }
+    hrtfparams->Gain = gain;
+}
+
+void MixHrtfBlend(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                  const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                  const ALsizei IrSize, const HrtfParams *oldparams,
+                  MixHrtfParams *newparams, HrtfState *hrtfstate,
+                  ALsizei BufferSize)
+{
+    const ALfloat (*OldCoeffs)[2] = ASSUME_ALIGNED(oldparams->Coeffs, 16);
+    const ALsizei OldDelay[2] = { oldparams->Delay[0], oldparams->Delay[1] };
+    ALfloat oldGain = oldparams->Gain;
+    ALfloat oldGainStep = -oldGain / (ALfloat)BufferSize;
+    const ALfloat (*NewCoeffs)[2] = ASSUME_ALIGNED(newparams->Coeffs, 16);
+    const ALsizei NewDelay[2] = { newparams->Delay[0], newparams->Delay[1] };
+    ALfloat newGain = newparams->Gain;
+    ALfloat newGainStep = newparams->GainStep;
+    ALfloat left, right;
+    ALsizei i;
+
+    LeftOut  += OutPos;
+    RightOut += OutPos;
+    for(i = 0;i < BufferSize;i++)
+    {
+        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f;
+        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f;
+
+        hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++);
+
+        left = hrtfstate->History[(Offset-OldDelay[0])&HRTF_HISTORY_MASK]*oldGain;
+        right = hrtfstate->History[(Offset-OldDelay[1])&HRTF_HISTORY_MASK]*oldGain;
+        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, OldCoeffs, left, right);
+
+        left = hrtfstate->History[(Offset-NewDelay[0])&HRTF_HISTORY_MASK]*newGain;
+        right = hrtfstate->History[(Offset-NewDelay[1])&HRTF_HISTORY_MASK]*newGain;
+        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, NewCoeffs, left, right);
+
+        *(LeftOut++)  += hrtfstate->Values[Offset&HRIR_MASK][0];
+        *(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1];
+
+        oldGain += oldGainStep;
+        newGain += newGainStep;
+        Offset++;
+    }
+    newparams->Gain = newGain;
+}
+
+void MixDirectHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                   const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
+                   const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
+                   ALsizei BufferSize)
+{
+    ALfloat insample;
+    ALsizei i;
+
+    for(i = 0;i < BufferSize;i++)
+    {
+        Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f;
+        Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f;
+        Offset++;
+
+        insample = *(data++);
+        ApplyCoeffs(Offset, Values, IrSize, Coeffs, insample, insample);
+        *(LeftOut++)  += Values[Offset&HRIR_MASK][0];
+        *(RightOut++) += Values[Offset&HRIR_MASK][1];
+    }
+}

Alc/mixer_neon.c

@@ -0,0 +1,331 @@
+#include "config.h"
+
+#include <arm_neon.h>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "alMain.h"
+#include "alu.h"
+#include "hrtf.h"
+#include "mixer_defs.h"
+
+
+const ALfloat *Resample_lerp32_Neon(const InterpState* UNUSED(state),
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei numsamples)
+{
+    const int32x4_t increment4 = vdupq_n_s32(increment*4);
+    const float32x4_t fracOne4 = vdupq_n_f32(1.0f/FRACTIONONE);
+    const int32x4_t fracMask4 = vdupq_n_s32(FRACTIONMASK);
+    alignas(16) ALint pos_[4];
+    alignas(16) ALsizei frac_[4];
+    int32x4_t pos4;
+    int32x4_t frac4;
+    ALsizei i;
+
+    InitiatePositionArrays(frac, increment, frac_, pos_, 4);
+
+    frac4 = vld1q_s32(frac_);
+    pos4 = vld1q_s32(pos_);
+
+    for(i = 0;numsamples-i > 3;i += 4)
+    {
+        const float32x4_t val1 = (float32x4_t){src[pos_[0]], src[pos_[1]], src[pos_[2]], src[pos_[3]]};
+        const float32x4_t val2 = (float32x4_t){src[pos_[0]+1], src[pos_[1]+1], src[pos_[2]+1], src[pos_[3]+1]};
+
+        /* val1 + (val2-val1)*mu */
+        const float32x4_t r0 = vsubq_f32(val2, val1);
+        const float32x4_t mu = vmulq_f32(vcvtq_f32_s32(frac4), fracOne4);
+        const float32x4_t out = vmlaq_f32(val1, mu, r0);
+
+        vst1q_f32(&dst[i], out);
+
+        frac4 = vaddq_s32(frac4, increment4);
+        pos4 = vaddq_s32(pos4, vshrq_n_s32(frac4, FRACTIONBITS));
+        frac4 = vandq_s32(frac4, fracMask4);
+
+        vst1q_s32(pos_, pos4);
+    }
+
+    if(i < numsamples)
+    {
+        /* NOTE: These four elements represent the position *after* the last
+         * four samples, so the lowest element is the next position to
+         * resample.
+         */
+        ALint pos = pos_[0];
+        frac = vgetq_lane_s32(frac4, 0);
+        do {
+            dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
+
+            frac += increment;
+            pos  += frac>>FRACTIONBITS;
+            frac &= FRACTIONMASK;
+        } while(++i < numsamples);
+    }
+    return dst;
+}
+
+const ALfloat *Resample_fir4_32_Neon(const InterpState* UNUSED(state),
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei numsamples)
+{
+    const int32x4_t increment4 = vdupq_n_s32(increment*4);
+    const int32x4_t fracMask4 = vdupq_n_s32(FRACTIONMASK);
+    alignas(16) ALint pos_[4];
+    alignas(16) ALsizei frac_[4];
+    int32x4_t pos4;
+    int32x4_t frac4;
+    ALsizei i;
+
+    InitiatePositionArrays(frac, increment, frac_, pos_, 4);
+
+    frac4 = vld1q_s32(frac_);
+    pos4 = vld1q_s32(pos_);
+
+    --src;
+    for(i = 0;numsamples-i > 3;i += 4)
+    {
+        const float32x4_t val0 = vld1q_f32(&src[pos_[0]]);
+        const float32x4_t val1 = vld1q_f32(&src[pos_[1]]);
+        const float32x4_t val2 = vld1q_f32(&src[pos_[2]]);
+        const float32x4_t val3 = vld1q_f32(&src[pos_[3]]);
+        float32x4_t k0 = vld1q_f32(sinc4Tab[frac_[0]]);
+        float32x4_t k1 = vld1q_f32(sinc4Tab[frac_[1]]);
+        float32x4_t k2 = vld1q_f32(sinc4Tab[frac_[2]]);
+        float32x4_t k3 = vld1q_f32(sinc4Tab[frac_[3]]);
+        float32x4_t out;
+
+        k0 = vmulq_f32(k0, val0);
+        k1 = vmulq_f32(k1, val1);
+        k2 = vmulq_f32(k2, val2);
+        k3 = vmulq_f32(k3, val3);
+        k0 = vcombine_f32(vpadd_f32(vget_low_f32(k0), vget_high_f32(k0)),
+                          vpadd_f32(vget_low_f32(k1), vget_high_f32(k1)));
+        k2 = vcombine_f32(vpadd_f32(vget_low_f32(k2), vget_high_f32(k2)),
+                          vpadd_f32(vget_low_f32(k3), vget_high_f32(k3)));
+        out = vcombine_f32(vpadd_f32(vget_low_f32(k0), vget_high_f32(k0)),
+                           vpadd_f32(vget_low_f32(k2), vget_high_f32(k2)));
+
+        vst1q_f32(&dst[i], out);
+
+        frac4 = vaddq_s32(frac4, increment4);
+        pos4 = vaddq_s32(pos4, vshrq_n_s32(frac4, FRACTIONBITS));
+        frac4 = vandq_s32(frac4, fracMask4);
+
+        vst1q_s32(pos_, pos4);
+        vst1q_s32(frac_, frac4);
+    }
+
+    if(i < numsamples)
+    {
+        /* NOTE: These four elements represent the position *after* the last
+         * four samples, so the lowest element is the next position to
+         * resample.
+         */
+        ALint pos = pos_[0];
+        frac = frac_[0];
+        do {
+            dst[i] = resample_fir4(src[pos], src[pos+1], src[pos+2], src[pos+3], frac);
+
+            frac += increment;
+            pos  += frac>>FRACTIONBITS;
+            frac &= FRACTIONMASK;
+        } while(++i < numsamples);
+    }
+    return dst;
+}
+
+const ALfloat *Resample_bsinc32_Neon(const InterpState *state,
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei dstlen)
+{
+    const float32x4_t sf4 = vdupq_n_f32(state->bsinc.sf);
+    const ALsizei m = state->bsinc.m;
+    const ALfloat *fil, *scd, *phd, *spd;
+    ALsizei pi, i, j;
+    float32x4_t r4;
+    ALfloat pf;
+
+    src += state->bsinc.l;
+    for(i = 0;i < dstlen;i++)
+    {
+        // Calculate the phase index and factor.
+#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
+        pi = frac >> FRAC_PHASE_BITDIFF;
+        pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
+#undef FRAC_PHASE_BITDIFF
+
+        fil = ASSUME_ALIGNED(state->bsinc.coeffs[pi].filter, 16);
+        scd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].scDelta, 16);
+        phd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].phDelta, 16);
+        spd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].spDelta, 16);
+
+        // Apply the scale and phase interpolated filter.
+        r4 = vdupq_n_f32(0.0f);
+        {
+            const float32x4_t pf4 = vdupq_n_f32(pf);
+            for(j = 0;j < m;j+=4)
+            {
+                /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
+                const float32x4_t f4 = vmlaq_f32(vmlaq_f32(vld1q_f32(&fil[j]),
+                                                           sf4, vld1q_f32(&scd[j])),
+                    pf4, vmlaq_f32(vld1q_f32(&phd[j]),
+                        sf4, vld1q_f32(&spd[j])
+                    )
+                );
+                /* r += f*src */
+                r4 = vmlaq_f32(r4, f4, vld1q_f32(&src[j]));
+            }
+        }
+        r4 = vaddq_f32(r4, vcombine_f32(vrev64_f32(vget_high_f32(r4)),
+                                        vrev64_f32(vget_low_f32(r4))));
+        dst[i] = vget_lane_f32(vadd_f32(vget_low_f32(r4), vget_high_f32(r4)), 0);
+
+        frac += increment;
+        src  += frac>>FRACTIONBITS;
+        frac &= FRACTIONMASK;
+    }
+    return dst;
+}
+
+
+static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
+                               const ALsizei IrSize,
+                               const ALfloat (*restrict Coeffs)[2],
+                               ALfloat left, ALfloat right)
+{
+    ALsizei c;
+    float32x4_t leftright4;
+    {
+        float32x2_t leftright2 = vdup_n_f32(0.0);
+        leftright2 = vset_lane_f32(left, leftright2, 0);
+        leftright2 = vset_lane_f32(right, leftright2, 1);
+        leftright4 = vcombine_f32(leftright2, leftright2);
+    }
+    Values = ASSUME_ALIGNED(Values, 16);
+    Coeffs = ASSUME_ALIGNED(Coeffs, 16);
+    for(c = 0;c < IrSize;c += 2)
+    {
+        const ALsizei o0 = (Offset+c)&HRIR_MASK;
+        const ALsizei o1 = (o0+1)&HRIR_MASK;
+        float32x4_t vals = vcombine_f32(vld1_f32((float32_t*)&Values[o0][0]),
+                                        vld1_f32((float32_t*)&Values[o1][0]));
+        float32x4_t coefs = vld1q_f32((float32_t*)&Coeffs[c][0]);
+
+        vals = vmlaq_f32(vals, coefs, leftright4);
+
+        vst1_f32((float32_t*)&Values[o0][0], vget_low_f32(vals));
+        vst1_f32((float32_t*)&Values[o1][0], vget_high_f32(vals));
+    }
+}
+
+#define MixHrtf MixHrtf_Neon
+#define MixHrtfBlend MixHrtfBlend_Neon
+#define MixDirectHrtf MixDirectHrtf_Neon
+#include "mixer_inc.c"
+#undef MixHrtf
+
+
+void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+              ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+              ALsizei BufferSize)
+{
+    ALfloat gain, delta, step;
+    float32x4_t gain4;
+    ALsizei c;
+
+    data = ASSUME_ALIGNED(data, 16);
+    OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
+
+    delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
+
+    for(c = 0;c < OutChans;c++)
+    {
+        ALsizei pos = 0;
+        gain = CurrentGains[c];
+        step = (TargetGains[c] - gain) * delta;
+        if(fabsf(step) > FLT_EPSILON)
+        {
+            ALsizei minsize = mini(BufferSize, Counter);
+            /* Mix with applying gain steps in aligned multiples of 4. */
+            if(minsize-pos > 3)
+            {
+                float32x4_t step4;
+                gain4 = vsetq_lane_f32(gain, gain4, 0);
+                gain4 = vsetq_lane_f32(gain + step, gain4, 1);
+                gain4 = vsetq_lane_f32(gain + step + step, gain4, 2);
+                gain4 = vsetq_lane_f32(gain + step + step + step, gain4, 3);
+                step4 = vdupq_n_f32(step + step + step + step);
+                do {
+                    const float32x4_t val4 = vld1q_f32(&data[pos]);
+                    float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]);
+                    dry4 = vmlaq_f32(dry4, val4, gain4);
+                    gain4 = vaddq_f32(gain4, step4);
+                    vst1q_f32(&OutBuffer[c][OutPos+pos], dry4);
+                    pos += 4;
+                } while(minsize-pos > 3);
+                /* NOTE: gain4 now represents the next four gains after the
+                 * last four mixed samples, so the lowest element represents
+                 * the next gain to apply.
+                 */
+                gain = vgetq_lane_f32(gain4, 0);
+            }
+            /* Mix with applying left over gain steps that aren't aligned multiples of 4. */
+            for(;pos < minsize;pos++)
+            {
+                OutBuffer[c][OutPos+pos] += data[pos]*gain;
+                gain += step;
+            }
+            if(pos == Counter)
+                gain = TargetGains[c];
+            CurrentGains[c] = gain;
+
+            /* Mix until pos is aligned with 4 or the mix is done. */
+            minsize = mini(BufferSize, (pos+3)&~3);
+            for(;pos < minsize;pos++)
+                OutBuffer[c][OutPos+pos] += data[pos]*gain;
+        }
+
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+        gain4 = vdupq_n_f32(gain);
+        for(;BufferSize-pos > 3;pos += 4)
+        {
+            const float32x4_t val4 = vld1q_f32(&data[pos]);
+            float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]);
+            dry4 = vmlaq_f32(dry4, val4, gain4);
+            vst1q_f32(&OutBuffer[c][OutPos+pos], dry4);
+        }
+        for(;pos < BufferSize;pos++)
+            OutBuffer[c][OutPos+pos] += data[pos]*gain;
+    }
+}
+
+void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
+{
+    float32x4_t gain4;
+    ALsizei c;
+
+    data = ASSUME_ALIGNED(data, 16);
+    OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
+
+    for(c = 0;c < InChans;c++)
+    {
+        ALsizei pos = 0;
+        ALfloat gain = Gains[c];
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+
+        gain4 = vdupq_n_f32(gain);
+        for(;BufferSize-pos > 3;pos += 4)
+        {
+            const float32x4_t val4 = vld1q_f32(&data[c][InPos+pos]);
+            float32x4_t dry4 = vld1q_f32(&OutBuffer[pos]);
+            dry4 = vmlaq_f32(dry4, val4, gain4);
+            vst1q_f32(&OutBuffer[pos], dry4);
+        }
+        for(;pos < BufferSize;pos++)
+            OutBuffer[pos] += data[c][InPos+pos]*gain;
+    }
+}

Alc/mixer_sse.c

@@ -0,0 +1,230 @@
+#include "config.h"
+
+#include <xmmintrin.h>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "alMain.h"
+#include "alu.h"
+
+#include "alSource.h"
+#include "alAuxEffectSlot.h"
+#include "mixer_defs.h"
+
+
+const ALfloat *Resample_bsinc32_SSE(const InterpState *state, const ALfloat *restrict src,
+                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                    ALsizei dstlen)
+{
+    const __m128 sf4 = _mm_set1_ps(state->bsinc.sf);
+    const ALsizei m = state->bsinc.m;
+    const ALfloat *fil, *scd, *phd, *spd;
+    ALsizei pi, i, j;
+    ALfloat pf;
+    __m128 r4;
+
+    src += state->bsinc.l;
+    for(i = 0;i < dstlen;i++)
+    {
+        // Calculate the phase index and factor.
+#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
+        pi = frac >> FRAC_PHASE_BITDIFF;
+        pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
+#undef FRAC_PHASE_BITDIFF
+
+        fil = ASSUME_ALIGNED(state->bsinc.coeffs[pi].filter, 16);
+        scd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].scDelta, 16);
+        phd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].phDelta, 16);
+        spd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].spDelta, 16);
+
+        // Apply the scale and phase interpolated filter.
+        r4 = _mm_setzero_ps();
+        {
+            const __m128 pf4 = _mm_set1_ps(pf);
+#define LD4(x) _mm_load_ps(x)
+#define ULD4(x) _mm_loadu_ps(x)
+#define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z))
+            for(j = 0;j < m;j+=4)
+            {
+                /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
+                const __m128 f4 = MLA4(MLA4(LD4(&fil[j]), sf4, LD4(&scd[j])),
+                    pf4, MLA4(LD4(&phd[j]), sf4, LD4(&spd[j]))
+                );
+                /* r += f*src */
+                r4 = MLA4(r4, f4, ULD4(&src[j]));
+            }
+#undef MLA4
+#undef ULD4
+#undef LD4
+        }
+        r4 = _mm_add_ps(r4, _mm_shuffle_ps(r4, r4, _MM_SHUFFLE(0, 1, 2, 3)));
+        r4 = _mm_add_ps(r4, _mm_movehl_ps(r4, r4));
+        dst[i] = _mm_cvtss_f32(r4);
+
+        frac += increment;
+        src  += frac>>FRACTIONBITS;
+        frac &= FRACTIONMASK;
+    }
+    return dst;
+}
+
+
+static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
+                               const ALsizei IrSize,
+                               const ALfloat (*restrict Coeffs)[2],
+                               ALfloat left, ALfloat right)
+{
+    const __m128 lrlr = _mm_setr_ps(left, right, left, right);
+    __m128 vals = _mm_setzero_ps();
+    __m128 coeffs;
+    ALsizei i;
+
+    Values = ASSUME_ALIGNED(Values, 16);
+    Coeffs = ASSUME_ALIGNED(Coeffs, 16);
+    if((Offset&1))
+    {
+        const ALsizei o0 = Offset&HRIR_MASK;
+        const ALsizei o1 = (Offset+IrSize-1)&HRIR_MASK;
+        __m128 imp0, imp1;
+
+        coeffs = _mm_load_ps(&Coeffs[0][0]);
+        vals = _mm_loadl_pi(vals, (__m64*)&Values[o0][0]);
+        imp0 = _mm_mul_ps(lrlr, coeffs);
+        vals = _mm_add_ps(imp0, vals);
+        _mm_storel_pi((__m64*)&Values[o0][0], vals);
+        for(i = 1;i < IrSize-1;i += 2)
+        {
+            const ALsizei o2 = (Offset+i)&HRIR_MASK;
+
+            coeffs = _mm_load_ps(&Coeffs[i+1][0]);
+            vals = _mm_load_ps(&Values[o2][0]);
+            imp1 = _mm_mul_ps(lrlr, coeffs);
+            imp0 = _mm_shuffle_ps(imp0, imp1, _MM_SHUFFLE(1, 0, 3, 2));
+            vals = _mm_add_ps(imp0, vals);
+            _mm_store_ps(&Values[o2][0], vals);
+            imp0 = imp1;
+        }
+        vals = _mm_loadl_pi(vals, (__m64*)&Values[o1][0]);
+        imp0 = _mm_movehl_ps(imp0, imp0);
+        vals = _mm_add_ps(imp0, vals);
+        _mm_storel_pi((__m64*)&Values[o1][0], vals);
+    }
+    else
+    {
+        for(i = 0;i < IrSize;i += 2)
+        {
+            const ALsizei o = (Offset + i)&HRIR_MASK;
+
+            coeffs = _mm_load_ps(&Coeffs[i][0]);
+            vals = _mm_load_ps(&Values[o][0]);
+            vals = _mm_add_ps(vals, _mm_mul_ps(lrlr, coeffs));
+            _mm_store_ps(&Values[o][0], vals);
+        }
+    }
+}
+
+#define MixHrtf MixHrtf_SSE
+#define MixHrtfBlend MixHrtfBlend_SSE
+#define MixDirectHrtf MixDirectHrtf_SSE
+#include "mixer_inc.c"
+#undef MixHrtf
+
+
+void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+             ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+             ALsizei BufferSize)
+{
+    ALfloat gain, delta, step;
+    __m128 gain4;
+    ALsizei c;
+
+    delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
+
+    for(c = 0;c < OutChans;c++)
+    {
+        ALsizei pos = 0;
+        gain = CurrentGains[c];
+        step = (TargetGains[c] - gain) * delta;
+        if(fabsf(step) > FLT_EPSILON)
+        {
+            ALsizei minsize = mini(BufferSize, Counter);
+            /* Mix with applying gain steps in aligned multiples of 4. */
+            if(minsize-pos > 3)
+            {
+                __m128 step4;
+                gain4 = _mm_setr_ps(
+                    gain,
+                    gain + step,
+                    gain + step + step,
+                    gain + step + step + step
+                );
+                step4 = _mm_set1_ps(step + step + step + step);
+                do {
+                    const __m128 val4 = _mm_load_ps(&data[pos]);
+                    __m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]);
+                    dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
+                    gain4 = _mm_add_ps(gain4, step4);
+                    _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
+                    pos += 4;
+                } while(minsize-pos > 3);
+                /* NOTE: gain4 now represents the next four gains after the
+                 * last four mixed samples, so the lowest element represents
+                 * the next gain to apply.
+                 */
+                gain = _mm_cvtss_f32(gain4);
+            }
+            /* Mix with applying left over gain steps that aren't aligned multiples of 4. */
+            for(;pos < minsize;pos++)
+            {
+                OutBuffer[c][OutPos+pos] += data[pos]*gain;
+                gain += step;
+            }
+            if(pos == Counter)
+                gain = TargetGains[c];
+            CurrentGains[c] = gain;
+
+            /* Mix until pos is aligned with 4 or the mix is done. */
+            minsize = mini(BufferSize, (pos+3)&~3);
+            for(;pos < minsize;pos++)
+                OutBuffer[c][OutPos+pos] += data[pos]*gain;
+        }
+
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+        gain4 = _mm_set1_ps(gain);
+        for(;BufferSize-pos > 3;pos += 4)
+        {
+            const __m128 val4 = _mm_load_ps(&data[pos]);
+            __m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]);
+            dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
+            _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
+        }
+        for(;pos < BufferSize;pos++)
+            OutBuffer[c][OutPos+pos] += data[pos]*gain;
+    }
+}
+
+void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
+{
+    __m128 gain4;
+    ALsizei c;
+
+    for(c = 0;c < InChans;c++)
+    {
+        ALsizei pos = 0;
+        ALfloat gain = Gains[c];
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+
+        gain4 = _mm_set1_ps(gain);
+        for(;BufferSize-pos > 3;pos += 4)
+        {
+            const __m128 val4 = _mm_load_ps(&data[c][InPos+pos]);
+            __m128 dry4 = _mm_load_ps(&OutBuffer[pos]);
+            dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
+            _mm_store_ps(&OutBuffer[pos], dry4);
+        }
+        for(;pos < BufferSize;pos++)
+            OutBuffer[pos] += data[c][InPos+pos]*gain;
+    }
+}

Alc/mixer_sse2.c

@@ -0,0 +1,82 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <xmmintrin.h>
+#include <emmintrin.h>
+
+#include "alu.h"
+#include "mixer_defs.h"
+
+
+const ALfloat *Resample_lerp32_SSE2(const InterpState* UNUSED(state),
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei numsamples)
+{
+    const __m128i increment4 = _mm_set1_epi32(increment*4);
+    const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE);
+    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
+    union { alignas(16) ALint i[4]; float f[4]; } pos_;
+    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
+    __m128i frac4, pos4;
+    ALint pos;
+    ALsizei i;
+
+    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
+
+    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
+    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
+
+    for(i = 0;numsamples-i > 3;i += 4)
+    {
+        const __m128 val1 = _mm_setr_ps(src[pos_.i[0]], src[pos_.i[1]], src[pos_.i[2]], src[pos_.i[3]]);
+        const __m128 val2 = _mm_setr_ps(src[pos_.i[0]+1], src[pos_.i[1]+1], src[pos_.i[2]+1], src[pos_.i[3]+1]);
+
+        /* val1 + (val2-val1)*mu */
+        const __m128 r0 = _mm_sub_ps(val2, val1);
+        const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4);
+        const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0));
+
+        _mm_store_ps(&dst[i], out);
+
+        frac4 = _mm_add_epi32(frac4, increment4);
+        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
+        frac4 = _mm_and_si128(frac4, fracMask4);
+
+        _mm_store_ps(pos_.f, _mm_castsi128_ps(pos4));
+    }
+
+    /* NOTE: These four elements represent the position *after* the last four
+     * samples, so the lowest element is the next position to resample.
+     */
+    pos = pos_.i[0];
+    frac = _mm_cvtsi128_si32(frac4);
+
+    for(;i < numsamples;i++)
+    {
+        dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
+
+        frac += increment;
+        pos  += frac>>FRACTIONBITS;
+        frac &= FRACTIONMASK;
+    }
+    return dst;
+}

Alc/mixer_sse3.c

@@ -0,0 +1,97 @@
+/**
+ * OpenAL cross platform audio library, SSE3 mixer functions
+ *
+ * Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
+ * Copyright (C) 2015 by Chris Robinson <chris.kcat@gmail.com>.
+ *
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <xmmintrin.h>
+#include <emmintrin.h>
+#include <pmmintrin.h>
+
+#include "alu.h"
+#include "mixer_defs.h"
+
+
+const ALfloat *Resample_fir4_32_SSE3(const InterpState* UNUSED(state),
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei numsamples)
+{
+    const __m128i increment4 = _mm_set1_epi32(increment*4);
+    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
+    union { alignas(16) ALint i[4]; float f[4]; } pos_;
+    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
+    __m128i frac4, pos4;
+    ALint pos;
+    ALsizei i;
+
+    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
+
+    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
+    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
+
+    --src;
+    for(i = 0;numsamples-i > 3;i += 4)
+    {
+        const __m128 val0 = _mm_loadu_ps(&src[pos_.i[0]]);
+        const __m128 val1 = _mm_loadu_ps(&src[pos_.i[1]]);
+        const __m128 val2 = _mm_loadu_ps(&src[pos_.i[2]]);
+        const __m128 val3 = _mm_loadu_ps(&src[pos_.i[3]]);
+        __m128 k0 = _mm_load_ps(sinc4Tab[frac_.i[0]]);
+        __m128 k1 = _mm_load_ps(sinc4Tab[frac_.i[1]]);
+        __m128 k2 = _mm_load_ps(sinc4Tab[frac_.i[2]]);
+        __m128 k3 = _mm_load_ps(sinc4Tab[frac_.i[3]]);
+        __m128 out;
+
+        k0 = _mm_mul_ps(k0, val0);
+        k1 = _mm_mul_ps(k1, val1);
+        k2 = _mm_mul_ps(k2, val2);
+        k3 = _mm_mul_ps(k3, val3);
+        k0 = _mm_hadd_ps(k0, k1);
+        k2 = _mm_hadd_ps(k2, k3);
+        out = _mm_hadd_ps(k0, k2);
+
+        _mm_store_ps(&dst[i], out);
+
+        frac4 = _mm_add_epi32(frac4, increment4);
+        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
+        frac4 = _mm_and_si128(frac4, fracMask4);
+
+        _mm_store_ps(pos_.f, _mm_castsi128_ps(pos4));
+        _mm_store_ps(frac_.f, _mm_castsi128_ps(frac4));
+    }
+
+    /* NOTE: These four elements represent the position *after* the last four
+     * samples, so the lowest element is the next position to resample.
+     */
+    pos = pos_.i[0];
+    frac = frac_.i[0];
+
+    for(;i < numsamples;i++)
+    {
+        dst[i] = resample_fir4(src[pos], src[pos+1], src[pos+2], src[pos+3], frac);
+
+        frac += increment;
+        pos  += frac>>FRACTIONBITS;
+        frac &= FRACTIONMASK;
+    }
+    return dst;
+}

Alc/mixer_sse41.c

@@ -0,0 +1,154 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <xmmintrin.h>
+#include <emmintrin.h>
+#include <smmintrin.h>
+
+#include "alu.h"
+#include "mixer_defs.h"
+
+
+const ALfloat *Resample_lerp32_SSE41(const InterpState* UNUSED(state),
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei numsamples)
+{
+    const __m128i increment4 = _mm_set1_epi32(increment*4);
+    const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE);
+    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
+    union { alignas(16) ALint i[4]; float f[4]; } pos_;
+    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
+    __m128i frac4, pos4;
+    ALint pos;
+    ALsizei i;
+
+    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
+
+    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
+    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
+
+    for(i = 0;numsamples-i > 3;i += 4)
+    {
+        const __m128 val1 = _mm_setr_ps(src[pos_.i[0]], src[pos_.i[1]], src[pos_.i[2]], src[pos_.i[3]]);
+        const __m128 val2 = _mm_setr_ps(src[pos_.i[0]+1], src[pos_.i[1]+1], src[pos_.i[2]+1], src[pos_.i[3]+1]);
+
+        /* val1 + (val2-val1)*mu */
+        const __m128 r0 = _mm_sub_ps(val2, val1);
+        const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4);
+        const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0));
+
+        _mm_store_ps(&dst[i], out);
+
+        frac4 = _mm_add_epi32(frac4, increment4);
+        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
+        frac4 = _mm_and_si128(frac4, fracMask4);
+
+        pos_.i[0] = _mm_extract_epi32(pos4, 0);
+        pos_.i[1] = _mm_extract_epi32(pos4, 1);
+        pos_.i[2] = _mm_extract_epi32(pos4, 2);
+        pos_.i[3] = _mm_extract_epi32(pos4, 3);
+    }
+
+    /* NOTE: These four elements represent the position *after* the last four
+     * samples, so the lowest element is the next position to resample.
+     */
+    pos = pos_.i[0];
+    frac = _mm_cvtsi128_si32(frac4);
+
+    for(;i < numsamples;i++)
+    {
+        dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
+
+        frac += increment;
+        pos  += frac>>FRACTIONBITS;
+        frac &= FRACTIONMASK;
+    }
+    return dst;
+}
+
+const ALfloat *Resample_fir4_32_SSE41(const InterpState* UNUSED(state),
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei numsamples)
+{
+    const __m128i increment4 = _mm_set1_epi32(increment*4);
+    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
+    union { alignas(16) ALint i[4]; float f[4]; } pos_;
+    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
+    __m128i frac4, pos4;
+    ALint pos;
+    ALsizei i;
+
+    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
+
+    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
+    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
+
+    --src;
+    for(i = 0;numsamples-i > 3;i += 4)
+    {
+        const __m128 val0 = _mm_loadu_ps(&src[pos_.i[0]]);
+        const __m128 val1 = _mm_loadu_ps(&src[pos_.i[1]]);
+        const __m128 val2 = _mm_loadu_ps(&src[pos_.i[2]]);
+        const __m128 val3 = _mm_loadu_ps(&src[pos_.i[3]]);
+        __m128 k0 = _mm_load_ps(sinc4Tab[frac_.i[0]]);
+        __m128 k1 = _mm_load_ps(sinc4Tab[frac_.i[1]]);
+        __m128 k2 = _mm_load_ps(sinc4Tab[frac_.i[2]]);
+        __m128 k3 = _mm_load_ps(sinc4Tab[frac_.i[3]]);
+        __m128 out;
+
+        k0 = _mm_mul_ps(k0, val0);
+        k1 = _mm_mul_ps(k1, val1);
+        k2 = _mm_mul_ps(k2, val2);
+        k3 = _mm_mul_ps(k3, val3);
+        k0 = _mm_hadd_ps(k0, k1);
+        k2 = _mm_hadd_ps(k2, k3);
+        out = _mm_hadd_ps(k0, k2);
+
+        _mm_store_ps(&dst[i], out);
+
+        frac4 = _mm_add_epi32(frac4, increment4);
+        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
+        frac4 = _mm_and_si128(frac4, fracMask4);
+
+        pos_.i[0] = _mm_extract_epi32(pos4, 0);
+        pos_.i[1] = _mm_extract_epi32(pos4, 1);
+        pos_.i[2] = _mm_extract_epi32(pos4, 2);
+        pos_.i[3] = _mm_extract_epi32(pos4, 3);
+        frac_.i[0] = _mm_extract_epi32(frac4, 0);
+        frac_.i[1] = _mm_extract_epi32(frac4, 1);
+        frac_.i[2] = _mm_extract_epi32(frac4, 2);
+        frac_.i[3] = _mm_extract_epi32(frac4, 3);
+    }
+
+    pos = pos_.i[0];
+    frac = frac_.i[0];
+
+    for(;i < numsamples;i++)
+    {
+        dst[i] = resample_fir4(src[pos], src[pos+1], src[pos+2], src[pos+3], frac);
+
+        frac += increment;
+        pos  += frac>>FRACTIONBITS;
+        frac &= FRACTIONMASK;
+    }
+    return dst;
+}

Alc/nfcfilter.c

@@ -0,0 +1,418 @@
+
+#include "config.h"
+
+#include "nfcfilter.h"
+
+#include "alu.h"
+
+
+/* Near-field control filters are the basis for handling the near-field effect.
+ * The near-field effect is a bass-boost present in the directional components
+ * of a recorded signal, created as a result of the wavefront curvature (itself
+ * a function of sound distance). Proper reproduction dictates this be
+ * compensated for using a bass-cut given the playback speaker distance, to
+ * avoid excessive bass in the playback.
+ *
+ * For real-time rendered audio, emulating the near-field effect based on the
+ * sound source's distance, and subsequently compensating for it at output
+ * based on the speaker distances, can create a more realistic perception of
+ * sound distance beyond a simple 1/r attenuation.
+ *
+ * These filters do just that. Each one applies a low-shelf filter, created as
+ * the combination of a bass-boost for a given sound source distance (near-
+ * field emulation) along with a bass-cut for a given control/speaker distance
+ * (near-field compensation).
+ *
+ * Note that it is necessary to apply a cut along with the boost, since the
+ * boost alone is unstable in higher-order ambisonics as it causes an infinite
+ * DC gain (even first-order ambisonics requires there to be no DC offset for
+ * the boost to work). Consequently, ambisonics requires a control parameter to
+ * be used to avoid an unstable boost-only filter. NFC-HOA defines this control
+ * as a reference delay, calculated with:
+ *
+ * reference_delay = control_distance / speed_of_sound
+ *
+ * This means w0 (for input) or w1 (for output) should be set to:
+ *
+ * wN = 1 / (reference_delay * sample_rate)
+ *
+ * when dealing with NFC-HOA content. For FOA input content, which does not
+ * specify a reference_delay variable, w0 should be set to 0 to apply only
+ * near-field compensation for output. It's important that w1 be a finite,
+ * positive, non-0 value or else the bass-boost will become unstable again.
+ * Also, w0 should not be too large compared to w1, to avoid excessively loud
+ * low frequencies.
+ */
+
+static const float B[4][3] = {
+    {    0.0f                             },
+    {    1.0f                             },
+    {    3.0f,     3.0f                   },
+    { 3.6778f,  6.4595f, 2.3222f          },
+  /*{ 4.2076f, 11.4877f, 5.7924f, 9.1401f }*/
+};
+
+void NfcFilterCreate1(NfcFilter *nfc, const float w0, const float w1)
+{
+    float b_00, g_0;
+    float r;
+
+    memset(nfc, 0, sizeof(*nfc));
+
+    nfc->g = 1.0f;
+    nfc->coeffs[0] = 1.0f;
+
+    /* Calculate bass-boost coefficients. */
+    r = 0.5f * w0;
+    b_00 = B[1][0] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->coeffs[0] *= g_0;
+    nfc->coeffs[1] = (2.0f * b_00) / g_0;
+
+    /* Calculate bass-cut coefficients. */
+    r = 0.5f * w1;
+    b_00 = B[1][0] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->g /= g_0;
+    nfc->coeffs[0] /= g_0;
+    nfc->coeffs[1+1] = (2.0f * b_00) / g_0;
+}
+
+void NfcFilterAdjust1(NfcFilter *nfc, const float w0)
+{
+    float b_00, g_0;
+    float r;
+
+    r = 0.5f * w0;
+    b_00 = B[1][0] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->coeffs[0] = nfc->g * g_0;
+    nfc->coeffs[1] = (2.0f * b_00) / g_0;
+}
+
+void NfcFilterUpdate1(NfcFilter *nfc, ALfloat *restrict dst, const float *restrict src, const int count)
+{
+    const float b0 = nfc->coeffs[0];
+    const float a0 = nfc->coeffs[1];
+    const float a1 = nfc->coeffs[2];
+    float z1 = nfc->history[0];
+    int i;
+
+    for(i = 0;i < count;i++)
+    {
+        float out = src[i] * b0;
+        float y;
+
+        y = out - (a1*z1);
+        out = y + (a0*z1);
+        z1 += y;
+
+        dst[i] = out;
+    }
+    nfc->history[0] = z1;
+}
+
+
+void NfcFilterCreate2(NfcFilter *nfc, const float w0, const float w1)
+{
+    float b_10, b_11, g_1;
+    float r;
+
+    memset(nfc, 0, sizeof(*nfc));
+
+    nfc->g = 1.0f;
+    nfc->coeffs[0] = 1.0f;
+
+    /* Calculate bass-boost coefficients. */
+    r = 0.5f * w0;
+    b_10 = B[2][0] * r;
+    b_11 = B[2][1] * r * r;
+    g_1 = 1.0f + b_10 + b_11;
+
+    nfc->coeffs[0] *= g_1;
+    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+    nfc->coeffs[2] = (4.0f * b_11) / g_1;
+
+    /* Calculate bass-cut coefficients. */
+    r = 0.5f * w1;
+    b_10 = B[2][0] * r;
+    b_11 = B[2][1] * r * r;
+    g_1 = 1.0f + b_10 + b_11;
+
+    nfc->g /= g_1;
+    nfc->coeffs[0] /= g_1;
+    nfc->coeffs[2+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+    nfc->coeffs[2+2] = (4.0f * b_11) / g_1;
+}
+
+void NfcFilterAdjust2(NfcFilter *nfc, const float w0)
+{
+    float b_10, b_11, g_1;
+    float r;
+
+    r = 0.5f * w0;
+    b_10 = B[2][0] * r;
+    b_11 = B[2][1] * r * r;
+    g_1 = 1.0f + b_10 + b_11;
+
+    nfc->coeffs[0] = nfc->g * g_1;
+    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+    nfc->coeffs[2] = (4.0f * b_11) / g_1;
+}
+
+void NfcFilterUpdate2(NfcFilter *nfc, ALfloat *restrict dst, const float *restrict src, const int count)
+{
+    const float b0 = nfc->coeffs[0];
+    const float a00 = nfc->coeffs[1];
+    const float a01 = nfc->coeffs[2];
+    const float a10 = nfc->coeffs[3];
+    const float a11 = nfc->coeffs[4];
+    float z1 = nfc->history[0];
+    float z2 = nfc->history[1];
+    int i;
+
+    for(i = 0;i < count;i++)
+    {
+        float out = src[i] * b0;
+        float y;
+
+        y = out - (a10*z1) - (a11*z2);
+        out = y + (a00*z1) + (a01*z2);
+        z2 += z1;
+        z1 += y;
+
+        dst[i] = out;
+    }
+    nfc->history[0] = z1;
+    nfc->history[1] = z2;
+}
+
+
+void NfcFilterCreate3(NfcFilter *nfc, const float w0, const float w1)
+{
+    float b_10, b_11, g_1;
+    float b_00, g_0;
+    float r;
+
+    memset(nfc, 0, sizeof(*nfc));
+
+    nfc->g = 1.0f;
+    nfc->coeffs[0] = 1.0f;
+
+    /* Calculate bass-boost coefficients. */
+    r = 0.5f * w0;
+    b_10 = B[3][0] * r;
+    b_11 = B[3][1] * r * r;
+    g_1 = 1.0f + b_10 + b_11;
+
+    nfc->coeffs[0] *= g_1;
+    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+    nfc->coeffs[2] = (4.0f * b_11) / g_1;
+
+    b_00 = B[3][2] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->coeffs[0] *= g_0;
+    nfc->coeffs[2+1] = (2.0f * b_00) / g_0;
+
+    /* Calculate bass-cut coefficients. */
+    r = 0.5f * w1;
+    b_10 = B[3][0] * r;
+    b_11 = B[3][1] * r * r;
+    g_1 = 1.0f + b_10 + b_11;
+
+    nfc->g /= g_1;
+    nfc->coeffs[0] /= g_1;
+    nfc->coeffs[3+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+    nfc->coeffs[3+2] = (4.0f * b_11) / g_1;
+    
+    b_00 = B[3][2] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->g /= g_0;
+    nfc->coeffs[0] /= g_0;
+    nfc->coeffs[3+2+1] = (2.0f * b_00) / g_0;
+}
+
+void NfcFilterAdjust3(NfcFilter *nfc, const float w0)
+{
+    float b_10, b_11, g_1;
+    float b_00, g_0;
+    float r;
+
+    r = 0.5f * w0;
+    b_10 = B[3][0] * r;
+    b_11 = B[3][1] * r * r;
+    g_1 = 1.0f + b_10 + b_11;
+
+    nfc->coeffs[0] = nfc->g * g_1;
+    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+    nfc->coeffs[2] = (4.0f * b_11) / g_1;
+
+    b_00 = B[3][2] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->coeffs[0] *= g_0;
+    nfc->coeffs[2+1] = (2.0f * b_00) / g_0;
+}
+
+void NfcFilterUpdate3(NfcFilter *nfc, ALfloat *restrict dst, const float *restrict src, const int count)
+{
+    const float b0 = nfc->coeffs[0];
+    const float a00 = nfc->coeffs[1];
+    const float a01 = nfc->coeffs[2];
+    const float a02 = nfc->coeffs[3];
+    const float a10 = nfc->coeffs[4];
+    const float a11 = nfc->coeffs[5];
+    const float a12 = nfc->coeffs[6];
+    float z1 = nfc->history[0];
+    float z2 = nfc->history[1];
+    float z3 = nfc->history[2];
+    int i;
+
+    for(i = 0;i < count;i++)
+    {
+        float out = src[i] * b0;
+        float y;
+
+        y = out - (a10*z1) - (a11*z2);
+        out = y + (a00*z1) + (a01*z2);
+        z2 += z1;
+        z1 += y;
+
+        y = out - (a12*z3);
+        out = y + (a02*z3);
+        z3 += y;
+
+        dst[i] = out;
+    }
+    nfc->history[0] = z1;
+    nfc->history[1] = z2;
+    nfc->history[2] = z3;
+}
+
+
+#if 0 /* Original methods the above are derived from. */
+static void NfcFilterCreate(NfcFilter *nfc, const ALsizei order, const float src_dist, const float ctl_dist, const float rate)
+{
+    static const float B[4][5] = {
+        {                                     },
+        {    1.0f                             },
+        {    3.0f,     3.0f                   },
+        { 3.6778f,  6.4595f, 2.3222f          },
+        { 4.2076f, 11.4877f, 5.7924f, 9.1401f }
+    };
+    float w0 = SPEEDOFSOUNDMETRESPERSEC / (src_dist * rate);
+    float w1 = SPEEDOFSOUNDMETRESPERSEC / (ctl_dist * rate);
+    ALsizei i;
+    float r;
+
+    nfc->g = 1.0f;
+    nfc->coeffs[0] = 1.0f;
+
+    /* NOTE: Slight adjustment from the literature to raise the center
+     * frequency a bit (0.5 -> 1.0).
+     */
+    r = 1.0f * w0;
+    for(i = 0; i < (order-1);i += 2)
+    {
+        float b_10 = B[order][i  ] * r;
+        float b_11 = B[order][i+1] * r * r;
+        float g_1 = 1.0f + b_10 + b_11;
+
+        nfc->b[i] = b_10;
+        nfc->b[i + 1] = b_11;
+        nfc->coeffs[0] *= g_1;
+        nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+        nfc->coeffs[i+2] = (4.0f * b_11) / g_1;
+    }
+    if(i < order)
+    {
+        float b_00 = B[order][i] * r;
+        float g_0 = 1.0f + b_00;
+
+        nfc->b[i] = b_00;
+        nfc->coeffs[0] *= g_0;
+        nfc->coeffs[i+1] = (2.0f * b_00) / g_0;
+    }
+
+    r = 1.0f * w1;
+    for(i = 0;i < (order-1);i += 2)
+    {
+        float b_10 = B[order][i  ] * r;
+        float b_11 = B[order][i+1] * r * r;
+        float g_1 = 1.0f + b_10 + b_11;
+
+        nfc->g /= g_1;
+        nfc->coeffs[0] /= g_1;
+        nfc->coeffs[order+i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+        nfc->coeffs[order+i+2] = (4.0f * b_11) / g_1;
+    }
+    if(i < order)
+    {
+        float b_00 = B[order][i] * r;
+        float g_0 = 1.0f + b_00;
+
+        nfc->g /= g_0;
+        nfc->coeffs[0] /= g_0;
+        nfc->coeffs[order+i+1] = (2.0f * b_00) / g_0;
+    }
+
+    for(i = 0; i < MAX_AMBI_ORDER; i++)
+        nfc->history[i] = 0.0f;
+}
+
+static void NfcFilterAdjust(NfcFilter *nfc, const float distance)
+{
+    int i;
+
+    nfc->coeffs[0] = nfc->g;
+
+    for(i = 0;i < (nfc->order-1);i += 2)
+    {
+        float b_10 = nfc->b[i] / distance;
+        float b_11 = nfc->b[i+1] / (distance * distance);
+        float g_1 = 1.0f + b_10 + b_11;
+
+        nfc->coeffs[0] *= g_1;
+        nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
+        nfc->coeffs[i+2] = (4.0f * b_11) / g_1;
+    }
+    if(i < nfc->order)
+    {
+        float b_00 = nfc->b[i] / distance;
+        float g_0 = 1.0f + b_00;
+
+        nfc->coeffs[0] *= g_0;
+        nfc->coeffs[i+1] = (2.0f * b_00) / g_0;
+    }
+}
+
+static float NfcFilterUpdate(const float in, NfcFilter *nfc)
+{
+    int i;
+    float out = in * nfc->coeffs[0];
+
+    for(i = 0;i < (nfc->order-1);i += 2)
+    {
+        float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) -
+                        (nfc->coeffs[nfc->order+i+2] * nfc->history[i+1]) + 1.0e-30f;
+        out = y + (nfc->coeffs[i+1]*nfc->history[i]) + (nfc->coeffs[i+2]*nfc->history[i+1]);
+
+        nfc->history[i+1] += nfc->history[i];
+        nfc->history[i] += y;
+    }
+    if(i < nfc->order)
+    {
+        float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) + 1.0e-30f;
+
+        out = y + (nfc->coeffs[i+1] * nfc->history[i]);
+        nfc->history[i] += y;
+    }
+
+    return out;
+}
+#endif

