Release 1.18.2

Apparently there is a bug with at least MinGW-W64 where fegetenv and fesetenv
do not properly save and restore the FPU rounding mode, resulting in the
rounding mode remaining as round-to-zero after certain function calls. I do not
know if this also affects MSVC, but better safe than sorry for now.

.gitignore

@@ -1,9 +1,7 @@
-build*/
+build
 winbuild
 win64build
-
-## kdevelop
-*.kdev4
-
-## qt-creator
-CMakeLists.txt.user*
+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

@@ -0,0 +1,652 @@
+/**
+ * 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
+ */
+
+#ifdef _WIN32
+#ifdef __MINGW32__
+#define _WIN32_IE 0x501
+#else
+#define _WIN32_IE 0x400
+#endif
+#endif
+
+#include "config.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <ctype.h>
+#include <string.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 {
+    ConfigEntry *entries;
+    unsigned int entryCount;
+} ConfigBlock;
+static ConfigBlock cfgBlock;
+
+
+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)
+{
+    char curSection[128] = "";
+    char *buffer = NULL;
+    size_t maxlen = 0;
+    ConfigEntry *ent;
+
+    while(readline(f, &buffer, &maxlen))
+    {
+        char *line, *comment;
+        char key[256] = "";
+        char value[256] = "";
+
+        line = rstrip(lstrip(buffer));
+        if(!line[0]) continue;
+
+        if(line[0] == '[')
+        {
+            char *section = line+1;
+            char *endsection;
+
+            endsection = strchr(section, ']');
+            if(!endsection || section == endsection)
+            {
+                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);
+        }
+
+        /* Check if we already have this option set */
+        ent = cfgBlock.entries;
+        while((unsigned int)(ent-cfgBlock.entries) < cfgBlock.entryCount)
+        {
+            if(strcasecmp(ent->key, key) == 0)
+                break;
+            ent++;
+        }
+
+        if((unsigned int)(ent-cfgBlock.entries) >= cfgBlock.entryCount)
+        {
+            /* Allocate a new option entry */
+            ent = realloc(cfgBlock.entries, (cfgBlock.entryCount+1)*sizeof(ConfigEntry));
+            if(!ent)
+            {
+                 ERR("config parse error: error reallocating config entries\n");
+                 continue;
+            }
+            cfgBlock.entries = ent;
+            ent = cfgBlock.entries + cfgBlock.entryCount;
+            cfgBlock.entryCount++;
+
+            ent->key = strdup(key);
+            ent->value = NULL;
+        }
+
+        free(ent->value);
+        ent->value = expdup(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;
+
+    if(SHGetSpecialFolderPathW(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE)
+    {
+        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
+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(!(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)
+    {
+        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);
+        }
+    }
+
+    if((str=getenv("XDG_CONFIG_HOME")) != NULL && str[0] != 0)
+        snprintf(buffer, sizeof(buffer), "%s/%s", str, "alsoft.conf");
+    else
+    {
+        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)
+{
+    unsigned int i;
+
+    for(i = 0;i < cfgBlock.entryCount;i++)
+    {
+        free(cfgBlock.entries[i].key);
+        free(cfgBlock.entries[i].value);
+    }
+    free(cfgBlock.entries);
+}
+
+const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def)
+{
+    unsigned int i;
+    char key[256];
+
+    if(!keyName)
+        return def;
+
+    if(blockName && strcasecmp(blockName, "general") != 0)
+    {
+        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
+        {
+            strncpy(key, keyName, sizeof(key)-1);
+            key[sizeof(key)-1] = 0;
+        }
+    }
+
+    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 ConfigValueExists(const char *devName, const char *blockName, const char *keyName)
+{
+    const char *val = GetConfigValue(devName, blockName, keyName, "");
+    return !!val[0];
+}
+
+int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret)
+{
+    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;
+
+#ifdef HAVE_STRTOF
+    *ret = strtof(val, NULL);
+#else
+    *ret = (float)strtod(val, NULL);
+#endif
+    return 1;
+}
+
+int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret)
+{
+    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 ||
+            strcasecmp(val, "on") == 0 || atoi(val) != 0);
+}