Alc/nfcfilter.h

@@ -0,0 +1,37 @@
+#ifndef NFCFILTER_H
+#define NFCFILTER_H
+
+#include "alMain.h"
+
+typedef struct NfcFilter {
+    float g;
+    float coeffs[MAX_AMBI_ORDER*2 + 1];
+    float history[MAX_AMBI_ORDER];
+} NfcFilter;
+
+/* NOTE:
+ * w0 = speed_of_sound / (source_distance * sample_rate);
+ * w1 = speed_of_sound / (control_distance * sample_rate);
+ *
+ * Generally speaking, the control distance should be approximately the average
+ * speaker distance, or based on the reference delay if outputing NFC-HOA. It
+ * must not be negative, 0, or infinite. The source distance should not be too
+ * small relative to the control distance.
+ */
+
+/* Near-field control filter for first-order ambisonic channels (1-3). */
+void NfcFilterCreate1(NfcFilter *nfc, const float w0, const float w1);
+void NfcFilterAdjust1(NfcFilter *nfc, const float w0);
+void NfcFilterUpdate1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
+
+/* Near-field control filter for second-order ambisonic channels (4-8). */
+void NfcFilterCreate2(NfcFilter *nfc, const float w0, const float w1);
+void NfcFilterAdjust2(NfcFilter *nfc, const float w0);
+void NfcFilterUpdate2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
+
+/* Near-field control filter for third-order ambisonic channels (9-15). */
+void NfcFilterCreate3(NfcFilter *nfc, const float w0, const float w1);
+void NfcFilterAdjust3(NfcFilter *nfc, const float w0);
+void NfcFilterUpdate3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
+
+#endif /* NFCFILTER_H */

Alc/oss.c

@@ -1,493 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <sys/ioctl.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-#include <unistd.h>
-#include <errno.h>
-#include <math.h>
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-
-#include <sys/soundcard.h>
-
-/*
- * The OSS documentation talks about SOUND_MIXER_READ, but the header
- * only contains MIXER_READ. Play safe. Same for WRITE.
- */
-#ifndef SOUND_MIXER_READ
-#define SOUND_MIXER_READ MIXER_READ
-#endif
-#ifndef SOUND_MIXER_WRITE
-#define SOUND_MIXER_WRITE MIXER_WRITE
-#endif
-
-static const ALCchar oss_device[] = "OSS Software";
-static const ALCchar oss_device_capture[] = "OSS Capture";
-
-typedef struct {
-    int fd;
-    volatile int killNow;
-    ALvoid *thread;
-
-    ALubyte *mix_data;
-    int data_size;
-
-    RingBuffer *ring;
-    int doCapture;
-} oss_data;
-
-
-static int log2i(ALCuint x)
-{
-    int y = 0;
-    while (x > 1)
-    {
-        x >>= 1;
-        y++;
-    }
-    return y;
-}
-
-
-static ALuint OSSProc(ALvoid *ptr)
-{
-    ALCdevice *pDevice = (ALCdevice*)ptr;
-    oss_data *data = (oss_data*)pDevice->ExtraData;
-    ALint frameSize;
-    ssize_t wrote;
-
-    frameSize = aluChannelsFromFormat(pDevice->Format) *
-                aluBytesFromFormat(pDevice->Format);
-
-    while(!data->killNow && pDevice->Connected)
-    {
-        ALint len = data->data_size;
-        ALubyte *WritePtr = data->mix_data;
-
-        aluMixData(pDevice, WritePtr, len/frameSize);
-        while(len > 0 && !data->killNow)
-        {
-            wrote = write(data->fd, WritePtr, len);
-            if(wrote < 0)
-            {
-                if(errno != EAGAIN && errno != EWOULDBLOCK && errno != EINTR)
-                {
-                    AL_PRINT("write failed: %s\n", strerror(errno));
-                    aluHandleDisconnect(pDevice);
-                    len = 0;
-                }
-                else
-                    Sleep(1);
-                continue;
-            }
-
-            len -= wrote;
-            WritePtr += wrote;
-        }
-    }
-
-    return 0;
-}
-
-static ALuint OSSCaptureProc(ALvoid *ptr)
-{
-    ALCdevice *pDevice = (ALCdevice*)ptr;
-    oss_data *data = (oss_data*)pDevice->ExtraData;
-    int frameSize;
-    int amt;
-
-    frameSize  = aluBytesFromFormat(pDevice->Format);
-    frameSize *= aluChannelsFromFormat(pDevice->Format);
-
-    while(!data->killNow)
-    {
-        amt = read(data->fd, data->mix_data, data->data_size);
-        if(amt < 0)
-        {
-            AL_PRINT("read failed: %s\n", strerror(errno));
-            aluHandleDisconnect(pDevice);
-            break;
-        }
-        if(amt == 0)
-        {
-            Sleep(1);
-            continue;
-        }
-        if(data->doCapture)
-            WriteRingBuffer(data->ring, data->mix_data, amt/frameSize);
-    }
-
-    return 0;
-}
-
-static ALCboolean oss_open_playback(ALCdevice *device, const ALCchar *deviceName)
-{
-    char driver[64];
-    oss_data *data;
-
-    strncpy(driver, GetConfigValue("oss", "device", "/dev/dsp"), sizeof(driver)-1);
-    driver[sizeof(driver)-1] = 0;
-    if(!deviceName)
-        deviceName = oss_device;
-    else if(strcmp(deviceName, oss_device) != 0)
-        return ALC_FALSE;
-
-    data = (oss_data*)calloc(1, sizeof(oss_data));
-    data->killNow = 0;
-
-    data->fd = open(driver, O_WRONLY);
-    if(data->fd == -1)
-    {
-        free(data);
-        AL_PRINT("Could not open %s: %s\n", driver, strerror(errno));
-        return ALC_FALSE;
-    }
-
-    device->szDeviceName = strdup(deviceName);
-    device->ExtraData = data;
-    return ALC_TRUE;
-}
-
-static void oss_close_playback(ALCdevice *device)
-{
-    oss_data *data = (oss_data*)device->ExtraData;
-
-    close(data->fd);
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean oss_reset_playback(ALCdevice *device)
-{
-    oss_data *data = (oss_data*)device->ExtraData;
-    int numFragmentsLogSize;
-    int log2FragmentSize;
-    unsigned int periods;
-    audio_buf_info info;
-    ALuint frameSize;
-    int numChannels;
-    int ossFormat;
-    int ossSpeed;
-    char *err;
-    int i;
-
-    switch(aluBytesFromFormat(device->Format))
-    {
-        case 1:
-            ossFormat = AFMT_U8;
-            break;
-        case 4:
-            switch(aluChannelsFromFormat(device->Format))
-            {
-                case 1: device->Format = AL_FORMAT_MONO16; break;
-                case 2: device->Format = AL_FORMAT_STEREO16; break;
-                case 4: device->Format = AL_FORMAT_QUAD16; break;
-                case 6: device->Format = AL_FORMAT_51CHN16; break;
-                case 7: device->Format = AL_FORMAT_61CHN16; break;
-                case 8: device->Format = AL_FORMAT_71CHN16; break;
-            }
-            /* fall-through */
-        case 2:
-            ossFormat = AFMT_S16_NE;
-            break;
-        default:
-            AL_PRINT("Unknown format: 0x%x\n", device->Format);
-            return ALC_FALSE;
-    }
-
-    periods = device->NumUpdates;
-    numChannels = aluChannelsFromFormat(device->Format);
-    frameSize = numChannels * aluBytesFromFormat(device->Format);
-
-    ossSpeed = device->Frequency;
-    log2FragmentSize = log2i(device->UpdateSize * frameSize);
-
-    /* according to the OSS spec, 16 bytes are the minimum */
-    if (log2FragmentSize < 4)
-        log2FragmentSize = 4;
-    /* Subtract one period since the temp mixing buffer counts as one. Still
-     * need at least two on the card, though. */
-    if(periods > 2) periods--;
-    numFragmentsLogSize = (periods << 16) | log2FragmentSize;
-
-#define ok(func, str) (i=(func),((i<0)?(err=(str)),0:1))
-    /* Don't fail if SETFRAGMENT fails. We can handle just about anything
-     * that's reported back via GETOSPACE */
-    ioctl(data->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize);
-    if (!(ok(ioctl(data->fd, SNDCTL_DSP_SETFMT, &ossFormat), "set format") &&
-          ok(ioctl(data->fd, SNDCTL_DSP_CHANNELS, &numChannels), "set channels") &&
-          ok(ioctl(data->fd, SNDCTL_DSP_SPEED, &ossSpeed), "set speed") &&
-          ok(ioctl(data->fd, SNDCTL_DSP_GETOSPACE, &info), "get space")))
-    {
-        AL_PRINT("%s failed: %s\n", err, strerror(errno));
-        return ALC_FALSE;
-    }
-#undef ok
-
-    if((int)aluChannelsFromFormat(device->Format) != numChannels)
-    {
-        AL_PRINT("Could not set %d channels, got %d instead\n", aluChannelsFromFormat(device->Format), numChannels);
-        return ALC_FALSE;
-    }
-
-    if(!((ossFormat == AFMT_U8 && aluBytesFromFormat(device->Format) == 1) ||
-         (ossFormat == AFMT_S16_NE && aluBytesFromFormat(device->Format) == 2)))
-    {
-        AL_PRINT("Could not set %d-bit output, got format %#x\n", aluBytesFromFormat(device->Format)*8, ossFormat);
-        return ALC_FALSE;
-    }
-
-    device->Frequency = ossSpeed;
-    device->UpdateSize = info.fragsize / frameSize;
-    device->NumUpdates = info.fragments + 1;
-
-    data->data_size = device->UpdateSize * frameSize;
-    data->mix_data = calloc(1, data->data_size);
-
-    data->thread = StartThread(OSSProc, device);
-    if(data->thread == NULL)
-    {
-        free(data->mix_data);
-        data->mix_data = NULL;
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void oss_stop_playback(ALCdevice *device)
-{
-    oss_data *data = (oss_data*)device->ExtraData;
-
-    if(!data->thread)
-        return;
-
-    data->killNow = 1;
-    StopThread(data->thread);
-    data->thread = NULL;
-
-    if(ioctl(data->fd, SNDCTL_DSP_RESET) != 0)
-        AL_PRINT("Error resetting device: %s\n", strerror(errno));
-
-    free(data->mix_data);
-    data->mix_data = NULL;
-}
-
-
-static ALCboolean oss_open_capture(ALCdevice *device, const ALCchar *deviceName)
-{
-    int numFragmentsLogSize;
-    int log2FragmentSize;
-    unsigned int periods;
-    audio_buf_info info;
-    ALuint frameSize;
-    int numChannels;
-    char driver[64];
-    oss_data *data;
-    int ossFormat;
-    int ossSpeed;
-    char *err;
-    int i;
-
-    strncpy(driver, GetConfigValue("oss", "capture", "/dev/dsp"), sizeof(driver)-1);
-    driver[sizeof(driver)-1] = 0;
-    if(!deviceName)
-        deviceName = oss_device_capture;
-    else if(strcmp(deviceName, oss_device_capture) != 0)
-        return ALC_FALSE;
-
-    data = (oss_data*)calloc(1, sizeof(oss_data));
-    data->killNow = 0;
-
-    data->fd = open(driver, O_RDONLY);
-    if(data->fd == -1)
-    {
-        free(data);
-        AL_PRINT("Could not open %s: %s\n", driver, strerror(errno));
-        return ALC_FALSE;
-    }
-
-    switch(aluBytesFromFormat(device->Format))
-    {
-        case 1:
-            ossFormat = AFMT_U8;
-            break;
-        case 2:
-            ossFormat = AFMT_S16_NE;
-            break;
-        default:
-            AL_PRINT("Unknown format: 0x%x\n", device->Format);
-            close(data->fd);
-            free(data);
-            return ALC_FALSE;
-    }
-
-    periods = 4;
-    numChannels = aluChannelsFromFormat(device->Format);
-    frameSize = numChannels * aluBytesFromFormat(device->Format);
-    ossSpeed = device->Frequency;
-    log2FragmentSize = log2i(device->UpdateSize * device->NumUpdates *
-                             frameSize / periods);
-
-    /* according to the OSS spec, 16 bytes are the minimum */
-    if (log2FragmentSize < 4)
-        log2FragmentSize = 4;
-    numFragmentsLogSize = (periods << 16) | log2FragmentSize;
-
-#define ok(func, str) (i=(func),((i<0)?(err=(str)),0:1))
-    if (!(ok(ioctl(data->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize), "set fragment") &&
-          ok(ioctl(data->fd, SNDCTL_DSP_SETFMT, &ossFormat), "set format") &&
-          ok(ioctl(data->fd, SNDCTL_DSP_CHANNELS, &numChannels), "set channels") &&
-          ok(ioctl(data->fd, SNDCTL_DSP_SPEED, &ossSpeed), "set speed") &&
-          ok(ioctl(data->fd, SNDCTL_DSP_GETISPACE, &info), "get space")))
-    {
-        AL_PRINT("%s failed: %s\n", err, strerror(errno));
-        close(data->fd);
-        free(data);
-        return ALC_FALSE;
-    }
-#undef ok
-
-    if((int)aluChannelsFromFormat(device->Format) != numChannels)
-    {
-        AL_PRINT("Could not set %d channels, got %d instead\n", aluChannelsFromFormat(device->Format), numChannels);
-        close(data->fd);
-        free(data);
-        return ALC_FALSE;
-    }
-
-    if(!((ossFormat == AFMT_U8 && aluBytesFromFormat(device->Format) == 1) ||
-         (ossFormat == AFMT_S16_NE && aluBytesFromFormat(device->Format) == 2)))
-    {
-        AL_PRINT("Could not set %d-bit input, got format %#x\n", aluBytesFromFormat(device->Format)*8, ossFormat);
-        close(data->fd);
-        free(data);
-        return ALC_FALSE;
-    }
-
-    data->ring = CreateRingBuffer(frameSize, device->UpdateSize * device->NumUpdates);
-    if(!data->ring)
-    {
-        AL_PRINT("ring buffer create failed\n");
-        close(data->fd);
-        free(data);
-        return ALC_FALSE;
-    }
-
-    data->data_size = info.fragsize;
-    data->mix_data = calloc(1, data->data_size);
-
-    device->ExtraData = data;
-    data->thread = StartThread(OSSCaptureProc, device);
-    if(data->thread == NULL)
-    {
-        device->ExtraData = NULL;
-        free(data->mix_data);
-        free(data);
-        return ALC_FALSE;
-    }
-
-    device->szDeviceName = strdup(deviceName);
-    return ALC_TRUE;
-}
-
-static void oss_close_capture(ALCdevice *device)
-{
-    oss_data *data = (oss_data*)device->ExtraData;
-    data->killNow = 1;
-    StopThread(data->thread);
-
-    close(data->fd);
-
-    DestroyRingBuffer(data->ring);
-
-    free(data->mix_data);
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static void oss_start_capture(ALCdevice *pDevice)
-{
-    oss_data *data = (oss_data*)pDevice->ExtraData;
-    data->doCapture = 1;
-}
-
-static void oss_stop_capture(ALCdevice *pDevice)
-{
-    oss_data *data = (oss_data*)pDevice->ExtraData;
-    data->doCapture = 0;
-}
-
-static void oss_capture_samples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
-{
-    oss_data *data = (oss_data*)pDevice->ExtraData;
-    if(lSamples <= (ALCuint)RingBufferSize(data->ring))
-        ReadRingBuffer(data->ring, pBuffer, lSamples);
-    else
-        alcSetError(ALC_INVALID_VALUE);
-}
-
-static ALCuint oss_available_samples(ALCdevice *pDevice)
-{
-    oss_data *data = (oss_data*)pDevice->ExtraData;
-    return RingBufferSize(data->ring);
-}
-
-
-BackendFuncs oss_funcs = {
-    oss_open_playback,
-    oss_close_playback,
-    oss_reset_playback,
-    oss_stop_playback,
-    oss_open_capture,
-    oss_close_capture,
-    oss_start_capture,
-    oss_stop_capture,
-    oss_capture_samples,
-    oss_available_samples
-};
-
-void alc_oss_init(BackendFuncs *func_list)
-{
-    *func_list = oss_funcs;
-}
-
-void alc_oss_deinit(void)
-{
-}
-
-void alc_oss_probe(int type)
-{
-    if(type == DEVICE_PROBE)
-        AppendDeviceList(oss_device);
-    else if(type == ALL_DEVICE_PROBE)
-        AppendAllDeviceList(oss_device);
-    else if(type == CAPTURE_DEVICE_PROBE)
-        AppendCaptureDeviceList(oss_device_capture);
-}

Alc/portaudio.c

@@ -1,298 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-#ifdef HAVE_DLFCN_H
-#include <dlfcn.h>
-#endif
-
-#include <portaudio.h>
-
-static void *pa_handle;
-#define MAKE_FUNC(x) static typeof(x) * p##x
-MAKE_FUNC(Pa_Initialize);
-MAKE_FUNC(Pa_Terminate);
-MAKE_FUNC(Pa_GetErrorText);
-MAKE_FUNC(Pa_StartStream);
-MAKE_FUNC(Pa_StopStream);
-MAKE_FUNC(Pa_OpenStream);
-MAKE_FUNC(Pa_CloseStream);
-MAKE_FUNC(Pa_GetDefaultOutputDevice);
-MAKE_FUNC(Pa_GetStreamInfo);
-#undef MAKE_FUNC
-
-
-static const ALCchar pa_device[] = "PortAudio Software";
-static volatile ALuint load_count;
-
-
-void pa_load(void)
-{
-    const char *str;
-    PaError err;
-
-    if(load_count == 0)
-    {
-#ifdef HAVE_DLFCN_H
-#if defined(__APPLE__) && defined(__MACH__)
-# define PALIB "libportaudio.2.dylib"
-#else
-# define PALIB "libportaudio.so.2"
-#endif
-        pa_handle = dlopen(PALIB, RTLD_NOW);
-        if(!pa_handle)
-            return;
-        dlerror();
-
-#define LOAD_FUNC(f) do { \
-    p##f = (typeof(f)*)dlsym(pa_handle, #f); \
-    if((str=dlerror()) != NULL) \
-    { \
-        dlclose(pa_handle); \
-        pa_handle = NULL; \
-        AL_PRINT("Could not load %s from "PALIB": %s\n", #f, str); \
-        return; \
-    } \
-} while(0)
-#else
-        str = NULL;
-        pa_handle = (void*)0xDEADBEEF;
-#define LOAD_FUNC(f) p##f = f
-#endif
-
-LOAD_FUNC(Pa_Initialize);
-LOAD_FUNC(Pa_Terminate);
-LOAD_FUNC(Pa_GetErrorText);
-LOAD_FUNC(Pa_StartStream);
-LOAD_FUNC(Pa_StopStream);
-LOAD_FUNC(Pa_OpenStream);
-LOAD_FUNC(Pa_CloseStream);
-LOAD_FUNC(Pa_GetDefaultOutputDevice);
-LOAD_FUNC(Pa_GetStreamInfo);
-
-#undef LOAD_FUNC
-
-        if((err=pPa_Initialize()) != paNoError)
-        {
-            AL_PRINT("Pa_Initialize() returned an error: %s\n", pPa_GetErrorText(err));
-#ifdef HAVE_DLFCN_H
-            dlclose(pa_handle);
-#endif
-            pa_handle = NULL;
-            return;
-        }
-    }
-
-    ++load_count;
-}
-
-void pa_unload(void)
-{
-    if(load_count == 0 || --load_count > 0)
-        return;
-
-    pPa_Terminate();
-#ifdef HAVE_DLFCN_H
-    dlclose(pa_handle);
-#endif
-    pa_handle = NULL;
-}
-
-
-typedef struct {
-    PaStream *stream;
-} pa_data;
-
-
-static int pa_callback(const void *inputBuffer, void *outputBuffer,
-                       unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
-                       const PaStreamCallbackFlags statusFlags, void *userData)
-{
-    ALCdevice *device = (ALCdevice*)userData;
-
-    (void)inputBuffer;
-    (void)timeInfo;
-    (void)statusFlags;
-
-    aluMixData(device, outputBuffer, framesPerBuffer);
-    return 0;
-}
-
-
-static ALCboolean pa_open_playback(ALCdevice *device, const ALCchar *deviceName)
-{
-    const PaStreamInfo *streamInfo;
-    PaStreamParameters outParams;
-    pa_data *data;
-    PaError err;
-
-    if(!deviceName)
-        deviceName = pa_device;
-    else if(strcmp(deviceName, pa_device) != 0)
-        return ALC_FALSE;
-
-    pa_load();
-    if(pa_handle == NULL)
-        return ALC_FALSE;
-
-    data = (pa_data*)calloc(1, sizeof(pa_data));
-    device->ExtraData = data;
-
-    outParams.device = GetConfigValueInt("port", "device", -1);
-    if(outParams.device < 0)
-        outParams.device = pPa_GetDefaultOutputDevice();
-    outParams.suggestedLatency = (device->UpdateSize*device->NumUpdates) /
-                                 (float)device->Frequency;
-    outParams.hostApiSpecificStreamInfo = NULL;
-
-    switch(aluBytesFromFormat(device->Format))
-    {
-        case 1:
-            outParams.sampleFormat = paUInt8;
-            break;
-        case 2:
-            outParams.sampleFormat = paInt16;
-            break;
-        case 4:
-            outParams.sampleFormat = paFloat32;
-            break;
-        default:
-            AL_PRINT("Unknown format: 0x%x\n", device->Format);
-            device->ExtraData = NULL;
-            free(data);
-            pa_unload();
-            return ALC_FALSE;
-    }
-    outParams.channelCount = aluChannelsFromFormat(device->Format);
-
-    err = pPa_OpenStream(&data->stream, NULL, &outParams, device->Frequency,
-                         device->UpdateSize, paNoFlag, pa_callback, device);
-    if(err != paNoError)
-    {
-        AL_PRINT("Pa_OpenStream() returned an error: %s\n", pPa_GetErrorText(err));
-        device->ExtraData = NULL;
-        free(data);
-        pa_unload();
-        return ALC_FALSE;
-    }
-    streamInfo = pPa_GetStreamInfo(data->stream);
-
-    err = pPa_StartStream(data->stream);
-    if(err != paNoError)
-    {
-        AL_PRINT("Pa_StartStream() returned an error: %s\n", pPa_GetErrorText(err));
-        pPa_CloseStream(data->stream);
-        device->ExtraData = NULL;
-        free(data);
-        pa_unload();
-        return ALC_FALSE;
-    }
-
-    device->szDeviceName = strdup(deviceName);
-    device->Frequency = streamInfo->sampleRate;
-    return ALC_TRUE;
-}
-
-static void pa_close_playback(ALCdevice *device)
-{
-    pa_data *data = (pa_data*)device->ExtraData;
-    PaError err;
-
-    err = pPa_StopStream(data->stream);
-    if(err != paNoError)
-        fprintf(stderr, "Error stopping stream: %s\n", pPa_GetErrorText(err));
-
-    err = pPa_CloseStream(data->stream);
-    if(err != paNoError)
-        fprintf(stderr, "Error closing stream: %s\n", pPa_GetErrorText(err));
-
-    free(data);
-    device->ExtraData = NULL;
-
-    pa_unload();
-}
-
-static ALCboolean pa_reset_playback(ALCdevice *device)
-{
-    pa_data *data = (pa_data*)device->ExtraData;
-    const PaStreamInfo *streamInfo;
-
-    streamInfo = pPa_GetStreamInfo(data->stream);
-    device->Frequency = streamInfo->sampleRate;
-
-    return ALC_TRUE;
-}
-
-static void pa_stop_playback(ALCdevice *device)
-{
-    (void)device;
-}
-
-
-static ALCboolean pa_open_capture(ALCdevice *device, const ALCchar *deviceName)
-{
-    return ALC_FALSE;
-    (void)device;
-    (void)deviceName;
-}
-
-
-
-static const BackendFuncs pa_funcs = {
-    pa_open_playback,
-    pa_close_playback,
-    pa_reset_playback,
-    pa_stop_playback,
-    pa_open_capture,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL
-};
-
-void alc_pa_init(BackendFuncs *func_list)
-{
-    *func_list = pa_funcs;
-}
-
-void alc_pa_deinit(void)
-{
-}
-
-void alc_pa_probe(int type)
-{
-    pa_load();
-    if(!pa_handle) return;
-
-    if(type == DEVICE_PROBE)
-        AppendDeviceList(pa_device);
-    else if(type == ALL_DEVICE_PROBE)
-        AppendAllDeviceList(pa_device);
-
-    pa_unload();
-}

Alc/pulseaudio.c

@@ -1,814 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2009 by Konstantinos Natsakis <konstantinos.natsakis@gmail.com>
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include "alMain.h"
-#ifdef HAVE_DLFCN_H
-#include <dlfcn.h>
-#endif
-
-#include <pulse/pulseaudio.h>
-
-#if PA_API_VERSION == 11
-#define PA_STREAM_ADJUST_LATENCY 0x2000U
-static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
-{
-    return (x == PA_STREAM_CREATING || x == PA_STREAM_READY);
-}
-static inline int PA_CONTEXT_IS_GOOD(pa_context_state_t x)
-{
-    return (x == PA_CONTEXT_CONNECTING || x == PA_CONTEXT_AUTHORIZING ||
-            x == PA_CONTEXT_SETTING_NAME || x == PA_CONTEXT_READY);
-}
-#define PA_STREAM_IS_GOOD PA_STREAM_IS_GOOD
-#define PA_CONTEXT_IS_GOOD PA_CONTEXT_IS_GOOD
-#elif PA_API_VERSION != 12
-#error Invalid PulseAudio API version
-#endif
-
-static void *pa_handle;
-#define MAKE_FUNC(x) static typeof(x) * p##x
-MAKE_FUNC(pa_context_unref);
-MAKE_FUNC(pa_sample_spec_valid);
-MAKE_FUNC(pa_stream_drop);
-MAKE_FUNC(pa_strerror);
-MAKE_FUNC(pa_context_get_state);
-MAKE_FUNC(pa_stream_get_state);
-MAKE_FUNC(pa_threaded_mainloop_signal);
-MAKE_FUNC(pa_stream_peek);
-MAKE_FUNC(pa_threaded_mainloop_wait);
-MAKE_FUNC(pa_threaded_mainloop_unlock);
-MAKE_FUNC(pa_threaded_mainloop_in_thread);
-MAKE_FUNC(pa_context_new);
-MAKE_FUNC(pa_threaded_mainloop_stop);
-MAKE_FUNC(pa_context_disconnect);
-MAKE_FUNC(pa_threaded_mainloop_start);
-MAKE_FUNC(pa_threaded_mainloop_get_api);
-MAKE_FUNC(pa_context_set_state_callback);
-MAKE_FUNC(pa_stream_write);
-MAKE_FUNC(pa_xfree);
-MAKE_FUNC(pa_stream_connect_record);
-MAKE_FUNC(pa_stream_connect_playback);
-MAKE_FUNC(pa_path_get_filename);
-MAKE_FUNC(pa_get_binary_name);
-MAKE_FUNC(pa_threaded_mainloop_free);
-MAKE_FUNC(pa_context_errno);
-MAKE_FUNC(pa_xmalloc0);
-MAKE_FUNC(pa_stream_unref);
-MAKE_FUNC(pa_threaded_mainloop_accept);
-MAKE_FUNC(pa_stream_set_write_callback);
-MAKE_FUNC(pa_threaded_mainloop_new);
-MAKE_FUNC(pa_context_connect);
-MAKE_FUNC(pa_stream_get_buffer_attr);
-MAKE_FUNC(pa_stream_set_buffer_attr_callback);
-MAKE_FUNC(pa_stream_set_read_callback);
-MAKE_FUNC(pa_stream_set_state_callback);
-MAKE_FUNC(pa_stream_new);
-MAKE_FUNC(pa_stream_disconnect);
-MAKE_FUNC(pa_threaded_mainloop_lock);
-MAKE_FUNC(pa_channel_map_init_auto);
-MAKE_FUNC(pa_channel_map_parse);
-#undef MAKE_FUNC
-
-#ifndef PATH_MAX
-#define PATH_MAX 4096
-#endif
-
-typedef struct {
-    ALCuint samples;
-    ALCuint frame_size;
-
-    RingBuffer *ring;
-
-    pa_buffer_attr attr;
-    pa_sample_spec spec;
-
-    char path_name[PATH_MAX];
-    const char *context_name;
-    const char *stream_name;
-
-    pa_threaded_mainloop *loop;
-
-    pa_stream *stream;
-    pa_context *context;
-} pulse_data;
-
-static const ALCchar pulse_device[] = "PulseAudio Software";
-static const ALCchar pulse_capture_device[] = "PulseAudio Capture";
-static volatile ALuint load_count;
-
-
-void pulse_load(void) //{{{
-{
-    if(load_count == 0)
-    {
-#ifdef _WIN32
-        pa_handle = LoadLibrary("libpulse-0.dll");
-#define LOAD_FUNC(x) do { \
-    p##x = GetProcAddress(pa_handle, #x); \
-    if(!(p##x)) { \
-        AL_PRINT("Could not load %s from libpulse-0.dll\n", #x); \
-        FreeLibrary(pa_handle); \
-        pa_handle = NULL; \
-        return; \
-    } \
-} while(0)
-
-#elif defined (HAVE_DLFCN_H)
-
-        const char *err;
-#if defined(__APPLE__) && defined(__MACH__)
-        pa_handle = dlopen("libpulse.0.dylib", RTLD_NOW);
-#else
-        pa_handle = dlopen("libpulse.so.0", RTLD_NOW);
-#endif
-        dlerror();
-
-#define LOAD_FUNC(x) do { \
-    p##x = dlsym(pa_handle, #x); \
-    if((err=dlerror()) != NULL) { \
-        AL_PRINT("Could not load %s from libpulse: %s\n", #x, err); \
-        dlclose(pa_handle); \
-        pa_handle = NULL; \
-        return; \
-    } \
-} while(0)
-
-#else
-
-        pa_handle = (void*)0xDEADBEEF;
-#define LOAD_FUNC(x) p##x = (x)
-
-#endif
-        if(!pa_handle)
-            return;
-
-LOAD_FUNC(pa_context_unref);
-LOAD_FUNC(pa_sample_spec_valid);
-LOAD_FUNC(pa_stream_drop);
-LOAD_FUNC(pa_strerror);
-LOAD_FUNC(pa_context_get_state);
-LOAD_FUNC(pa_stream_get_state);
-LOAD_FUNC(pa_threaded_mainloop_signal);
-LOAD_FUNC(pa_stream_peek);
-LOAD_FUNC(pa_threaded_mainloop_wait);
-LOAD_FUNC(pa_threaded_mainloop_unlock);
-LOAD_FUNC(pa_threaded_mainloop_in_thread);
-LOAD_FUNC(pa_context_new);
-LOAD_FUNC(pa_threaded_mainloop_stop);
-LOAD_FUNC(pa_context_disconnect);
-LOAD_FUNC(pa_threaded_mainloop_start);
-LOAD_FUNC(pa_threaded_mainloop_get_api);
-LOAD_FUNC(pa_context_set_state_callback);
-LOAD_FUNC(pa_stream_write);
-LOAD_FUNC(pa_xfree);
-LOAD_FUNC(pa_stream_connect_record);
-LOAD_FUNC(pa_stream_connect_playback);
-LOAD_FUNC(pa_path_get_filename);
-LOAD_FUNC(pa_get_binary_name);
-LOAD_FUNC(pa_threaded_mainloop_free);
-LOAD_FUNC(pa_context_errno);
-LOAD_FUNC(pa_xmalloc0);
-LOAD_FUNC(pa_stream_unref);
-LOAD_FUNC(pa_threaded_mainloop_accept);
-LOAD_FUNC(pa_stream_set_write_callback);
-LOAD_FUNC(pa_threaded_mainloop_new);
-LOAD_FUNC(pa_context_connect);
-LOAD_FUNC(pa_stream_get_buffer_attr);
-LOAD_FUNC(pa_stream_set_buffer_attr_callback);
-LOAD_FUNC(pa_stream_set_read_callback);
-LOAD_FUNC(pa_stream_set_state_callback);
-LOAD_FUNC(pa_stream_new);
-LOAD_FUNC(pa_stream_disconnect);
-LOAD_FUNC(pa_threaded_mainloop_lock);
-LOAD_FUNC(pa_channel_map_init_auto);
-LOAD_FUNC(pa_channel_map_parse);
-
-#undef LOAD_FUNC
-    }
-    ++load_count;
-} //}}}
-
-void pulse_unload(void) //{{{
-{
-    if(load_count == 0 || --load_count > 0)
-        return;
-
-#ifdef _WIN32
-    FreeLibrary(pa_handle);
-#elif defined (HAVE_DLFCN_H)
-    dlclose(pa_handle);
-#endif
-    pa_handle = NULL;
-} //}}}
-
-
-// PulseAudio Event Callbacks //{{{
-static void context_state_callback(pa_context *context, void *pdata) //{{{
-{
-    ALCdevice *Device = pdata;
-    pulse_data *data = Device->ExtraData;
-    (void)context;
-
-    if(ppa_threaded_mainloop_in_thread(data->loop))
-        ppa_threaded_mainloop_signal(data->loop, 1);
-}//}}}
-
-static void stream_state_callback(pa_stream *stream, void *pdata) //{{{
-{
-    ALCdevice *Device = pdata;
-    pulse_data *data = Device->ExtraData;
-    (void)stream;
-
-    if(ppa_threaded_mainloop_in_thread(data->loop))
-        ppa_threaded_mainloop_signal(data->loop, 1);
-}//}}}
-
-static void stream_buffer_attr_callback(pa_stream *stream, void *pdata) //{{{
-{
-    ALCdevice *Device = pdata;
-    pulse_data *data = Device->ExtraData;
-
-    SuspendContext(NULL);
-
-    data->attr = *(ppa_stream_get_buffer_attr(stream));
-    if((data->attr.tlength%data->attr.minreq) != 0)
-        AL_PRINT("new tlength (%d) is not a multiple of minreq (%d)!\n",
-                 data->attr.tlength, data->attr.minreq);
-    Device->UpdateSize = data->attr.minreq;
-    Device->NumUpdates = data->attr.tlength/data->attr.minreq;
-
-    ProcessContext(NULL);
-}//}}}
-
-static void context_state_callback2(pa_context *context, void *pdata) //{{{
-{
-    ALCdevice *Device = pdata;
-
-    if(ppa_context_get_state(context) == PA_CONTEXT_FAILED)
-    {
-        AL_PRINT("Received context failure!\n");
-        aluHandleDisconnect(Device);
-    }
-}//}}}
-
-static void stream_state_callback2(pa_stream *stream, void *pdata) //{{{
-{
-    ALCdevice *Device = pdata;
-
-    if(ppa_stream_get_state(stream) == PA_STREAM_FAILED)
-    {
-        AL_PRINT("Received stream failure!\n");
-        aluHandleDisconnect(Device);
-    }
-}//}}}
-
-//}}}
-
-// PulseAudio I/O Callbacks //{{{
-static void stream_write_callback(pa_stream *stream, size_t len, void *pdata) //{{{
-{
-    ALCdevice *Device = pdata;
-    pulse_data *data = Device->ExtraData;
-
-    len -= len%data->attr.minreq;
-    if(len > 0)
-    {
-        void *buf = ppa_xmalloc0(len);
-        aluMixData(Device, buf, len/data->frame_size);
-        ppa_stream_write(stream, buf, len, ppa_xfree, 0, PA_SEEK_RELATIVE);
-    }
-} //}}}
-
-static void stream_read_callback(pa_stream *stream, size_t length, void *pdata) //{{{
-{
-    ALCdevice *Device = pdata;
-    pulse_data *data = Device->ExtraData;
-    const void *buf;
-
-    if(ppa_stream_peek(stream, &buf, &length) < 0)
-    {
-        AL_PRINT("pa_stream_peek() failed: %s\n",
-                 ppa_strerror(ppa_context_errno(data->context)));
-        return;
-    }
-
-    assert(buf);
-    assert(length);
-
-    length /= data->frame_size;
-
-    if(data->samples < length)
-        AL_PRINT("stream_read_callback: buffer overflow!\n");
-
-    WriteRingBuffer(data->ring, buf, (length<data->samples) ? length : data->samples);
-
-    ppa_stream_drop(stream);
-} //}}}
-//}}}
-
-static ALCboolean pulse_open(ALCdevice *device, const ALCchar *device_name) //{{{
-{
-    pulse_data *data = ppa_xmalloc0(sizeof(pulse_data));
-    pa_context_state_t state;
-
-    if(ppa_get_binary_name(data->path_name, sizeof(data->path_name)))
-        data->context_name = ppa_path_get_filename(data->path_name);
-    else
-        data->context_name = "OpenAL Soft";
-
-    if(!(data->loop = ppa_threaded_mainloop_new()))
-    {
-        AL_PRINT("pa_threaded_mainloop_new() failed!\n");
-        goto out;
-    }
-
-    if(ppa_threaded_mainloop_start(data->loop) < 0)
-    {
-        AL_PRINT("pa_threaded_mainloop_start() failed\n");
-        goto out;
-    }
-
-    ppa_threaded_mainloop_lock(data->loop);
-    device->ExtraData = data;
-
-    data->context = ppa_context_new(ppa_threaded_mainloop_get_api(data->loop), data->context_name);
-    if(!data->context)
-    {
-        AL_PRINT("pa_context_new() failed: %s\n",
-                 ppa_strerror(ppa_context_errno(data->context)));
-
-        ppa_threaded_mainloop_unlock(data->loop);
-        goto out;
-    }
-
-    ppa_context_set_state_callback(data->context, context_state_callback, device);
-
-    if(ppa_context_connect(data->context, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL) < 0)
-    {
-        AL_PRINT("Context did not connect: %s\n",
-                 ppa_strerror(ppa_context_errno(data->context)));
-
-        ppa_context_unref(data->context);
-        data->context = NULL;
-
-        ppa_threaded_mainloop_unlock(data->loop);
-        goto out;
-    }
-
-    while((state=ppa_context_get_state(data->context)) != PA_CONTEXT_READY)
-    {
-        if(!PA_CONTEXT_IS_GOOD(state))
-        {
-            AL_PRINT("Context did not get ready: %s\n",
-                     ppa_strerror(ppa_context_errno(data->context)));
-
-            ppa_context_unref(data->context);
-            data->context = NULL;
-
-            ppa_threaded_mainloop_unlock(data->loop);
-            goto out;
-        }
-
-        ppa_threaded_mainloop_wait(data->loop);
-        ppa_threaded_mainloop_accept(data->loop);
-    }
-    ppa_context_set_state_callback(data->context, context_state_callback2, device);
-
-    device->szDeviceName = strdup(device_name);
-
-    ppa_threaded_mainloop_unlock(data->loop);
-    return ALC_TRUE;
-
-out:
-    if(data->loop)
-    {
-        ppa_threaded_mainloop_stop(data->loop);
-        ppa_threaded_mainloop_free(data->loop);
-    }
-
-    device->ExtraData = NULL;
-    ppa_xfree(data);
-    return ALC_FALSE;
-} //}}}
-
-static void pulse_close(ALCdevice *device) //{{{
-{
-    pulse_data *data = device->ExtraData;
-
-    ppa_threaded_mainloop_lock(data->loop);
-
-    if(data->stream)
-    {
-        ppa_stream_disconnect(data->stream);
-        ppa_stream_unref(data->stream);
-    }
-
-    ppa_context_disconnect(data->context);
-    ppa_context_unref(data->context);
-
-    ppa_threaded_mainloop_unlock(data->loop);
-
-    ppa_threaded_mainloop_stop(data->loop);
-    ppa_threaded_mainloop_free(data->loop);
-
-    DestroyRingBuffer(data->ring);
-
-    device->ExtraData = NULL;
-    ppa_xfree(data);
-} //}}}
-//}}}
-
-// OpenAL {{{
-static ALCboolean pulse_open_playback(ALCdevice *device, const ALCchar *device_name) //{{{
-{
-    if(!device_name)
-        device_name = pulse_device;
-    else if(strcmp(device_name, pulse_device) != 0)
-        return ALC_FALSE;
-
-    pulse_load();
-    if(!pa_handle)
-        return ALC_FALSE;
-
-    if(pulse_open(device, device_name) != ALC_FALSE)
-        return ALC_TRUE;
-
-    pulse_unload();
-    return ALC_FALSE;
-} //}}}
-
-static void pulse_close_playback(ALCdevice *device) //{{{
-{
-    pulse_close(device);
-    pulse_unload();
-} //}}}
-
-static ALCboolean pulse_reset_playback(ALCdevice *device) //{{{
-{
-    pulse_data *data = device->ExtraData;
-    pa_stream_state_t state;
-    pa_channel_map chanmap;
-
-    ppa_threaded_mainloop_lock(data->loop);
-
-    data->frame_size = aluBytesFromFormat(device->Format) *
-                       aluChannelsFromFormat(device->Format);
-    data->attr.minreq = data->frame_size * device->UpdateSize;
-    data->attr.prebuf = -1;
-    data->attr.maxlength = -1;
-    data->attr.fragsize = -1;
-    data->attr.tlength = data->attr.minreq * device->NumUpdates;
-    data->stream_name = "Playback Stream";
-
-    switch(aluBytesFromFormat(device->Format))
-    {
-        case 1:
-            data->spec.format = PA_SAMPLE_U8;
-            break;
-        case 2:
-            data->spec.format = PA_SAMPLE_S16NE;
-            break;
-        case 4:
-            data->spec.format = PA_SAMPLE_FLOAT32NE;
-            break;
-        default:
-            AL_PRINT("Unknown format: 0x%x\n", device->Format);
-            ppa_threaded_mainloop_unlock(data->loop);
-            return ALC_FALSE;
-    }
-    data->spec.rate = device->Frequency;
-    data->spec.channels = aluChannelsFromFormat(device->Format);
-
-    if(ppa_sample_spec_valid(&data->spec) == 0)
-    {
-        AL_PRINT("Invalid sample format\n");
-        ppa_threaded_mainloop_unlock(data->loop);
-        return ALC_FALSE;
-    }
-
-#ifdef _WIN32
-    if(!ppa_channel_map_init_auto(&chanmap, data->spec.channels, PA_CHANNEL_MAP_WAVEEX))
-    {
-        AL_PRINT("Couldn't build map for channel count (%d)!", data->spec.channels);
-        ppa_threaded_mainloop_unlock(data->loop);
-        return ALC_FALSE;
-    }
-#else
-    switch(data->spec.channels)
-    {
-    case 1:
-        ppa_channel_map_parse(&chanmap, "mono");
-        break;
-    case 2:
-        ppa_channel_map_parse(&chanmap, "front-left,front-right");
-        break;
-    case 4:
-        ppa_channel_map_parse(&chanmap, "front-left,front-right,rear-left,rear-right");
-        break;
-    case 6:
-        ppa_channel_map_parse(&chanmap, "front-left,front-right,rear-left,rear-right,front-center,lfe");
-        break;
-    case 7:
-        ppa_channel_map_parse(&chanmap, "front-left,front-right,front-center,lfe,rear-center,side-left,side-right");
-        break;
-    case 8:
-        ppa_channel_map_parse(&chanmap, "front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right");
-        break;
-    default:
-        AL_PRINT("Got unhandled channel count (%d)!", data->spec.channels);
-        ppa_threaded_mainloop_unlock(data->loop);
-        return ALC_FALSE;
-    }
-#endif
-
-    data->stream = ppa_stream_new(data->context, data->stream_name, &data->spec, &chanmap);
-    if(!data->stream)
-    {
-        AL_PRINT("pa_stream_new() failed: %s\n",
-                 ppa_strerror(ppa_context_errno(data->context)));
-
-        ppa_threaded_mainloop_unlock(data->loop);
-        return ALC_FALSE;
-    }
-
-    ppa_stream_set_state_callback(data->stream, stream_state_callback, device);
-    ppa_stream_set_write_callback(data->stream, stream_write_callback, device);
-
-    if(ppa_stream_connect_playback(data->stream, NULL, &data->attr, PA_STREAM_ADJUST_LATENCY, NULL, NULL) < 0)
-    {
-        AL_PRINT("Stream did not connect: %s\n",
-                 ppa_strerror(ppa_context_errno(data->context)));
-
-        ppa_stream_unref(data->stream);
-        data->stream = NULL;
-
-        ppa_threaded_mainloop_unlock(data->loop);
-        return ALC_FALSE;
-    }
-
-    while((state=ppa_stream_get_state(data->stream)) != PA_STREAM_READY)
-    {
-        if(!PA_STREAM_IS_GOOD(state))
-        {
-            AL_PRINT("Stream did not get ready: %s\n",
-                     ppa_strerror(ppa_context_errno(data->context)));
-
-            ppa_stream_unref(data->stream);
-            data->stream = NULL;
-
-            ppa_threaded_mainloop_unlock(data->loop);
-            return ALC_FALSE;
-        }
-
-        ppa_threaded_mainloop_wait(data->loop);
-        ppa_threaded_mainloop_accept(data->loop);
-    }
-    ppa_stream_set_state_callback(data->stream, stream_state_callback2, device);
-
-    stream_buffer_attr_callback(data->stream, device);
-    ppa_stream_set_buffer_attr_callback(data->stream, stream_buffer_attr_callback, device);
-
-    ppa_threaded_mainloop_unlock(data->loop);
-    return ALC_TRUE;
-} //}}}
-
-static void pulse_stop_playback(ALCdevice *device) //{{{
-{
-    pulse_data *data = device->ExtraData;
-
-    if(!data->stream)
-        return;
-
-    ppa_threaded_mainloop_lock(data->loop);
-
-    ppa_stream_disconnect(data->stream);
-    ppa_stream_unref(data->stream);
-    data->stream = NULL;
-
-    ppa_threaded_mainloop_unlock(data->loop);
-} //}}}
-
-
-static ALCboolean pulse_open_capture(ALCdevice *device, const ALCchar *device_name) //{{{
-{
-    pulse_data *data;
-    pa_stream_state_t state;
-
-    if(!device_name)
-        device_name = pulse_capture_device;
-    else if(strcmp(device_name, pulse_capture_device) != 0)
-        return ALC_FALSE;
-
-    pulse_load();
-    if(!pa_handle)
-        return ALC_FALSE;
-
-    if(pulse_open(device, device_name) == ALC_FALSE)
-    {
-        pulse_unload();
-        return ALC_FALSE;
-    }
-
-    data = device->ExtraData;
-    ppa_threaded_mainloop_lock(data->loop);
-
-    data->samples = device->UpdateSize * device->NumUpdates;
-    data->frame_size = aluBytesFromFormat(device->Format) *
-                       aluChannelsFromFormat(device->Format);
-
-    if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples)))
-    {
-        ppa_threaded_mainloop_unlock(data->loop);
-        pulse_close(device);
-        pulse_unload();
-        return ALC_FALSE;
-    }
-
-    data->attr.minreq = -1;
-    data->attr.prebuf = -1;
-    data->attr.maxlength = -1;
-    data->attr.tlength = -1;
-    data->attr.fragsize = data->frame_size * data->samples / 2;
-    data->stream_name = "Capture Stream";
-
-    data->spec.rate = device->Frequency;
-    data->spec.channels = aluChannelsFromFormat(device->Format);
-
-    switch(aluBytesFromFormat(device->Format))
-    {
-        case 1:
-            data->spec.format = PA_SAMPLE_U8;
-            break;
-        case 2:
-            data->spec.format = PA_SAMPLE_S16NE;
-            break;
-        case 4:
-            data->spec.format = PA_SAMPLE_FLOAT32NE;
-            break;
-        default:
-            AL_PRINT("Unknown format: 0x%x\n", device->Format);
-            ppa_threaded_mainloop_unlock(data->loop);
-            pulse_close(device);
-            pulse_unload();
-            return ALC_FALSE;
-    }
-
-    if(ppa_sample_spec_valid(&data->spec) == 0)
-    {
-        AL_PRINT("Invalid sample format\n");
-        ppa_threaded_mainloop_unlock(data->loop);
-        pulse_close(device);
-        pulse_unload();
-        return ALC_FALSE;
-    }
-
-    data->stream = ppa_stream_new(data->context, data->stream_name, &data->spec, NULL);
-    if(!data->stream)
-    {
-        AL_PRINT("pa_stream_new() failed: %s\n",
-                 ppa_strerror(ppa_context_errno(data->context)));
-
-        ppa_threaded_mainloop_unlock(data->loop);
-        pulse_close(device);
-        pulse_unload();
-        return ALC_FALSE;
-    }
-
-    ppa_stream_set_state_callback(data->stream, stream_state_callback, device);
-
-    if(ppa_stream_connect_record(data->stream, NULL, &data->attr, PA_STREAM_ADJUST_LATENCY) < 0)
-    {
-        AL_PRINT("Stream did not connect: %s\n",
-                 ppa_strerror(ppa_context_errno(data->context)));
-
-        ppa_stream_unref(data->stream);
-        ppa_threaded_mainloop_unlock(data->loop);
-
-        data->stream = NULL;
-        pulse_close(device);
-        pulse_unload();
-        return ALC_FALSE;
-    }
-
-    while((state=ppa_stream_get_state(data->stream)) != PA_STREAM_READY)
-    {
-        if(!PA_STREAM_IS_GOOD(state))
-        {
-            AL_PRINT("Stream did not get ready: %s\n",
-                     ppa_strerror(ppa_context_errno(data->context)));
-
-            ppa_stream_unref(data->stream);
-            ppa_threaded_mainloop_unlock(data->loop);
-
-            data->stream = NULL;
-            pulse_close(device);
-            pulse_unload();
-            return ALC_FALSE;
-        }
-
-        ppa_threaded_mainloop_wait(data->loop);
-        ppa_threaded_mainloop_accept(data->loop);
-    }
-    ppa_stream_set_state_callback(data->stream, stream_state_callback2, device);
-
-    ppa_threaded_mainloop_unlock(data->loop);
-    return ALC_TRUE;
-} //}}}
-
-static void pulse_close_capture(ALCdevice *device) //{{{
-{
-    pulse_close(device);
-    pulse_unload();
-} //}}}
-
-static void pulse_start_capture(ALCdevice *device) //{{{
-{
-    pulse_data *data = device->ExtraData;
-
-    ppa_threaded_mainloop_lock(data->loop);
-    ppa_stream_set_read_callback(data->stream, stream_read_callback, device);
-    ppa_threaded_mainloop_unlock(data->loop);
-} //}}}
-
-static void pulse_stop_capture(ALCdevice *device) //{{{
-{
-    pulse_data *data = device->ExtraData;
-
-    ppa_threaded_mainloop_lock(data->loop);
-    ppa_stream_set_read_callback(data->stream, NULL, NULL);
-    ppa_threaded_mainloop_unlock(data->loop);
-} //}}}
-
-static void pulse_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples) //{{{
-{
-    pulse_data *data = device->ExtraData;
-    ALCuint available = RingBufferSize(data->ring);
-
-    if(available < samples)
-        alcSetError(ALC_INVALID_VALUE);
-    else
-        ReadRingBuffer(data->ring, buffer, samples);
-} //}}}
-
-static ALCuint pulse_available_samples(ALCdevice *device) //{{{
-{
-    pulse_data *data = device->ExtraData;
-    return RingBufferSize(data->ring);
-} //}}}
-
-BackendFuncs pulse_funcs = { //{{{
-    pulse_open_playback,
-    pulse_close_playback,
-    pulse_reset_playback,
-    pulse_stop_playback,
-    pulse_open_capture,
-    pulse_close_capture,
-    pulse_start_capture,
-    pulse_stop_capture,
-    pulse_capture_samples,
-    pulse_available_samples
-}; //}}}
-
-void alc_pulse_init(BackendFuncs *func_list) //{{{
-{
-    *func_list = pulse_funcs;
-} //}}}
-
-void alc_pulse_deinit(void) //{{{
-{
-} //}}}
-
-void alc_pulse_probe(int type) //{{{
-{
-    pulse_load();
-    if(!pa_handle) return;
-
-    if(type == DEVICE_PROBE)
-        AppendDeviceList(pulse_device);
-    else if(type == ALL_DEVICE_PROBE)
-        AppendAllDeviceList(pulse_device);
-    else if(type == CAPTURE_DEVICE_PROBE)
-        AppendCaptureDeviceList(pulse_capture_device);
-
-    pulse_unload();
-} //}}}
-//}}}

Alc/solaris.c

@@ -1,292 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <sys/ioctl.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-#include <unistd.h>
-#include <errno.h>
-#include <math.h>
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-
-#include <sys/audioio.h>
-
-
-static const ALCchar solaris_device[] = "Solaris Software";
-
-typedef struct {
-    int fd;
-    volatile int killNow;
-    ALvoid *thread;
-
-    ALubyte *mix_data;
-    int data_size;
-} solaris_data;
-
-
-static ALuint SolarisProc(ALvoid *ptr)
-{
-    ALCdevice *pDevice = (ALCdevice*)ptr;
-    solaris_data *data = (solaris_data*)pDevice->ExtraData;
-    int remaining = 0;
-    ALint frameSize;
-    int wrote;
-
-    frameSize = aluChannelsFromFormat(pDevice->Format) *
-                aluBytesFromFormat(pDevice->Format);
-
-    while(!data->killNow && pDevice->Connected)
-    {
-        ALint len = data->data_size;
-        ALubyte *WritePtr = data->mix_data;
-
-        aluMixData(pDevice, WritePtr, len/frameSize);
-        while(len > 0 && !data->killNow)
-        {
-            wrote = write(data->fd, WritePtr, len);
-            if(wrote < 0)
-            {
-                if(errno != EAGAIN && errno != EWOULDBLOCK && errno != EINTR)
-                {
-                    AL_PRINT("write failed: %s\n", strerror(errno));
-                    aluHandleDisconnect(pDevice);
-                    len = 0;
-                }
-                else
-                    Sleep(1);
-                continue;
-            }
-
-            len -= wrote;
-            WritePtr += wrote;
-        }
-    }
-
-    return 0;
-}
-
-
-static ALCboolean solaris_open_playback(ALCdevice *device, const ALCchar *deviceName)
-{
-    char driver[64];
-    solaris_data *data;
-
-    strncpy(driver, GetConfigValue("solaris", "device", "/dev/audio"), sizeof(driver)-1);
-    driver[sizeof(driver)-1] = 0;
-    if(!deviceName)
-        deviceName = solaris_device;
-    else if(strcmp(deviceName, solaris_device) != 0)
-        return ALC_FALSE;
-
-    data = (solaris_data*)calloc(1, sizeof(solaris_data));
-    data->killNow = 0;
-
-    data->fd = open(driver, O_WRONLY);
-    if(data->fd == -1)
-    {
-        free(data);
-        AL_PRINT("Could not open %s: %s\n", driver, strerror(errno));
-        return ALC_FALSE;
-    }
-
-    device->szDeviceName = strdup(deviceName);
-    device->ExtraData = data;
-    return ALC_TRUE;
-}
-
-static void solaris_close_playback(ALCdevice *device)
-{
-    solaris_data *data = (solaris_data*)device->ExtraData;
-
-    close(data->fd);
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean solaris_reset_playback(ALCdevice *device)
-{
-    solaris_data *data = (solaris_data*)device->ExtraData;
-    audio_info_t info;
-    ALuint frameSize;
-    int numChannels;
-
-    AUDIO_INITINFO(&info);
-
-    switch(aluBytesFromFormat(device->Format))
-    {
-        case 1:
-            info.play.precision = 8;
-            info.play.encoding = AUDIO_ENCODING_LINEAR8;
-            break;
-        case 4:
-            switch(numChannels)
-            {
-                case 1: device->Format = AL_FORMAT_MONO16; break;
-                case 2: device->Format = AL_FORMAT_STEREO16; break;
-                case 4: device->Format = AL_FORMAT_QUAD16; break;
-                case 6: device->Format = AL_FORMAT_51CHN16; break;
-                case 7: device->Format = AL_FORMAT_61CHN16; break;
-                case 8: device->Format = AL_FORMAT_71CHN16; break;
-            }
-            /* fall-through */
-        case 2:
-            info.play.precision = 16;
-            info.play.encoding = AUDIO_ENCODING_LINEAR;
-            break;
-        default:
-            AL_PRINT("Unknown format: 0x%x\n", device->Format);
-            return ALC_FALSE;
-    }
-
-    numChannels = aluChannelsFromFormat(device->Format);
-    info.play.sample_rate = device->Frequency;
-    info.play.channels = numChannels;
-
-    frameSize = numChannels * aluBytesFromFormat(device->Format);
-    info.play.buffer_size = device->UpdateSize*device->NumUpdates * frameSize;
-
-    if(ioctl(data->fd, AUDIO_SETINFO, &info) < 0)
-    {
-        AL_PRINT("ioctl failed: %s\n", strerror(errno));
-        return ALC_FALSE;
-    }
-
-    if(aluChannelsFromFormat(device->Format) != info.play.channels)
-    {
-        AL_PRINT("Could not set %d channels, got %d instead\n", aluChannelsFromFormat(device->Format), info.play.channels);
-        return ALC_FALSE;
-    }
-
-    if(!((info.play.precision == 8 && aluBytesFromFormat(device->Format) == 1) ||
-         (info.play.precision == 16 && aluBytesFromFormat(device->Format) == 2)))
-    {
-        AL_PRINT("Could not set %d-bit output, got %d\n", aluBytesFromFormat(device->Format)*8, info.play.precision);
-        return ALC_FALSE;
-    }
-
-    device->Frequency = info.play.sample_rate;
-    device->UpdateSize = (info.play.buffer_size/device->NumUpdates) + 1;
-
-    data->data_size = device->UpdateSize * frameSize;
-    data->mix_data = calloc(1, data->data_size);
-
-    data->thread = StartThread(SolarisProc, device);
-    if(data->thread == NULL)
-    {
-        free(data->mix_data);
-        data->mix_data = NULL;
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void solaris_stop_playback(ALCdevice *device)
-{
-    solaris_data *data = (solaris_data*)device->ExtraData;
-
-    if(!data->thread)
-        return;
-
-    data->killNow = 1;
-    StopThread(data->thread);
-    data->thread = NULL;
-
-    free(data->mix_data);
-    data->mix_data = NULL;
-}
-
-
-static ALCboolean solaris_open_capture(ALCdevice *device, const ALCchar *deviceName, ALCuint frequency, ALCenum format, ALCsizei SampleSize)
-{
-    (void)device;
-    (void)deviceName;
-    (void)frequency;
-    (void)format;
-    (void)SampleSize;
-    return ALC_FALSE;
-}
-
-static void solaris_close_capture(ALCdevice *device)
-{
-    (void)device;
-}
-
-static void solaris_start_capture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void solaris_stop_capture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void solaris_capture_samples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
-{
-    (void)pDevice;
-    (void)pBuffer;
-    (void)lSamples;
-}
-
-static ALCuint solaris_available_samples(ALCdevice *pDevice)
-{
-    (void)pDevice;
-    return 0;
-}
-
-
-BackendFuncs solaris_funcs = {
-    solaris_open_playback,
-    solaris_close_playback,
-    solaris_reset_playback,
-    solaris_stop_playback,
-    solaris_open_capture,
-    solaris_close_capture,
-    solaris_start_capture,
-    solaris_stop_capture,
-    solaris_capture_samples,
-    solaris_available_samples
-};
-
-void alc_solaris_init(BackendFuncs *func_list)
-{
-    *func_list = solaris_funcs;
-}
-
-void alc_solaris_deinit(void)
-{
-}
-
-void alc_solaris_probe(ALCboolean capture)
-{
-    if(type == DEVICE_PROBE)
-        AppendDeviceList(solaris_device);
-    else if(type == ALL_DEVICE_PROBE)
-        AppendAllDeviceList(solaris_device);
-}

Alc/uhjfilter.c

@@ -0,0 +1,134 @@
+
+#include "config.h"
+
+#include "alu.h"
+#include "uhjfilter.h"
+
+/* This is the maximum number of samples processed for each inner loop
+ * iteration. */
+#define MAX_UPDATE_SAMPLES  128
+
+
+static const ALfloat Filter1Coeff[4] = {
+    0.6923878f, 0.9360654322959f, 0.9882295226860f, 0.9987488452737f
+};
+static const ALfloat Filter2Coeff[4] = {
+    0.4021921162426f, 0.8561710882420f, 0.9722909545651f, 0.9952884791278f
+};
+
+static void allpass_process(AllPassState *state, ALfloat *restrict dst, const ALfloat *restrict src, const ALfloat aa, ALsizei todo)
+{
+    ALsizei i;
+
+    if(todo > 1)
+    {
+        dst[0] = aa*(src[0] + state->y[1]) - state->x[1];
+        dst[1] = aa*(src[1] + state->y[0]) - state->x[0];
+        for(i = 2;i < todo;i++)
+            dst[i] = aa*(src[i] + dst[i-2]) - src[i-2];
+        state->x[1] = src[i-2];
+        state->x[0] = src[i-1];
+        state->y[1] = dst[i-2];
+        state->y[0] = dst[i-1];
+    }
+    else if(todo == 1)
+    {
+        dst[0] = aa*(src[0] + state->y[1]) - state->x[1];
+        state->x[1] = state->x[0];
+        state->x[0] = src[0];
+        state->y[1] = state->y[0];
+        state->y[0] = dst[0];
+    }
+}
+
+
+/* NOTE: There seems to be a bit of an inconsistency in how this encoding is
+ * supposed to work. Some references, such as
+ *
+ * http://members.tripod.com/martin_leese/Ambisonic/UHJ_file_format.html
+ *
+ * specify a pre-scaling of sqrt(2) on the W channel input, while other
+ * references, such as
+ *
+ * https://en.wikipedia.org/wiki/Ambisonic_UHJ_format#Encoding.5B1.5D
+ * and
+ * https://wiki.xiph.org/Ambisonics#UHJ_format
+ *
+ * do not. The sqrt(2) scaling is in line with B-Format decoder coefficients
+ * which include such a scaling for the W channel input, however the original
+ * source for this equation is a 1985 paper by Michael Gerzon, which does not
+ * apparently include the scaling. Applying the extra scaling creates a louder
+ * result with a narrower stereo image compared to not scaling, and I don't
+ * know which is the intended result.
+ */
+
+void EncodeUhj2(Uhj2Encoder *enc, ALfloat *restrict LeftOut, ALfloat *restrict RightOut, ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
+{
+    ALfloat D[MAX_UPDATE_SAMPLES], S[MAX_UPDATE_SAMPLES];
+    ALfloat temp[2][MAX_UPDATE_SAMPLES];
+    ALsizei base, i;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALsizei todo = mini(SamplesToDo - base, MAX_UPDATE_SAMPLES);
+
+        /* D = 0.6554516*Y */
+        for(i = 0;i < todo;i++)
+            temp[0][i] = 0.6554516f*InSamples[2][base+i];
+        allpass_process(&enc->Filter1_Y[0], temp[1], temp[0],
+                        Filter1Coeff[0]*Filter1Coeff[0], todo);
+        allpass_process(&enc->Filter1_Y[1], temp[0], temp[1],
+                        Filter1Coeff[1]*Filter1Coeff[1], todo);
+        allpass_process(&enc->Filter1_Y[2], temp[1], temp[0],
+                        Filter1Coeff[2]*Filter1Coeff[2], todo);
+        /* NOTE: Filter1 requires a 1 sample delay for the final output, so
+         * take the last processed sample from the previous run as the first
+         * output sample.
+         */
+        D[0] = enc->Filter1_Y[3].y[0];
+        allpass_process(&enc->Filter1_Y[3], temp[0], temp[1],
+                        Filter1Coeff[3]*Filter1Coeff[3], todo);
+        for(i = 1;i < todo;i++)
+            D[i] = temp[0][i-1];
+
+        /* D += j(-0.3420201*W + 0.5098604*X) */
+        for(i = 0;i < todo;i++)
+            temp[0][i] = -0.3420201f*InSamples[0][base+i] +
+                          0.5098604f*InSamples[1][base+i];
+        allpass_process(&enc->Filter2_WX[0], temp[1], temp[0],
+                        Filter2Coeff[0]*Filter2Coeff[0], todo);
+        allpass_process(&enc->Filter2_WX[1], temp[0], temp[1],
+                        Filter2Coeff[1]*Filter2Coeff[1], todo);
+        allpass_process(&enc->Filter2_WX[2], temp[1], temp[0],
+                        Filter2Coeff[2]*Filter2Coeff[2], todo);
+        allpass_process(&enc->Filter2_WX[3], temp[0], temp[1],
+                        Filter2Coeff[3]*Filter2Coeff[3], todo);
+        for(i = 0;i < todo;i++)
+            D[i] += temp[0][i];
+
+        /* S = 0.9396926*W + 0.1855740*X */
+        for(i = 0;i < todo;i++)
+            temp[0][i] = 0.9396926f*InSamples[0][base+i] +
+                         0.1855740f*InSamples[1][base+i];
+        allpass_process(&enc->Filter1_WX[0], temp[1], temp[0],
+                        Filter1Coeff[0]*Filter1Coeff[0], todo);
+        allpass_process(&enc->Filter1_WX[1], temp[0], temp[1],
+                        Filter1Coeff[1]*Filter1Coeff[1], todo);
+        allpass_process(&enc->Filter1_WX[2], temp[1], temp[0],
+                        Filter1Coeff[2]*Filter1Coeff[2], todo);
+        S[0] = enc->Filter1_WX[3].y[0];
+        allpass_process(&enc->Filter1_WX[3], temp[0], temp[1],
+                        Filter1Coeff[3]*Filter1Coeff[3], todo);
+        for(i = 1;i < todo;i++)
+            S[i] = temp[0][i-1];
+
+        /* Left = (S + D)/2.0 */
+        for(i = 0;i < todo;i++)
+            *(LeftOut++) += (S[i] + D[i]) * 0.5f;
+        /* Right = (S - D)/2.0 */
+        for(i = 0;i < todo;i++)
+            *(RightOut++) += (S[i] - D[i]) * 0.5f;
+
+        base += todo;
+    }
+}

Alc/uhjfilter.h

@@ -0,0 +1,49 @@
+#ifndef UHJFILTER_H
+#define UHJFILTER_H
+
+#include "AL/al.h"
+
+#include "alMain.h"
+
+typedef struct AllPassState {
+    ALfloat x[2]; /* Last two input samples */
+    ALfloat y[2]; /* Last two output samples */
+} AllPassState;
+
+/* Encoding 2-channel UHJ from B-Format is done as:
+ *
+ * S = 0.9396926*W + 0.1855740*X
+ * D = j(-0.3420201*W + 0.5098604*X) + 0.6554516*Y
+ *
+ * Left = (S + D)/2.0
+ * Right = (S - D)/2.0
+ *
+ * where j is a wide-band +90 degree phase shift.
+ *
+ * The phase shift is done using a Hilbert transform, described here:
+ * https://web.archive.org/web/20060708031958/http://www.biochem.oulu.fi/~oniemita/dsp/hilbert/
+ * It works using 2 sets of 4 chained filters. The first filter chain produces
+ * a phase shift of varying magnitude over a wide range of frequencies, while
+ * the second filter chain produces a phase shift 90 degrees ahead of the
+ * first over the same range.
+ *
+ * Combining these two stages requires the use of three filter chains. S-
+ * channel output uses a Filter1 chain on the W and X channel mix, while the D-
+ * channel output uses a Filter1 chain on the Y channel plus a Filter2 chain on
+ * the W and X channel mix. This results in the W and X input mix on the D-
+ * channel output having the required +90 degree phase shift relative to the
+ * other inputs.
+ */
+
+typedef struct Uhj2Encoder {
+    AllPassState Filter1_WX[4];
+    AllPassState Filter1_Y[4];
+    AllPassState Filter2_WX[4];
+} Uhj2Encoder;
+
+/* Encodes a 2-channel UHJ (stereo-compatible) signal from a B-Format input
+ * signal. The input must use FuMa channel ordering and scaling.
+ */
+void EncodeUhj2(Uhj2Encoder *enc, ALfloat *restrict LeftOut, ALfloat *restrict RightOut, ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
+
+#endif /* UHJFILTER_H */

Alc/vector.h

@@ -0,0 +1,93 @@
+#ifndef AL_VECTOR_H
+#define AL_VECTOR_H
+
+#include <stdlib.h>
+
+#include <AL/al.h>
+
+#include "almalloc.h"
+
+
+#define TYPEDEF_VECTOR(T, N) typedef struct {                                 \
+    size_t Capacity;                                                          \
+    size_t Size;                                                              \
+    T Data[];                                                                 \
+} _##N;                                                                       \
+typedef _##N* N;                                                              \
+typedef const _##N* const_##N;
+
+#define VECTOR(T) struct {                                                    \
+    size_t Capacity;                                                          \
+    size_t Size;                                                              \
+    T Data[];                                                                 \
+}*
+
+#define VECTOR_INIT(_x)       do { (_x) = NULL; } while(0)
+#define VECTOR_INIT_STATIC()  NULL
+#define VECTOR_DEINIT(_x)     do { al_free((_x)); (_x) = NULL; } while(0)
+
+#define VECTOR_RESIZE(_x, _s, _c) do {                                        \
+    size_t _size = (_s);                                                      \
+    size_t _cap = (_c);                                                       \
+    if(_size > _cap)                                                          \
+        _cap = _size;                                                         \
+                                                                              \
+    if(!(_x) && _cap == 0)                                                    \
+        break;                                                                \
+                                                                              \
+    if(((_x) ? (_x)->Capacity : 0) < _cap)                                    \
+    {                                                                         \
+        ptrdiff_t data_offset = (char*)((_x)->Data) - (char*)(_x);            \
+        size_t old_size = ((_x) ? (_x)->Size : 0);                            \
+        void *temp;                                                           \
+                                                                              \
+        temp = al_calloc(16, data_offset + sizeof((_x)->Data[0])*_cap);       \
+        assert(temp != NULL);                                                 \
+        if((_x))                                                              \
+            memcpy(((char*)temp)+data_offset, (_x)->Data,                     \
+                   sizeof((_x)->Data[0])*old_size);                           \
+                                                                              \
+        al_free((_x));                                                        \
+        (_x) = temp;                                                          \
+        (_x)->Capacity = _cap;                                                \
+    }                                                                         \
+    (_x)->Size = _size;                                                       \
+} while(0)                                                                    \
+
+#define VECTOR_CAPACITY(_x) ((_x) ? (_x)->Capacity : 0)
+#define VECTOR_SIZE(_x)     ((_x) ? (_x)->Size : 0)
+
+#define VECTOR_BEGIN(_x) ((_x) ? (_x)->Data + 0 : NULL)
+#define VECTOR_END(_x)   ((_x) ? (_x)->Data + (_x)->Size : NULL)
+
+#define VECTOR_PUSH_BACK(_x, _obj) do {      \
+    size_t _pbsize = VECTOR_SIZE(_x)+1;      \
+    VECTOR_RESIZE(_x, _pbsize, _pbsize);     \
+    (_x)->Data[(_x)->Size-1] = (_obj);       \
+} while(0)
+#define VECTOR_POP_BACK(_x) ((void)((_x)->Size--))
+
+#define VECTOR_BACK(_x)  ((_x)->Data[(_x)->Size-1])
+#define VECTOR_FRONT(_x) ((_x)->Data[0])
+
+#define VECTOR_ELEM(_x, _o) ((_x)->Data[(_o)])
+
+#define VECTOR_FOR_EACH(_t, _x, _f)  do {                                     \
+    _t *_iter = VECTOR_BEGIN((_x));                                           \
+    _t *_end = VECTOR_END((_x));                                              \
+    for(;_iter != _end;++_iter)                                               \
+        _f(_iter);                                                            \
+} while(0)
+
+#define VECTOR_FIND_IF(_i, _t, _x, _f)  do {                                  \
+    _t *_iter = VECTOR_BEGIN((_x));                                           \
+    _t *_end = VECTOR_END((_x));                                              \
+    for(;_iter != _end;++_iter)                                               \
+    {                                                                         \
+        if(_f(_iter))                                                         \
+            break;                                                            \
+    }                                                                         \
+    (_i) = _iter;                                                             \
+} while(0)
+
+#endif /* AL_VECTOR_H */

Alc/wave.c

@@ -1,347 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-
-
-typedef struct {
-    FILE *f;
-    long DataStart;
-
-    ALvoid *buffer;
-    ALuint size;
-
-    volatile int killNow;
-    ALvoid *thread;
-} wave_data;
-
-
-static const ALCchar waveDevice[] = "Wave File Writer";
-
-
-static ALuint WaveProc(ALvoid *ptr)
-{
-    ALCdevice *pDevice = (ALCdevice*)ptr;
-    wave_data *data = (wave_data*)pDevice->ExtraData;
-    ALuint frameSize;
-    ALuint now, last;
-    size_t fs;
-    ALuint avail;
-    union {
-        short s;
-        char b[sizeof(short)];
-    } uSB;
-
-    uSB.s = 1;
-    frameSize = aluBytesFromFormat(pDevice->Format) *
-                aluChannelsFromFormat(pDevice->Format);
-
-    last = timeGetTime();
-    while(!data->killNow && pDevice->Connected)
-    {
-        now = timeGetTime();
-
-        avail = (now-last) * pDevice->Frequency / 1000;
-        if(avail < pDevice->UpdateSize)
-        {
-            Sleep(1);
-            continue;
-        }
-
-        while(avail >= pDevice->UpdateSize)
-        {
-            aluMixData(pDevice, data->buffer, pDevice->UpdateSize);
-
-            if(uSB.b[0] != 1 && aluBytesFromFormat(pDevice->Format) > 1)
-            {
-                ALubyte *bytes = data->buffer;
-                ALuint i;
-
-                for(i = 0;i < data->size;i++)
-                    fputc(bytes[i^1], data->f);
-            }
-            else
-                fs = fwrite(data->buffer, frameSize, pDevice->UpdateSize,
-                            data->f);
-            if(ferror(data->f))
-            {
-                AL_PRINT("Error writing to file\n");
-                aluHandleDisconnect(pDevice);
-                break;
-            }
-
-            avail -= pDevice->UpdateSize;
-        }
-        last = now;
-    }
-
-    return 0;
-}
-
-static ALCboolean wave_open_playback(ALCdevice *device, const ALCchar *deviceName)
-{
-    wave_data *data;
-    const char *fname;
-
-    fname = GetConfigValue("wave", "file", "");
-    if(!fname[0])
-        return ALC_FALSE;
-
-    if(!deviceName)
-        deviceName = waveDevice;
-    else if(strcmp(deviceName, waveDevice) != 0)
-        return ALC_FALSE;
-
-    data = (wave_data*)calloc(1, sizeof(wave_data));
-
-    data->f = fopen(fname, "wb");
-    if(!data->f)
-    {
-        free(data);
-        AL_PRINT("Could not open file '%s': %s\n", fname, strerror(errno));
-        return ALC_FALSE;
-    }
-
-    device->szDeviceName = strdup(deviceName);
-    device->ExtraData = data;
-    return ALC_TRUE;
-}
-
-static void wave_close_playback(ALCdevice *device)
-{
-    wave_data *data = (wave_data*)device->ExtraData;
-
-    fclose(data->f);
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean wave_reset_playback(ALCdevice *device)
-{
-    wave_data *data = (wave_data*)device->ExtraData;
-    ALuint channels, bits, i;
-
-    fseek(data->f, 0, SEEK_SET);
-    clearerr(data->f);
-
-    bits = aluBytesFromFormat(device->Format) * 8;
-    channels = aluChannelsFromFormat(device->Format);
-    switch(bits)
-    {
-        case 8:
-        case 16:
-        case 32:
-            if(channels == 0)
-            {
-                AL_PRINT("Unknown format?! %x\n", device->Format);
-                return ALC_FALSE;
-            }
-            break;
-
-        default:
-            AL_PRINT("Unknown format?! %x\n", device->Format);
-            return ALC_FALSE;
-    }
-
-    fprintf(data->f, "RIFF");
-    fputc(0, data->f); // 'RIFF' header len; filled in at close
-    fputc(0, data->f);
-    fputc(0, data->f);
-    fputc(0, data->f);
-
-    fprintf(data->f, "WAVE");
-
-    fprintf(data->f, "fmt ");
-    fputc(16, data->f); // 'fmt ' header len; 16 bytes for PCM
-    fputc(0, data->f);
-    fputc(0, data->f);
-    fputc(0, data->f);
-    // 16-bit val, format type id (PCM: 1)
-    fputc(1, data->f);
-    fputc(0, data->f);
-    // 16-bit val, channel count
-    fputc(channels&0xff, data->f);
-    fputc((channels>>8)&0xff, data->f);
-    // 32-bit val, frequency
-    fputc(device->Frequency&0xff, data->f);
-    fputc((device->Frequency>>8)&0xff, data->f);
-    fputc((device->Frequency>>16)&0xff, data->f);
-    fputc((device->Frequency>>24)&0xff, data->f);
-    // 32-bit val, bytes per second
-    i = device->Frequency * channels * bits / 8;
-    fputc(i&0xff, data->f);
-    fputc((i>>8)&0xff, data->f);
-    fputc((i>>16)&0xff, data->f);
-    fputc((i>>24)&0xff, data->f);
-    // 16-bit val, frame size
-    i = channels * bits / 8;
-    fputc(i&0xff, data->f);
-    fputc((i>>8)&0xff, data->f);
-    // 16-bit val, bits per sample
-    fputc(bits&0xff, data->f);
-    fputc((bits>>8)&0xff, data->f);
-
-    fprintf(data->f, "data");
-    fputc(0, data->f); // 'data' header len; filled in at close
-    fputc(0, data->f);
-    fputc(0, data->f);
-    fputc(0, data->f);
-
-    if(ferror(data->f))
-    {
-        AL_PRINT("Error writing header: %s\n", strerror(errno));
-        return ALC_FALSE;
-    }
-
-    data->DataStart = ftell(data->f);
-
-    data->size = device->UpdateSize * channels * bits / 8;
-    data->buffer = malloc(data->size);
-    if(!data->buffer)
-    {
-        AL_PRINT("buffer malloc failed\n");
-        return ALC_FALSE;
-    }
-
-    data->thread = StartThread(WaveProc, device);
-    if(data->thread == NULL)
-    {
-        free(data->buffer);
-        data->buffer = NULL;
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void wave_stop_playback(ALCdevice *device)
-{
-    wave_data *data = (wave_data*)device->ExtraData;
-    ALuint dataLen;
-    long size;
-
-    if(!data->thread)
-        return;
-
-    data->killNow = 1;
-    StopThread(data->thread);
-    data->thread = NULL;
-
-    free(data->buffer);
-    data->buffer = NULL;
-
-    size = ftell(data->f);
-    if(size > 0)
-    {
-        dataLen = size - data->DataStart;
-        if(fseek(data->f, data->DataStart-4, SEEK_SET) == 0)
-        {
-            fputc(dataLen&0xff, data->f); // 'data' header len
-            fputc((dataLen>>8)&0xff, data->f);
-            fputc((dataLen>>16)&0xff, data->f);
-            fputc((dataLen>>24)&0xff, data->f);
-        }
-        if(fseek(data->f, 4, SEEK_SET) == 0)
-        {
-            size -= 8;
-            fputc(size&0xff, data->f); // 'WAVE' header len
-            fputc((size>>8)&0xff, data->f);
-            fputc((size>>16)&0xff, data->f);
-            fputc((size>>24)&0xff, data->f);
-        }
-    }
-}
-
-
-static ALCboolean wave_open_capture(ALCdevice *pDevice, const ALCchar *deviceName)
-{
-    (void)pDevice;
-    (void)deviceName;
-    return ALC_FALSE;
-}
-
-static void wave_close_capture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void wave_start_capture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void wave_stop_capture(ALCdevice *pDevice)
-{
-    (void)pDevice;
-}
-
-static void wave_capture_samples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
-{
-    (void)pDevice;
-    (void)pBuffer;
-    (void)lSamples;
-}
-
-static ALCuint wave_available_samples(ALCdevice *pDevice)
-{
-    (void)pDevice;
-    return 0;
-}
-
-
-BackendFuncs wave_funcs = {
-    wave_open_playback,
-    wave_close_playback,
-    wave_reset_playback,
-    wave_stop_playback,
-    wave_open_capture,
-    wave_close_capture,
-    wave_start_capture,
-    wave_stop_capture,
-    wave_capture_samples,
-    wave_available_samples
-};
-
-void alc_wave_init(BackendFuncs *func_list)
-{
-    *func_list = wave_funcs;
-}
-
-void alc_wave_deinit(void)
-{
-}
-
-void alc_wave_probe(int type)
-{
-    if(*(GetConfigValue("wave", "file", "")) == 0)
-        return;
-
-    if(type == DEVICE_PROBE)
-        AppendDeviceList(waveDevice);
-    else if(type == ALL_DEVICE_PROBE)
-        AppendAllDeviceList(waveDevice);
-}