Alc/alcRing.c

@@ -0,0 +1,317 @@
+/**
+ * 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 <string.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "threads.h"
+#include "almalloc.h"
+#include "compat.h"
+
+
+/* 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;
+
+    alignas(16) char buf[];
+};
+
+/* 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)
+{
+    ll_ringbuffer_t *rb;
+    ALuint power_of_two;
+
+    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;
+}
+
+/* Free all data associated with the ringbuffer `rb'. */
+void ll_ringbuffer_free(ll_ringbuffer_t *rb)
+{
+    if(rb)
+    {
+#ifdef USE_MLOCK
+        if(rb->mlocked)
+            munlock(rb, sizeof(*rb) + rb->size*rb->elem_size);
+#endif /* USE_MLOCK */
+        al_free(rb);
+    }
+}
+
+/* Lock the data block of `rb' using the system call 'mlock'. */
+int ll_ringbuffer_mlock(ll_ringbuffer_t *rb)
+{
+#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;
+}
+
+/* Reset the read and write pointers to zero. This is not thread safe. */
+void ll_ringbuffer_reset(ll_ringbuffer_t *rb)
+{
+    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);
+}
+
+/* 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;
+}
+
+/* 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;
+
+    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)
+    {
+        n1 = rb->size - read_ptr;
+        n2 = cnt2 & rb->size_mask;
+    }
+    else
+    {
+        n1 = to_read;
+        n2 = 0;
+    }
+
+    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;
+}
+
+/* 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)
+{
+    size_t free_cnt;
+    size_t cnt2;
+    size_t to_read;
+    size_t n1, n2;
+    size_t read_ptr;
+
+    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)
+    {
+        n1 = rb->size - read_ptr;
+        n2 = cnt2 & rb->size_mask;
+    }
+    else
+    {
+        n1 = to_read;
+        n2 = 0;
+    }
+
+    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/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 +1,81 @@
+
+#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))
+{
+}

Alc/backends/base.h

@@ -0,0 +1,155 @@
+#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 *ALCqsaBackendFactory_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);
+
+#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 +1,640 @@
+/**
+ * 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);
+static __typeof(jack_error_callback) * pjack_error_callback;
+#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
+#define jack_error_callback (*pjack_error_callback)
+#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
+        /* Optional symbols. These don't exist in all versions of JACK. */
+#define LOAD_SYM(f) p##f = GetSymbol(jack_handle, #f)
+        LOAD_SYM(jack_error_callback);
+#undef LOAD_SYM
+
+        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))
+{
+    void (*old_error_cb)(const char*);
+    jack_client_t *client;
+    jack_status_t status;
+
+    if(!jack_load())
+        return ALC_FALSE;
+
+    if(!GetConfigValueBool(NULL, "jack", "spawn-server", 0))
+        ClientOptions |= JackNoStartServer;
+
+    old_error_cb = (&jack_error_callback ? jack_error_callback : NULL);
+    jack_set_error_function(jack_msg_handler);
+    client = jack_client_open("alsoft", ClientOptions, &status, NULL);
+    jack_set_error_function(old_error_cb);
+    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 +1,868 @@
+/**
+ * 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)
+    {
+        TRACE("Could not open /dev/mixer: %s\n", strerror(errno));
+        return;
+    }
+    if(ioctl(fd, SNDCTL_SYSINFO, &si) == -1)
+    {
+        TRACE("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)
+            {
+#ifdef HAVE_STAT
+                struct stat buf;
+                if(stat(cur->path, &buf) == 0)
+#endif
+                    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)
+            {
+#ifdef HAVE_STAT
+                struct stat buf;
+                if(stat(cur->path, &buf) == 0)
+#endif
+                    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/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 +1,453 @@
+/**
+ * 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((device->FmtType == DevFmtFloat) ?
+                 (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 +1,612 @@
+
+#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;
+}
+
+
+void splitterap_init(SplitterAllpass *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->z1 = 0.0f;
+}
+
+void splitterap_clear(SplitterAllpass *splitter)
+{
+    splitter->z1 = 0.0f;
+}
+
+void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count)
+{
+    ALfloat coeff, d, x;
+    ALfloat z1;
+    ALsizei i;
+
+    coeff = splitter->coeff;
+    z1 = splitter->z1;
+    for(i = 0;i < count;i++)
+    {
+        x = samples[i];
+
+        d = x - coeff*z1;
+        x = z1 + coeff*d;
+        z1 = d;
+
+        samples[i] = x;
+    }
+    splitter->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 +1,75 @@
+#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);
+
+/* The all-pass portion of the band splitter. Applies the same phase shift
+ * without splitting the signal.
+ */
+typedef struct SplitterAllpass {
+    ALfloat coeff;
+    ALfloat z1;
+} SplitterAllpass;
+
+void splitterap_init(SplitterAllpass *splitter, ALfloat freq_mult);
+void splitterap_clear(SplitterAllpass *splitter);
+void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count);
+
+#endif /* BFORMATDEC_H */

Alc/bs2b.c

@@ -0,0 +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 <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 +1,466 @@
+
+#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;
+    ALsizei step;
+
+    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;
+
+    /* Have to set the mixer FPU mode since that's what the resampler code expects. */
+    START_MIXER_MODE();
+    step = fastf2i(minf((ALdouble)srcRate / dstRate, MAX_PITCH)*FRACTIONONE + 0.5f);
+    converter->mIncrement = maxi(step, 1);
+    if(converter->mIncrement == FRACTIONONE)
+        converter->mResample = Resample_copy32_C;
+    else
+    {
+        /* TODO: Allow other resamplers. */
+        BsincPrepare(converter->mIncrement, &converter->mState.bsinc);
+        converter->mResample = SelectResampler(BSincResampler);
+    }
+    END_MIXER_MODE();
+
+    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 Sample_ALint(val - INT_MAX - 1); }
+
+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 fastf2i(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 fastf2i(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 fastf2i(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(srcframes < 1)
+    {
+        /* No output samples if there's no input samples. */
+        return 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)/increment, 1, BUFFERSIZE);
+}
+
+
+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;
+
+    START_MIXER_MODE();
+    while(pos < dstframes && *srcframes > 0)
+    {
+        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;
+            converter->mSrcPrepCount = 0;
+            continue;
+        }
+        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)/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->mDstTypeSize*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(&converter->mState,
+                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;
+    }
+    END_MIXER_MODE();
+
+    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 +1,55 @@
+#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;
+    InterpState mState;
+    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/effects/chorus.c