Alc/winmm.c

@@ -1,470 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#define _WIN32_WINNT 0x0500
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-
-#include <windows.h>
-#include <mmsystem.h>
-
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-
-
-typedef struct {
-    // MMSYSTEM Capture Device
-    ALboolean        bWaveInShutdown;
-    HANDLE           hWaveInHdrEvent;
-    HANDLE           hWaveInThreadEvent;
-    HANDLE           hWaveInThread;
-    DWORD            ulWaveInThreadID;
-    ALint            lWaveInBuffersCommitted;
-    HWAVEIN          hWaveInHandle;
-    WAVEHDR          WaveInBuffer[4];
-    ALCchar          *pCapturedSampleData;
-    ALuint           ulCapturedDataSize;
-    ALuint           ulReadCapturedDataPos;
-    ALuint           ulWriteCapturedDataPos;
-} WinMMData;
-
-
-static ALCchar **CaptureDeviceList;
-static ALuint  NumCaptureDevices;
-
-/*
-    WaveInProc
-
-    Posts a message to 'CaptureThreadProc' everytime a WaveIn Buffer is completed and
-    returns to the application (with more data)
-*/
-static void CALLBACK WaveInProc(HWAVEIN hDevice,UINT uMsg,DWORD_PTR dwInstance,DWORD_PTR dwParam1,DWORD_PTR dwParam2)
-{
-    ALCdevice *pDevice = (ALCdevice *)dwInstance;
-    WinMMData *pData = pDevice->ExtraData;
-
-    (void)hDevice;
-    (void)dwParam2;
-
-    if ((uMsg==WIM_DATA))
-    {
-        // Decrement number of buffers in use
-        pData->lWaveInBuffersCommitted--;
-
-        if (pData->bWaveInShutdown == AL_FALSE)
-        {
-            // Notify Wave Processor Thread that a Wave Header has returned
-            PostThreadMessage(pData->ulWaveInThreadID,uMsg,0,dwParam1);
-        }
-        else
-        {
-            if (pData->lWaveInBuffersCommitted == 0)
-            {
-                // Signal Wave Buffers Returned event
-                if (pData->hWaveInHdrEvent)
-                    SetEvent(pData->hWaveInHdrEvent);
-
-                // Post 'Quit' Message to WaveIn Processor Thread
-                PostThreadMessage(pData->ulWaveInThreadID,WM_QUIT,0,0);
-            }
-        }
-    }
-}
-
-/*
-    CaptureThreadProc
-
-    Used by "MMSYSTEM" Device.  Called when a WaveIn buffer had been filled with new
-    audio data.
-*/
-DWORD WINAPI CaptureThreadProc(LPVOID lpParameter)
-{
-    ALCdevice *pDevice = (ALCdevice*)lpParameter;
-    WinMMData *pData = pDevice->ExtraData;
-    ALuint ulOffset, ulMaxSize, ulSection;
-    LPWAVEHDR pWaveHdr;
-    MSG msg;
-
-    while (GetMessage(&msg, NULL, 0, 0))
-    {
-        if ((msg.message==WIM_DATA)&&(!pData->bWaveInShutdown))
-        {
-            SuspendContext(NULL);
-
-            pWaveHdr = ((LPWAVEHDR)msg.lParam);
-
-            // Calculate offset in local buffer to write data to
-            ulOffset = pData->ulWriteCapturedDataPos % pData->ulCapturedDataSize;
-
-            if ((ulOffset + pWaveHdr->dwBytesRecorded) > pData->ulCapturedDataSize)
-            {
-                ulSection = pData->ulCapturedDataSize - ulOffset;
-                memcpy(pData->pCapturedSampleData + ulOffset, pWaveHdr->lpData, ulSection);
-                memcpy(pData->pCapturedSampleData, pWaveHdr->lpData + ulSection, pWaveHdr->dwBytesRecorded - ulSection);
-            }
-            else
-            {
-                memcpy(pData->pCapturedSampleData + ulOffset, pWaveHdr->lpData, pWaveHdr->dwBytesRecorded);
-            }
-
-            pData->ulWriteCapturedDataPos += pWaveHdr->dwBytesRecorded;
-
-            if (pData->ulWriteCapturedDataPos > (pData->ulReadCapturedDataPos + pData->ulCapturedDataSize))
-            {
-                // Application has not read enough audio data from the capture buffer so data has been
-                // overwritten.  Reset ReadPosition.
-                pData->ulReadCapturedDataPos = pData->ulWriteCapturedDataPos - pData->ulCapturedDataSize;
-            }
-
-            // To prevent an over-flow prevent the offset values from getting too large
-            ulMaxSize = pData->ulCapturedDataSize << 4;
-            if ((pData->ulReadCapturedDataPos > ulMaxSize) && (pData->ulWriteCapturedDataPos > ulMaxSize))
-            {
-                pData->ulReadCapturedDataPos -= ulMaxSize;
-                pData->ulWriteCapturedDataPos -= ulMaxSize;
-            }
-
-            // Send buffer back to capture more data
-            waveInAddBuffer(pData->hWaveInHandle,pWaveHdr,sizeof(WAVEHDR));
-            pData->lWaveInBuffersCommitted++;
-
-            ProcessContext(NULL);
-        }
-    }
-
-    // Signal Wave Thread completed event
-    if (pData->hWaveInThreadEvent)
-        SetEvent(pData->hWaveInThreadEvent);
-
-    ExitThread(0);
-
-    return 0;
-}
-
-
-static ALCboolean WinMMOpenPlayback(ALCdevice *device, const ALCchar *deviceName)
-{
-    (void)device;
-    (void)deviceName;
-    return ALC_FALSE;
-}
-
-static void WinMMClosePlayback(ALCdevice *device)
-{
-    (void)device;
-}
-
-
-static ALCboolean WinMMOpenCapture(ALCdevice *pDevice, const ALCchar *deviceName)
-{
-    WAVEFORMATEX wfexCaptureFormat;
-    WinMMData *pData = NULL;
-    ALint lDeviceID = 0;
-    ALint lBufferSize;
-    ALuint i;
-
-    // Find the Device ID matching the deviceName if valid
-    if (deviceName)
-    {
-        for(i = 0;i < NumCaptureDevices;i++)
-        {
-            if (!strcmp(deviceName, CaptureDeviceList[i]))
-            {
-                lDeviceID = i;
-                break;
-            }
-        }
-        if(i == NumCaptureDevices)
-            return ALC_FALSE;
-    }
-
-    pData = calloc(1, sizeof(*pData));
-    if(!pData)
-    {
-        alcSetError(ALC_OUT_OF_MEMORY);
-        return ALC_FALSE;
-    }
-
-    memset(&wfexCaptureFormat, 0, sizeof(WAVEFORMATEX));
-    wfexCaptureFormat.wFormatTag = WAVE_FORMAT_PCM;
-    wfexCaptureFormat.nChannels = aluChannelsFromFormat(pDevice->Format);
-    wfexCaptureFormat.wBitsPerSample = aluBytesFromFormat(pDevice->Format) * 8;
-    wfexCaptureFormat.nBlockAlign = wfexCaptureFormat.wBitsPerSample *
-                                    wfexCaptureFormat.nChannels / 8;
-    wfexCaptureFormat.nSamplesPerSec = pDevice->Frequency;
-    wfexCaptureFormat.nAvgBytesPerSec = wfexCaptureFormat.nSamplesPerSec *
-                                        wfexCaptureFormat.nBlockAlign;
-    wfexCaptureFormat.cbSize = 0;
-
-    if (waveInOpen(&pData->hWaveInHandle, lDeviceID, &wfexCaptureFormat, (DWORD_PTR)&WaveInProc, (DWORD_PTR)pDevice, CALLBACK_FUNCTION) != MMSYSERR_NOERROR)
-        goto failure;
-
-    pData->hWaveInHdrEvent = CreateEvent(NULL, AL_TRUE, AL_FALSE, "WaveInAllHeadersReturned");
-    if (pData->hWaveInHdrEvent == NULL)
-        goto failure;
-
-    pData->hWaveInThreadEvent = CreateEvent(NULL, AL_TRUE, AL_FALSE, "WaveInThreadDestroyed");
-    if (pData->hWaveInThreadEvent == NULL)
-        goto failure;
-
-    // Allocate circular memory buffer for the captured audio
-    pData->ulCapturedDataSize = pDevice->UpdateSize*pDevice->NumUpdates *
-                                wfexCaptureFormat.nBlockAlign;
-
-    // Make sure circular buffer is at least 100ms in size (and an exact multiple of
-    // the block alignment
-    if (pData->ulCapturedDataSize < (wfexCaptureFormat.nAvgBytesPerSec / 10))
-    {
-        pData->ulCapturedDataSize = wfexCaptureFormat.nAvgBytesPerSec / 10;
-        pData->ulCapturedDataSize -= (pData->ulCapturedDataSize % wfexCaptureFormat.nBlockAlign);
-    }
-
-    pData->pCapturedSampleData = (ALCchar*)malloc(pData->ulCapturedDataSize);
-    pData->lWaveInBuffersCommitted=0;
-
-    // Create 4 Buffers of 50ms each
-    lBufferSize = wfexCaptureFormat.nAvgBytesPerSec / 20;
-    lBufferSize -= (lBufferSize % wfexCaptureFormat.nBlockAlign);
-
-    for (i=0;i<4;i++)
-    {
-        memset(&pData->WaveInBuffer[i], 0, sizeof(WAVEHDR));
-        pData->WaveInBuffer[i].dwBufferLength = lBufferSize;
-        pData->WaveInBuffer[i].lpData = calloc(1,pData->WaveInBuffer[i].dwBufferLength);
-        pData->WaveInBuffer[i].dwFlags = 0;
-        pData->WaveInBuffer[i].dwLoops = 0;
-        waveInPrepareHeader(pData->hWaveInHandle, &pData->WaveInBuffer[i], sizeof(WAVEHDR));
-        waveInAddBuffer(pData->hWaveInHandle, &pData->WaveInBuffer[i], sizeof(WAVEHDR));
-        pData->lWaveInBuffersCommitted++;
-    }
-
-    pData->ulReadCapturedDataPos = 0;
-    pData->ulWriteCapturedDataPos = 0;
-
-    pDevice->ExtraData = pData;
-
-    pData->hWaveInThread = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)CaptureThreadProc, (LPVOID)pDevice, 0, &pData->ulWaveInThreadID);
-    if (pData->hWaveInThread == NULL)
-        goto failure;
-
-    pDevice->szDeviceName = strdup(CaptureDeviceList[lDeviceID]);
-    return ALC_TRUE;
-
-failure:
-    for (i=0;i<4;i++)
-    {
-        if(pData->WaveInBuffer[i].lpData)
-        {
-            waveInUnprepareHeader(pData->hWaveInHandle, &pData->WaveInBuffer[i], sizeof(WAVEHDR));
-            free(pData->WaveInBuffer[i].lpData);
-        }
-    }
-
-    free(pData->pCapturedSampleData);
-    if(pData->hWaveInHandle)
-        waveInClose(pData->hWaveInHandle);
-    if(pData->hWaveInThread)
-        CloseHandle(pData->hWaveInThread);
-    if (pData->hWaveInHdrEvent)
-        CloseHandle(pData->hWaveInHdrEvent);
-    if (pData->hWaveInThreadEvent)
-        CloseHandle(pData->hWaveInThreadEvent);
-
-    free(pData);
-    pDevice->ExtraData = NULL;
-    return ALC_FALSE;
-}
-
-static void WinMMCloseCapture(ALCdevice *pDevice)
-{
-    WinMMData *pData = (WinMMData*)pDevice->ExtraData;
-    int i;
-
-    // Call waveOutReset to shutdown wave device
-    pData->bWaveInShutdown = AL_TRUE;
-    waveInReset(pData->hWaveInHandle);
-
-    // Wait for signal that all Wave Buffers have returned
-    WaitForSingleObjectEx(pData->hWaveInHdrEvent, 5000, FALSE);
-
-    // Wait for signal that Wave Thread has been destroyed
-    WaitForSingleObjectEx(pData->hWaveInThreadEvent, 5000, FALSE);
-
-    // Release the wave buffers
-    for (i=0;i<4;i++)
-    {
-        waveInUnprepareHeader(pData->hWaveInHandle, &pData->WaveInBuffer[i], sizeof(WAVEHDR));
-        free(pData->WaveInBuffer[i].lpData);
-    }
-
-    // Free Audio Buffer data
-    free(pData->pCapturedSampleData);
-    pData->pCapturedSampleData = NULL;
-
-    // Close the Wave device
-    waveInClose(pData->hWaveInHandle);
-    pData->hWaveInHandle = 0;
-
-    CloseHandle(pData->hWaveInThread);
-    pData->hWaveInThread = 0;
-
-    if (pData->hWaveInHdrEvent)
-    {
-        CloseHandle(pData->hWaveInHdrEvent);
-        pData->hWaveInHdrEvent = 0;
-    }
-
-    if (pData->hWaveInThreadEvent)
-    {
-        CloseHandle(pData->hWaveInThreadEvent);
-        pData->hWaveInThreadEvent = 0;
-    }
-
-    free(pData);
-    pDevice->ExtraData = NULL;
-}
-
-static void WinMMStartCapture(ALCdevice *pDevice)
-{
-    WinMMData *pData = (WinMMData*)pDevice->ExtraData;
-    waveInStart(pData->hWaveInHandle);
-}
-
-static void WinMMStopCapture(ALCdevice *pDevice)
-{
-    WinMMData *pData = (WinMMData*)pDevice->ExtraData;
-    waveInStop(pData->hWaveInHandle);
-}
-
-static void WinMMCaptureSamples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
-{
-    WinMMData *pData = (WinMMData*)pDevice->ExtraData;
-    ALuint ulSamples = (unsigned long)lSamples;
-    ALuint ulBytes, ulBytesToCopy;
-    ALuint ulCapturedSamples;
-    ALuint ulReadOffset;
-    ALuint frameSize = aluBytesFromFormat(pDevice->Format) *
-                       aluChannelsFromFormat(pDevice->Format);
-
-    // Check that we have the requested numbers of Samples
-    ulCapturedSamples = (pData->ulWriteCapturedDataPos -
-                         pData->ulReadCapturedDataPos) /
-                        frameSize;
-    if(ulSamples > ulCapturedSamples)
-    {
-        alcSetError(ALC_INVALID_VALUE);
-        return;
-    }
-
-    ulBytes = ulSamples * frameSize;
-
-    // Get Read Offset
-    ulReadOffset = (pData->ulReadCapturedDataPos % pData->ulCapturedDataSize);
-
-    // Check for wrap-around condition
-    if ((ulReadOffset + ulBytes) > pData->ulCapturedDataSize)
-    {
-        // Copy data from last Read position to end of data
-        ulBytesToCopy = pData->ulCapturedDataSize - ulReadOffset;
-        memcpy(pBuffer, pData->pCapturedSampleData + ulReadOffset, ulBytesToCopy);
-
-        // Copy rest of the data from the start of the captured data
-        memcpy(((char *)pBuffer) + ulBytesToCopy, pData->pCapturedSampleData, ulBytes - ulBytesToCopy);
-    }
-    else
-    {
-        // Copy data from the read position in the captured data
-        memcpy(pBuffer, pData->pCapturedSampleData + ulReadOffset, ulBytes);
-    }
-
-    // Update Read Position
-    pData->ulReadCapturedDataPos += ulBytes;
-}
-
-static ALCuint WinMMAvailableSamples(ALCdevice *pDevice)
-{
-    WinMMData *pData = (WinMMData*)pDevice->ExtraData;
-    ALCuint lCapturedBytes = (pData->ulWriteCapturedDataPos - pData->ulReadCapturedDataPos);
-    return lCapturedBytes / (aluBytesFromFormat(pDevice->Format) *
-                             aluChannelsFromFormat(pDevice->Format));
-}
-
-
-BackendFuncs WinMMFuncs = {
-    WinMMOpenPlayback,
-    WinMMClosePlayback,
-    NULL,
-    NULL,
-    WinMMOpenCapture,
-    WinMMCloseCapture,
-    WinMMStartCapture,
-    WinMMStopCapture,
-    WinMMCaptureSamples,
-    WinMMAvailableSamples
-};
-
-void alcWinMMInit(BackendFuncs *FuncList)
-{
-    *FuncList = WinMMFuncs;
-}
-
-void alcWinMMDeinit()
-{
-    ALuint lLoop;
-
-    for(lLoop = 0; lLoop < NumCaptureDevices; lLoop++)
-        free(CaptureDeviceList[lLoop]);
-    free(CaptureDeviceList);
-    CaptureDeviceList = NULL;
-
-    NumCaptureDevices = 0;
-}
-
-void alcWinMMProbe(int type)
-{
-    ALuint lLoop;
-
-    if(type != CAPTURE_DEVICE_PROBE)
-        return;
-
-    for(lLoop = 0; lLoop < NumCaptureDevices; lLoop++)
-        free(CaptureDeviceList[lLoop]);
-
-    NumCaptureDevices = waveInGetNumDevs();
-    CaptureDeviceList = realloc(CaptureDeviceList, sizeof(ALCchar*) * NumCaptureDevices);
-    for(lLoop = 0; lLoop < NumCaptureDevices; lLoop++)
-    {
-        WAVEINCAPS WaveInCaps;
-
-        if(waveInGetDevCaps(lLoop, &WaveInCaps, sizeof(WAVEINCAPS)) == MMSYSERR_NOERROR)
-        {
-            char name[128];
-            snprintf(name, sizeof(name), "WaveIn on %s", WaveInCaps.szPname);
-            AppendCaptureDeviceList(name);
-            CaptureDeviceList[lLoop] = strdup(name);
-        }
-        else
-            CaptureDeviceList[lLoop] = strdup("");
-    }
-}

COPYING

@@ -1,17 +1,15 @@
- 
-		  GNU LIBRARY GENERAL PUBLIC LICENSE
-		       Version 2, June 1991
-
+                  GNU LIBRARY GENERAL PUBLIC LICENSE
+                       Version 2, June 1991
 
  Copyright (C) 1991 Free Software Foundation, Inc.
-                    675 Mass Ave, Cambridge, MA 02139, USA
+ 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
  Everyone is permitted to copy and distribute verbatim copies
  of this license document, but changing it is not allowed.
 
 [This is the first released version of the library GPL.  It is
  numbered 2 because it goes with version 2 of the ordinary GPL.]
 
-			    Preamble
+                            Preamble
 
   The licenses for most software are designed to take away your
 freedom to share and change it.  By contrast, the GNU General Public
@@ -101,7 +99,7 @@ works together with the library.
   Note that it is possible for a library to be covered by the ordinary
 General Public License rather than by this special one.
 
-		  GNU LIBRARY GENERAL PUBLIC LICENSE
+                  GNU LIBRARY GENERAL PUBLIC LICENSE
    TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
 
   0. This License Agreement applies to any software library which
@@ -413,7 +411,7 @@ decision will be guided by the two goals of preserving the free status
 of all derivatives of our free software and of promoting the sharing
 and reuse of software generally.
 
-			    NO WARRANTY
+                            NO WARRANTY
 
   15. BECAUSE THE LIBRARY IS LICENSED FREE OF CHARGE, THERE IS NO
 WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW.
@@ -436,9 +434,9 @@ FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF
 SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
 DAMAGES.
 
-		     END OF TERMS AND CONDITIONS
+                     END OF TERMS AND CONDITIONS
 
-     Appendix: How to Apply These Terms to Your New Libraries
+           How to Apply These Terms to Your New Libraries
 
   If you develop a new library, and you want it to be of the greatest
 possible use to the public, we recommend making it free software that
@@ -465,8 +463,8 @@ convey the exclusion of warranty; and each file should have at least the
     Library General Public License for more details.
 
     You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
 
 Also add information on how to contact you by electronic and paper mail.
 
@@ -481,4 +479,3 @@ necessary.  Here is a sample; alter the names:
   Ty Coon, President of Vice
 
 That's all there is to it!
-

ChangeLog

@@ -0,0 +1,271 @@
+openal-soft-1.18.0:
+
+    Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
+
+    Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
+    AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
+
+    Implemented 3D processing for some effects. Currently implemented for
+    Reverb, Compressor, Equalizer, and Ring Modulator.
+
+    Implemented 2-channel UHJ output encoding. This needs to be enabled with a
+    config option to be used.
+
+    Implemented dual-band processing for high-quality ambisonic decoding.
+
+    Implemented distance-compensation for surround sound output.
+
+    Implemented near-field emulation and compensation with ambisonic rendering.
+    Currently only applies when using the high-quality ambisonic decoder or
+    ambisonic output, with appropriate config options.
+
+    Implemented an output limiter to reduce the amount of distortion from
+    clipping.
+
+    Implemented dithering for 8-bit and 16-bit output.
+
+    Implemented a config option to select a preferred HRTF.
+
+    Implemented a run-time check for NEON extensions using /proc/cpuinfo.
+
+    Implemented experimental capture support for the OpenSL backend.
+
+    Fixed building on compilers with NEON support but don't default to having
+    NEON enabled.
+
+    Fixed support for JACK on Windows.
+
+    Fixed starting a source while alcSuspendContext is in effect.
+
+    Fixed detection of headsets as headphones, with MMDevAPI.
+
+    Added support for AmbDec config files, for custom ambisonic decoder
+    configurations. Version 3 files only.
+
+    Added backend-specific options to alsoft-config.
+
+    Added first-, second-, and third-order ambisonic output formats. Currently
+    only works with backends that don't rely on channel labels, like JACK,
+    ALSA, and OSS.
+
+    Added a build option to embed the default HRTFs into the lib.
+
+    Added AmbDec presets to enable high-quality ambisonic decoding.
+
+    Added an AmbDec preset for 3D7.1 speaker setups.
+
+    Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
+    the provided ambdec presets.
+
+    Added the ability for MMDevAPI to open devices given a Device ID or GUID
+    string.
+
+    Added an option to the example apps to open a specific device.
+
+    Increased the maximum auxiliary send limit to 16 (up from 4). Requires
+    requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
+    attribute.
+
+    Increased the default auxiliary effect slot count to 64 (up from 4).
+
+    Reduced the default period count to 3 (down from 4).
+
+    Slightly improved automatic naming for enumerated HRTFs.
+
+    Improved B-Format decoding with HRTF output.
+
+    Improved internal property handling for better batching behavior.
+
+    Improved performance of certain filter uses.
+
+    Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
+    extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
+
+openal-soft-1.17.2:
+
+    Implemented device enumeration for OSSv4.
+
+    Fixed building on OSX.
+
+    Fixed building on non-Windows systems without POSIX-2008.
+
+    Fixed Dedicated Dialog and Dedicated LFE effect output.
+
+    Added a build option to override the share install dir.
+
+    Added a build option to static-link libgcc for MinGW.
+
+openal-soft-1.17.1:
+
+    Fixed building with JACK and without PulseAudio.
+
+    Fixed building on FreeBSD.
+
+    Fixed the ALSA backend's allow-resampler option.
+
+    Fixed handling of inexact ALSA period counts.
+
+    Altered device naming scheme on Windows backends to better match other
+    drivers.
+
+    Updated the CoreAudio backend to use the AudioComponent API. This clears up
+    deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
+
+openal-soft-1.17.0:
+
+    Implemented a JACK playback backend.
+
+    Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
+
+    Implemented the ALC_SOFT_HRTF extension.
+
+    Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
+
+    Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
+    24-point Sinc resampling, and performs anti-aliasing.
+
+    Implemented B-Format output support for the wave file writer. This creates
+    FuMa-style first-order Ambisonics wave files (AMB format).
+
+    Implemented a stereo-mode config option for treating stereo modes as either
+    speakers or headphones.
+
+    Implemented per-device configuration options.
+
+    Fixed handling of PulseAudio and MMDevAPI devices that have identical
+    descriptions.
+
+    Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
+
+    Fixed logging of Unicode characters on Windows.
+
+    Fixed 5.1 surround sound channels. By default it will now use the side
+    channels for the surround output. A configuration using rear channels is
+    still available.
+
+    Fixed the QSA backend potentially altering the capture format.
+
+    Fixed detecting MMDevAPI's default device.
+
+    Fixed returning the default capture device name.
+
+    Fixed mixing property calculations when deferring context updates.
+
+    Altered the behavior of alcSuspendContext and alcProcessContext to better
+    match certain Windows drivers.
+
+    Altered the panning algorithm, utilizing Ambisonics for better side and
+    back positioning cues with surround sound output.
+
+    Improved support for certain older Windows apps.
+
+    Improved the alffplay example to support surround sound streams.
+
+    Improved support for building as a sub-project.
+
+    Added an HRTF playback example.
+
+    Added a tone generator output test.
+
+    Added a toolchain to help with cross-compiling to Android.
+
+openal-soft-1.16.0:
+
+    Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
+    effects.
+
+    Implemented high-pass and band-pass EFX filters.
+
+    Implemented the high-pass filter for the EAXReverb effect.
+
+    Implemented SSE2 and SSE4.1 linear resamplers.
+
+    Implemented Neon-enhanced non-HRTF mixers.
+
+    Implemented a QSA backend, for QNX.
+
+    Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
+    AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
+    extensions.
+
+    Fixed resetting mmdevapi backend devices.
+
+    Fixed clamping when converting 32-bit float samples to integer.
+
+    Fixed modulation range in the Modulator effect.
+
+    Several fixes for the OpenSL playback backend.
+
+    Fixed device specifier names that have Unicode characters on Windows.
+
+    Added support for filenames and paths with Unicode (UTF-8) characters on
+    Windows.
+
+    Added support for alsoft.conf config files found in XDG Base Directory
+    Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
+    defaults) on non-Windows systems.
+
+    Added a GUI configuration utility (requires Qt 4.8).
+
+    Added support for environment variable expansion in config options (not
+    keys or section names).
+
+    Added an example that uses SDL2 and ffmpeg.
+
+    Modified examples to use SDL_sound.
+
+    Modified CMake config option names for better sorting.
+
+    HRTF data sets specified in the hrtf_tables config option may now be
+    relative or absolute filenames.
+
+    Made the default HRTF data set an external file, and added a data set for
+    48khz playback in addition to 44.1khz.
+
+    Added support for C11 atomic methods.
+
+    Improved support for some non-GNU build systems.
+
+openal-soft-1.15.1:
+
+    Fixed a regression with retrieving the source's AL_GAIN property.
+
+openal-soft-1.15:
+
+    Fixed device enumeration with the OSS backend.
+
+    Reorganized internal mixing logic, so unneeded steps can potentially be
+    skipped for better performance.
+
+    Removed the lookup table for calculating the mixing pans. The panning is
+    now calculated directly for better precision.
+
+    Improved the panning of stereo source channels when using stereo output.
+
+    Improved source filter quality on send paths.
+
+    Added a config option to allow PulseAudio to move streams between devices.
+
+    The PulseAudio backend will now attempt to spawn a server by default.
+
+    Added a workaround for a DirectSound bug relating to float32 output.
+
+    Added SSE-based mixers, for HRTF and non-HRTF mixing.
+
+    Added support for the new AL_SOFT_source_latency extension.
+
+    Improved ALSA capture by avoiding an extra buffer when using sizes
+    supported by the underlying device.
+
+    Improved the makehrtf utility to support new options and input formats.
+
+    Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
+    the header includes can optionally be omitted.
+
+    Added a couple example code programs to show how to apply reverb, and
+    retrieve latency.
+
+    The configuration sample is now installed into the share/openal/ directory
+    instead of /etc/openal.
+
+    The configuration sample now gets installed by default.

OpenAL32/Include/alAuxEffectSlot.h

@@ -1,75 +1,187 @@
 #ifndef _AL_AUXEFFECTSLOT_H_
 #define _AL_AUXEFFECTSLOT_H_
 
-#include "AL/al.h"
+#include "alMain.h"
 #include "alEffect.h"
-#include "alFilter.h"
+
+#include "atomic.h"
+#include "align.h"
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-#define AL_EFFECTSLOT_EFFECT                               0x0001
-#define AL_EFFECTSLOT_GAIN                                 0x0002
-#define AL_EFFECTSLOT_AUXILIARY_SEND_AUTO                  0x0003
+struct ALeffectStateVtable;
+struct ALeffectslot;
 
-#define AL_EFFECTSLOT_NULL                                 0x0000
+typedef struct ALeffectState {
+    RefCount Ref;
+    const struct ALeffectStateVtable *vtbl;
 
-typedef struct ALeffectState ALeffectState;
+    ALfloat (*OutBuffer)[BUFFERSIZE];
+    ALsizei OutChannels;
+} ALeffectState;
 
-typedef struct ALeffectslot
-{
-    ALeffect effect;
+void ALeffectState_Construct(ALeffectState *state);
+void ALeffectState_Destruct(ALeffectState *state);
 
-    ALfloat Gain;
+struct ALeffectStateVtable {
+    void (*const Destruct)(ALeffectState *state);
+
+    ALboolean (*const deviceUpdate)(ALeffectState *state, ALCdevice *device);
+    void (*const update)(ALeffectState *state, const ALCdevice *device, const struct ALeffectslot *slot, const union ALeffectProps *props);
+    void (*const process)(ALeffectState *state, ALsizei samplesToDo, const ALfloat (*restrict samplesIn)[BUFFERSIZE], ALfloat (*restrict samplesOut)[BUFFERSIZE], ALsizei numChannels);
+
+    void (*const Delete)(void *ptr);
+};
+
+#define DEFINE_ALEFFECTSTATE_VTABLE(T)                                        \
+DECLARE_THUNK(T, ALeffectState, void, Destruct)                               \
+DECLARE_THUNK1(T, ALeffectState, ALboolean, deviceUpdate, ALCdevice*)         \
+DECLARE_THUNK3(T, ALeffectState, void, update, const ALCdevice*, const ALeffectslot*, const ALeffectProps*) \
+DECLARE_THUNK4(T, ALeffectState, void, process, ALsizei, const ALfloatBUFFERSIZE*restrict, ALfloatBUFFERSIZE*restrict, ALsizei) \
+static void T##_ALeffectState_Delete(void *ptr)                               \
+{ return T##_Delete(STATIC_UPCAST(T, ALeffectState, (ALeffectState*)ptr)); }  \
+                                                                              \
+static const struct ALeffectStateVtable T##_ALeffectState_vtable = {          \
+    T##_ALeffectState_Destruct,                                               \
+                                                                              \
+    T##_ALeffectState_deviceUpdate,                                           \
+    T##_ALeffectState_update,                                                 \
+    T##_ALeffectState_process,                                                \
+                                                                              \
+    T##_ALeffectState_Delete,                                                 \
+}
+
+
+struct ALeffectStateFactoryVtable;
+
+typedef struct ALeffectStateFactory {
+    const struct ALeffectStateFactoryVtable *vtbl;
+} ALeffectStateFactory;
+
+struct ALeffectStateFactoryVtable {
+    ALeffectState *(*const create)(ALeffectStateFactory *factory);
+};
+
+#define DEFINE_ALEFFECTSTATEFACTORY_VTABLE(T)                                 \
+DECLARE_THUNK(T, ALeffectStateFactory, ALeffectState*, create)                \
+                                                                              \
+static const struct ALeffectStateFactoryVtable T##_ALeffectStateFactory_vtable = { \
+    T##_ALeffectStateFactory_create,                                          \
+}
+
+
+#define MAX_EFFECT_CHANNELS (4)
+
+
+struct ALeffectslotArray {
+    ALsizei count;
+    struct ALeffectslot *slot[];
+};
+
+
+struct ALeffectslotProps {
+    ALfloat   Gain;
     ALboolean AuxSendAuto;
 
-    ALeffectState *EffectState;
+    ALenum Type;
+    ALeffectProps Props;
 
-    ALfloat WetBuffer[BUFFERSIZE];
+    ALeffectState *State;
 
-    ALuint refcount;
+    ATOMIC(struct ALeffectslotProps*) next;
+};
 
-    // Index to itself
-    ALuint effectslot;
 
-    struct ALeffectslot *next;
+typedef struct ALeffectslot {
+    ALfloat   Gain;
+    ALboolean AuxSendAuto;
+
+    struct {
+        ALenum Type;
+        ALeffectProps Props;
+
+        ALeffectState *State;
+    } Effect;
+
+    ATOMIC_FLAG PropsClean;
+
+    RefCount ref;
+
+    ATOMIC(struct ALeffectslotProps*) Update;
+    ATOMIC(struct ALeffectslotProps*) FreeList;
+
+    struct {
+        ALfloat   Gain;
+        ALboolean AuxSendAuto;
+
+        ALenum EffectType;
+        ALeffectState *EffectState;
+
+        ALfloat RoomRolloff; /* Added to the source's room rolloff, not multiplied. */
+        ALfloat DecayTime;
+        ALfloat DecayHFRatio;
+        ALboolean DecayHFLimit;
+        ALfloat AirAbsorptionGainHF;
+    } Params;
+
+    /* Self ID */
+    ALuint id;
+
+    ALsizei NumChannels;
+    BFChannelConfig ChanMap[MAX_EFFECT_CHANNELS];
+    /* Wet buffer configuration is ACN channel order with N3D scaling:
+     * * Channel 0 is the unattenuated mono signal.
+     * * Channel 1 is OpenAL -X
+     * * Channel 2 is OpenAL Y
+     * * Channel 3 is OpenAL -Z
+     * Consequently, effects that only want to work with mono input can use
+     * channel 0 by itself. Effects that want multichannel can process the
+     * ambisonics signal and make a B-Format pan (ComputeFirstOrderGains) for
+     * first-order device output (FOAOut).
+     */
+    alignas(16) ALfloat WetBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE];
 } ALeffectslot;
 
-ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots);
-ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots);
-ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot);
+inline void LockEffectSlotsRead(ALCcontext *context)
+{ LockUIntMapRead(&context->EffectSlotMap); }
+inline void UnlockEffectSlotsRead(ALCcontext *context)
+{ UnlockUIntMapRead(&context->EffectSlotMap); }
+inline void LockEffectSlotsWrite(ALCcontext *context)
+{ LockUIntMapWrite(&context->EffectSlotMap); }
+inline void UnlockEffectSlotsWrite(ALCcontext *context)
+{ UnlockUIntMapWrite(&context->EffectSlotMap); }
 
-ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint iValue);
-ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *piValues);
-ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat flValue);
-ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *pflValues);
-
-ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *piValue);
-ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *piValues);
-ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *pflValue);
-ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *pflValues);
+inline struct ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id)
+{ return (struct ALeffectslot*)LookupUIntMapKeyNoLock(&context->EffectSlotMap, id); }
+inline struct ALeffectslot *RemoveEffectSlot(ALCcontext *context, ALuint id)
+{ return (struct ALeffectslot*)RemoveUIntMapKeyNoLock(&context->EffectSlotMap, id); }
 
+ALenum InitEffectSlot(ALeffectslot *slot);
+void DeinitEffectSlot(ALeffectslot *slot);
+void UpdateEffectSlotProps(ALeffectslot *slot);
+void UpdateAllEffectSlotProps(ALCcontext *context);
 ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context);
 
 
-struct ALeffectState {
-    ALvoid (*Destroy)(ALeffectState *State);
-    ALboolean (*DeviceUpdate)(ALeffectState *State, ALCdevice *Device);
-    ALvoid (*Update)(ALeffectState *State, ALCcontext *Context, const ALeffect *Effect);
-    ALvoid (*Process)(ALeffectState *State, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[OUTPUTCHANNELS]);
-};
+ALeffectStateFactory *ALnullStateFactory_getFactory(void);
+ALeffectStateFactory *ALreverbStateFactory_getFactory(void);
+ALeffectStateFactory *ALchorusStateFactory_getFactory(void);
+ALeffectStateFactory *ALcompressorStateFactory_getFactory(void);
+ALeffectStateFactory *ALdistortionStateFactory_getFactory(void);
+ALeffectStateFactory *ALechoStateFactory_getFactory(void);
+ALeffectStateFactory *ALequalizerStateFactory_getFactory(void);
+ALeffectStateFactory *ALflangerStateFactory_getFactory(void);
+ALeffectStateFactory *ALmodulatorStateFactory_getFactory(void);
 
-ALeffectState *NoneCreate(void);
-ALeffectState *EAXVerbCreate(void);
-ALeffectState *VerbCreate(void);
-ALeffectState *EchoCreate(void);
+ALeffectStateFactory *ALdedicatedStateFactory_getFactory(void);
 
-#define ALEffect_Destroy(a)         ((a)->Destroy((a)))
-#define ALEffect_DeviceUpdate(a,b)  ((a)->DeviceUpdate((a),(b)))
-#define ALEffect_Update(a,b,c)      ((a)->Update((a),(b),(c)))
-#define ALEffect_Process(a,b,c,d,e) ((a)->Process((a),(b),(c),(d),(e)))
 
+ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect);
+
+void InitEffectFactoryMap(void);
+void DeinitEffectFactoryMap(void);
 