@@ -0,0 +1,410 @@
+/**
+ * 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 = fastf2i(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 void GetTriangleDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
+                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
+                              const ALsizei todo)
+{
+    ALsizei i;
+    for(i = 0;i < todo;i++)
+    {
+        delays[i] = fastf2i((1.0f - fabsf(2.0f - lfo_scale*offset)) * depth) + delay;
+        offset = (offset+1)%lfo_range;
+    }
+}
+
+static void GetSinusoidDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
+                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
+                              const ALsizei todo)
+{
+    ALsizei i;
+    for(i = 0;i < todo;i++)
+    {
+        delays[i] = fastf2i(sinf(lfo_scale*offset) * depth) + delay;
+        offset = (offset+1)%lfo_range;
+    }
+}
+
+
+static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALfloat *restrict leftbuf = state->SampleBuffer[0];
+    ALfloat *restrict rightbuf = state->SampleBuffer[1];
+    const ALsizei bufmask = state->BufferLength-1;
+    const ALfloat feedback = state->feedback;
+    ALsizei offset = state->offset;
+    ALsizei i, c;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        const ALsizei todo = mini(128, SamplesToDo-base);
+        ALfloat temps[128][2];
+        ALint moddelays[2][128];
+
+        switch(state->waveform)
+        {
+            case CWF_Triangle:
+                GetTriangleDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
+                                  state->lfo_scale, state->depth, state->delay, todo);
+                GetTriangleDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
+                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
+                                  todo);
+                break;
+            case CWF_Sinusoid:
+                GetSinusoidDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
+                                  state->lfo_scale, state->depth, state->delay, todo);
+                GetSinusoidDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
+                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
+                                  todo);
+                break;
+        }
+
+        for(i = 0;i < todo;i++)
+        {
+            leftbuf[offset&bufmask] = SamplesIn[0][base+i];
+            temps[i][0] = leftbuf[(offset-moddelays[0][i])&bufmask] * feedback;
+            leftbuf[offset&bufmask] += temps[i][0];
+
+            rightbuf[offset&bufmask] = SamplesIn[0][base+i];
+            temps[i][1] = rightbuf[(offset-moddelays[1][i])&bufmask] * feedback;
+            rightbuf[offset&bufmask] += temps[i][1];
+
+            offset++;
+        }
+
+        for(c = 0;c < NumChannels;c++)
+        {
+            ALfloat gain = state->Gain[0][c];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(i = 0;i < todo;i++)
+                    SamplesOut[c][i+base] += temps[i][0] * gain;
+            }
+
+            gain = state->Gain[1][c];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(i = 0;i < todo;i++)
+                    SamplesOut[c][i+base] += temps[i][1] * gain;
+            }
+        }
+
+        base += todo;
+    }
+
+    state->offset = offset;
+}
+
+
+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 +1,408 @@
+/**
+ * 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 void GetTriangleDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
+                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
+                              const ALsizei todo)
+{
+    ALsizei i;
+    for(i = 0;i < todo;i++)
+    {
+        delays[i] = fastf2i((1.0f - fabsf(2.0f - lfo_scale*offset)) * depth) + delay;
+        offset = (offset+1)%lfo_range;
+    }
+}
+
+static void GetSinusoidDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
+                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
+                              const ALsizei todo)
+{
+    ALsizei i;
+    for(i = 0;i < todo;i++)
+    {
+        delays[i] = fastf2i(sinf(lfo_scale*offset) * depth) + delay;
+        offset = (offset+1)%lfo_range;
+    }
+}
+
+static ALvoid ALflangerState_process(ALflangerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALfloat *restrict leftbuf = state->SampleBuffer[0];
+    ALfloat *restrict rightbuf = state->SampleBuffer[1];
+    const ALsizei bufmask = state->BufferLength-1;
+    const ALfloat feedback = state->feedback;
+    ALsizei offset = state->offset;
+    ALsizei i, c;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        const ALsizei todo = mini(128, SamplesToDo-base);
+        ALfloat temps[128][2];
+        ALint moddelays[2][128];
+
+        switch(state->waveform)
+        {
+            case FWF_Triangle:
+                GetTriangleDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
+                                  state->lfo_scale, state->depth, state->delay, todo);
+                GetTriangleDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
+                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
+                                  todo);
+                break;
+            case FWF_Sinusoid:
+                GetSinusoidDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
+                                  state->lfo_scale, state->depth, state->delay, todo);
+                GetSinusoidDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
+                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
+                                  todo);
+                break;
+        }
+
+        for(i = 0;i < todo;i++)
+        {
+            leftbuf[offset&bufmask] = SamplesIn[0][base+i];
+            temps[i][0] = leftbuf[(offset-moddelays[0][i])&bufmask] * feedback;
+            leftbuf[offset&bufmask] += temps[i][0];
+
+            rightbuf[offset&bufmask] = SamplesIn[0][base+i];
+            temps[i][1] = rightbuf[(offset-moddelays[1][i])&bufmask] * feedback;
+            rightbuf[offset&bufmask] += temps[i][1];
+
+            offset++;
+        }
+
+        for(c = 0;c < NumChannels;c++)
+        {
+            ALfloat gain = state->Gain[0][c];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(i = 0;i < todo;i++)
+                    SamplesOut[c][i+base] += temps[i][0] * gain;
+            }
+
+            gain = state->Gain[1][c];
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
+            {
+                for(i = 0;i < todo;i++)
+                    SamplesOut[c][i+base] += temps[i][1] * gain;
+            }
+        }
+
+        base += todo;
+    }
+
+    state->offset = offset;
+}
+
+
+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 = fastf2i(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 +1,52 @@
+#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"
+
+
+/* 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] = fastf2i(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 +1,671 @@
+/**
+ * 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
+                    {
+                        ALfloat *nfcsamples = Device->NFCtrlData;
+                        ALsizei 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];
+#define APPLY_NFC_MIX(order)                                                  \
+    if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
+    {                                                                         \
+        NfcFilterUpdate##order(&parms->NFCtrlFilter[order-1], nfcsamples,     \
+                               samples, DstBufferSize);                       \
+        MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
+            voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
+            parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
+        );                                                                    \
+        chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
+    }
+                        APPLY_NFC_MIX(1)
+                        APPLY_NFC_MIX(2)
+                        APPLY_NFC_MIX(3)
+#undef APPLY_NFC_MIX
+                    }
+                }
+                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/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 */