 #ifdef __cplusplus
 }

OpenAL32/Include/alBuffer.h

@@ -1,32 +1,123 @@
 #ifndef _AL_BUFFER_H_
 #define _AL_BUFFER_H_
 
-#include "AL/al.h"
+#include "alMain.h"
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-#define BUFFER_PADDING 2
+/* User formats */
+enum UserFmtType {
+    UserFmtByte   = AL_BYTE_SOFT,
+    UserFmtUByte  = AL_UNSIGNED_BYTE_SOFT,
+    UserFmtShort  = AL_SHORT_SOFT,
+    UserFmtUShort = AL_UNSIGNED_SHORT_SOFT,
+    UserFmtInt    = AL_INT_SOFT,
+    UserFmtUInt   = AL_UNSIGNED_INT_SOFT,
+    UserFmtFloat  = AL_FLOAT_SOFT,
+    UserFmtDouble = AL_DOUBLE_SOFT,
+    UserFmtMulaw  = AL_MULAW_SOFT,
+    UserFmtAlaw   = 0x10000000,
+    UserFmtIMA4,
+    UserFmtMSADPCM,
+};
+enum UserFmtChannels {
+    UserFmtMono      = AL_MONO_SOFT,
+    UserFmtStereo    = AL_STEREO_SOFT,
+    UserFmtRear      = AL_REAR_SOFT,
+    UserFmtQuad      = AL_QUAD_SOFT,
+    UserFmtX51       = AL_5POINT1_SOFT, /* (WFX order) */
+    UserFmtX61       = AL_6POINT1_SOFT, /* (WFX order) */
+    UserFmtX71       = AL_7POINT1_SOFT, /* (WFX order) */
+    UserFmtBFormat2D = AL_BFORMAT2D_SOFT, /* WXY */
+    UserFmtBFormat3D = AL_BFORMAT3D_SOFT, /* WXYZ */
+};
 
-typedef struct ALbuffer
+ALsizei BytesFromUserFmt(enum UserFmtType type);
+ALsizei ChannelsFromUserFmt(enum UserFmtChannels chans);
+inline ALsizei FrameSizeFromUserFmt(enum UserFmtChannels chans, enum UserFmtType type)
 {
-    ALshort *data;
-    ALsizei  size;
+    return ChannelsFromUserFmt(chans) * BytesFromUserFmt(type);
+}
 
-    ALenum   format;
-    ALenum   eOriginalFormat;
-    ALsizei  frequency;
 
-    ALuint   refcount; // Number of sources using this buffer (deletion can only occur when this is 0)
+/* Storable formats */
+enum FmtType {
+    FmtByte  = UserFmtByte,
+    FmtShort = UserFmtShort,
+    FmtFloat = UserFmtFloat,
+};
+enum FmtChannels {
+    FmtMono   = UserFmtMono,
+    FmtStereo = UserFmtStereo,
+    FmtRear   = UserFmtRear,
+    FmtQuad   = UserFmtQuad,
+    FmtX51    = UserFmtX51,
+    FmtX61    = UserFmtX61,
+    FmtX71    = UserFmtX71,
+    FmtBFormat2D = UserFmtBFormat2D,
+    FmtBFormat3D = UserFmtBFormat3D,
+};
+#define MAX_INPUT_CHANNELS  (8)
 
-    // Index to itself
-    ALuint buffer;
+ALsizei BytesFromFmt(enum FmtType type);
+ALsizei ChannelsFromFmt(enum FmtChannels chans);
+inline ALsizei FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type)
+{
+    return ChannelsFromFmt(chans) * BytesFromFmt(type);
+}
 
-    struct ALbuffer *next;
+
+typedef struct ALbuffer {
+    ALvoid  *data;
+
+    ALsizei  Frequency;
+    ALenum   Format;
+    ALsizei  SampleLen;
+
+    enum FmtChannels FmtChannels;
+    enum FmtType     FmtType;
+    ALuint BytesAlloc;
+
+    enum UserFmtChannels OriginalChannels;
+    enum UserFmtType     OriginalType;
+    ALsizei              OriginalSize;
+    ALsizei              OriginalAlign;
+
+    ALsizei LoopStart;
+    ALsizei LoopEnd;
+
+    ATOMIC(ALsizei) UnpackAlign;
+    ATOMIC(ALsizei) PackAlign;
+
+    /* Number of times buffer was attached to a source (deletion can only occur when 0) */
+    RefCount ref;
+
+    RWLock lock;
+
+    /* Self ID */
+    ALuint id;
 } ALbuffer;
 
-ALvoid ALAPIENTRY alBufferSubDataEXT(ALuint buffer,ALenum format,const ALvoid *data,ALsizei offset,ALsizei length);
+ALbuffer *NewBuffer(ALCcontext *context);
+void DeleteBuffer(ALCdevice *device, ALbuffer *buffer);
+
+ALenum LoadData(ALbuffer *buffer, ALuint freq, ALenum NewFormat, ALsizei frames, enum UserFmtChannels SrcChannels, enum UserFmtType SrcType, const ALvoid *data, ALsizei align, ALboolean storesrc);
+
+inline void LockBuffersRead(ALCdevice *device)
+{ LockUIntMapRead(&device->BufferMap); }
+inline void UnlockBuffersRead(ALCdevice *device)
+{ UnlockUIntMapRead(&device->BufferMap); }
+inline void LockBuffersWrite(ALCdevice *device)
+{ LockUIntMapWrite(&device->BufferMap); }
+inline void UnlockBuffersWrite(ALCdevice *device)
+{ UnlockUIntMapWrite(&device->BufferMap); }
+
+inline struct ALbuffer *LookupBuffer(ALCdevice *device, ALuint id)
+{ return (struct ALbuffer*)LookupUIntMapKeyNoLock(&device->BufferMap, id); }
+inline struct ALbuffer *RemoveBuffer(ALCdevice *device, ALuint id)
+{ return (struct ALbuffer*)RemoveUIntMapKeyNoLock(&device->BufferMap, id); }
 
 ALvoid ReleaseALBuffers(ALCdevice *device);
 

OpenAL32/Include/alDatabuffer.h

@@ -1,55 +0,0 @@
-#ifndef _AL_DATABUFFER_H_
-#define _AL_DATABUFFER_H_
-
-#include "AL/al.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#define UNMAPPED 0
-#define MAPPED   1
-
-typedef struct ALdatabuffer
-{
-    ALubyte *data;
-    ALuint   size;
-    ALenum   state;
-
-    ALenum   usage;
-
-    /* Index to self */
-    ALuint   databuffer;
-
-    struct ALdatabuffer *next;
-} ALdatabuffer;
-
-ALvoid ALAPIENTRY alGenDatabuffersEXT(ALsizei n,ALuint *puiBuffers);
-ALvoid ALAPIENTRY alDeleteDatabuffersEXT(ALsizei n, const ALuint *puiBuffers);
-ALboolean ALAPIENTRY alIsDatabufferEXT(ALuint uiBuffer);
-
-ALvoid ALAPIENTRY alDatabufferDataEXT(ALuint buffer,const ALvoid *data,ALsizei size,ALenum usage);
-ALvoid ALAPIENTRY alDatabufferSubDataEXT(ALuint buffer, ALuint start, ALsizei length, const ALvoid *data);
-ALvoid ALAPIENTRY alGetDatabufferSubDataEXT(ALuint buffer, ALuint start, ALsizei length, ALvoid *data);
-
-ALvoid ALAPIENTRY alDatabufferfEXT(ALuint buffer, ALenum eParam, ALfloat flValue);
-ALvoid ALAPIENTRY alDatabufferfvEXT(ALuint buffer, ALenum eParam, const ALfloat* flValues);
-ALvoid ALAPIENTRY alDatabufferiEXT(ALuint buffer, ALenum eParam, ALint lValue);
-ALvoid ALAPIENTRY alDatabufferivEXT(ALuint buffer, ALenum eParam, const ALint* plValues);
-ALvoid ALAPIENTRY alGetDatabufferfEXT(ALuint buffer, ALenum eParam, ALfloat *pflValue);
-ALvoid ALAPIENTRY alGetDatabufferfvEXT(ALuint buffer, ALenum eParam, ALfloat* pflValues);
-ALvoid ALAPIENTRY alGetDatabufferiEXT(ALuint buffer, ALenum eParam, ALint *plValue);
-ALvoid ALAPIENTRY alGetDatabufferivEXT(ALuint buffer, ALenum eParam, ALint* plValues);
-
-ALvoid ALAPIENTRY alSelectDatabufferEXT(ALenum target, ALuint uiBuffer);
-
-ALvoid* ALAPIENTRY alMapDatabufferEXT(ALuint uiBuffer, ALuint start, ALsizei length, ALenum access);
-ALvoid ALAPIENTRY alUnmapDatabufferEXT(ALuint uiBuffer);
-
-ALvoid ReleaseALDatabuffers(ALCdevice *device);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alEffect.h

@@ -1,211 +1,67 @@
-// NOTE:  The effect structure is getting too large, it may be a good idea to
-//        start using a union or another form of unified storage.
 #ifndef _AL_EFFECT_H_
 #define _AL_EFFECT_H_
 
-#include "AL/al.h"
+#include "alMain.h"
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-#define AL_EFFECT_TYPE                                     0x8001
-
-#define AL_EFFECT_NULL                                     0x0000
-#define AL_EFFECT_REVERB                                   0x0001
-#define AL_EFFECT_CHORUS                                   0x0002
-#define AL_EFFECT_DISTORTION                               0x0003
-#define AL_EFFECT_ECHO                                     0x0004
-#define AL_EFFECT_FLANGER                                  0x0005
-#define AL_EFFECT_FREQUENCY_SHIFTER                        0x0006
-#define AL_EFFECT_VOCAL_MORPHER                            0x0007
-#define AL_EFFECT_PITCH_SHIFTER                            0x0008
-#define AL_EFFECT_RING_MODULATOR                           0x0009
-#define AL_EFFECT_AUTOWAH                                  0x000A
-#define AL_EFFECT_COMPRESSOR                               0x000B
-#define AL_EFFECT_EQUALIZER                                0x000C
-#define AL_EFFECT_EAXREVERB                                0x8000
-
-#define AL_REVERB_DENSITY                                  0x0001
-#define AL_REVERB_DIFFUSION                                0x0002
-#define AL_REVERB_GAIN                                     0x0003
-#define AL_REVERB_GAINHF                                   0x0004
-#define AL_REVERB_DECAY_TIME                               0x0005
-#define AL_REVERB_DECAY_HFRATIO                            0x0006
-#define AL_REVERB_REFLECTIONS_GAIN                         0x0007
-#define AL_REVERB_REFLECTIONS_DELAY                        0x0008
-#define AL_REVERB_LATE_REVERB_GAIN                         0x0009
-#define AL_REVERB_LATE_REVERB_DELAY                        0x000A
-#define AL_REVERB_AIR_ABSORPTION_GAINHF                    0x000B
-#define AL_REVERB_ROOM_ROLLOFF_FACTOR                      0x000C
-#define AL_REVERB_DECAY_HFLIMIT                            0x000D
-
-#define AL_REVERB_MIN_DENSITY                              (0.0f)
-#define AL_REVERB_MAX_DENSITY                              (1.0f)
-#define AL_REVERB_DEFAULT_DENSITY                          (1.0f)
-#define AL_REVERB_MIN_DIFFUSION                            (0.0f)
-#define AL_REVERB_MAX_DIFFUSION                            (1.0f)
-#define AL_REVERB_DEFAULT_DIFFUSION                        (1.0f)
-#define AL_REVERB_MIN_GAIN                                 (0.0f)
-#define AL_REVERB_MAX_GAIN                                 (1.0f)
-#define AL_REVERB_DEFAULT_GAIN                             (0.32f)
-#define AL_REVERB_MIN_GAINHF                               (0.0f)
-#define AL_REVERB_MAX_GAINHF                               (1.0f)
-#define AL_REVERB_DEFAULT_GAINHF                           (0.89f)
-#define AL_REVERB_MIN_DECAY_TIME                           (0.1f)
-#define AL_REVERB_MAX_DECAY_TIME                           (20.0f)
-#define AL_REVERB_DEFAULT_DECAY_TIME                       (1.49f)
-#define AL_REVERB_MIN_DECAY_HFRATIO                        (0.1f)
-#define AL_REVERB_MAX_DECAY_HFRATIO                        (2.0f)
-#define AL_REVERB_DEFAULT_DECAY_HFRATIO                    (0.83f)
-#define AL_REVERB_MIN_REFLECTIONS_GAIN                     (0.0f)
-#define AL_REVERB_MAX_REFLECTIONS_GAIN                     (3.16f)
-#define AL_REVERB_DEFAULT_REFLECTIONS_GAIN                 (0.05f)
-#define AL_REVERB_MIN_REFLECTIONS_DELAY                    (0.0f)
-#define AL_REVERB_MAX_REFLECTIONS_DELAY                    (0.3f)
-#define AL_REVERB_DEFAULT_REFLECTIONS_DELAY                (0.007f)
-#define AL_REVERB_MIN_LATE_REVERB_GAIN                     (0.0f)
-#define AL_REVERB_MAX_LATE_REVERB_GAIN                     (10.0f)
-#define AL_REVERB_DEFAULT_LATE_REVERB_GAIN                 (1.26f)
-#define AL_REVERB_MIN_LATE_REVERB_DELAY                    (0.0f)
-#define AL_REVERB_MAX_LATE_REVERB_DELAY                    (0.1f)
-#define AL_REVERB_DEFAULT_LATE_REVERB_DELAY                (0.011f)
-#define AL_REVERB_MIN_AIR_ABSORPTION_GAINHF                (0.892f)
-#define AL_REVERB_MAX_AIR_ABSORPTION_GAINHF                (1.0f)
-#define AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF            (0.994f)
-#define AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR                  (0.0f)
-#define AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR                  (10.0f)
-#define AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR              (0.0f)
-#define AL_REVERB_MIN_DECAY_HFLIMIT                        (AL_FALSE)
-#define AL_REVERB_MAX_DECAY_HFLIMIT                        (AL_TRUE)
-#define AL_REVERB_DEFAULT_DECAY_HFLIMIT                    (AL_TRUE)
-
-#define AL_ECHO_DELAY                                      0x0001
-#define AL_ECHO_LRDELAY                                    0x0002
-#define AL_ECHO_DAMPING                                    0x0003
-#define AL_ECHO_FEEDBACK                                   0x0004
-#define AL_ECHO_SPREAD                                     0x0005
-
-#define AL_ECHO_MIN_DELAY                                  (0.0f)
-#define AL_ECHO_MAX_DELAY                                  (0.207f)
-#define AL_ECHO_DEFAULT_DELAY                              (0.1f)
-#define AL_ECHO_MIN_LRDELAY                                (0.0f)
-#define AL_ECHO_MAX_LRDELAY                                (0.404f)
-#define AL_ECHO_DEFAULT_LRDELAY                            (0.1f)
-#define AL_ECHO_MIN_DAMPING                                (0.0f)
-#define AL_ECHO_MAX_DAMPING                                (0.99f)
-#define AL_ECHO_DEFAULT_DAMPING                            (0.5f)
-#define AL_ECHO_MIN_FEEDBACK                               (0.0f)
-#define AL_ECHO_MAX_FEEDBACK                               (1.0f)
-#define AL_ECHO_DEFAULT_FEEDBACK                           (0.5f)
-#define AL_ECHO_MIN_SPREAD                                 (-1.0f)
-#define AL_ECHO_MAX_SPREAD                                 (1.0f)
-#define AL_ECHO_DEFAULT_SPREAD                             (-1.0f)
-
-#define AL_EAXREVERB_DENSITY                               0x0001
-#define AL_EAXREVERB_DIFFUSION                             0x0002
-#define AL_EAXREVERB_GAIN                                  0x0003
-#define AL_EAXREVERB_GAINHF                                0x0004
-#define AL_EAXREVERB_GAINLF                                0x0005
-#define AL_EAXREVERB_DECAY_TIME                            0x0006
-#define AL_EAXREVERB_DECAY_HFRATIO                         0x0007
-#define AL_EAXREVERB_DECAY_LFRATIO                         0x0008
-#define AL_EAXREVERB_REFLECTIONS_GAIN                      0x0009
-#define AL_EAXREVERB_REFLECTIONS_DELAY                     0x000A
-#define AL_EAXREVERB_REFLECTIONS_PAN                       0x000B
-#define AL_EAXREVERB_LATE_REVERB_GAIN                      0x000C
-#define AL_EAXREVERB_LATE_REVERB_DELAY                     0x000D
-#define AL_EAXREVERB_LATE_REVERB_PAN                       0x000E
-#define AL_EAXREVERB_ECHO_TIME                             0x000F
-#define AL_EAXREVERB_ECHO_DEPTH                            0x0010
-#define AL_EAXREVERB_MODULATION_TIME                       0x0011
-#define AL_EAXREVERB_MODULATION_DEPTH                      0x0012
-#define AL_EAXREVERB_AIR_ABSORPTION_GAINHF                 0x0013
-#define AL_EAXREVERB_HFREFERENCE                           0x0014
-#define AL_EAXREVERB_LFREFERENCE                           0x0015
-#define AL_EAXREVERB_ROOM_ROLLOFF_FACTOR                   0x0016
-#define AL_EAXREVERB_DECAY_HFLIMIT                         0x0017
-
-#define AL_EAXREVERB_MIN_DENSITY                           (0.0f)
-#define AL_EAXREVERB_MAX_DENSITY                           (1.0f)
-#define AL_EAXREVERB_DEFAULT_DENSITY                       (1.0f)
-#define AL_EAXREVERB_MIN_DIFFUSION                         (0.0f)
-#define AL_EAXREVERB_MAX_DIFFUSION                         (1.0f)
-#define AL_EAXREVERB_DEFAULT_DIFFUSION                     (1.0f)
-#define AL_EAXREVERB_MIN_GAIN                              (0.0f)
-#define AL_EAXREVERB_MAX_GAIN                              (1.0f)
-#define AL_EAXREVERB_DEFAULT_GAIN                          (0.32f)
-#define AL_EAXREVERB_MIN_GAINHF                            (0.0f)
-#define AL_EAXREVERB_MAX_GAINHF                            (1.0f)
-#define AL_EAXREVERB_DEFAULT_GAINHF                        (0.89f)
-#define AL_EAXREVERB_MIN_GAINLF                            (0.0f)
-#define AL_EAXREVERB_MAX_GAINLF                            (1.0f)
-#define AL_EAXREVERB_DEFAULT_GAINLF                        (1.0f)
-#define AL_EAXREVERB_MIN_DECAY_TIME                        (0.1f)
-#define AL_EAXREVERB_MAX_DECAY_TIME                        (20.0f)
-#define AL_EAXREVERB_DEFAULT_DECAY_TIME                    (1.49f)
-#define AL_EAXREVERB_MIN_DECAY_HFRATIO                     (0.1f)
-#define AL_EAXREVERB_MAX_DECAY_HFRATIO                     (2.0f)
-#define AL_EAXREVERB_DEFAULT_DECAY_HFRATIO                 (0.83f)
-#define AL_EAXREVERB_MIN_DECAY_LFRATIO                     (0.1f)
-#define AL_EAXREVERB_MAX_DECAY_LFRATIO                     (2.0f)
-#define AL_EAXREVERB_DEFAULT_DECAY_LFRATIO                 (1.0f)
-#define AL_EAXREVERB_MIN_REFLECTIONS_GAIN                  (0.0f)
-#define AL_EAXREVERB_MAX_REFLECTIONS_GAIN                  (3.16f)
-#define AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN              (0.05f)
-#define AL_EAXREVERB_MIN_REFLECTIONS_DELAY                 (0.0f)
-#define AL_EAXREVERB_MAX_REFLECTIONS_DELAY                 (0.3f)
-#define AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY             (0.007f)
-#define AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ           (0.0f)
-#define AL_EAXREVERB_MIN_LATE_REVERB_GAIN                  (0.0f)
-#define AL_EAXREVERB_MAX_LATE_REVERB_GAIN                  (10.0f)
-#define AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN              (1.26f)
-#define AL_EAXREVERB_MIN_LATE_REVERB_DELAY                 (0.0f)
-#define AL_EAXREVERB_MAX_LATE_REVERB_DELAY                 (0.1f)
-#define AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY             (0.011f)
-#define AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ           (0.0f)
-#define AL_EAXREVERB_MIN_ECHO_TIME                         (0.075f)
-#define AL_EAXREVERB_MAX_ECHO_TIME                         (0.25f)
-#define AL_EAXREVERB_DEFAULT_ECHO_TIME                     (0.25f)
-#define AL_EAXREVERB_MIN_ECHO_DEPTH                        (0.0f)
-#define AL_EAXREVERB_MAX_ECHO_DEPTH                        (1.0f)
-#define AL_EAXREVERB_DEFAULT_ECHO_DEPTH                    (0.0f)
-#define AL_EAXREVERB_MIN_MODULATION_TIME                   (0.04f)
-#define AL_EAXREVERB_MAX_MODULATION_TIME                   (4.0f)
-#define AL_EAXREVERB_DEFAULT_MODULATION_TIME               (0.25f)
-#define AL_EAXREVERB_MIN_MODULATION_DEPTH                  (0.0f)
-#define AL_EAXREVERB_MAX_MODULATION_DEPTH                  (1.0f)
-#define AL_EAXREVERB_DEFAULT_MODULATION_DEPTH              (0.0f)
-#define AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF             (0.892f)
-#define AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF             (1.0f)
-#define AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF         (0.994f)
-#define AL_EAXREVERB_MIN_HFREFERENCE                       (1000.0f)
-#define AL_EAXREVERB_MAX_HFREFERENCE                       (20000.0f)
-#define AL_EAXREVERB_DEFAULT_HFREFERENCE                   (5000.0f)
-#define AL_EAXREVERB_MIN_LFREFERENCE                       (20.0f)
-#define AL_EAXREVERB_MAX_LFREFERENCE                       (1000.0f)
-#define AL_EAXREVERB_DEFAULT_LFREFERENCE                   (250.0f)
-#define AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR               (0.0f)
-#define AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR               (10.0f)
-#define AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR           (0.0f)
-#define AL_EAXREVERB_MIN_DECAY_HFLIMIT                     (AL_FALSE)
-#define AL_EAXREVERB_MAX_DECAY_HFLIMIT                     (AL_TRUE)
-#define AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT                 (AL_TRUE)
+struct ALeffect;
 
 enum {
-    EAXREVERB = 0,
-    REVERB,
-    ECHO,
+    AL__EAXREVERB = 0,
+    AL__REVERB,
+    AL__CHORUS,
+    AL__COMPRESSOR,
+    AL__DISTORTION,
+    AL__ECHO,
+    AL__EQUALIZER,
+    AL__FLANGER,
+    AL__MODULATOR,
+    AL__DEDICATED,
 
     MAX_EFFECTS
 };
 extern ALboolean DisabledEffects[MAX_EFFECTS];
 
-typedef struct ALeffect
-{
-    // Effect type (AL_EFFECT_NULL, ...)
-    ALenum type;
+extern ALfloat ReverbBoost;
+extern ALboolean EmulateEAXReverb;
 
+struct ALeffectVtable {
+    void (*const setParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
+    void (*const setParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
+    void (*const setParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
+    void (*const setParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals);
+
+    void (*const getParami)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *val);
+    void (*const getParamiv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals);
+    void (*const getParamf)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
+    void (*const getParamfv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
+};
+
+#define DEFINE_ALEFFECT_VTABLE(T)           \
+const struct ALeffectVtable T##_vtable = {  \
+    T##_setParami, T##_setParamiv,          \
+    T##_setParamf, T##_setParamfv,          \
+    T##_getParami, T##_getParamiv,          \
+    T##_getParamf, T##_getParamfv,          \
+}
+
+extern const struct ALeffectVtable ALeaxreverb_vtable;
+extern const struct ALeffectVtable ALreverb_vtable;
+extern const struct ALeffectVtable ALchorus_vtable;
+extern const struct ALeffectVtable ALcompressor_vtable;
+extern const struct ALeffectVtable ALdistortion_vtable;
+extern const struct ALeffectVtable ALecho_vtable;
+extern const struct ALeffectVtable ALequalizer_vtable;
+extern const struct ALeffectVtable ALflanger_vtable;
+extern const struct ALeffectVtable ALmodulator_vtable;
+extern const struct ALeffectVtable ALnull_vtable;
+extern const struct ALeffectVtable ALdedicated_vtable;
+
+
+typedef union ALeffectProps {
     struct {
         // Shared Reverb Properties
         ALfloat Density;
@@ -235,6 +91,27 @@ typedef struct ALeffect
         ALfloat LFReference;
     } Reverb;
 
+    struct {
+        ALint Waveform;
+        ALint Phase;
+        ALfloat Rate;
+        ALfloat Depth;
+        ALfloat Feedback;
+        ALfloat Delay;
+    } Chorus;
+
+    struct {
+        ALboolean OnOff;
+    } Compressor;
+
+    struct {
+        ALfloat Edge;
+        ALfloat Gain;
+        ALfloat LowpassCutoff;
+        ALfloat EQCenter;
+        ALfloat EQBandwidth;
+    } Distortion;
+
     struct {
         ALfloat Delay;
         ALfloat LRDelay;
@@ -245,28 +122,73 @@ typedef struct ALeffect
         ALfloat Spread;
     } Echo;
 
-    // Index to itself
-    ALuint effect;
+    struct {
+        ALfloat LowCutoff;
+        ALfloat LowGain;
+        ALfloat Mid1Center;
+        ALfloat Mid1Gain;
+        ALfloat Mid1Width;
+        ALfloat Mid2Center;
+        ALfloat Mid2Gain;
+        ALfloat Mid2Width;
+        ALfloat HighCutoff;
+        ALfloat HighGain;
+    } Equalizer;
 
-    struct ALeffect *next;
+    struct {
+        ALint Waveform;
+        ALint Phase;
+        ALfloat Rate;
+        ALfloat Depth;
+        ALfloat Feedback;
+        ALfloat Delay;
+    } Flanger;
+
+    struct {
+        ALfloat Frequency;
+        ALfloat HighPassCutoff;
+        ALint Waveform;
+    } Modulator;
+
+    struct {
+        ALfloat Gain;
+    } Dedicated;
+} ALeffectProps;
+
+typedef struct ALeffect {
+    // Effect type (AL_EFFECT_NULL, ...)
+    ALenum type;
+
+    ALeffectProps Props;
+
+    const struct ALeffectVtable *vtbl;
+
+    /* Self ID */
+    ALuint id;
 } ALeffect;
 
-ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects);
-ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, ALuint *effects);
-ALboolean AL_APIENTRY alIsEffect(ALuint effect);
+inline void LockEffectsRead(ALCdevice *device)
+{ LockUIntMapRead(&device->EffectMap); }
+inline void UnlockEffectsRead(ALCdevice *device)
+{ UnlockUIntMapRead(&device->EffectMap); }
+inline void LockEffectsWrite(ALCdevice *device)
+{ LockUIntMapWrite(&device->EffectMap); }
+inline void UnlockEffectsWrite(ALCdevice *device)
+{ UnlockUIntMapWrite(&device->EffectMap); }
 
-ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint iValue);
-ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, ALint *piValues);
-ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat flValue);
-ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, ALfloat *pflValues);
+inline struct ALeffect *LookupEffect(ALCdevice *device, ALuint id)
+{ return (struct ALeffect*)LookupUIntMapKeyNoLock(&device->EffectMap, id); }
+inline struct ALeffect *RemoveEffect(ALCdevice *device, ALuint id)
+{ return (struct ALeffect*)RemoveUIntMapKeyNoLock(&device->EffectMap, id); }
 
-ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *piValue);
-ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *piValues);
-ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *pflValue);
-ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *pflValues);
+inline ALboolean IsReverbEffect(ALenum type)
+{ return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; }
 
+ALenum InitEffect(ALeffect *effect);
 ALvoid ReleaseALEffects(ALCdevice *device);
 
+ALvoid LoadReverbPreset(const char *name, ALeffect *effect);
+
 #ifdef __cplusplus
 }
 #endif

OpenAL32/Include/alError.h

@@ -1,13 +1,30 @@
 #ifndef _AL_ERROR_H_
 #define _AL_ERROR_H_
 
-#include "AL/al.h"
+#include "alMain.h"
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-ALvoid alSetError(ALenum errorCode);
+extern ALboolean TrapALError;
+
+ALvoid alSetError(ALCcontext *Context, ALenum errorCode);
+
+#define SET_ERROR_AND_RETURN(ctx, err) do {                                    \
+    alSetError((ctx), (err));                                                  \
+    return;                                                                    \
+} while(0)
+
+#define SET_ERROR_AND_RETURN_VALUE(ctx, err, val) do {                         \
+    alSetError((ctx), (err));                                                  \
+    return (val);                                                              \
+} while(0)
+
+#define SET_ERROR_AND_GOTO(ctx, err, lbl) do {                                 \
+    alSetError((ctx), (err));                                                  \
+    goto lbl;                                                                  \
+} while(0)
 
 #ifdef __cplusplus
 }

OpenAL32/Include/alExtension.h

@@ -1,33 +0,0 @@
-#ifndef _AL_EXTENSION_H_
-#define _AL_EXTENSION_H_
-
-#include "AL/al.h"
-#include "AL/alc.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-typedef struct ALextension_struct
-{
-	ALchar		*extName;
-	ALvoid		*address;
-} ALextension;
-
-typedef struct ALfunction_struct
-{
-	ALchar		*funcName;
-	ALvoid		*address;
-} ALfunction;
-
-typedef struct ALenum_struct
-{
-	ALchar		*enumName;
-	ALenum		value;
-} ALenums;
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alFilter.h

@@ -1,93 +1,159 @@
 #ifndef _AL_FILTER_H_
 #define _AL_FILTER_H_
 
-#include "AL/al.h"
-#include "alu.h"
+#include "alMain.h"
+
+#include "math_defs.h"
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-typedef struct {
-    ALfloat coeff;
-#ifndef _MSC_VER
-    ALfloat history[0];
-#else
-    ALfloat history[1];
-#endif
-} FILTER;
+#define LOWPASSFREQREF  (5000.0f)
+#define HIGHPASSFREQREF  (250.0f)
 
-static __inline ALfloat lpFilter4P(FILTER *iir, ALuint offset, ALfloat input)
+
+/* Filters implementation is based on the "Cookbook formulae for audio
+ * EQ biquad filter coefficients" by Robert Bristow-Johnson
+ * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
+ */
+/* Implementation note: For the shelf filters, the specified gain is for the
+ * reference frequency, which is the centerpoint of the transition band. This
+ * better matches EFX filter design. To set the gain for the shelf itself, use
+ * the square root of the desired linear gain (or halve the dB gain).
+ */
+
+typedef enum ALfilterType {
+    /** EFX-style low-pass filter, specifying a gain and reference frequency. */
+    ALfilterType_HighShelf,
+    /** EFX-style high-pass filter, specifying a gain and reference frequency. */
+    ALfilterType_LowShelf,
+    /** Peaking filter, specifying a gain and reference frequency. */
+    ALfilterType_Peaking,
+
+    /** Low-pass cut-off filter, specifying a cut-off frequency. */
+    ALfilterType_LowPass,
+    /** High-pass cut-off filter, specifying a cut-off frequency. */
+    ALfilterType_HighPass,
+    /** Band-pass filter, specifying a center frequency. */
+    ALfilterType_BandPass,
+} ALfilterType;
+
+typedef struct ALfilterState {
+    ALfloat x[2]; /* History of two last input samples  */
+    ALfloat y[2]; /* History of two last output samples */
+    ALfloat b0, b1, b2; /* Transfer function coefficients "b" */
+    ALfloat a1, a2; /* Transfer function coefficients "a" (a0 is pre-applied) */
+} ALfilterState;
+/* Currently only a C-based filter process method is implemented. */
+#define ALfilterState_process ALfilterState_processC
+
+/* Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
+ * reference gain and shelf slope parameter.
+ * 0 < gain
+ * 0 < slope <= 1
+ */
+inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope)
 {
-    ALfloat *history = &iir->history[offset];
-    ALfloat a = iir->coeff;
-    ALfloat output = input;
-
-    output = output + (history[0]-output)*a;
-    history[0] = output;
-    output = output + (history[1]-output)*a;
-    history[1] = output;
-    output = output + (history[2]-output)*a;
-    history[2] = output;
-    output = output + (history[3]-output)*a;
-    history[3] = output;
-
-    return output;
+    return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f);
+}
+/* Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the frequency
+ * multiple (i.e. ref_freq / sampling_freq) and bandwidth.
+ * 0 < freq_mult < 0.5.
+ */
+inline ALfloat calc_rcpQ_from_bandwidth(ALfloat freq_mult, ALfloat bandwidth)
+{
+    ALfloat w0 = F_TAU * freq_mult;
+    return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
 }
 
-static __inline ALfloat lpFilter2P(FILTER *iir, ALuint offset, ALfloat input)
+inline void ALfilterState_clear(ALfilterState *filter)
 {
-    ALfloat *history = &iir->history[offset];
-    ALfloat a = iir->coeff;
-    ALfloat output = input;
+    filter->x[0] = 0.0f;
+    filter->x[1] = 0.0f;
+    filter->y[0] = 0.0f;
+    filter->y[1] = 0.0f;
+}
 
-    output = output + (history[0]-output)*a;
-    history[0] = output;
-    output = output + (history[1]-output)*a;
-    history[1] = output;
+void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat rcpQ);
 
-    return output;
+inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src)
+{
+    dst->b0 = src->b0;
+    dst->b1 = src->b1;
+    dst->b2 = src->b2;
+    dst->a1 = src->a1;
+    dst->a2 = src->a2;
+}
+
+void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples);
+
+inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples)
+{
+    if(numsamples >= 2)
+    {
+        filter->x[1] = src[numsamples-2];
+        filter->x[0] = src[numsamples-1];
+        filter->y[1] = src[numsamples-2];
+        filter->y[0] = src[numsamples-1];
+    }
+    else if(numsamples == 1)
+    {
+        filter->x[1] = filter->x[0];
+        filter->x[0] = src[0];
+        filter->y[1] = filter->y[0];
+        filter->y[0] = src[0];
+    }
 }
 
 
-#define AL_FILTER_TYPE                                     0x8001
-
-#define AL_FILTER_NULL                                     0x0000
-#define AL_FILTER_LOWPASS                                  0x0001
-#define AL_FILTER_HIGHPASS                                 0x0002
-#define AL_FILTER_BANDPASS                                 0x0003
-
-#define AL_LOWPASS_GAIN                                    0x0001
-#define AL_LOWPASS_GAINHF                                  0x0002
-
-
-typedef struct ALfilter
-{
+typedef struct ALfilter {
     // Filter type (AL_FILTER_NULL, ...)
     ALenum type;
 
     ALfloat Gain;
     ALfloat GainHF;
+    ALfloat HFReference;
+    ALfloat GainLF;
+    ALfloat LFReference;
 
-    // Index to itself
-    ALuint filter;
+    void (*SetParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint val);
+    void (*SetParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals);
+    void (*SetParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val);
+    void (*SetParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals);
 
-    struct ALfilter *next;
+    void (*GetParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *val);
+    void (*GetParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals);
+    void (*GetParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val);
+    void (*GetParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals);
+
+    /* Self ID */
+    ALuint id;
 } ALfilter;
 
-ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters);
-ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, ALuint *filters);
-ALboolean AL_APIENTRY alIsFilter(ALuint filter);
+#define ALfilter_SetParami(x, c, p, v)  ((x)->SetParami((x),(c),(p),(v)))
+#define ALfilter_SetParamiv(x, c, p, v) ((x)->SetParamiv((x),(c),(p),(v)))
+#define ALfilter_SetParamf(x, c, p, v)  ((x)->SetParamf((x),(c),(p),(v)))
+#define ALfilter_SetParamfv(x, c, p, v) ((x)->SetParamfv((x),(c),(p),(v)))
 
-ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint iValue);
-ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, ALint *piValues);
-ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat flValue);
-ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, ALfloat *pflValues);
+#define ALfilter_GetParami(x, c, p, v)  ((x)->GetParami((x),(c),(p),(v)))
+#define ALfilter_GetParamiv(x, c, p, v) ((x)->GetParamiv((x),(c),(p),(v)))
+#define ALfilter_GetParamf(x, c, p, v)  ((x)->GetParamf((x),(c),(p),(v)))
+#define ALfilter_GetParamfv(x, c, p, v) ((x)->GetParamfv((x),(c),(p),(v)))
 
-ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *piValue);
-ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *piValues);
-ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *pflValue);
-ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *pflValues);
+inline void LockFiltersRead(ALCdevice *device)
+{ LockUIntMapRead(&device->FilterMap); }
+inline void UnlockFiltersRead(ALCdevice *device)
+{ UnlockUIntMapRead(&device->FilterMap); }
+inline void LockFiltersWrite(ALCdevice *device)
+{ LockUIntMapWrite(&device->FilterMap); }
+inline void UnlockFiltersWrite(ALCdevice *device)
+{ UnlockUIntMapWrite(&device->FilterMap); }
+
+inline struct ALfilter *LookupFilter(ALCdevice *device, ALuint id)
+{ return (struct ALfilter*)LookupUIntMapKeyNoLock(&device->FilterMap, id); }
+inline struct ALfilter *RemoveFilter(ALCdevice *device, ALuint id)
+{ return (struct ALfilter*)RemoveUIntMapKeyNoLock(&device->FilterMap, id); }
 
 ALvoid ReleaseALFilters(ALCdevice *device);
 

OpenAL32/Include/alListener.h

@@ -1,24 +1,64 @@
 #ifndef _AL_LISTENER_H_
 #define _AL_LISTENER_H_
 
-#include "AL/al.h"
+#include "alMain.h"
+#include "alu.h"
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-#define AL_METERS_PER_UNIT                                 0x20004
-
-typedef struct ALlistener_struct
-{
+struct ALlistenerProps {
     ALfloat Position[3];
     ALfloat Velocity[3];
     ALfloat Forward[3];
     ALfloat Up[3];
     ALfloat Gain;
     ALfloat MetersPerUnit;
+
+    ALfloat DopplerFactor;
+    ALfloat DopplerVelocity;
+    ALfloat SpeedOfSound;
+    ALboolean SourceDistanceModel;
+    enum DistanceModel DistanceModel;
+
+    ATOMIC(struct ALlistenerProps*) next;
+};
+
+typedef struct ALlistener {
+    ALfloat Position[3];
+    ALfloat Velocity[3];
+    ALfloat Forward[3];
+    ALfloat Up[3];
+    ALfloat Gain;
+    ALfloat MetersPerUnit;
+
+    /* Pointer to the most recent property values that are awaiting an update.
+     */
+    ATOMIC(struct ALlistenerProps*) Update;
+
+    /* A linked list of unused property containers, free to use for future
+     * updates.
+     */
+    ATOMIC(struct ALlistenerProps*) FreeList;
+
+    struct {
+        aluMatrixf Matrix;
+        aluVector  Velocity;
+
+        ALfloat Gain;
+        ALfloat MetersPerUnit;
+
+        ALfloat DopplerFactor;
+        ALfloat SpeedOfSound;
+
+        ALboolean SourceDistanceModel;
+        enum DistanceModel DistanceModel;
+    } Params;
 } ALlistener;
 
+void UpdateListenerProps(ALCcontext *context);
+
 #ifdef __cplusplus
 }
 #endif

OpenAL32/Include/alSource.h

@@ -1,68 +1,51 @@
 #ifndef _AL_SOURCE_H_
 #define _AL_SOURCE_H_
 
-#define AL_NUM_SOURCE_PARAMS    128
-
-#define MAX_SENDS                 2
-
-#include "alFilter.h"
+#include "bool.h"
+#include "alMain.h"
 #include "alu.h"
-#include "AL/al.h"
+#include "hrtf.h"
+#include "atomic.h"
 
-#define AL_DIRECT_FILTER                                   0x20005
-#define AL_AUXILIARY_SEND_FILTER                           0x20006
-#define AL_AIR_ABSORPTION_FACTOR                           0x20007
-#define AL_ROOM_ROLLOFF_FACTOR                             0x20008
-#define AL_CONE_OUTER_GAINHF                               0x20009
-#define AL_DIRECT_FILTER_GAINHF_AUTO                       0x2000A
-#define AL_AUXILIARY_SEND_FILTER_GAIN_AUTO                 0x2000B
-#define AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO               0x2000C
+#define MAX_SENDS      16
+#define DEFAULT_SENDS  2
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-typedef struct ALbufferlistitem
-{
-    struct ALbuffer         *buffer;
-    struct ALbufferlistitem *next;
+struct ALbuffer;
+struct ALsource;
+
+
+typedef struct ALbufferlistitem {
+    struct ALbuffer *buffer;
+    ATOMIC(struct ALbufferlistitem*) next;
 } ALbufferlistitem;
 
-typedef struct ALsource
-{
-    ALfloat      flPitch;
-    ALfloat      flGain;
-    ALfloat      flOuterGain;
-    ALfloat      flMinGain;
-    ALfloat      flMaxGain;
-    ALfloat      flInnerAngle;
-    ALfloat      flOuterAngle;
-    ALfloat      flRefDistance;
-    ALfloat      flMaxDistance;
-    ALfloat      flRollOffFactor;
-    ALfloat      vPosition[3];
-    ALfloat      vVelocity[3];
-    ALfloat      vOrientation[3];
-    ALboolean    bHeadRelative;
-    ALboolean    bLooping;
-    ALenum       DistanceModel;
 