COPYING

@@ -0,0 +1,481 @@
+                  GNU LIBRARY GENERAL PUBLIC LICENSE
+                       Version 2, June 1991
+
+ Copyright (C) 1991 Free Software Foundation, Inc.
+ 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
+
+  The licenses for most software are designed to take away your
+freedom to share and change it.  By contrast, the GNU General Public
+Licenses are intended to guarantee your freedom to share and change
+free software--to make sure the software is free for all its users.
+
+  This license, the Library General Public License, applies to some
+specially designated Free Software Foundation software, and to any
+other libraries whose authors decide to use it.  You can use it for
+your libraries, too.
+
+  When we speak of free software, we are referring to freedom, not
+price.  Our General Public Licenses are designed to make sure that you
+have the freedom to distribute copies of free software (and charge for
+this service if you wish), that you receive source code or can get it
+if you want it, that you can change the software or use pieces of it
+in new free programs; and that you know you can do these things.
+
+  To protect your rights, we need to make restrictions that forbid
+anyone to deny you these rights or to ask you to surrender the rights.
+These restrictions translate to certain responsibilities for you if
+you distribute copies of the library, or if you modify it.
+
+  For example, if you distribute copies of the library, whether gratis
+or for a fee, you must give the recipients all the rights that we gave
+you.  You must make sure that they, too, receive or can get the source
+code.  If you link a program with the library, you must provide
+complete object files to the recipients so that they can relink them
+with the library, after making changes to the library and recompiling
+it.  And you must show them these terms so they know their rights.
+
+  Our method of protecting your rights has two steps: (1) copyright
+the library, and (2) offer you this license which gives you legal
+permission to copy, distribute and/or modify the library.
+
+  Also, for each distributor's protection, we want to make certain
+that everyone understands that there is no warranty for this free
+library.  If the library is modified by someone else and passed on, we
+want its recipients to know that what they have is not the original
+version, so that any problems introduced by others will not reflect on
+the original authors' reputations.
+
+  Finally, any free program is threatened constantly by software
+patents.  We wish to avoid the danger that companies distributing free
+software will individually obtain patent licenses, thus in effect
+transforming the program into proprietary software.  To prevent this,
+we have made it clear that any patent must be licensed for everyone's
+free use or not licensed at all.
+
+  Most GNU software, including some libraries, is covered by the ordinary
+GNU General Public License, which was designed for utility programs.  This
+license, the GNU Library General Public License, applies to certain
+designated libraries.  This license is quite different from the ordinary
+one; be sure to read it in full, and don't assume that anything in it is
+the same as in the ordinary license.
+
+  The reason we have a separate public license for some libraries is that
+they blur the distinction we usually make between modifying or adding to a
+program and simply using it.  Linking a program with a library, without
+changing the library, is in some sense simply using the library, and is
+analogous to running a utility program or application program.  However, in
+a textual and legal sense, the linked executable is a combined work, a
+derivative of the original library, and the ordinary General Public License
+treats it as such.
+
+  Because of this blurred distinction, using the ordinary General
+Public License for libraries did not effectively promote software
+sharing, because most developers did not use the libraries.  We
+concluded that weaker conditions might promote sharing better.
+
+  However, unrestricted linking of non-free programs would deprive the
+users of those programs of all benefit from the free status of the
+libraries themselves.  This Library General Public License is intended to
+permit developers of non-free programs to use free libraries, while
+preserving your freedom as a user of such programs to change the free
+libraries that are incorporated in them.  (We have not seen how to achieve
+this as regards changes in header files, but we have achieved it as regards
+changes in the actual functions of the Library.)  The hope is that this
+will lead to faster development of free libraries.