-    ALenum       state;
-    ALuint       position;
-    ALuint       position_fraction;
-
-    struct ALbuffer *Buffer;
-
-    struct ALbufferlistitem *queue; // Linked list of buffers in queue
-    ALuint       BuffersInQueue;    // Number of buffers in queue
-    ALuint       BuffersPlayed;     // Number of buffers played on this loop
-
-    ALfilter DirectFilter;
-
-    struct {
-        struct ALeffectslot *Slot;
-        ALfilter WetFilter;
-    } Send[MAX_SENDS];
+typedef struct ALsource {
+    /** Source properties. */
+    ALfloat   Pitch;
+    ALfloat   Gain;
+    ALfloat   OuterGain;
+    ALfloat   MinGain;
+    ALfloat   MaxGain;
+    ALfloat   InnerAngle;
+    ALfloat   OuterAngle;
+    ALfloat   RefDistance;
+    ALfloat   MaxDistance;
+    ALfloat   RolloffFactor;
+    ALfloat   Position[3];
+    ALfloat   Velocity[3];
+    ALfloat   Direction[3];
+    ALfloat   Orientation[2][3];
+    ALboolean HeadRelative;
+    ALboolean Looping;
+    enum DistanceModel DistanceModel;
+    enum Resampler Resampler;
+    ALboolean DirectChannels;
+    enum SpatializeMode Spatialize;
 
     ALboolean DryGainHFAuto;
     ALboolean WetGainAuto;
@@ -73,38 +56,69 @@ typedef struct ALsource
     ALfloat RoomRolloffFactor;
     ALfloat DopplerFactor;
 
-    ALint  lOffset;
-    ALint  lOffsetType;
+    /* NOTE: Stereo pan angles are specified in radians, counter-clockwise
+     * rather than clockwise.
+     */
+    ALfloat StereoPan[2];
 
-    // Source Type (Static, Streaming, or Undetermined)
-    ALint  lSourceType;
+    ALfloat Radius;
 
-    // Current gains, which are ramped while mixed
-    ALfloat DryGains[OUTPUTCHANNELS];
-    ALfloat WetGains[MAX_SENDS];
-    ALboolean FirstStart;
-
-    // Current target parameters used for mixing
+    /** Direct filter and auxiliary send info. */
     struct {
-        ALfloat DryGains[OUTPUTCHANNELS];
-        ALfloat WetGains[MAX_SENDS];
-        ALfloat Pitch;
+        ALfloat Gain;
+        ALfloat GainHF;
+        ALfloat HFReference;
+        ALfloat GainLF;
+        ALfloat LFReference;
+    } Direct;
+    struct {
+        struct ALeffectslot *Slot;
+        ALfloat Gain;
+        ALfloat GainHF;
+        ALfloat HFReference;
+        ALfloat GainLF;
+        ALfloat LFReference;
+    } *Send;
 
-        struct {
-            FILTER iirFilter;
-            ALfloat history[2];
-        } Send[MAX_SENDS];
+    /**
+     * Last user-specified offset, and the offset type (bytes, samples, or
+     * seconds).
+     */
+    ALdouble Offset;
+    ALenum   OffsetType;
 
-        FILTER iirFilter;
-        ALfloat history[OUTPUTCHANNELS*2];
-    } Params;
+    /** Source type (static, streaming, or undetermined) */
+    ALint SourceType;
 
-    // Index to itself
-    ALuint source;
+    /** Source state (initial, playing, paused, or stopped) */
+    ATOMIC(ALenum) state;
 
-    struct ALsource *next;
+    /** Source Buffer Queue head. */
+    RWLock queue_lock;
+    ALbufferlistitem *queue;
+
+    ATOMIC_FLAG PropsClean;
+
+    /** Self ID */
+    ALuint id;
 } ALsource;
 
+inline void LockSourcesRead(ALCcontext *context)
+{ LockUIntMapRead(&context->SourceMap); }
+inline void UnlockSourcesRead(ALCcontext *context)
+{ UnlockUIntMapRead(&context->SourceMap); }
+inline void LockSourcesWrite(ALCcontext *context)
+{ LockUIntMapWrite(&context->SourceMap); }
+inline void UnlockSourcesWrite(ALCcontext *context)
+{ UnlockUIntMapWrite(&context->SourceMap); }
+
+inline struct ALsource *LookupSource(ALCcontext *context, ALuint id)
+{ return (struct ALsource*)LookupUIntMapKeyNoLock(&context->SourceMap, id); }
+inline struct ALsource *RemoveSource(ALCcontext *context, ALuint id)
+{ return (struct ALsource*)RemoveUIntMapKeyNoLock(&context->SourceMap, id); }
+
+void UpdateAllSourceProps(ALCcontext *context);
+
 ALvoid ReleaseALSources(ALCcontext *Context);
 
 #ifdef __cplusplus

OpenAL32/Include/alState.h

@@ -1,14 +0,0 @@
-#ifndef _AL_STATE_H_
-#define _AL_STATE_H_
-
-#include "AL/al.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alThunk.h

@@ -1,42 +1,20 @@
-#ifndef _AL_THUNK_H_
-#define _AL_THUNK_H_
+#ifndef ALTHUNK_H
+#define ALTHUNK_H
 
-#include "config.h"
-
-#include "AL/al.h"
-#include "AL/alc.h"
+#include "alMain.h"
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-void alThunkInit(void);
-void alThunkExit(void);
-ALuint alThunkAddEntry(ALvoid * ptr);
-void alThunkRemoveEntry(ALuint index);
-ALvoid *alThunkLookupEntry(ALuint index);
-
-#if (SIZEOF_VOIDP > SIZEOF_UINT)
-
-#define ALTHUNK_INIT()          alThunkInit()
-#define ALTHUNK_EXIT()          alThunkExit()
-#define ALTHUNK_ADDENTRY(p)     alThunkAddEntry(p)
-#define ALTHUNK_REMOVEENTRY(i)  alThunkRemoveEntry(i)
-#define ALTHUNK_LOOKUPENTRY(i)  alThunkLookupEntry(i)
-
-#else
-
-#define ALTHUNK_INIT()
-#define ALTHUNK_EXIT()
-#define ALTHUNK_ADDENTRY(p)     ((ALuint)p)
-#define ALTHUNK_REMOVEENTRY(i)
-#define ALTHUNK_LOOKUPENTRY(i)  ((ALvoid*)(i))
-
-#endif // (SIZEOF_VOIDP > SIZEOF_INT)
+void ThunkInit(void);
+void ThunkExit(void);
+ALenum NewThunkEntry(ALuint *index);
+void FreeThunkEntry(ALuint index);
 
 #ifdef __cplusplus
 }
 #endif
 
-#endif //_AL_THUNK_H_
+#endif //ALTHUNK_H
 

OpenAL32/Include/alu.h

@@ -1,157 +1,514 @@
 #ifndef _ALU_H_
 #define _ALU_H_
 
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "AL/alext.h"
-
+#include <limits.h>
+#include <math.h>
 #ifdef HAVE_FLOAT_H
 #include <float.h>
 #endif
-
-#ifndef M_PI
-#define M_PI           3.14159265358979323846  /* pi */
-#define M_PI_2         1.57079632679489661923  /* pi/2 */
+#ifdef HAVE_IEEEFP_H
+#include <ieeefp.h>
 #endif
 
-#ifdef HAVE_SQRTF
-#define aluSqrt(x) ((ALfloat)sqrtf((float)(x)))
-#else
-#define aluSqrt(x) ((ALfloat)sqrt((double)(x)))
-#endif
+#include "alMain.h"
+#include "alBuffer.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
 
-#ifdef HAVE_ACOSF
-#define aluAcos(x) ((ALfloat)acosf((float)(x)))
-#else
-#define aluAcos(x) ((ALfloat)acos((double)(x)))
-#endif
+#include "hrtf.h"
+#include "align.h"
+#include "nfcfilter.h"
+#include "math_defs.h"
 
-#ifdef HAVE_ATANF
-#define aluAtan(x) ((ALfloat)atanf((float)(x)))
-#else
-#define aluAtan(x) ((ALfloat)atan((double)(x)))
-#endif
 
-#ifdef HAVE_FABSF
-#define aluFabs(x) ((ALfloat)fabsf((float)(x)))
-#else
-#define aluFabs(x) ((ALfloat)fabs((double)(x)))
-#endif
+#define MAX_PITCH  (255)
+
+/* Maximum number of buffer samples before the current pos needed for resampling. */
+#define MAX_PRE_SAMPLES 12
+
+/* Maximum number of buffer samples after the current pos needed for resampling. */
+#define MAX_POST_SAMPLES 12
 
-// fixes for mingw32.
-#if defined(max) && !defined(__max)
-#define __max max
-#endif
-#if defined(min) && !defined(__min)
-#define __min min
-#endif
 
 #ifdef __cplusplus
 extern "C" {
 #endif
 
-enum {
-    FRONT_LEFT = 0,
-    FRONT_RIGHT,
-    FRONT_CENTER,
-    SIDE_LEFT,
-    SIDE_RIGHT,
-    BACK_LEFT,
-    BACK_RIGHT,
-    BACK_CENTER,
-    LFE,
+struct ALsource;
+struct ALbufferlistitem;
+struct ALvoice;
+struct ALeffectslot;
 
-    OUTPUTCHANNELS
+
+#define DITHER_RNG_SEED 22222
+
+
+enum SpatializeMode {
+    SpatializeOff = AL_FALSE,
+    SpatializeOn = AL_TRUE,
+    SpatializeAuto = AL_AUTO_SOFT
 };
 
-#define BUFFERSIZE 24000
+enum Resampler {
+    PointResampler,
+    LinearResampler,
+    FIR4Resampler,
+    BSincResampler,
 
-extern ALboolean DuplicateStereo;
+    ResamplerMax = BSincResampler
+};
+extern enum Resampler ResamplerDefault;
 
-/* NOTE: The AL_FORMAT_REAR* enums aren't handled here be cause they're
- *       converted to AL_FORMAT_QUAD* when loaded */
-static __inline ALuint aluBytesFromFormat(ALenum format)
+/* The number of distinct scale and phase intervals within the filter table. */
+#define BSINC_SCALE_BITS  4
+#define BSINC_SCALE_COUNT (1<<BSINC_SCALE_BITS)
+#define BSINC_PHASE_BITS  4
+#define BSINC_PHASE_COUNT (1<<BSINC_PHASE_BITS)
+
+/* Interpolator state.  Kind of a misnomer since the interpolator itself is
+ * stateless.  This just keeps it from having to recompute scale-related
+ * mappings for every sample.
+ */
+typedef struct BsincState {
+    ALfloat sf; /* Scale interpolation factor. */
+    ALuint m;   /* Coefficient count. */
+    ALint l;    /* Left coefficient offset. */
+    struct {
+        const ALfloat *filter;   /* Filter coefficients. */
+        const ALfloat *scDelta;  /* Scale deltas. */
+        const ALfloat *phDelta;  /* Phase deltas. */
+        const ALfloat *spDelta;  /* Scale-phase deltas. */
+    } coeffs[BSINC_PHASE_COUNT];
+} BsincState;
+
+typedef union InterpState {
+    BsincState bsinc;
+} InterpState;
+
+typedef const ALfloat* (*ResamplerFunc)(const InterpState *state,
+    const ALfloat *restrict src, ALsizei frac, ALint increment,
+    ALfloat *restrict dst, ALsizei dstlen
+);
+
+
+typedef union aluVector {
+    alignas(16) ALfloat v[4];
+} aluVector;
+
+inline void aluVectorSet(aluVector *vector, ALfloat x, ALfloat y, ALfloat z, ALfloat w)
 {
-    switch(format)
-    {
-        case AL_FORMAT_MONO8:
-        case AL_FORMAT_STEREO8:
-        case AL_FORMAT_QUAD8_LOKI:
-        case AL_FORMAT_QUAD8:
-        case AL_FORMAT_51CHN8:
-        case AL_FORMAT_61CHN8:
-        case AL_FORMAT_71CHN8:
-            return 1;
-
-        case AL_FORMAT_MONO16:
-        case AL_FORMAT_STEREO16:
-        case AL_FORMAT_QUAD16_LOKI:
-        case AL_FORMAT_QUAD16:
-        case AL_FORMAT_51CHN16:
-        case AL_FORMAT_61CHN16:
-        case AL_FORMAT_71CHN16:
-            return 2;
-
-        case AL_FORMAT_MONO_FLOAT32:
-        case AL_FORMAT_STEREO_FLOAT32:
-        case AL_FORMAT_QUAD32:
-        case AL_FORMAT_51CHN32:
-        case AL_FORMAT_61CHN32:
-        case AL_FORMAT_71CHN32:
-            return 4;
-
-        default:
-            return 0;
-    }
-}
-static __inline ALuint aluChannelsFromFormat(ALenum format)
-{
-    switch(format)
-    {
-        case AL_FORMAT_MONO8:
-        case AL_FORMAT_MONO16:
-        case AL_FORMAT_MONO_FLOAT32:
-            return 1;
-
-        case AL_FORMAT_STEREO8:
-        case AL_FORMAT_STEREO16:
-        case AL_FORMAT_STEREO_FLOAT32:
-            return 2;
-
-        case AL_FORMAT_QUAD8_LOKI:
-        case AL_FORMAT_QUAD16_LOKI:
-        case AL_FORMAT_QUAD8:
-        case AL_FORMAT_QUAD16:
-        case AL_FORMAT_QUAD32:
-            return 4;
-
-        case AL_FORMAT_51CHN8:
-        case AL_FORMAT_51CHN16:
-        case AL_FORMAT_51CHN32:
-            return 6;
-
-        case AL_FORMAT_61CHN8:
-        case AL_FORMAT_61CHN16:
-        case AL_FORMAT_61CHN32:
-            return 7;
-
-        case AL_FORMAT_71CHN8:
-        case AL_FORMAT_71CHN16:
-        case AL_FORMAT_71CHN32:
-            return 8;
-
-        default:
-            return 0;
-    }
+    vector->v[0] = x;
+    vector->v[1] = y;
+    vector->v[2] = z;
+    vector->v[3] = w;
 }
 
-ALvoid aluInitPanning(ALCcontext *Context);
-ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size);
-ALvoid aluHandleDisconnect(ALCdevice *device);
+
+typedef union aluMatrixf {
+    alignas(16) ALfloat m[4][4];
+} aluMatrixf;
+extern const aluMatrixf IdentityMatrixf;
+
+inline void aluMatrixfSetRow(aluMatrixf *matrix, ALuint row,
+                             ALfloat m0, ALfloat m1, ALfloat m2, ALfloat m3)
+{
+    matrix->m[row][0] = m0;
+    matrix->m[row][1] = m1;
+    matrix->m[row][2] = m2;
+    matrix->m[row][3] = m3;
+}
+
+inline void aluMatrixfSet(aluMatrixf *matrix, ALfloat m00, ALfloat m01, ALfloat m02, ALfloat m03,
+                                              ALfloat m10, ALfloat m11, ALfloat m12, ALfloat m13,
+                                              ALfloat m20, ALfloat m21, ALfloat m22, ALfloat m23,
+                                              ALfloat m30, ALfloat m31, ALfloat m32, ALfloat m33)
+{
+    aluMatrixfSetRow(matrix, 0, m00, m01, m02, m03);
+    aluMatrixfSetRow(matrix, 1, m10, m11, m12, m13);
+    aluMatrixfSetRow(matrix, 2, m20, m21, m22, m23);
+    aluMatrixfSetRow(matrix, 3, m30, m31, m32, m33);
+}
+
+
+enum ActiveFilters {
+    AF_None = 0,
+    AF_LowPass = 1,
+    AF_HighPass = 2,
+    AF_BandPass = AF_LowPass | AF_HighPass
+};
+
+
+typedef struct MixHrtfParams {
+    const ALfloat (*Coeffs)[2];
+    ALsizei Delay[2];
+    ALfloat Gain;
+    ALfloat GainStep;
+} MixHrtfParams;
+
+
+typedef struct DirectParams {
+    ALfilterState LowPass;
+    ALfilterState HighPass;
+
+    NfcFilter NFCtrlFilter[MAX_AMBI_ORDER];
+
+    struct {
+        HrtfParams Old;
+        HrtfParams Target;
+        HrtfState State;
+    } Hrtf;
+
+    struct {
+        ALfloat Current[MAX_OUTPUT_CHANNELS];
+        ALfloat Target[MAX_OUTPUT_CHANNELS];
+    } Gains;
+} DirectParams;
+
+typedef struct SendParams {
+    ALfilterState LowPass;
+    ALfilterState HighPass;
+
+    struct {
+        ALfloat Current[MAX_OUTPUT_CHANNELS];
+        ALfloat Target[MAX_OUTPUT_CHANNELS];
+    } Gains;
+} SendParams;
+
+
+struct ALvoiceProps {
+    ATOMIC(struct ALvoiceProps*) next;
+
+    ALfloat Pitch;
+    ALfloat Gain;
+    ALfloat OuterGain;
+    ALfloat MinGain;
+    ALfloat MaxGain;
+    ALfloat InnerAngle;
+    ALfloat OuterAngle;
+    ALfloat RefDistance;
+    ALfloat MaxDistance;
+    ALfloat RolloffFactor;
+    ALfloat Position[3];
+    ALfloat Velocity[3];
+    ALfloat Direction[3];
+    ALfloat Orientation[2][3];
+    ALboolean HeadRelative;
+    enum DistanceModel DistanceModel;
+    enum Resampler Resampler;
+    ALboolean DirectChannels;
+    enum SpatializeMode SpatializeMode;
+
+    ALboolean DryGainHFAuto;
+    ALboolean WetGainAuto;
+    ALboolean WetGainHFAuto;
+    ALfloat   OuterGainHF;
+
+    ALfloat AirAbsorptionFactor;
+    ALfloat RoomRolloffFactor;
+    ALfloat DopplerFactor;
+
+    ALfloat StereoPan[2];
+
+    ALfloat Radius;
+
+    /** Direct filter and auxiliary send info. */
+    struct {
+        ALfloat Gain;
+        ALfloat GainHF;
+        ALfloat HFReference;
+        ALfloat GainLF;
+        ALfloat LFReference;
+    } Direct;
+    struct {
+        struct ALeffectslot *Slot;
+        ALfloat Gain;
+        ALfloat GainHF;
+        ALfloat HFReference;
+        ALfloat GainLF;
+        ALfloat LFReference;
+    } Send[];
+};
+
+/* If not 'fading', gain targets are used directly without fading. */
+#define VOICE_IS_FADING (1<<0)
+#define VOICE_HAS_HRTF  (1<<1)
+#define VOICE_HAS_NFC   (1<<2)
+
+typedef struct ALvoice {
+    struct ALvoiceProps *Props;
+
+    ATOMIC(struct ALvoiceProps*) Update;
+    ATOMIC(struct ALvoiceProps*) FreeList;
+
+    ATOMIC(struct ALsource*) Source;
+    ATOMIC(bool) Playing;
+
+    /**
+     * Source offset in samples, relative to the currently playing buffer, NOT
+     * the whole queue, and the fractional (fixed-point) offset to the next
+     * sample.
+     */
+    ATOMIC(ALuint) position;
+    ATOMIC(ALsizei) position_fraction;
+
+    /* Current buffer queue item being played. */
+    ATOMIC(struct ALbufferlistitem*) current_buffer;
+
+    /* Buffer queue item to loop to at end of queue (will be NULL for non-
+     * looping voices).
+     */
+    ATOMIC(struct ALbufferlistitem*) loop_buffer;
+
+    /**
+     * Number of channels and bytes-per-sample for the attached source's
+     * buffer(s).
+     */
+    ALsizei NumChannels;
+    ALsizei SampleSize;
+
+    /** Current target parameters used for mixing. */
+    ALint Step;
+
+    ResamplerFunc Resampler;
+
+    ALuint Flags;
+
+    ALuint Offset; /* Number of output samples mixed since starting. */
+
+    alignas(16) ALfloat PrevSamples[MAX_INPUT_CHANNELS][MAX_PRE_SAMPLES];
+
+    InterpState ResampleState;
+
+    struct {
+        enum ActiveFilters FilterType;
+        DirectParams Params[MAX_INPUT_CHANNELS];
+
+        ALfloat (*Buffer)[BUFFERSIZE];
+        ALsizei Channels;
+        ALsizei ChannelsPerOrder[MAX_AMBI_ORDER+1];
+    } Direct;
+
+    struct {
+        enum ActiveFilters FilterType;
+        SendParams Params[MAX_INPUT_CHANNELS];
+
+        ALfloat (*Buffer)[BUFFERSIZE];
+        ALsizei Channels;
+    } Send[];
+} ALvoice;
+
+void DeinitVoice(ALvoice *voice);
+
+
+typedef void (*MixerFunc)(const ALfloat *data, ALsizei OutChans,
+                          ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains,
+                          const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+                          ALsizei BufferSize);
+typedef void (*RowMixerFunc)(ALfloat *OutBuffer, const ALfloat *gains,
+                             const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
+                             ALsizei InPos, ALsizei BufferSize);
+typedef void (*HrtfMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                              const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                              const ALsizei IrSize, MixHrtfParams *hrtfparams,
+                              HrtfState *hrtfstate, ALsizei BufferSize);
+typedef void (*HrtfMixerBlendFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                                   const ALfloat *data, ALsizei Offset, ALsizei OutPos,
+                                   const ALsizei IrSize, const HrtfParams *oldparams,
+                                   MixHrtfParams *newparams, HrtfState *hrtfstate,
+                                   ALsizei BufferSize);
+typedef void (*HrtfDirectMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
+                                    const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
+                                    const ALfloat (*restrict Coeffs)[2],
+                                    ALfloat (*restrict Values)[2], ALsizei BufferSize);
+
+
+#define GAIN_MIX_MAX  (16.0f) /* +24dB */
+
+#define GAIN_SILENCE_THRESHOLD  (0.00001f) /* -100dB */
+
+#define SPEEDOFSOUNDMETRESPERSEC  (343.3f)
+#define AIRABSORBGAINHF           (0.99426f) /* -0.05dB */
+
+/* Target gain for the reverb decay feedback reaching the decay time. */
+#define REVERB_DECAY_GAIN  (0.001f) /* -60 dB */
+
+#define FRACTIONBITS (12)
+#define FRACTIONONE  (1<<FRACTIONBITS)
+#define FRACTIONMASK (FRACTIONONE-1)
+
+
+inline ALfloat minf(ALfloat a, ALfloat b)
+{ return ((a > b) ? b : a); }
+inline ALfloat maxf(ALfloat a, ALfloat b)
+{ return ((a > b) ? a : b); }
+inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max)
+{ return minf(max, maxf(min, val)); }
+
+inline ALdouble mind(ALdouble a, ALdouble b)
+{ return ((a > b) ? b : a); }
+inline ALdouble maxd(ALdouble a, ALdouble b)
+{ return ((a > b) ? a : b); }
+inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max)
+{ return mind(max, maxd(min, val)); }
+
+inline ALuint minu(ALuint a, ALuint b)
+{ return ((a > b) ? b : a); }
+inline ALuint maxu(ALuint a, ALuint b)
+{ return ((a > b) ? a : b); }
+inline ALuint clampu(ALuint val, ALuint min, ALuint max)
+{ return minu(max, maxu(min, val)); }
+
+inline ALint mini(ALint a, ALint b)
+{ return ((a > b) ? b : a); }
+inline ALint maxi(ALint a, ALint b)
+{ return ((a > b) ? a : b); }
+inline ALint clampi(ALint val, ALint min, ALint max)
+{ return mini(max, maxi(min, val)); }
+
+inline ALint64 mini64(ALint64 a, ALint64 b)
+{ return ((a > b) ? b : a); }
+inline ALint64 maxi64(ALint64 a, ALint64 b)
+{ return ((a > b) ? a : b); }
+inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max)
+{ return mini64(max, maxi64(min, val)); }
+
+inline ALuint64 minu64(ALuint64 a, ALuint64 b)
+{ return ((a > b) ? b : a); }
+inline ALuint64 maxu64(ALuint64 a, ALuint64 b)
+{ return ((a > b) ? a : b); }
+inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max)
+{ return minu64(max, maxu64(min, val)); }
+
+
+extern alignas(16) const ALfloat bsincTab[18840];
+extern alignas(16) const ALfloat sinc4Tab[FRACTIONONE][4];
+
+
+inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu)
+{
+    return val1 + (val2-val1)*mu;
+}
+inline ALfloat resample_fir4(ALfloat val0, ALfloat val1, ALfloat val2, ALfloat val3, ALsizei frac)
+{
+    return sinc4Tab[frac][0]*val0 + sinc4Tab[frac][1]*val1 +
+           sinc4Tab[frac][2]*val2 + sinc4Tab[frac][3]*val3;
+}
+
+
+enum HrtfRequestMode {
+    Hrtf_Default = 0,
+    Hrtf_Enable = 1,
+    Hrtf_Disable = 2,
+};
+
+void aluInitMixer(void);
+
+MixerFunc SelectMixer(void);
+RowMixerFunc SelectRowMixer(void);
+ResamplerFunc SelectResampler(enum Resampler resampler);
+
+/* aluInitRenderer
+ *
+ * Set up the appropriate panning method and mixing method given the device
+ * properties.
+ */
+void aluInitRenderer(ALCdevice *device, ALint hrtf_id, enum HrtfRequestMode hrtf_appreq, enum HrtfRequestMode hrtf_userreq);
+
+void aluInitEffectPanning(struct ALeffectslot *slot);
+
+/**
+ * CalcDirectionCoeffs
+ *
+ * Calculates ambisonic coefficients based on a direction vector. The vector
+ * must be normalized (unit length), and the spread is the angular width of the
+ * sound (0...tau).
+ */
+void CalcDirectionCoeffs(const ALfloat dir[3], ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]);
+
+/**
+ * CalcAngleCoeffs
+ *
+ * Calculates ambisonic coefficients based on azimuth and elevation. The
+ * azimuth and elevation parameters are in radians, going right and up
+ * respectively.
+ */
+inline void CalcAngleCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS])
+{
+    ALfloat dir[3] = {
+        sinf(azimuth) * cosf(elevation),
+        sinf(elevation),
+        -cosf(azimuth) * cosf(elevation)
+    };
+    CalcDirectionCoeffs(dir, spread, coeffs);
+}
+
+/**
+ * CalcAnglePairwiseCoeffs
+ *
+ * Calculates ambisonic coefficients based on azimuth and elevation. The
+ * azimuth and elevation parameters are in radians, going right and up
+ * respectively. This pairwise variant warps the result such that +30 azimuth
+ * is full right, and -30 azimuth is full left.
+ */
+void CalcAnglePairwiseCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]);
+
+/**
+ * ComputeAmbientGains
+ *
+ * Computes channel gains for ambient, omni-directional sounds.
+ */
+#define ComputeAmbientGains(b, g, o) do {                                     \
+    if((b).CoeffCount > 0)                                                    \
+        ComputeAmbientGainsMC((b).Ambi.Coeffs, (b).NumChannels, g, o);        \
+    else                                                                      \
+        ComputeAmbientGainsBF((b).Ambi.Map, (b).NumChannels, g, o);           \
+} while (0)
+void ComputeAmbientGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+void ComputeAmbientGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+
+/**
+ * ComputePanningGains
+ *
+ * Computes panning gains using the given channel decoder coefficients and the
+ * pre-calculated direction or angle coefficients.
+ */
+#define ComputePanningGains(b, c, g, o) do {                                  \
+    if((b).CoeffCount > 0)                                                    \
+        ComputePanningGainsMC((b).Ambi.Coeffs, (b).NumChannels, (b).CoeffCount, c, g, o);\
+    else                                                                      \
+        ComputePanningGainsBF((b).Ambi.Map, (b).NumChannels, c, g, o);        \
+} while (0)
+void ComputePanningGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALsizei numcoeffs, const ALfloat coeffs[MAX_AMBI_COEFFS], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+void ComputePanningGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat coeffs[MAX_AMBI_COEFFS], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+
+/**
+ * ComputeFirstOrderGains
+ *
+ * Sets channel gains for a first-order ambisonics input channel. The matrix is
+ * a 1x4 'slice' of a transform matrix for the input channel, used to scale and
+ * orient the sound samples.
+ */
+#define ComputeFirstOrderGains(b, m, g, o) do {                               \
+    if((b).CoeffCount > 0)                                                    \
+        ComputeFirstOrderGainsMC((b).Ambi.Coeffs, (b).NumChannels, m, g, o);  \
+    else                                                                      \
+        ComputeFirstOrderGainsBF((b).Ambi.Map, (b).NumChannels, m, g, o);     \
+} while (0)
+void ComputeFirstOrderGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+void ComputeFirstOrderGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+
+
+ALboolean MixSource(struct ALvoice *voice, struct ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo);
+
+void aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size);
+/* Caller must lock the device. */
+void aluHandleDisconnect(ALCdevice *device);
+
+extern ALfloat ConeScale;
+extern ALfloat ZScale;
 
 #ifdef __cplusplus
 }
 #endif
 
 #endif
-

OpenAL32/Include/bs2b.h

@@ -1,109 +1,94 @@
-/*-
- * Copyright (c) 2005 Boris Mikhaylov
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject to
- * the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#ifndef BS2B_H
-#define BS2B_H
-
-/* Number of crossfeed levels */
-#define BS2B_CLEVELS           3
-
-/* Normal crossfeed levels */
-#define BS2B_HIGH_CLEVEL       3
-#define BS2B_MIDDLE_CLEVEL     2
-#define BS2B_LOW_CLEVEL        1
-
-/* Easy crossfeed levels */
-#define BS2B_HIGH_ECLEVEL      BS2B_HIGH_CLEVEL    + BS2B_CLEVELS
-#define BS2B_MIDDLE_ECLEVEL    BS2B_MIDDLE_CLEVEL  + BS2B_CLEVELS
-#define BS2B_LOW_ECLEVEL       BS2B_LOW_CLEVEL     + BS2B_CLEVELS
-
-/* Default crossfeed levels */
-#define BS2B_DEFAULT_CLEVEL    BS2B_HIGH_ECLEVEL
-/* Default sample rate (Hz) */
-#define BS2B_DEFAULT_SRATE     44100
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-struct bs2b {
-    int level;  /* Crossfeed level */
-    int srate;   /* Sample rate (Hz) */
-
-    /* Lowpass IIR filter coefficients */
-    double a0_lo;
-    double b1_lo;
-
-    /* Highboost IIR filter coefficients */
-    double a0_hi;
-    double a1_hi;
-    double b1_hi;
-
-    /* Global gain against overloading */
-    double gain;
-
-    /* Buffer of last filtered sample.
-     * [0] - first channel, [1] - second channel
-     */
-    struct t_last_sample {
-        double asis[2];
-        double lo[2];
-        double hi[2];
-    } last_sample;
-};
-
-/* Clear buffers and set new coefficients with new crossfeed level value.
- * level - crossfeed level of *LEVEL values.
- */
-void bs2b_set_level(struct bs2b *bs2b, int level);
-
-/* Return current crossfeed level value */
-int bs2b_get_level(struct bs2b *bs2b);
-
-/* Clear buffers and set new coefficients with new sample rate value.
- * srate - sample rate by Hz.
- */
-void bs2b_set_srate(struct bs2b *bs2b, int srate);
-
-/* Return current sample rate value */
-int bs2b_get_srate(struct bs2b *bs2b);
-
-/* Clear buffer */
-void bs2b_clear(struct bs2b *bs2b);
-
-/* Return 1 if buffer is clear */
-int bs2b_is_clear(struct bs2b *bs2b);
-
-/* Crossfeeds one stereo sample that are pointed by sample.
- * [0] - first channel, [1] - second channel.
- * Returns crossfided samle by sample pointer.
- */
-
-/* sample poits to floats */
-void bs2b_cross_feed(struct bs2b *bs2b, float *sample);
-
-#ifdef __cplusplus
-}    /* extern "C" */
-#endif /* __cplusplus */
-
-#endif /* BS2B_H */
+/*-
+ * Copyright (c) 2005 Boris Mikhaylov
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files (the
+ * "Software"), to deal in the Software without restriction, including
+ * without limitation the rights to use, copy, modify, merge, publish,
+ * distribute, sublicense, and/or sell copies of the Software, and to
+ * permit persons to whom the Software is furnished to do so, subject to
+ * the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
+ * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
+ * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
+ * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+#ifndef BS2B_H
+#define BS2B_H
+
+/* Number of crossfeed levels */
+#define BS2B_CLEVELS           3
+
+/* Normal crossfeed levels */
+#define BS2B_HIGH_CLEVEL       3
+#define BS2B_MIDDLE_CLEVEL     2
+#define BS2B_LOW_CLEVEL        1
+
+/* Easy crossfeed levels */
+#define BS2B_HIGH_ECLEVEL      BS2B_HIGH_CLEVEL    + BS2B_CLEVELS
+#define BS2B_MIDDLE_ECLEVEL    BS2B_MIDDLE_CLEVEL  + BS2B_CLEVELS
+#define BS2B_LOW_ECLEVEL       BS2B_LOW_CLEVEL     + BS2B_CLEVELS
+
+/* Default crossfeed levels */
+#define BS2B_DEFAULT_CLEVEL    BS2B_HIGH_ECLEVEL
+/* Default sample rate (Hz) */
+#define BS2B_DEFAULT_SRATE     44100
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+struct bs2b {
+    int level;  /* Crossfeed level */
+    int srate;   /* Sample rate (Hz) */
+
+    /* Lowpass IIR filter coefficients */
+    float a0_lo;
+    float b1_lo;
+
+    /* Highboost IIR filter coefficients */
+    float a0_hi;
+    float a1_hi;
+    float b1_hi;
+
+    /* Buffer of last filtered sample.
+     * [0] - first channel, [1] - second channel
+     */
+    struct t_last_sample {
+        float asis;
+        float lo;
+        float hi;
+    } last_sample[2];
+};
+
+/* Clear buffers and set new coefficients with new crossfeed level and sample
+ * rate values.
+ * level - crossfeed level of *LEVEL values.
+ * srate - sample rate by Hz.
+ */
+void bs2b_set_params(struct bs2b *bs2b, int level, int srate);
+
+/* Return current crossfeed level value */
+int bs2b_get_level(struct bs2b *bs2b);
+
+/* Return current sample rate value */
+int bs2b_get_srate(struct bs2b *bs2b);
+
+/* Clear buffer */
+void bs2b_clear(struct bs2b *bs2b);
+
+void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo);
+
+#ifdef __cplusplus
+}    /* extern "C" */
+#endif /* __cplusplus */
+
+#endif /* BS2B_H */

OpenAL32/Include/sample_cvt.h

@@ -0,0 +1,9 @@
+#ifndef SAMPLE_CVT_H
+#define SAMPLE_CVT_H
+
+#include "AL/al.h"
+#include "alBuffer.h"
+
+void ConvertData(ALvoid *dst, enum UserFmtType dstType, const ALvoid *src, enum UserFmtType srcType, ALsizei numchans, ALsizei len, ALsizei align);
+
+#endif /* SAMPLE_CVT_H */