+
+  The precise terms and conditions for copying, distribution and
+modification follow.  Pay close attention to the difference between a
+"work based on the library" and a "work that uses the library".  The
+former contains code derived from the library, while the latter only
+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
+   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
+
+  0. This License Agreement applies to any software library which
+contains a notice placed by the copyright holder or other authorized
+party saying it may be distributed under the terms of this Library
+General Public License (also called "this License").  Each licensee is
+addressed as "you".
+
+  A "library" means a collection of software functions and/or data
+prepared so as to be conveniently linked with application programs
+(which use some of those functions and data) to form executables.
+
+  The "Library", below, refers to any such software library or work
+which has been distributed under these terms.  A "work based on the
+Library" means either the Library or any derivative work under
+copyright law: that is to say, a work containing the Library or a
+portion of it, either verbatim or with modifications and/or translated
+straightforwardly into another language.  (Hereinafter, translation is
+included without limitation in the term "modification".)
+
+  "Source code" for a work means the preferred form of the work for
+making modifications to it.  For a library, complete source code means
+all the source code for all modules it contains, plus any associated
+interface definition files, plus the scripts used to control compilation
+and installation of the library.
+
+  Activities other than copying, distribution and modification are not
+covered by this License; they are outside its scope.  The act of
+running a program using the Library is not restricted, and output from
+such a program is covered only if its contents constitute a work based
+on the Library (independent of the use of the Library in a tool for
+writing it).  Whether that is true depends on what the Library does
+and what the program that uses the Library does.
+  
+  1. You may copy and distribute verbatim copies of the Library's
+complete source code as you receive it, in any medium, provided that
+you conspicuously and appropriately publish on each copy an
+appropriate copyright notice and disclaimer of warranty; keep intact
+all the notices that refer to this License and to the absence of any
+warranty; and distribute a copy of this License along with the
+Library.
+
+  You may charge a fee for the physical act of transferring a copy,
+and you may at your option offer warranty protection in exchange for a
+fee.
+
+  2. You may modify your copy or copies of the Library or any portion
+of it, thus forming a work based on the Library, and copy and
+distribute such modifications or work under the terms of Section 1
+above, provided that you also meet all of these conditions:
+
+    a) The modified work must itself be a software library.
+
+    b) You must cause the files modified to carry prominent notices
+    stating that you changed the files and the date of any change.
+
+    c) You must cause the whole of the work to be licensed at no
+    charge to all third parties under the terms of this License.
+
+    d) If a facility in the modified Library refers to a function or a
+    table of data to be supplied by an application program that uses
+    the facility, other than as an argument passed when the facility
+    is invoked, then you must make a good faith effort to ensure that,
+    in the event an application does not supply such function or
+    table, the facility still operates, and performs whatever part of
+    its purpose remains meaningful.
+
+    (For example, a function in a library to compute square roots has
+    a purpose that is entirely well-defined independent of the
+    application.  Therefore, Subsection 2d requires that any
+    application-supplied function or table used by this function must
+    be optional: if the application does not supply it, the square
+    root function must still compute square roots.)
+
+These requirements apply to the modified work as a whole.  If
+identifiable sections of that work are not derived from the Library,
+and can be reasonably considered independent and separate works in
+themselves, then this License, and its terms, do not apply to those
+sections when you distribute them as separate works.  But when you
+distribute the same sections as part of a whole which is a work based
+on the Library, the distribution of the whole must be on the terms of
+this License, whose permissions for other licensees extend to the