OpenAL32/alDatabuffer.c

@@ -1,639 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <assert.h>
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "AL/alext.h"
-#include "alError.h"
-#include "alDatabuffer.h"
-#include "alThunk.h"
-
-
-/*
-*    alGenDatabuffersEXT(ALsizei n, ALuint *puiBuffers)
-*
-*    Generates n AL Databuffers, and stores the Databuffers Names in the array pointed to by puiBuffers
-*/
-ALvoid ALAPIENTRY alGenDatabuffersEXT(ALsizei n,ALuint *puiBuffers)
-{
-    ALCcontext *Context;
-    ALsizei i=0;
-
-    Context = GetContextSuspended();
-    if(!Context) return;
-
-    /* Check that we are actually generation some Databuffers */
-    if(n > 0)
-    {
-        ALCdevice *device = Context->Device;
-
-        /* Check the pointer is valid (and points to enough memory to store
-         * Databuffer Names) */
-        if(!IsBadWritePtr((void*)puiBuffers, n * sizeof(ALuint)))
-        {
-            ALdatabuffer **list = &device->Databuffers;
-            while(*list)
-                list = &(*list)->next;
-
-            /* Create all the new Databuffers */
-            while(i < n)
-            {
-                *list = calloc(1, sizeof(ALdatabuffer));
-                if(!(*list))
-                {
-                    alDeleteDatabuffersEXT(i, puiBuffers);
-                    alSetError(AL_OUT_OF_MEMORY);
-                    break;
-                }
-
-                puiBuffers[i] = (ALuint)ALTHUNK_ADDENTRY(*list);
-                (*list)->databuffer = puiBuffers[i];
-                (*list)->state = UNMAPPED;
-                device->DatabufferCount++;
-                i++;
-
-                list = &(*list)->next;
-            }
-        }
-        else
-            alSetError(AL_INVALID_VALUE);
-    }
-
-    ProcessContext(Context);
-}
-
-/*
-*    alDatabeleteBuffersEXT(ALsizei n, ALuint *puiBuffers)
-*
-*    Deletes the n AL Databuffers pointed to by puiBuffers
-*/
-ALvoid ALAPIENTRY alDeleteDatabuffersEXT(ALsizei n, const ALuint *puiBuffers)
-{
-    ALCcontext *Context;
-    ALdatabuffer *ALBuf;
-    ALsizei i;
-    ALboolean bFailed = AL_FALSE;
-
-    Context = GetContextSuspended();
-    if(!Context) return;
-
-    /* Check we are actually Deleting some Databuffers */
-    if(n >= 0)
-    {
-        ALCdevice *device = Context->Device;
-
-        /* Check that all the databuffers are valid and can actually be
-         * deleted */
-        for(i = 0;i < n;i++)
-        {
-            /* Check for valid Buffer ID (can be NULL buffer) */
-            if(alIsDatabufferEXT(puiBuffers[i]))
-            {
-                /* If not the NULL buffer, check that it's unmapped */
-                ALBuf = ((ALdatabuffer *)ALTHUNK_LOOKUPENTRY(puiBuffers[i]));
-                if(ALBuf)
-                {
-                    if(ALBuf->state != UNMAPPED)
-                    {
-                        /* Databuffer still in use, cannot be deleted */
-                        alSetError(AL_INVALID_OPERATION);
-                        bFailed = AL_TRUE;
-                    }
-                }
-            }
-            else
-            {
-                /* Invalid Databuffer */
-                alSetError(AL_INVALID_NAME);
-                bFailed = AL_TRUE;
-            }
-        }
-
-        /* If all the Databuffers were valid (and unmapped), then we can
-         * delete them */
-        if(!bFailed)
-        {
-            for(i = 0;i < n;i++)
-            {
-                if(puiBuffers[i] && alIsDatabufferEXT(puiBuffers[i]))
-                {
-                    ALdatabuffer **list = &device->Databuffers;
-
-                    ALBuf = (ALdatabuffer*)ALTHUNK_LOOKUPENTRY(puiBuffers[i]);
-                    while(*list && *list != ALBuf)
-                        list = &(*list)->next;
-
-                    if(*list)
-                        *list = (*list)->next;
-
-                    if(ALBuf == Context->SampleSource)
-                        Context->SampleSource = NULL;
-                    if(ALBuf == Context->SampleSink)
-                        Context->SampleSink = NULL;
-
-                    // Release the memory used to store audio data
-                    free(ALBuf->data);
-
-                    // Release buffer structure
-                    ALTHUNK_REMOVEENTRY(puiBuffers[i]);
-                    memset(ALBuf, 0, sizeof(ALdatabuffer));
-                    device->DatabufferCount--;
-                    free(ALBuf);
-                }
-            }
-        }
-    }
-    else
-        alSetError(AL_INVALID_VALUE);
-
-    ProcessContext(Context);
-
-    return;
-
-}
-
-/*
-*    alIsDatabufferEXT(ALuint uiBuffer)
-*
-*    Checks if ulBuffer is a valid Databuffer Name
-*/
-ALboolean ALAPIENTRY alIsDatabufferEXT(ALuint uiBuffer)
-{
-    ALCcontext *Context;
-    ALdatabuffer *ALBuf;
-
-    Context = GetContextSuspended();
-    if(!Context) return AL_FALSE;
-
-    /* Check through list of generated databuffers for uiBuffer */
-    ALBuf = Context->Device->Databuffers;
-    while(ALBuf && ALBuf->databuffer != uiBuffer)
-        ALBuf = ALBuf->next;
-
-    ProcessContext(Context);
-
-    return ((ALBuf || !uiBuffer) ? AL_TRUE : AL_FALSE);
-}
-
-/*
-*    alDatabufferDataEXT(ALuint buffer,ALvoid *data,ALsizei size,ALenum usage)
-*
-*    Fill databuffer with data
-*/
-ALvoid ALAPIENTRY alDatabufferDataEXT(ALuint buffer,const ALvoid *data,ALsizei size,ALenum usage)
-{
-    ALCcontext *Context;
-    ALdatabuffer *ALBuf;
-    ALvoid *temp;
-
-    Context = GetContextSuspended();
-    if(!Context) return;
-
-    if(alIsDatabufferEXT(buffer) && buffer != 0)
-    {
-        ALBuf = (ALdatabuffer*)ALTHUNK_LOOKUPENTRY(buffer);
-        if(ALBuf->state == UNMAPPED)
-        {
-            if(usage == AL_STREAM_WRITE_EXT || usage == AL_STREAM_READ_EXT ||
-               usage == AL_STREAM_COPY_EXT || usage == AL_STATIC_WRITE_EXT ||
-               usage == AL_STATIC_READ_EXT || usage == AL_STATIC_COPY_EXT ||
-               usage == AL_DYNAMIC_WRITE_EXT || usage == AL_DYNAMIC_READ_EXT ||
-               usage == AL_DYNAMIC_COPY_EXT)
-            {
-                /* (Re)allocate data */
-                temp = realloc(ALBuf->data, size);
-                if(temp)
-                {
-                    ALBuf->data = temp;
-                    ALBuf->size = size;
-                    ALBuf->usage = usage;
-                    if(data)
-                        memcpy(ALBuf->data, data, size);
-                }
-                else
-                    alSetError(AL_OUT_OF_MEMORY);
-            }
-            else
-                alSetError(AL_INVALID_ENUM);
-        }
-        else
-            alSetError(AL_INVALID_OPERATION);
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(Context);
-}
-
-ALvoid ALAPIENTRY alDatabufferSubDataEXT(ALuint uiBuffer, ALuint start, ALsizei length, const ALvoid *data)
-{
-    ALCcontext    *pContext;
-    ALdatabuffer  *pBuffer;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(uiBuffer) && uiBuffer != 0)
-    {
-        pBuffer = (ALdatabuffer*)ALTHUNK_LOOKUPENTRY(uiBuffer);
-
-        if(length >= 0 && start+length <= pBuffer->size)
-        {
-            if(pBuffer->state == UNMAPPED)
-                memcpy(pBuffer->data+start, data, length);
-            else
-                alSetError(AL_INVALID_OPERATION);
-        }
-        else
-            alSetError(AL_INVALID_VALUE);
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-ALvoid ALAPIENTRY alGetDatabufferSubDataEXT(ALuint uiBuffer, ALuint start, ALsizei length, ALvoid *data)
-{
-    ALCcontext    *pContext;
-    ALdatabuffer  *pBuffer;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(uiBuffer) && uiBuffer != 0)
-    {
-        pBuffer = (ALdatabuffer*)ALTHUNK_LOOKUPENTRY(uiBuffer);
-
-        if(length >= 0 && start+length <= pBuffer->size)
-        {
-            if(pBuffer->state == UNMAPPED)
-                memcpy(data, pBuffer->data+start, length);
-            else
-                alSetError(AL_INVALID_OPERATION);
-        }
-        else
-            alSetError(AL_INVALID_VALUE);
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-
-ALvoid ALAPIENTRY alDatabufferfEXT(ALuint buffer, ALenum eParam, ALfloat flValue)
-{
-    ALCcontext    *pContext;
-
-    (void)flValue;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(buffer) && buffer != 0)
-    {
-        switch(eParam)
-        {
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-ALvoid ALAPIENTRY alDatabufferfvEXT(ALuint buffer, ALenum eParam, const ALfloat* flValues)
-{
-    ALCcontext    *pContext;
-
-    (void)flValues;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(buffer) && buffer != 0)
-    {
-        switch(eParam)
-        {
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-
-ALvoid ALAPIENTRY alDatabufferiEXT(ALuint buffer, ALenum eParam, ALint lValue)
-{
-    ALCcontext    *pContext;
-
-    (void)lValue;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(buffer) && buffer != 0)
-    {
-        switch(eParam)
-        {
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-ALvoid ALAPIENTRY alDatabufferivEXT(ALuint buffer, ALenum eParam, const ALint* plValues)
-{
-    ALCcontext    *pContext;
-
-    (void)plValues;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(buffer) && buffer != 0)
-    {
-        switch(eParam)
-        {
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-
-ALvoid ALAPIENTRY alGetDatabufferfEXT(ALuint buffer, ALenum eParam, ALfloat *pflValue)
-{
-    ALCcontext    *pContext;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(pflValue)
-    {
-        if(alIsDatabufferEXT(buffer) && buffer != 0)
-        {
-            switch(eParam)
-            {
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-            }
-        }
-        else
-            alSetError(AL_INVALID_NAME);
-    }
-    else
-        alSetError(AL_INVALID_VALUE);
-
-    ProcessContext(pContext);
-}
-
-ALvoid ALAPIENTRY alGetDatabufferfvEXT(ALuint buffer, ALenum eParam, ALfloat* pflValues)
-{
-    ALCcontext    *pContext;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(pflValues)
-    {
-        if(alIsDatabufferEXT(buffer) && buffer != 0)
-        {
-            switch(eParam)
-            {
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-            }
-        }
-        else
-            alSetError(AL_INVALID_NAME);
-    }
-    else
-        alSetError(AL_INVALID_VALUE);
-
-    ProcessContext(pContext);
-}
-
-ALvoid ALAPIENTRY alGetDatabufferiEXT(ALuint buffer, ALenum eParam, ALint *plValue)
-{
-    ALCcontext    *pContext;
-    ALdatabuffer  *pBuffer;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(plValue)
-    {
-        if(alIsDatabufferEXT(buffer) && buffer != 0)
-        {
-            pBuffer = (ALdatabuffer*)ALTHUNK_LOOKUPENTRY(buffer);
-
-            switch(eParam)
-            {
-            case AL_SIZE:
-                *plValue = pBuffer->size;
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-            }
-        }
-        else
-            alSetError(AL_INVALID_NAME);
-    }
-    else
-        alSetError(AL_INVALID_VALUE);
-
-    ProcessContext(pContext);
-}
-
-ALvoid ALAPIENTRY alGetDatabufferivEXT(ALuint buffer, ALenum eParam, ALint* plValues)
-{
-    ALCcontext    *pContext;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(plValues)
-    {
-        if(alIsDatabufferEXT(buffer) && buffer != 0)
-        {
-            switch (eParam)
-            {
-            case AL_SIZE:
-                alGetBufferi(buffer, eParam, plValues);
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-            }
-        }
-        else
-            alSetError(AL_INVALID_NAME);
-    }
-    else
-        alSetError(AL_INVALID_VALUE);
-
-    ProcessContext(pContext);
-}
-
-
-ALvoid ALAPIENTRY alSelectDatabufferEXT(ALenum target, ALuint uiBuffer)
-{
-    ALCcontext    *pContext;
-    ALdatabuffer  *pBuffer;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(uiBuffer))
-    {
-        pBuffer = (ALdatabuffer*)(uiBuffer ? ALTHUNK_LOOKUPENTRY(uiBuffer) : NULL);
-        if(target == AL_SAMPLE_SOURCE_EXT)
-            pContext->SampleSource = pBuffer;
-        else if(target == AL_SAMPLE_SINK_EXT)
-            pContext->SampleSink = pBuffer;
-        else
-            alSetError(AL_INVALID_VALUE);
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-
-ALvoid* ALAPIENTRY alMapDatabufferEXT(ALuint uiBuffer, ALuint start, ALsizei length, ALenum access)
-{
-    ALCcontext    *pContext;
-    ALdatabuffer  *pBuffer;
-    ALvoid        *ret = NULL;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return NULL;
-
-    if(alIsDatabufferEXT(uiBuffer) && uiBuffer != 0)
-    {
-        pBuffer = (ALdatabuffer*)ALTHUNK_LOOKUPENTRY(uiBuffer);
-
-        if(length >= 0 && start+length <= pBuffer->size)
-        {
-            if(access == AL_READ_ONLY_EXT || access == AL_WRITE_ONLY_EXT ||
-               access == AL_READ_WRITE_EXT)
-            {
-                if(pBuffer->state == UNMAPPED)
-                {
-                    ret = pBuffer->data + start;
-                    pBuffer->state = MAPPED;
-                }
-                else
-                    alSetError(AL_INVALID_OPERATION);
-            }
-            else
-                alSetError(AL_INVALID_ENUM);
-        }
-        else
-            alSetError(AL_INVALID_VALUE);
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-
-    return ret;
-}
-
-ALvoid ALAPIENTRY alUnmapDatabufferEXT(ALuint uiBuffer)
-{
-    ALCcontext    *pContext;
-    ALdatabuffer  *pBuffer;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(alIsDatabufferEXT(uiBuffer) && uiBuffer != 0)
-    {
-        pBuffer = (ALdatabuffer*)ALTHUNK_LOOKUPENTRY(uiBuffer);
-
-        if(pBuffer->state == MAPPED)
-            pBuffer->state = UNMAPPED;
-        else
-            alSetError(AL_INVALID_OPERATION);
-    }
-    else
-        alSetError(AL_INVALID_NAME);
-
-    ProcessContext(pContext);
-}
-
-
-/*
-*    ReleaseALDatabuffers()
-*
-*    INTERNAL FN : Called by DLLMain on exit to destroy any buffers that still exist
-*/
-ALvoid ReleaseALDatabuffers(ALCdevice *device)
-{
-    ALdatabuffer *ALBuffer;
-    ALdatabuffer *ALBufferTemp;
-
-    ALBuffer = device->Databuffers;
-    while(ALBuffer)
-    {
-        // Release sample data
-        free(ALBuffer->data);
-
-        // Release Buffer structure
-        ALBufferTemp = ALBuffer;
-        ALBuffer = ALBuffer->next;
-        memset(ALBufferTemp, 0, sizeof(ALdatabuffer));
-        free(ALBufferTemp);
-    }
-    device->Databuffers = NULL;
-    device->DatabufferCount = 0;
-}

OpenAL32/alError.c

@@ -13,42 +13,70 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
 #include "config.h"
 
+#include <signal.h>
+
+#ifdef HAVE_WINDOWS_H
+#define WIN32_LEAN_AND_MEAN
+#include <windows.h>
+#endif
+
 #include "alMain.h"
 #include "AL/alc.h"
 #include "alError.h"
 
-ALAPI ALenum ALAPIENTRY alGetError(ALvoid)
+ALboolean TrapALError = AL_FALSE;
+
+ALvoid alSetError(ALCcontext *Context, ALenum errorCode)
+{
+    ALenum curerr = AL_NO_ERROR;
+
+    WARN("Error generated on context %p, code 0x%04x\n", Context, errorCode);
+    if(TrapALError)
+    {
+#ifdef _WIN32
+        /* DebugBreak will cause an exception if there is no debugger */
+        if(IsDebuggerPresent())
+            DebugBreak();
+#elif defined(SIGTRAP)
+        raise(SIGTRAP);
+#endif
+    }
+
+    (void)(ATOMIC_COMPARE_EXCHANGE_STRONG_SEQ(&Context->LastError, &curerr, errorCode));
+}
+
+AL_API ALenum AL_APIENTRY alGetError(void)
 {
     ALCcontext *Context;
     ALenum errorCode;
 
-    Context = GetContextSuspended();
-    if(!Context) return AL_INVALID_OPERATION;
+    Context = GetContextRef();
+    if(!Context)
+    {
+        WARN("Querying error state on null context (implicitly 0x%04x)\n",
+             AL_INVALID_OPERATION);
+        if(TrapALError)
+        {
+#ifdef _WIN32
+            if(IsDebuggerPresent())
+                DebugBreak();
+#elif defined(SIGTRAP)
+            raise(SIGTRAP);
+#endif
+        }
+        return AL_INVALID_OPERATION;
+    }
 
-    errorCode = Context->LastError;
-    Context->LastError = AL_NO_ERROR;
+    errorCode = ATOMIC_EXCHANGE_SEQ(&Context->LastError, AL_NO_ERROR);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 
     return errorCode;
 }
-
-ALvoid alSetError(ALenum errorCode)
-{
-    ALCcontext *Context;
-
-    Context = GetContextSuspended();
-    if(Context)
-    {
-        if(Context->LastError == AL_NO_ERROR)
-            Context->LastError = errorCode;
-        ProcessContext(Context);
-    }
-}

OpenAL32/alExtension.c

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -23,346 +23,55 @@
 #include <stdlib.h>
 #include <string.h>
 #include <ctype.h>
-#include "alExtension.h"
+
 #include "alError.h"
 #include "alMain.h"
 #include "alFilter.h"
 #include "alEffect.h"
 #include "alAuxEffectSlot.h"
-#include "alDatabuffer.h"
 #include "alSource.h"
 #include "alBuffer.h"
 #include "AL/al.h"
 #include "AL/alc.h"
 
-static ALfunction  function[]=   {
-    { "alEnable",                   (ALvoid *) alEnable                  },
-    { "alDisable",                  (ALvoid *) alDisable                 },
-    { "alIsEnabled",                (ALvoid *) alIsEnabled               },
-    { "alGetString",                (ALvoid *) alGetString               },
-    { "alGetBooleanv",              (ALvoid *) alGetBooleanv             },
-    { "alGetIntegerv",              (ALvoid *) alGetIntegerv             },
-    { "alGetFloatv",                (ALvoid *) alGetFloatv               },
-    { "alGetDoublev",               (ALvoid *) alGetDoublev              },
-    { "alGetBoolean",               (ALvoid *) alGetBoolean              },
-    { "alGetInteger",               (ALvoid *) alGetInteger              },
-    { "alGetFloat",                 (ALvoid *) alGetFloat                },
-    { "alGetDouble",                (ALvoid *) alGetDouble               },
-    { "alGetError",                 (ALvoid *) alGetError                },
-    { "alIsExtensionPresent",       (ALvoid *) alIsExtensionPresent      },
-    { "alGetProcAddress",           (ALvoid *) alGetProcAddress          },
-    { "alGetEnumValue",             (ALvoid *) alGetEnumValue            },
-    { "alListenerf",                (ALvoid *) alListenerf               },
-    { "alListener3f",               (ALvoid *) alListener3f              },
-    { "alListenerfv",               (ALvoid *) alListenerfv              },
-    { "alListeneri",                (ALvoid *) alListeneri               },
-    { "alListener3i",               (ALvoid *) alListener3i              },
-    { "alListeneriv",               (ALvoid *) alListeneriv              },
-    { "alGetListenerf",             (ALvoid *) alGetListenerf            },
-    { "alGetListener3f",            (ALvoid *) alGetListener3f           },
-    { "alGetListenerfv",            (ALvoid *) alGetListenerfv           },
-    { "alGetListeneri",             (ALvoid *) alGetListeneri            },
-    { "alGetListener3i",            (ALvoid *) alGetListener3i           },
-    { "alGetListeneriv",            (ALvoid *) alGetListeneriv           },
-    { "alGenSources",               (ALvoid *) alGenSources              },
-    { "alDeleteSources",            (ALvoid *) alDeleteSources           },
-    { "alIsSource",                 (ALvoid *) alIsSource                },
-    { "alSourcef",                  (ALvoid *) alSourcef                 },
-    { "alSource3f",                 (ALvoid *) alSource3f                },
-    { "alSourcefv",                 (ALvoid *) alSourcefv                },
-    { "alSourcei",                  (ALvoid *) alSourcei                 },
-    { "alSource3i",                 (ALvoid *) alSource3i                },
-    { "alSourceiv",                 (ALvoid *) alSourceiv                },
-    { "alGetSourcef",               (ALvoid *) alGetSourcef              },
-    { "alGetSource3f",              (ALvoid *) alGetSource3f             },
-    { "alGetSourcefv",              (ALvoid *) alGetSourcefv             },
-    { "alGetSourcei",               (ALvoid *) alGetSourcei              },
-    { "alGetSource3i",              (ALvoid *) alGetSource3i             },
-    { "alGetSourceiv",              (ALvoid *) alGetSourceiv             },
-    { "alSourcePlayv",              (ALvoid *) alSourcePlayv             },
-    { "alSourceStopv",              (ALvoid *) alSourceStopv             },
-    { "alSourceRewindv",            (ALvoid *) alSourceRewindv           },
-    { "alSourcePausev",             (ALvoid *) alSourcePausev            },
-    { "alSourcePlay",               (ALvoid *) alSourcePlay              },
-    { "alSourceStop",               (ALvoid *) alSourceStop              },
-    { "alSourceRewind",             (ALvoid *) alSourceRewind            },
-    { "alSourcePause",              (ALvoid *) alSourcePause             },
-    { "alSourceQueueBuffers",       (ALvoid *) alSourceQueueBuffers      },
-    { "alSourceUnqueueBuffers",     (ALvoid *) alSourceUnqueueBuffers    },
-    { "alGenBuffers",               (ALvoid *) alGenBuffers              },
-    { "alDeleteBuffers",            (ALvoid *) alDeleteBuffers           },
-    { "alIsBuffer",                 (ALvoid *) alIsBuffer                },
-    { "alBufferData",               (ALvoid *) alBufferData              },
-    { "alBufferf",                  (ALvoid *) alBufferf                 },
-    { "alBuffer3f",                 (ALvoid *) alBuffer3f                },
-    { "alBufferfv",                 (ALvoid *) alBufferfv                },
-    { "alBufferi",                  (ALvoid *) alBufferi                 },
-    { "alBuffer3i",                 (ALvoid *) alBuffer3i                },
-    { "alBufferiv",                 (ALvoid *) alBufferiv                },
-    { "alGetBufferf",               (ALvoid *) alGetBufferf              },
-    { "alGetBuffer3f",              (ALvoid *) alGetBuffer3f             },
-    { "alGetBufferfv",              (ALvoid *) alGetBufferfv             },
-    { "alGetBufferi",               (ALvoid *) alGetBufferi              },
-    { "alGetBuffer3i",              (ALvoid *) alGetBuffer3i             },
-    { "alGetBufferiv",              (ALvoid *) alGetBufferiv             },
-    { "alDopplerFactor",            (ALvoid *) alDopplerFactor           },
-    { "alDopplerVelocity",          (ALvoid *) alDopplerVelocity         },
-    { "alSpeedOfSound",             (ALvoid *) alSpeedOfSound            },
-    { "alDistanceModel",            (ALvoid *) alDistanceModel           },
 
-    { "alGenFilters",               (ALvoid *) alGenFilters              },
-    { "alDeleteFilters",            (ALvoid *) alDeleteFilters           },
-    { "alIsFilter",                 (ALvoid *) alIsFilter                },
-    { "alFilteri",                  (ALvoid *) alFilteri                 },
-    { "alFilteriv",                 (ALvoid *) alFilteriv                },
-    { "alFilterf",                  (ALvoid *) alFilterf                 },
-    { "alFilterfv",                 (ALvoid *) alFilterfv                },
-    { "alGetFilteri",               (ALvoid *) alGetFilteri              },
-    { "alGetFilteriv",              (ALvoid *) alGetFilteriv             },
-    { "alGetFilterf",               (ALvoid *) alGetFilterf              },
-    { "alGetFilterfv",              (ALvoid *) alGetFilterfv             },
-
-    { "alGenEffects",               (ALvoid *) alGenEffects              },
-    { "alDeleteEffects",            (ALvoid *) alDeleteEffects           },
-    { "alIsEffect",                 (ALvoid *) alIsEffect                },
-    { "alEffecti",                  (ALvoid *) alEffecti                 },
-    { "alEffectiv",                 (ALvoid *) alEffectiv                },
-    { "alEffectf",                  (ALvoid *) alEffectf                 },
-    { "alEffectfv",                 (ALvoid *) alEffectfv                },
-    { "alGetEffecti",               (ALvoid *) alGetEffecti              },
-    { "alGetEffectiv",              (ALvoid *) alGetEffectiv             },
-    { "alGetEffectf",               (ALvoid *) alGetEffectf              },
-    { "alGetEffectfv",              (ALvoid *) alGetEffectfv             },
-
-    { "alGenAuxiliaryEffectSlots",  (ALvoid *) alGenAuxiliaryEffectSlots },
-    { "alDeleteAuxiliaryEffectSlots",(ALvoid *) alDeleteAuxiliaryEffectSlots},
-    { "alIsAuxiliaryEffectSlot",    (ALvoid *) alIsAuxiliaryEffectSlot   },
-    { "alAuxiliaryEffectSloti",     (ALvoid *) alAuxiliaryEffectSloti    },
-    { "alAuxiliaryEffectSlotiv",    (ALvoid *) alAuxiliaryEffectSlotiv   },
-    { "alAuxiliaryEffectSlotf",     (ALvoid *) alAuxiliaryEffectSlotf    },
-    { "alAuxiliaryEffectSlotfv",    (ALvoid *) alAuxiliaryEffectSlotfv   },
-    { "alGetAuxiliaryEffectSloti",  (ALvoid *) alGetAuxiliaryEffectSloti },
-    { "alGetAuxiliaryEffectSlotiv", (ALvoid *) alGetAuxiliaryEffectSlotiv},
-    { "alGetAuxiliaryEffectSlotf",  (ALvoid *) alGetAuxiliaryEffectSlotf },
-    { "alGetAuxiliaryEffectSlotfv", (ALvoid *) alGetAuxiliaryEffectSlotfv},
-
-    { "alBufferSubDataEXT",         (ALvoid *) alBufferSubDataEXT        },
-
-    { "alGenDatabuffersEXT",        (ALvoid *) alGenDatabuffersEXT       },
-    { "alDeleteDatabuffersEXT",     (ALvoid *) alDeleteDatabuffersEXT    },
-    { "alIsDatabufferEXT",          (ALvoid *) alIsDatabufferEXT         },
-    { "alDatabufferDataEXT",        (ALvoid *) alDatabufferDataEXT       },
-    { "alDatabufferSubDataEXT",     (ALvoid *) alDatabufferSubDataEXT    },
-    { "alGetDatabufferSubDataEXT",  (ALvoid *) alGetDatabufferSubDataEXT },
-    { "alDatabufferfEXT",           (ALvoid *) alDatabufferfEXT          },
-    { "alDatabufferfvEXT",          (ALvoid *) alDatabufferfvEXT         },
-    { "alDatabufferiEXT",           (ALvoid *) alDatabufferiEXT          },
-    { "alDatabufferivEXT",          (ALvoid *) alDatabufferivEXT         },
-    { "alGetDatabufferfEXT",        (ALvoid *) alGetDatabufferfEXT       },
-    { "alGetDatabufferfvEXT",       (ALvoid *) alGetDatabufferfvEXT      },
-    { "alGetDatabufferiEXT",        (ALvoid *) alGetDatabufferiEXT       },
-    { "alGetDatabufferivEXT",       (ALvoid *) alGetDatabufferivEXT      },
-    { "alSelectDatabufferEXT",      (ALvoid *) alSelectDatabufferEXT     },
-    { "alMapDatabufferEXT",         (ALvoid *) alMapDatabufferEXT        },
-    { "alUnmapDatabufferEXT",       (ALvoid *) alUnmapDatabufferEXT      },
-
-    { NULL,                         (ALvoid *) NULL                      } };
-
-static ALenums enumeration[]={
-    // Types
-    { (ALchar *)"AL_INVALID",                           AL_INVALID                          },
-    { (ALchar *)"AL_NONE",                              AL_NONE                             },
-    { (ALchar *)"AL_FALSE",                             AL_FALSE                            },
-    { (ALchar *)"AL_TRUE",                              AL_TRUE                             },
-
-    // Source and Listener Properties
-    { (ALchar *)"AL_SOURCE_RELATIVE",                   AL_SOURCE_RELATIVE                  },
-    { (ALchar *)"AL_CONE_INNER_ANGLE",                  AL_CONE_INNER_ANGLE                 },
-    { (ALchar *)"AL_CONE_OUTER_ANGLE",                  AL_CONE_OUTER_ANGLE                 },
-    { (ALchar *)"AL_PITCH",                             AL_PITCH                            },
-    { (ALchar *)"AL_POSITION",                          AL_POSITION                         },
-    { (ALchar *)"AL_DIRECTION",                         AL_DIRECTION                        },
-    { (ALchar *)"AL_VELOCITY",                          AL_VELOCITY                         },
-    { (ALchar *)"AL_LOOPING",                           AL_LOOPING                          },
-    { (ALchar *)"AL_BUFFER",                            AL_BUFFER                           },
-    { (ALchar *)"AL_GAIN",                              AL_GAIN                             },
-    { (ALchar *)"AL_MIN_GAIN",                          AL_MIN_GAIN                         },
-    { (ALchar *)"AL_MAX_GAIN",                          AL_MAX_GAIN                         },
-    { (ALchar *)"AL_ORIENTATION",                       AL_ORIENTATION                      },
-    { (ALchar *)"AL_REFERENCE_DISTANCE",                AL_REFERENCE_DISTANCE               },
-    { (ALchar *)"AL_ROLLOFF_FACTOR",                    AL_ROLLOFF_FACTOR                   },
-    { (ALchar *)"AL_CONE_OUTER_GAIN",                   AL_CONE_OUTER_GAIN                  },
-    { (ALchar *)"AL_MAX_DISTANCE",                      AL_MAX_DISTANCE                     },
-    { (ALchar *)"AL_SEC_OFFSET",                        AL_SEC_OFFSET                       },
-    { (ALchar *)"AL_SAMPLE_OFFSET",                     AL_SAMPLE_OFFSET                    },
-    { (ALchar *)"AL_BYTE_OFFSET",                       AL_BYTE_OFFSET                      },
-    { (ALchar *)"AL_SOURCE_TYPE",                       AL_SOURCE_TYPE                      },
-    { (ALchar *)"AL_STATIC",                            AL_STATIC                           },
-    { (ALchar *)"AL_STREAMING",                         AL_STREAMING                        },
-    { (ALchar *)"AL_UNDETERMINED",                      AL_UNDETERMINED                     },
-    { (ALchar *)"AL_METERS_PER_UNIT",                   AL_METERS_PER_UNIT                  },
-
-    // Source EFX Properties
-    { (ALchar *)"AL_DIRECT_FILTER",                     AL_DIRECT_FILTER                    },
-    { (ALchar *)"AL_AUXILIARY_SEND_FILTER",             AL_AUXILIARY_SEND_FILTER            },
-    { (ALchar *)"AL_AIR_ABSORPTION_FACTOR",             AL_AIR_ABSORPTION_FACTOR            },
-    { (ALchar *)"AL_ROOM_ROLLOFF_FACTOR",               AL_ROOM_ROLLOFF_FACTOR              },
-    { (ALchar *)"AL_CONE_OUTER_GAINHF",                 AL_CONE_OUTER_GAINHF                },
-    { (ALchar *)"AL_DIRECT_FILTER_GAINHF_AUTO",         AL_DIRECT_FILTER_GAINHF_AUTO        },
-    { (ALchar *)"AL_AUXILIARY_SEND_FILTER_GAIN_AUTO",   AL_AUXILIARY_SEND_FILTER_GAIN_AUTO  },
-    { (ALchar *)"AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO", AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO},
-
-
-    // Source State information
-    { (ALchar *)"AL_SOURCE_STATE",                      AL_SOURCE_STATE                     },
-    { (ALchar *)"AL_INITIAL",                           AL_INITIAL                          },
-    { (ALchar *)"AL_PLAYING",                           AL_PLAYING                          },
-    { (ALchar *)"AL_PAUSED",                            AL_PAUSED                           },
-    { (ALchar *)"AL_STOPPED",                           AL_STOPPED                          },
-
-    // Queue information
-    { (ALchar *)"AL_BUFFERS_QUEUED",                    AL_BUFFERS_QUEUED                   },
-    { (ALchar *)"AL_BUFFERS_PROCESSED",                 AL_BUFFERS_PROCESSED                },
-
-    // Buffer Formats
-    { (ALchar *)"AL_FORMAT_MONO8",                      AL_FORMAT_MONO8                     },
-    { (ALchar *)"AL_FORMAT_MONO16",                     AL_FORMAT_MONO16                    },
-    { (ALchar *)"AL_FORMAT_MONO_FLOAT32",               AL_FORMAT_MONO_FLOAT32              },
-    { (ALchar *)"AL_FORMAT_STEREO8",                    AL_FORMAT_STEREO8                   },
-    { (ALchar *)"AL_FORMAT_STEREO16",                   AL_FORMAT_STEREO16                  },
-    { (ALchar *)"AL_FORMAT_STEREO_FLOAT32",             AL_FORMAT_STEREO_FLOAT32            },
-    { (ALchar *)"AL_FORMAT_MONO_IMA4",                  AL_FORMAT_MONO_IMA4                 },
-    { (ALchar *)"AL_FORMAT_STEREO_IMA4",                AL_FORMAT_STEREO_IMA4               },
-    { (ALchar *)"AL_FORMAT_QUAD8_LOKI",                 AL_FORMAT_QUAD8_LOKI                },
-    { (ALchar *)"AL_FORMAT_QUAD16_LOKI",                AL_FORMAT_QUAD16_LOKI               },
-    { (ALchar *)"AL_FORMAT_QUAD8",                      AL_FORMAT_QUAD8                     },
-    { (ALchar *)"AL_FORMAT_QUAD16",                     AL_FORMAT_QUAD16                    },
-    { (ALchar *)"AL_FORMAT_QUAD32",                     AL_FORMAT_QUAD32                    },
-    { (ALchar *)"AL_FORMAT_51CHN8",                     AL_FORMAT_51CHN8                    },
-    { (ALchar *)"AL_FORMAT_51CHN16",                    AL_FORMAT_51CHN16                   },
-    { (ALchar *)"AL_FORMAT_51CHN32",                    AL_FORMAT_51CHN32                   },
-    { (ALchar *)"AL_FORMAT_61CHN8",                     AL_FORMAT_61CHN8                    },
-    { (ALchar *)"AL_FORMAT_61CHN16",                    AL_FORMAT_61CHN16                   },
-    { (ALchar *)"AL_FORMAT_61CHN32",                    AL_FORMAT_61CHN32                   },
-    { (ALchar *)"AL_FORMAT_71CHN8",                     AL_FORMAT_71CHN8                    },
-    { (ALchar *)"AL_FORMAT_71CHN16",                    AL_FORMAT_71CHN16                   },
-    { (ALchar *)"AL_FORMAT_71CHN32",                    AL_FORMAT_71CHN32                   },
-    { (ALchar *)"AL_FORMAT_REAR8",                      AL_FORMAT_REAR8                     },
-    { (ALchar *)"AL_FORMAT_REAR16",                     AL_FORMAT_REAR16                    },
-    { (ALchar *)"AL_FORMAT_REAR32",                     AL_FORMAT_REAR32                    },
-
-    // Buffer attributes
-    { (ALchar *)"AL_FREQUENCY",                         AL_FREQUENCY                        },
-    { (ALchar *)"AL_BITS",                              AL_BITS                             },
-    { (ALchar *)"AL_CHANNELS",                          AL_CHANNELS                         },
-    { (ALchar *)"AL_SIZE",                              AL_SIZE                             },
-
-    // Buffer States (not supported yet)
-    { (ALchar *)"AL_UNUSED",                            AL_UNUSED                           },
-    { (ALchar *)"AL_PENDING",                           AL_PENDING                          },
-    { (ALchar *)"AL_PROCESSED",                         AL_PROCESSED                        },
-
-    // AL Error Messages
-    { (ALchar *)"AL_NO_ERROR",                          AL_NO_ERROR                         },
-    { (ALchar *)"AL_INVALID_NAME",                      AL_INVALID_NAME                     },
-    { (ALchar *)"AL_INVALID_ENUM",                      AL_INVALID_ENUM                     },
-    { (ALchar *)"AL_INVALID_VALUE",                     AL_INVALID_VALUE                    },
-    { (ALchar *)"AL_INVALID_OPERATION",                 AL_INVALID_OPERATION                },
-    { (ALchar *)"AL_OUT_OF_MEMORY",                     AL_OUT_OF_MEMORY                    },
-
-    // Context strings
-    { (ALchar *)"AL_VENDOR",                            AL_VENDOR                           },
-    { (ALchar *)"AL_VERSION",                           AL_VERSION                          },
-    { (ALchar *)"AL_RENDERER",                          AL_RENDERER                         },
-    { (ALchar *)"AL_EXTENSIONS",                        AL_EXTENSIONS                       },
-
-    // Global states
-    { (ALchar *)"AL_DOPPLER_FACTOR",                    AL_DOPPLER_FACTOR                   },
-    { (ALchar *)"AL_DOPPLER_VELOCITY",                  AL_DOPPLER_VELOCITY                 },
-    { (ALchar *)"AL_DISTANCE_MODEL",                    AL_DISTANCE_MODEL                   },
-    { (ALchar *)"AL_SPEED_OF_SOUND",                    AL_SPEED_OF_SOUND                   },
-
-    // Distance Models
-    { (ALchar *)"AL_INVERSE_DISTANCE",                  AL_INVERSE_DISTANCE                 },
-    { (ALchar *)"AL_INVERSE_DISTANCE_CLAMPED",          AL_INVERSE_DISTANCE_CLAMPED         },
-    { (ALchar *)"AL_LINEAR_DISTANCE",                   AL_LINEAR_DISTANCE                  },
-    { (ALchar *)"AL_LINEAR_DISTANCE_CLAMPED",           AL_LINEAR_DISTANCE_CLAMPED          },
-    { (ALchar *)"AL_EXPONENT_DISTANCE",                 AL_EXPONENT_DISTANCE                },
-    { (ALchar *)"AL_EXPONENT_DISTANCE_CLAMPED",         AL_EXPONENT_DISTANCE_CLAMPED        },
-
-    // Filter types
-    { (ALchar *)"AL_FILTER_TYPE",                       AL_FILTER_TYPE                      },
-    { (ALchar *)"AL_FILTER_NULL",                       AL_FILTER_NULL                      },
-    { (ALchar *)"AL_FILTER_LOWPASS",                    AL_FILTER_LOWPASS                   },
-    { (ALchar *)"AL_FILTER_HIGHPASS",                   AL_FILTER_HIGHPASS                  },
-    { (ALchar *)"AL_FILTER_BANDPASS",                   AL_FILTER_BANDPASS                  },
-
-    // Filter params
-    { (ALchar *)"AL_LOWPASS_GAIN",                      AL_LOWPASS_GAIN                     },
-    { (ALchar *)"AL_LOWPASS_GAINHF",                    AL_LOWPASS_GAINHF                   },
-
-    // Effect types
-    { (ALchar *)"AL_EFFECT_TYPE",                       AL_EFFECT_TYPE                      },
-    { (ALchar *)"AL_EFFECT_NULL",                       AL_EFFECT_NULL                      },
-    { (ALchar *)"AL_EFFECT_REVERB",                     AL_EFFECT_REVERB                    },
-    { (ALchar *)"AL_EFFECT_CHORUS",                     AL_EFFECT_CHORUS                    },
-    { (ALchar *)"AL_EFFECT_DISTORTION",                 AL_EFFECT_DISTORTION                },
-    { (ALchar *)"AL_EFFECT_ECHO",                       AL_EFFECT_ECHO                      },
-    { (ALchar *)"AL_EFFECT_FLANGER",                    AL_EFFECT_FLANGER                   },
-    { (ALchar *)"AL_EFFECT_FREQUENCY_SHIFTER",          AL_EFFECT_FREQUENCY_SHIFTER         },
-    { (ALchar *)"AL_EFFECT_VOCAL_MORPHER",              AL_EFFECT_VOCAL_MORPHER             },
-    { (ALchar *)"AL_EFFECT_PITCH_SHIFTER",              AL_EFFECT_PITCH_SHIFTER             },
-    { (ALchar *)"AL_EFFECT_RING_MODULATOR",             AL_EFFECT_RING_MODULATOR            },
-    { (ALchar *)"AL_EFFECT_AUTOWAH",                    AL_EFFECT_AUTOWAH                   },
-    { (ALchar *)"AL_EFFECT_COMPRESSOR",                 AL_EFFECT_COMPRESSOR                },
-    { (ALchar *)"AL_EFFECT_EQUALIZER",                  AL_EFFECT_EQUALIZER                 },
-
-    // Reverb params
-    { (ALchar *)"AL_REVERB_DENSITY",                    AL_REVERB_DENSITY                   },
-    { (ALchar *)"AL_REVERB_DIFFUSION",                  AL_REVERB_DIFFUSION                 },
-    { (ALchar *)"AL_REVERB_GAIN",                       AL_REVERB_GAIN                      },
-    { (ALchar *)"AL_REVERB_GAINHF",                     AL_REVERB_GAINHF                    },
-    { (ALchar *)"AL_REVERB_DECAY_TIME",                 AL_REVERB_DECAY_TIME                },
-    { (ALchar *)"AL_REVERB_DECAY_HFRATIO",              AL_REVERB_DECAY_HFRATIO             },
-    { (ALchar *)"AL_REVERB_REFLECTIONS_GAIN",           AL_REVERB_REFLECTIONS_GAIN          },
-    { (ALchar *)"AL_REVERB_REFLECTIONS_DELAY",          AL_REVERB_REFLECTIONS_DELAY         },
-    { (ALchar *)"AL_REVERB_LATE_REVERB_GAIN",           AL_REVERB_LATE_REVERB_GAIN          },
-    { (ALchar *)"AL_REVERB_LATE_REVERB_DELAY",          AL_REVERB_LATE_REVERB_DELAY         },
-    { (ALchar *)"AL_REVERB_AIR_ABSORPTION_GAINHF",      AL_REVERB_AIR_ABSORPTION_GAINHF     },
-    { (ALchar *)"AL_REVERB_ROOM_ROLLOFF_FACTOR",        AL_REVERB_ROOM_ROLLOFF_FACTOR       },
-    { (ALchar *)"AL_REVERB_DECAY_HFLIMIT",              AL_REVERB_DECAY_HFLIMIT             },
-
-
-    // Default
-    { (ALchar *)NULL,                                   (ALenum)0                           }
+const struct EffectList EffectList[] = {
+    { "eaxreverb",  AL__EAXREVERB,  "AL_EFFECT_EAXREVERB",      AL_EFFECT_EAXREVERB },
+    { "reverb",     AL__REVERB,     "AL_EFFECT_REVERB",         AL_EFFECT_REVERB },
+    { "chorus",     AL__CHORUS,     "AL_EFFECT_CHORUS",         AL_EFFECT_CHORUS },
+    { "compressor", AL__COMPRESSOR, "AL_EFFECT_COMPRESSOR",     AL_EFFECT_COMPRESSOR },
+    { "distortion", AL__DISTORTION, "AL_EFFECT_DISTORTION",     AL_EFFECT_DISTORTION },
+    { "echo",       AL__ECHO,       "AL_EFFECT_ECHO",           AL_EFFECT_ECHO },
+    { "equalizer",  AL__EQUALIZER,  "AL_EFFECT_EQUALIZER",      AL_EFFECT_EQUALIZER },
+    { "flanger",    AL__FLANGER,    "AL_EFFECT_FLANGER",        AL_EFFECT_FLANGER },
+    { "modulator",  AL__MODULATOR,  "AL_EFFECT_RING_MODULATOR", AL_EFFECT_RING_MODULATOR },
+    { "dedicated",  AL__DEDICATED,  "AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT", AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
+    { "dedicated",  AL__DEDICATED,  "AL_EFFECT_DEDICATED_DIALOGUE", AL_EFFECT_DEDICATED_DIALOGUE },
+    { NULL, 0, NULL, (ALenum)0 }
 };
 
 
-
-ALAPI ALboolean ALAPIENTRY alIsExtensionPresent(const ALchar *extName)
+AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName)
 {
-    ALboolean bIsSupported = AL_FALSE;
-    ALCcontext *pContext;
+    ALboolean ret = AL_FALSE;
+    ALCcontext *context;
     const char *ptr;
     size_t len;
 
-    if (!extName)
-    {
-        alSetError(AL_INVALID_VALUE);
-        return AL_FALSE;
-    }
+    context = GetContextRef();
+    if(!context) return AL_FALSE;
 
-    pContext = GetContextSuspended();
-    if(!pContext) return AL_FALSE;
+    if(!(extName))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
 
     len = strlen(extName);
-    ptr = pContext->ExtensionList;
+    ptr = context->ExtensionList;
     while(ptr && *ptr)
     {
         if(strncasecmp(ptr, extName, len) == 0 &&
            (ptr[len] == '\0' || isspace(ptr[len])))
         {
-            bIsSupported = AL_TRUE;
+            ret = AL_TRUE;
             break;
         }
         if((ptr=strchr(ptr, ' ')) != NULL)
@@ -373,31 +82,22 @@ ALAPI ALboolean ALAPIENTRY alIsExtensionPresent(const ALchar *extName)
         }
     }
 
-    ProcessContext(pContext);
-
-    return bIsSupported;
+done:
+    ALCcontext_DecRef(context);
+    return ret;
 }
 
 
-ALAPI ALvoid * ALAPIENTRY alGetProcAddress(const ALchar *funcName)
+AL_API ALvoid* AL_APIENTRY alGetProcAddress(const ALchar *funcName)
 {
-    ALsizei i = 0;
-
-    while(function[i].funcName &&
-          strcmp((char*)function[i].funcName, (char*)funcName) != 0)
-        i++;
-
-    return function[i].address;
+    if(!funcName)
+        return NULL;
+    return alcGetProcAddress(NULL, funcName);
 }
 
-/* NOTE: This function must be able to run without a context! */
-ALAPI ALenum ALAPIENTRY alGetEnumValue(const ALchar *enumName)
+AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *enumName)
 {
-    ALsizei i = 0;
-
-    while(enumeration[i].enumName &&
-          strcmp(enumeration[i].enumName, enumName) != 0)
-        i++;
-
-    return enumeration[i].value;
+    if(!enumName)
+        return (ALenum)0;
+    return alcGetEnumValue(NULL, enumName);
 }

OpenAL32/alFilter.c

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -22,404 +22,698 @@
 
 #include <stdlib.h>
 
-#include "AL/al.h"
-#include "AL/alc.h"
 #include "alMain.h"
+#include "alu.h"
 #include "alFilter.h"
 #include "alThunk.h"
 #include "alError.h"
 
 
+extern inline void LockFiltersRead(ALCdevice *device);
+extern inline void UnlockFiltersRead(ALCdevice *device);
+extern inline void LockFiltersWrite(ALCdevice *device);
+extern inline void UnlockFiltersWrite(ALCdevice *device);
+extern inline struct ALfilter *LookupFilter(ALCdevice *device, ALuint id);
+extern inline struct ALfilter *RemoveFilter(ALCdevice *device, ALuint id);
+extern inline void ALfilterState_clear(ALfilterState *filter);
+extern inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src);
+extern inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples);
+extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope);
+extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat freq_mult, ALfloat bandwidth);
+
 static void InitFilterParams(ALfilter *filter, ALenum type);
 
 
-ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
+AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
 {
-    ALCcontext *Context;
-    ALsizei i;
+    ALCdevice *device;
+    ALCcontext *context;
+    ALsizei cur = 0;
+    ALenum err;
 
-    Context = GetContextSuspended();
-    if(!Context) return;
+    context = GetContextRef();
+    if(!context) return;
 
-    if (n > 0)
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    device = context->Device;
+    for(cur = 0;cur < n;cur++)
     {
-        ALCdevice *device = Context->Device;
-
-        // Check that enough memory has been allocted in the 'filters' array for n Filters
-        if (!IsBadWritePtr((void*)filters, n * sizeof(ALuint)))
+        ALfilter *filter = al_calloc(16, sizeof(ALfilter));
+        if(!filter)
         {
-            ALfilter **list = &device->FilterList;
-            while(*list)
-                list = &(*list)->next;
-
-            i = 0;
-            while(i < n)
-            {
-                *list = calloc(1, sizeof(ALfilter));
-                if(!(*list))
-                {
-                    // We must have run out or memory
-                    alDeleteFilters(i, filters);
-                    alSetError(AL_OUT_OF_MEMORY);
-                    break;
-                }
-
-                filters[i] = (ALuint)ALTHUNK_ADDENTRY(*list);
-                (*list)->filter = filters[i];
-
-                InitFilterParams(*list, AL_FILTER_NULL);
-                device->FilterCount++;
-                i++;
-
-                list = &(*list)->next;
-            }
+            alDeleteFilters(cur, filters);
+            SET_ERROR_AND_GOTO(context, AL_OUT_OF_MEMORY, done);
         }
+        InitFilterParams(filter, AL_FILTER_NULL);
+
+        err = NewThunkEntry(&filter->id);
+        if(err == AL_NO_ERROR)
+            err = InsertUIntMapEntry(&device->FilterMap, filter->id, filter);
+        if(err != AL_NO_ERROR)
+        {
+            FreeThunkEntry(filter->id);
+            memset(filter, 0, sizeof(ALfilter));
+            al_free(filter);
+
+            alDeleteFilters(cur, filters);
+            SET_ERROR_AND_GOTO(context, err, done);
+        }
+
+        filters[cur] = filter->id;
     }
 
-    ProcessContext(Context);
+done:
+    ALCcontext_DecRef(context);
 }
 
-ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, ALuint *filters)
+AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters)
 {
-    ALCcontext *Context;
-    ALfilter *ALFilter;
+    ALCdevice *device;
+    ALCcontext *context;
+    ALfilter *filter;
     ALsizei i;
 
-    Context = GetContextSuspended();
-    if(!Context) return;
+    context = GetContextRef();
+    if(!context) return;
 
-    if (n >= 0)
+    device = context->Device;
+    LockFiltersWrite(device);
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    for(i = 0;i < n;i++)
     {
-        ALCdevice *device = Context->Device;
-
-        // Check that all filters are valid
-        for (i = 0; i < n; i++)
-        {
-            if (!alIsFilter(filters[i]))
-            {
-                alSetError(AL_INVALID_NAME);
-                break;
-            }
-        }
-
-        if (i == n)
-        {
-            // All filters are valid
-            for (i = 0; i < n; i++)
-            {
-                // Recheck that the filter is valid, because there could be duplicated names
-                if (filters[i] && alIsFilter(filters[i]))
-                {
-                    ALfilter **list;
-
-                    ALFilter = ((ALfilter*)ALTHUNK_LOOKUPENTRY(filters[i]));
-
-                    // Remove Source from list of Sources
-                    list = &device->FilterList;
-                    while(*list && *list != ALFilter)
-                         list = &(*list)->next;
-
-                    if(*list)
-                        *list = (*list)->next;
-                    ALTHUNK_REMOVEENTRY(ALFilter->filter);
-
-                    memset(ALFilter, 0, sizeof(ALfilter));
-                    free(ALFilter);
-
-                    device->FilterCount--;
-                }
-            }
-        }
+        if(filters[i] && LookupFilter(device, filters[i]) == NULL)
+            SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
     }
-    else
-        alSetError(AL_INVALID_VALUE);
+    for(i = 0;i < n;i++)
+    {
+        if((filter=RemoveFilter(device, filters[i])) == NULL)
+            continue;
+        FreeThunkEntry(filter->id);
 
-    ProcessContext(Context);
+        memset(filter, 0, sizeof(*filter));
+        al_free(filter);
+    }
+
+done:
+    UnlockFiltersWrite(device);
+    ALCcontext_DecRef(context);
 }
 
-ALboolean AL_APIENTRY alIsFilter(ALuint filter)
+AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter)
 {
     ALCcontext *Context;
-    ALfilter *list;
+    ALboolean  result;
 
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return AL_FALSE;
 
-    list = Context->Device->FilterList;
-    while(list && list->filter != filter)
-        list = list->next;
+    LockFiltersRead(Context->Device);
+    result = ((!filter || LookupFilter(Context->Device, filter)) ?
+              AL_TRUE : AL_FALSE);
+    UnlockFiltersRead(Context->Device);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 
-    return ((list || !filter) ? AL_TRUE : AL_FALSE);
+    return result;
 }
 
-ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint iValue)
+AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint value)
 {
     ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
 
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return;
 
-    if (filter && alIsFilter(filter))
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
     {
-        ALfilter *ALFilter = (ALfilter*)ALTHUNK_LOOKUPENTRY(filter);
-
-        switch(param)
+        if(param == AL_FILTER_TYPE)
         {
-        case AL_FILTER_TYPE:
-            if(iValue == AL_FILTER_NULL ||
-               iValue == AL_FILTER_LOWPASS)
-                InitFilterParams(ALFilter, iValue);
+            if(value == AL_FILTER_NULL || value == AL_FILTER_LOWPASS ||
+               value == AL_FILTER_HIGHPASS || value == AL_FILTER_BANDPASS)
+                InitFilterParams(ALFilter, value);
             else
-                alSetError(AL_INVALID_VALUE);
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
+                alSetError(Context, AL_INVALID_VALUE);
+        }
+        else
+        {
+            /* Call the appropriate handler */
+            ALfilter_SetParami(ALFilter, Context, param, value);
         }
     }
-    else
-        alSetError(AL_INVALID_NAME);
+    UnlockFiltersWrite(Device);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 }
 
-ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, ALint *piValues)
+AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *values)
 {
     ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
 
-    Context = GetContextSuspended();
-    if(!Context) return;
-
-    if (filter && alIsFilter(filter))
+    switch(param)
     {
-        switch(param)
-        {
         case AL_FILTER_TYPE:
-            alFilteri(filter, param, piValues[0]);
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
+            alFilteri(filter, param, values[0]);
+            return;
     }
-    else
-        alSetError(AL_INVALID_NAME);
 
-    ProcessContext(Context);
-}
-
-ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat flValue)
-{
-    ALCcontext *Context;
-
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return;
 
-    if (filter && alIsFilter(filter))
-    {
-        ALfilter *ALFilter = (ALfilter*)ALTHUNK_LOOKUPENTRY(filter);
-
-        switch(ALFilter->type)
-        {
-        case AL_FILTER_LOWPASS:
-            switch(param)
-            {
-            case AL_LOWPASS_GAIN:
-                if(flValue >= 0.0f && flValue <= 1.0f)
-                    ALFilter->Gain = flValue;
-                else
-                    alSetError(AL_INVALID_VALUE);
-                break;
-
-            case AL_LOWPASS_GAINHF:
-                if(flValue >= 0.0f && flValue <= 1.0f)
-                    ALFilter->GainHF = flValue;
-                else
-                    alSetError(AL_INVALID_VALUE);
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-            }
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
-    }
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
     else
-        alSetError(AL_INVALID_NAME);
+    {
+        /* Call the appropriate handler */
+        ALfilter_SetParamiv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersWrite(Device);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 }
 
-ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, ALfloat *pflValues)
+AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat value)
 {
     ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
 
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return;
 
-    if (filter && alIsFilter(filter))
-    {
-        switch(param)
-        {
-        default:
-            alFilterf(filter, param, pflValues[0]);
-            break;
-        }
-    }
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
     else
-        alSetError(AL_INVALID_NAME);
+    {
+        /* Call the appropriate handler */
+        ALfilter_SetParamf(ALFilter, Context, param, value);
+    }
+    UnlockFiltersWrite(Device);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 }
 
-ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *piValue)
+AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *values)
 {
     ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
 
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return;
 
-    if (filter && alIsFilter(filter))
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
     {
-        ALfilter *ALFilter = (ALfilter*)ALTHUNK_LOOKUPENTRY(filter);
+        /* Call the appropriate handler */
+        ALfilter_SetParamfv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersWrite(Device);
 
-        switch(param)
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *value)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockFiltersRead(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        if(param == AL_FILTER_TYPE)
+            *value = ALFilter->type;
+        else
         {
+            /* Call the appropriate handler */
+            ALfilter_GetParami(ALFilter, Context, param, value);
+        }
+    }
+    UnlockFiltersRead(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *values)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
+
+    switch(param)
+    {
         case AL_FILTER_TYPE:
-            *piValue = ALFilter->type;
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
+            alGetFilteri(filter, param, values);
+            return;
     }
-    else
-        alSetError(AL_INVALID_NAME);
 
-    ProcessContext(Context);
-}
-
-ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *piValues)
-{
-    ALCcontext *Context;
-
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return;
 
-    if (filter && alIsFilter(filter))
-    {
-        switch(param)
-        {
-        case AL_FILTER_TYPE:
-            alGetFilteri(filter, param, piValues);
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
-    }
+    Device = Context->Device;
+    LockFiltersRead(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
     else
-        alSetError(AL_INVALID_NAME);
+    {
+        /* Call the appropriate handler */
+        ALfilter_GetParamiv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersRead(Device);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 }
 
-ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *pflValue)
+AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *value)
 {
     ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
 
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return;
 
-    if (filter && alIsFilter(filter))
-    {
-        ALfilter *ALFilter = (ALfilter*)ALTHUNK_LOOKUPENTRY(filter);
-
-        switch(ALFilter->type)
-        {
-        case AL_FILTER_LOWPASS:
-            switch(param)
-            {
-            case AL_LOWPASS_GAIN:
-                *pflValue = ALFilter->Gain;
-                break;
-
-            case AL_LOWPASS_GAINHF:
-                *pflValue = ALFilter->GainHF;
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-            }
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-        }
-    }
+    Device = Context->Device;
+    LockFiltersRead(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
     else
-        alSetError(AL_INVALID_NAME);
+    {
+        /* Call the appropriate handler */
+        ALfilter_GetParamf(ALFilter, Context, param, value);
+    }
+    UnlockFiltersRead(Device);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 }
 
-ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *pflValues)
+AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *values)
 {
     ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
 
-    Context = GetContextSuspended();
+    Context = GetContextRef();
     if(!Context) return;
 
-    if (filter && alIsFilter(filter))
-    {
-        switch(param)
-        {
-        default:
-            alGetFilterf(filter, param, pflValues);
-            break;
-        }
-    }
+    Device = Context->Device;
+    LockFiltersRead(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
     else
-        alSetError(AL_INVALID_NAME);
+    {
+        /* Call the appropriate handler */
+        ALfilter_GetParamfv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersRead(Device);
 
-    ProcessContext(Context);
+    ALCcontext_DecRef(Context);
 }
 
 
+void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat rcpQ)
+{
+    ALfloat alpha, sqrtgain_alpha_2;
+    ALfloat w0, sin_w0, cos_w0;
+    ALfloat a[3] = { 1.0f, 0.0f, 0.0f };
+    ALfloat b[3] = { 1.0f, 0.0f, 0.0f };
+
+    // Limit gain to -100dB
+    assert(gain > 0.00001f);
+
+    w0 = F_TAU * freq_mult;
+    sin_w0 = sinf(w0);
+    cos_w0 = cosf(w0);
+    alpha = sin_w0/2.0f * rcpQ;
+
+    /* Calculate filter coefficients depending on filter type */
+    switch(type)
+    {
+        case ALfilterType_HighShelf:
+            sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
+            b[0] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
+            b[1] = -2.0f*gain*((gain-1.0f) + (gain+1.0f)*cos_w0                   );
+            b[2] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2);
+            a[0] =             (gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
+            a[1] =  2.0f*     ((gain-1.0f) - (gain+1.0f)*cos_w0                   );
+            a[2] =             (gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
+            break;
+        case ALfilterType_LowShelf:
+            sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
+            b[0] =       gain*((gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
+            b[1] =  2.0f*gain*((gain-1.0f) - (gain+1.0f)*cos_w0                   );
+            b[2] =       gain*((gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2);
+            a[0] =             (gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
+            a[1] = -2.0f*     ((gain-1.0f) + (gain+1.0f)*cos_w0                   );
+            a[2] =             (gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
+            break;
+        case ALfilterType_Peaking:
+            gain = sqrtf(gain);
+            b[0] =  1.0f + alpha * gain;
+            b[1] = -2.0f * cos_w0;
+            b[2] =  1.0f - alpha * gain;
+            a[0] =  1.0f + alpha / gain;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha / gain;
+            break;
+
+        case ALfilterType_LowPass:
+            b[0] = (1.0f - cos_w0) / 2.0f;
+            b[1] =  1.0f - cos_w0;
+            b[2] = (1.0f - cos_w0) / 2.0f;
+            a[0] =  1.0f + alpha;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha;
+            break;
+        case ALfilterType_HighPass:
+            b[0] =  (1.0f + cos_w0) / 2.0f;
+            b[1] = -(1.0f + cos_w0);
+            b[2] =  (1.0f + cos_w0) / 2.0f;
+            a[0] =   1.0f + alpha;
+            a[1] =  -2.0f * cos_w0;
+            a[2] =   1.0f - alpha;
+            break;
+        case ALfilterType_BandPass:
+            b[0] =  alpha;
+            b[1] =  0;
+            b[2] = -alpha;
+            a[0] =  1.0f + alpha;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha;
+            break;
+    }
+
+    filter->a1 = a[1] / a[0];
+    filter->a2 = a[2] / a[0];
+    filter->b0 = b[0] / a[0];
+    filter->b1 = b[1] / a[0];
+    filter->b2 = b[2] / a[0];
+}
+
+
+static void lp_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void lp_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void lp_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
+{
+    switch(param)
+    {
+        case AL_LOWPASS_GAIN:
+            if(!(val >= AL_LOWPASS_MIN_GAIN && val <= AL_LOWPASS_MAX_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            filter->Gain = val;
+            break;
+
+        case AL_LOWPASS_GAINHF:
+            if(!(val >= AL_LOWPASS_MIN_GAINHF && val <= AL_LOWPASS_MAX_GAINHF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            filter->GainHF = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+static void lp_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    lp_SetParamf(filter, context, param, vals[0]);
+}
+
+static void lp_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void lp_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void lp_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    switch(param)
+    {
+        case AL_LOWPASS_GAIN:
+            *val = filter->Gain;
+            break;
+
+        case AL_LOWPASS_GAINHF:
+            *val = filter->GainHF;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+static void lp_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    lp_GetParamf(filter, context, param, vals);
+}
+
+
+static void hp_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void hp_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void hp_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
+{
+    switch(param)
+    {
+        case AL_HIGHPASS_GAIN:
+            if(!(val >= AL_HIGHPASS_MIN_GAIN && val <= AL_HIGHPASS_MAX_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            filter->Gain = val;
+            break;
+
+        case AL_HIGHPASS_GAINLF:
+            if(!(val >= AL_HIGHPASS_MIN_GAINLF && val <= AL_HIGHPASS_MAX_GAINLF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            filter->GainLF = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+static void hp_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    hp_SetParamf(filter, context, param, vals[0]);
+}
+
+static void hp_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void hp_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void hp_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    switch(param)
+    {
+        case AL_HIGHPASS_GAIN:
+            *val = filter->Gain;
+            break;
+
+        case AL_HIGHPASS_GAINLF:
+            *val = filter->GainLF;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+static void hp_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    hp_GetParamf(filter, context, param, vals);
+}
+
+
+static void bp_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void bp_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void bp_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
+{
+    switch(param)
+    {
+        case AL_BANDPASS_GAIN:
+            if(!(val >= AL_BANDPASS_MIN_GAIN && val <= AL_BANDPASS_MAX_GAIN))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            filter->Gain = val;
+            break;
+
+        case AL_BANDPASS_GAINHF:
+            if(!(val >= AL_BANDPASS_MIN_GAINHF && val <= AL_BANDPASS_MAX_GAINHF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            filter->GainHF = val;
+            break;
+
+        case AL_BANDPASS_GAINLF:
+            if(!(val >= AL_BANDPASS_MIN_GAINLF && val <= AL_BANDPASS_MAX_GAINLF))
+                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
+            filter->GainLF = val;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+static void bp_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    bp_SetParamf(filter, context, param, vals[0]);
+}
+
+static void bp_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void bp_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void bp_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    switch(param)
+    {
+        case AL_BANDPASS_GAIN:
+            *val = filter->Gain;
+            break;
+
+        case AL_BANDPASS_GAINHF:
+            *val = filter->GainHF;
+            break;
+
+        case AL_BANDPASS_GAINLF:
+            *val = filter->GainLF;
+            break;
+
+        default:
+            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
+    }
+}
+static void bp_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    bp_GetParamf(filter, context, param, vals);
+}
+
+
+static void null_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void null_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void null_SetParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALfloat UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void null_SetParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALfloat *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+
+static void null_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void null_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void null_GetParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALfloat *UNUSED(val))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+static void null_GetParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALfloat *UNUSED(vals))
+{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
+
+
 ALvoid ReleaseALFilters(ALCdevice *device)
 {
-    ALfilter *list = device->FilterList;
-    while(list)
+    ALsizei i;
+    for(i = 0;i < device->FilterMap.size;i++)
     {
-        ALfilter *temp = list;
-        list = list->next;
+        ALfilter *temp = device->FilterMap.values[i];
+        device->FilterMap.values[i] = NULL;
 
         // Release filter structure
+        FreeThunkEntry(temp->id);
         memset(temp, 0, sizeof(ALfilter));
-        free(temp);
+        al_free(temp);
     }
-    device->FilterList = NULL;
-    device->FilterCount = 0;
 }
 
 
 static void InitFilterParams(ALfilter *filter, ALenum type)
 {
-    filter->type = type;
+    if(type == AL_FILTER_LOWPASS)
+    {
+        filter->Gain = AL_LOWPASS_DEFAULT_GAIN;
+        filter->GainHF = AL_LOWPASS_DEFAULT_GAINHF;
+        filter->HFReference = LOWPASSFREQREF;
+        filter->GainLF = 1.0f;
+        filter->LFReference = HIGHPASSFREQREF;
 
-    filter->Gain = 1.0;
-    filter->GainHF = 1.0;
+        filter->SetParami  = lp_SetParami;
+        filter->SetParamiv = lp_SetParamiv;
+        filter->SetParamf  = lp_SetParamf;
+        filter->SetParamfv = lp_SetParamfv;
+        filter->GetParami  = lp_GetParami;
+        filter->GetParamiv = lp_GetParamiv;
+        filter->GetParamf  = lp_GetParamf;
+        filter->GetParamfv = lp_GetParamfv;
+    }
+    else if(type == AL_FILTER_HIGHPASS)
+    {
+        filter->Gain = AL_HIGHPASS_DEFAULT_GAIN;
+        filter->GainHF = 1.0f;
+        filter->HFReference = LOWPASSFREQREF;
+        filter->GainLF = AL_HIGHPASS_DEFAULT_GAINLF;
+        filter->LFReference = HIGHPASSFREQREF;
+
+        filter->SetParami  = hp_SetParami;
+        filter->SetParamiv = hp_SetParamiv;
+        filter->SetParamf  = hp_SetParamf;
+        filter->SetParamfv = hp_SetParamfv;
+        filter->GetParami  = hp_GetParami;
+        filter->GetParamiv = hp_GetParamiv;
+        filter->GetParamf  = hp_GetParamf;
+        filter->GetParamfv = hp_GetParamfv;
+    }
+    else if(type == AL_FILTER_BANDPASS)
+    {
+        filter->Gain = AL_BANDPASS_DEFAULT_GAIN;
+        filter->GainHF = AL_BANDPASS_DEFAULT_GAINHF;
+        filter->HFReference = LOWPASSFREQREF;
+        filter->GainLF = AL_BANDPASS_DEFAULT_GAINLF;
+        filter->LFReference = HIGHPASSFREQREF;
+
+        filter->SetParami  = bp_SetParami;
+        filter->SetParamiv = bp_SetParamiv;
+        filter->SetParamf  = bp_SetParamf;
+        filter->SetParamfv = bp_SetParamfv;
+        filter->GetParami  = bp_GetParami;
+        filter->GetParamiv = bp_GetParamiv;
+        filter->GetParamf  = bp_GetParamf;
+        filter->GetParamfv = bp_GetParamfv;
+    }
+    else
+    {
+        filter->Gain = 1.0f;
+        filter->GainHF = 1.0f;
+        filter->HFReference = LOWPASSFREQREF;
+        filter->GainLF = 1.0f;
+        filter->LFReference = HIGHPASSFREQREF;
+
+        filter->SetParami  = null_SetParami;
+        filter->SetParamiv = null_SetParamiv;
+        filter->SetParamf  = null_SetParamf;
+        filter->SetParamfv = null_SetParamfv;
+        filter->GetParami  = null_GetParami;
+        filter->GetParamiv = null_GetParamiv;
+        filter->GetParamf  = null_GetParamf;
+        filter->GetParamfv = null_GetParamfv;
+    }
+    filter->type = type;
 }

OpenAL32/alListener.c

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -24,430 +24,487 @@
 #include "AL/alc.h"
 #include "alError.h"
 #include "alListener.h"
+#include "alSource.h"
 
-ALAPI ALvoid ALAPIENTRY alListenerf(ALenum eParam, ALfloat flValue)
+AL_API ALvoid AL_APIENTRY alListenerf(ALenum param, ALfloat value)
 {
-    ALCcontext *pContext;
+    ALCcontext *context;
 
-    pContext = GetContextSuspended();
-    if(!pContext) return;
+    context = GetContextRef();
+    if(!context) return;
 
-    switch(eParam)
+    WriteLock(&context->PropLock);
+    switch(param)
     {
+    case AL_GAIN:
+        if(!(value >= 0.0f && isfinite(value)))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        context->Listener->Gain = value;
+        break;
+
+    case AL_METERS_PER_UNIT:
+        if(!(value >= 0.0f && isfinite(value)))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        context->Listener->MetersPerUnit = value;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+
+done:
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alListener3f(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    switch(param)
+    {
+    case AL_POSITION:
+        if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        context->Listener->Position[0] = value1;
+        context->Listener->Position[1] = value2;
+        context->Listener->Position[2] = value3;
+        break;
+
+    case AL_VELOCITY:
+        if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        context->Listener->Velocity[0] = value1;
+        context->Listener->Velocity[1] = value2;
+        context->Listener->Velocity[2] = value3;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+
+done:
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alListenerfv(ALenum param, const ALfloat *values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        switch(param)
+        {
         case AL_GAIN:
-            if(flValue >= 0.0f)
-                pContext->Listener.Gain = flValue;
-            else
-                alSetError(AL_INVALID_VALUE);
-            break;
-
         case AL_METERS_PER_UNIT:
-            if(flValue > 0.0f)
-                pContext->Listener.MetersPerUnit = flValue;
-            else
-                alSetError(AL_INVALID_VALUE);
-            break;
+            alListenerf(param, values[0]);
+            return;
 
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-    }
-
-    ProcessContext(pContext);
-}
-
-
-ALAPI ALvoid ALAPIENTRY alListener3f(ALenum eParam, ALfloat flValue1, ALfloat flValue2, ALfloat flValue3)
-{
-    ALCcontext *pContext;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    switch(eParam)
-    {
-        case AL_POSITION:
-            pContext->Listener.Position[0] = flValue1;
-            pContext->Listener.Position[1] = flValue2;
-            pContext->Listener.Position[2] = flValue3;
-            break;
-
-        case AL_VELOCITY:
-            pContext->Listener.Velocity[0] = flValue1;
-            pContext->Listener.Velocity[1] = flValue2;
-            pContext->Listener.Velocity[2] = flValue3;
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-    }
-
-    ProcessContext(pContext);
-}
-
-
-ALAPI ALvoid ALAPIENTRY alListenerfv(ALenum eParam, const ALfloat *pflValues)
-{
-    ALCcontext *pContext;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(pflValues)
-    {
-        switch(eParam)
-        {
-            case AL_GAIN:
-                if(pflValues[0] >= 0.0f)
-                    pContext->Listener.Gain = pflValues[0];
-                else
-                    alSetError(AL_INVALID_VALUE);
-                break;
-
-            case AL_METERS_PER_UNIT:
-                if(pflValues[0] > 0.0f)
-                    pContext->Listener.MetersPerUnit = pflValues[0];
-                else
-                    alSetError(AL_INVALID_VALUE);
-                break;
-
-            case AL_POSITION:
-                pContext->Listener.Position[0] = pflValues[0];
-                pContext->Listener.Position[1] = pflValues[1];
-                pContext->Listener.Position[2] = pflValues[2];
-                break;
-
-            case AL_VELOCITY:
-                pContext->Listener.Velocity[0] = pflValues[0];
-                pContext->Listener.Velocity[1] = pflValues[1];
-                pContext->Listener.Velocity[2] = pflValues[2];
-                break;
-
-            case AL_ORIENTATION:
-                // AT then UP
-                pContext->Listener.Forward[0] = pflValues[0];
-                pContext->Listener.Forward[1] = pflValues[1];
-                pContext->Listener.Forward[2] = pflValues[2];
-                pContext->Listener.Up[0] = pflValues[3];
-                pContext->Listener.Up[1] = pflValues[4];
-                pContext->Listener.Up[2] = pflValues[5];
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-        }
-    }
-    else
-        alSetError(AL_INVALID_VALUE);
-
-    ProcessContext(pContext);
-}
-
-
-ALAPI ALvoid ALAPIENTRY alListeneri(ALenum eParam, ALint lValue)
-{
-    ALCcontext *pContext;
-
-    (void)lValue;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    switch(eParam)
-    {
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-    }
-
-    ProcessContext(pContext);
-}
-
-
-ALAPI void ALAPIENTRY alListener3i(ALenum eParam, ALint lValue1, ALint lValue2, ALint lValue3)
-{
-    ALCcontext *pContext;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    switch(eParam)
-    {
         case AL_POSITION:
         case AL_VELOCITY:
-            alListener3f(eParam, (ALfloat)lValue1, (ALfloat)lValue2, (ALfloat)lValue3);
-            break;
-
-        default:
-            alSetError(AL_INVALID_ENUM);
-            break;
-    }
-
-    ProcessContext(pContext);
-}
-
-
-ALAPI void ALAPIENTRY alListeneriv( ALenum eParam, const ALint* plValues )
-{
-    ALCcontext *pContext;
-    ALfloat flValues[6];
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(plValues)
-    {
-        switch(eParam)
-        {
-            case AL_POSITION:
-            case AL_VELOCITY:
-                flValues[0] = (ALfloat)plValues[0];
-                flValues[1] = (ALfloat)plValues[1];
-                flValues[2] = (ALfloat)plValues[2];
-                alListenerfv(eParam, flValues);
-                break;
-
-            case AL_ORIENTATION:
-                flValues[0] = (ALfloat)plValues[0];
-                flValues[1] = (ALfloat)plValues[1];
-                flValues[2] = (ALfloat)plValues[2];
-                flValues[3] = (ALfloat)plValues[3];
-                flValues[4] = (ALfloat)plValues[4];
-                flValues[5] = (ALfloat)plValues[5];
-                alListenerfv(eParam, flValues);
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
+            alListener3f(param, values[0], values[1], values[2]);
+            return;
         }
     }
-    else
-        alSetError(AL_INVALID_VALUE);
 
-    ProcessContext(pContext);
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
+    {
+    case AL_ORIENTATION:
+        if(!(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2]) &&
+             isfinite(values[3]) && isfinite(values[4]) && isfinite(values[5])))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        /* AT then UP */
+        context->Listener->Forward[0] = values[0];
+        context->Listener->Forward[1] = values[1];
+        context->Listener->Forward[2] = values[2];
+        context->Listener->Up[0] = values[3];
+        context->Listener->Up[1] = values[4];
+        context->Listener->Up[2] = values[5];
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+
+done:
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
 }
 
 
-ALAPI ALvoid ALAPIENTRY alGetListenerf(ALenum eParam, ALfloat *pflValue)
+AL_API ALvoid AL_APIENTRY alListeneri(ALenum param, ALint UNUSED(value))
 {
-    ALCcontext *pContext;
+    ALCcontext *context;
 
-    pContext = GetContextSuspended();
-    if(!pContext) return;
+    context = GetContextRef();
+    if(!context) return;
 
-    if(pflValue)
+    WriteLock(&context->PropLock);
+    switch(param)
     {
-        switch(eParam)
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+
+done:
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API void AL_APIENTRY alListener3i(ALenum param, ALint value1, ALint value2, ALint value3)
+{
+    ALCcontext *context;
+
+    switch(param)
+    {
+    case AL_POSITION:
+    case AL_VELOCITY:
+        alListener3f(param, (ALfloat)value1, (ALfloat)value2, (ALfloat)value3);
+        return;
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    switch(param)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+
+done:
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API void AL_APIENTRY alListeneriv(ALenum param, const ALint *values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        ALfloat fvals[6];
+        switch(param)
         {
-            case AL_GAIN:
-                *pflValue = pContext->Listener.Gain;
-                break;
+        case AL_POSITION:
+        case AL_VELOCITY:
+            alListener3f(param, (ALfloat)values[0], (ALfloat)values[1], (ALfloat)values[2]);
+            return;
 
-            case AL_METERS_PER_UNIT:
-                *pflValue = pContext->Listener.MetersPerUnit;
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
+        case AL_ORIENTATION:
+            fvals[0] = (ALfloat)values[0];
+            fvals[1] = (ALfloat)values[1];
+            fvals[2] = (ALfloat)values[2];
+            fvals[3] = (ALfloat)values[3];
+            fvals[4] = (ALfloat)values[4];
+            fvals[5] = (ALfloat)values[5];
+            alListenerfv(param, fvals);
+            return;
         }
     }
-    else
-        alSetError(AL_INVALID_VALUE);
 
-    ProcessContext(pContext);
-}
+    context = GetContextRef();
+    if(!context) return;
 
-
-ALAPI ALvoid ALAPIENTRY alGetListener3f(ALenum eParam, ALfloat *pflValue1, ALfloat *pflValue2, ALfloat *pflValue3)
-{
-    ALCcontext *pContext;
-
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(pflValue1 && pflValue2 && pflValue3)
+    WriteLock(&context->PropLock);
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
     {
-        switch(eParam)
-        {
-            case AL_POSITION:
-                *pflValue1 = pContext->Listener.Position[0];
-                *pflValue2 = pContext->Listener.Position[1];
-                *pflValue3 = pContext->Listener.Position[2];
-                break;
-
-            case AL_VELOCITY:
-                *pflValue1 = pContext->Listener.Velocity[0];
-                *pflValue2 = pContext->Listener.Velocity[1];
-                *pflValue3 = pContext->Listener.Velocity[2];
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-        }
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
     }
-    else
-        alSetError(AL_INVALID_VALUE);
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
 
-    ProcessContext(pContext);
+done:
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
 }
 
 
-ALAPI ALvoid ALAPIENTRY alGetListenerfv(ALenum eParam, ALfloat *pflValues)
+AL_API ALvoid AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value)
 {
-    ALCcontext *pContext;
+    ALCcontext *context;
 
-    pContext = GetContextSuspended();
-    if(!pContext) return;
+    context = GetContextRef();
+    if(!context) return;
 
-    if(pflValues)
+    ReadLock(&context->PropLock);
+    if(!(value))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
     {
-        switch(eParam)
-        {
-            case AL_GAIN:
-                pflValues[0] = pContext->Listener.Gain;
-                break;
+    case AL_GAIN:
+        *value = context->Listener->Gain;
+        break;
 
-            case AL_METERS_PER_UNIT:
-                pflValues[0] = pContext->Listener.MetersPerUnit;
-                break;
+    case AL_METERS_PER_UNIT:
+        *value = context->Listener->MetersPerUnit;
+        break;
 
-            case AL_POSITION:
-                pflValues[0] = pContext->Listener.Position[0];
-                pflValues[1] = pContext->Listener.Position[1];
-                pflValues[2] = pContext->Listener.Position[2];
-                break;
-
-            case AL_VELOCITY:
-                pflValues[0] = pContext->Listener.Velocity[0];
-                pflValues[1] = pContext->Listener.Velocity[1];
-                pflValues[2] = pContext->Listener.Velocity[2];
-                break;
-
-            case AL_ORIENTATION:
-                // AT then UP
-                pflValues[0] = pContext->Listener.Forward[0];
-                pflValues[1] = pContext->Listener.Forward[1];
-                pflValues[2] = pContext->Listener.Forward[2];
-                pflValues[3] = pContext->Listener.Up[0];
-                pflValues[4] = pContext->Listener.Up[1];
-                pflValues[5] = pContext->Listener.Up[2];
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-        }
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
     }
-    else
-        alSetError(AL_INVALID_VALUE);
 
-    ProcessContext(pContext);
+done:
+    ReadUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
 }
 
 
-ALAPI ALvoid ALAPIENTRY alGetListeneri(ALenum eParam, ALint *plValue)
+AL_API ALvoid AL_APIENTRY alGetListener3f(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3)
 {
-    ALCcontext *pContext;
+    ALCcontext *context;
 
-    pContext = GetContextSuspended();
-    if(!pContext) return;
+    context = GetContextRef();
+    if(!context) return;
 
-    if(plValue)
+    ReadLock(&context->PropLock);
+    if(!(value1 && value2 && value3))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
     {
-        switch(eParam)
-        {
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-        }
-    }
-    else
-        alSetError(AL_INVALID_VALUE);
+    case AL_POSITION:
+        *value1 = context->Listener->Position[0];
+        *value2 = context->Listener->Position[1];
+        *value3 = context->Listener->Position[2];
+        break;
 
-    ProcessContext(pContext);
+    case AL_VELOCITY:
+        *value1 = context->Listener->Velocity[0];
+        *value2 = context->Listener->Velocity[1];
+        *value3 = context->Listener->Velocity[2];
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ReadUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
 }
 
 
-ALAPI void ALAPIENTRY alGetListener3i(ALenum eParam, ALint *plValue1, ALint *plValue2, ALint *plValue3)
+AL_API ALvoid AL_APIENTRY alGetListenerfv(ALenum param, ALfloat *values)
 {
-    ALCcontext *pContext;
+    ALCcontext *context;
 
-    pContext = GetContextSuspended();
-    if(!pContext) return;
-
-    if(plValue1 && plValue2 && plValue3)
+    switch(param)
     {
-        switch (eParam)
-        {
-            case AL_POSITION:
-                *plValue1 = (ALint)pContext->Listener.Position[0];
-                *plValue2 = (ALint)pContext->Listener.Position[1];
-                *plValue3 = (ALint)pContext->Listener.Position[2];
-                break;
+    case AL_GAIN:
+    case AL_METERS_PER_UNIT:
+        alGetListenerf(param, values);
+        return;
 
-            case AL_VELOCITY:
-                *plValue1 = (ALint)pContext->Listener.Velocity[0];
-                *plValue2 = (ALint)pContext->Listener.Velocity[1];
-                *plValue3 = (ALint)pContext->Listener.Velocity[2];
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-        }
+    case AL_POSITION:
+    case AL_VELOCITY:
+        alGetListener3f(param, values+0, values+1, values+2);
+        return;
     }
-    else
-        alSetError(AL_INVALID_VALUE);
 
-    ProcessContext(pContext);
+    context = GetContextRef();
+    if(!context) return;
+
+    ReadLock(&context->PropLock);
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
+    {
+    case AL_ORIENTATION:
+        // AT then UP
+        values[0] = context->Listener->Forward[0];
+        values[1] = context->Listener->Forward[1];
+        values[2] = context->Listener->Forward[2];
+        values[3] = context->Listener->Up[0];
+        values[4] = context->Listener->Up[1];
+        values[5] = context->Listener->Up[2];
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ReadUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
 }
 
 
-ALAPI void ALAPIENTRY alGetListeneriv(ALenum eParam, ALint* plValues)
+AL_API ALvoid AL_APIENTRY alGetListeneri(ALenum param, ALint *value)
 {
-    ALCcontext *pContext;
+    ALCcontext *context;
 
-    pContext = GetContextSuspended();
-    if(!pContext) return;
+    context = GetContextRef();
+    if(!context) return;
 
-    if(plValues)
+    ReadLock(&context->PropLock);
+    if(!(value))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
     {
-        switch(eParam)
-        {
-            case AL_POSITION:
-                plValues[0] = (ALint)pContext->Listener.Position[0];
-                plValues[1] = (ALint)pContext->Listener.Position[1];
-                plValues[2] = (ALint)pContext->Listener.Position[2];
-                break;
-
-            case AL_VELOCITY:
-                plValues[0] = (ALint)pContext->Listener.Velocity[0];
-                plValues[1] = (ALint)pContext->Listener.Velocity[1];
-                plValues[2] = (ALint)pContext->Listener.Velocity[2];
-                break;
-
-            case AL_ORIENTATION:
-                // AT then UP
-                plValues[0] = (ALint)pContext->Listener.Forward[0];
-                plValues[1] = (ALint)pContext->Listener.Forward[1];
-                plValues[2] = (ALint)pContext->Listener.Forward[2];
-                plValues[3] = (ALint)pContext->Listener.Up[0];
-                plValues[4] = (ALint)pContext->Listener.Up[1];
-                plValues[5] = (ALint)pContext->Listener.Up[2];
-                break;
-
-            default:
-                alSetError(AL_INVALID_ENUM);
-                break;
-        }
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
     }
-    else
-        alSetError(AL_INVALID_VALUE);
 
-    ProcessContext(pContext);
+done:
+    ReadUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API void AL_APIENTRY alGetListener3i(ALenum param, ALint *value1, ALint *value2, ALint *value3)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    ReadLock(&context->PropLock);
+    if(!(value1 && value2 && value3))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch (param)
+    {
+    case AL_POSITION:
+        *value1 = (ALint)context->Listener->Position[0];
+        *value2 = (ALint)context->Listener->Position[1];
+        *value3 = (ALint)context->Listener->Position[2];
+        break;
+
+    case AL_VELOCITY:
+        *value1 = (ALint)context->Listener->Velocity[0];
+        *value2 = (ALint)context->Listener->Velocity[1];
+        *value3 = (ALint)context->Listener->Velocity[2];
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ReadUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API void AL_APIENTRY alGetListeneriv(ALenum param, ALint* values)
+{
+    ALCcontext *context;
+
+    switch(param)
+    {
+    case AL_POSITION:
+    case AL_VELOCITY:
+        alGetListener3i(param, values+0, values+1, values+2);
+        return;
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    ReadLock(&context->PropLock);
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
+    {
+    case AL_ORIENTATION:
+        // AT then UP
+        values[0] = (ALint)context->Listener->Forward[0];
+        values[1] = (ALint)context->Listener->Forward[1];
+        values[2] = (ALint)context->Listener->Forward[2];
+        values[3] = (ALint)context->Listener->Up[0];
+        values[4] = (ALint)context->Listener->Up[1];
+        values[5] = (ALint)context->Listener->Up[2];
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ReadUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+void UpdateListenerProps(ALCcontext *context)
+{
+    ALlistener *listener = context->Listener;
+    struct ALlistenerProps *props;
+
+    /* Get an unused proprty container, or allocate a new one as needed. */
+    props = ATOMIC_LOAD(&listener->FreeList, almemory_order_acquire);
+    if(!props)
+        props = al_calloc(16, sizeof(*props));
+    else
+    {
+        struct ALlistenerProps *next;
+        do {
+            next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
+        } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&listener->FreeList, &props, next,
+                almemory_order_seq_cst, almemory_order_acquire) == 0);
+    }
+
+    /* Copy in current property values. */
+    props->Position[0] = listener->Position[0];
+    props->Position[1] = listener->Position[1];
+    props->Position[2] = listener->Position[2];
+
+    props->Velocity[0] = listener->Velocity[0];
+    props->Velocity[1] = listener->Velocity[1];
+    props->Velocity[2] = listener->Velocity[2];
+
+    props->Forward[0] = listener->Forward[0];
+    props->Forward[1] = listener->Forward[1];
+    props->Forward[2] = listener->Forward[2];
+    props->Up[0] = listener->Up[0];
+    props->Up[1] = listener->Up[1];
+    props->Up[2] = listener->Up[2];
+
+    props->Gain = listener->Gain;
+    props->MetersPerUnit = listener->MetersPerUnit;
+
+    props->DopplerFactor = context->DopplerFactor;
+    props->DopplerVelocity = context->DopplerVelocity;
+    props->SpeedOfSound = context->SpeedOfSound;
+
+    props->SourceDistanceModel = context->SourceDistanceModel;
+    props->DistanceModel = context->DistanceModel;;
+
+    /* Set the new container for updating internal parameters. */
+    props = ATOMIC_EXCHANGE_PTR(&listener->Update, props, almemory_order_acq_rel);
+    if(props)
+    {
+        /* If there was an unused update container, put it back in the
+         * freelist.
+         */
+        ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &listener->FreeList, props);
+    }
 }