+entire whole, and thus to each and every part regardless of who wrote
+it.
+
+Thus, it is not the intent of this section to claim rights or contest
+your rights to work written entirely by you; rather, the intent is to
+exercise the right to control the distribution of derivative or
+collective works based on the Library.
+
+In addition, mere aggregation of another work not based on the Library
+with the Library (or with a work based on the Library) on a volume of
+a storage or distribution medium does not bring the other work under
+the scope of this License.
+
+  3. You may opt to apply the terms of the ordinary GNU General Public
+License instead of this License to a given copy of the Library.  To do
+this, you must alter all the notices that refer to this License, so
+that they refer to the ordinary GNU General Public License, version 2,
+instead of to this License.  (If a newer version than version 2 of the
+ordinary GNU General Public License has appeared, then you can specify
+that version instead if you wish.)  Do not make any other change in
+these notices.
+
+  Once this change is made in a given copy, it is irreversible for
+that copy, so the ordinary GNU General Public License applies to all
+subsequent copies and derivative works made from that copy.
+
+  This option is useful when you wish to copy part of the code of
+the Library into a program that is not a library.
+
+  4. You may copy and distribute the Library (or a portion or
+derivative of it, under Section 2) in object code or executable form
+under the terms of Sections 1 and 2 above provided that you accompany
+it with the complete corresponding machine-readable source code, which
+must be distributed under the terms of Sections 1 and 2 above on a
+medium customarily used for software interchange.
+
+  If distribution of object code is made by offering access to copy
+from a designated place, then offering equivalent access to copy the
+source code from the same place satisfies the requirement to
+distribute the source code, even though third parties are not
+compelled to copy the source along with the object code.
+
+  5. A program that contains no derivative of any portion of the
+Library, but is designed to work with the Library by being compiled or
+linked with it, is called a "work that uses the Library".  Such a
+work, in isolation, is not a derivative work of the Library, and
+therefore falls outside the scope of this License.
+
+  However, linking a "work that uses the Library" with the Library
+creates an executable that is a derivative of the Library (because it
+contains portions of the Library), rather than a "work that uses the
+library".  The executable is therefore covered by this License.
+Section 6 states terms for distribution of such executables.
+
+  When a "work that uses the Library" uses material from a header file
+that is part of the Library, the object code for the work may be a
+derivative work of the Library even though the source code is not.
+Whether this is true is especially significant if the work can be
+linked without the Library, or if the work is itself a library.  The
+threshold for this to be true is not precisely defined by law.
+
+  If such an object file uses only numerical parameters, data
+structure layouts and accessors, and small macros and small inline
+functions (ten lines or less in length), then the use of the object
+file is unrestricted, regardless of whether it is legally a derivative
+work.  (Executables containing this object code plus portions of the
+Library will still fall under Section 6.)
+
+  Otherwise, if the work is a derivative of the Library, you may
+distribute the object code for the work under the terms of Section 6.
+Any executables containing that work also fall under Section 6,
+whether or not they are linked directly with the Library itself.
+
+  6. As an exception to the Sections above, you may also compile or
+link a "work that uses the Library" with the Library to produce a
+work containing portions of the Library, and distribute that work
+under terms of your choice, provided that the terms permit
+modification of the work for the customer's own use and reverse
+engineering for debugging such modifications.
+
+  You must give prominent notice with each copy of the work that the
+Library is used in it and that the Library and its use are covered by
+this License.  You must supply a copy of this License.  If the work
+during execution displays copyright notices, you must include the
+copyright notice for the Library among them, as well as a reference
+directing the user to the copy of this License.  Also, you must do one
+of these things:
+
+    a) Accompany the work with the complete corresponding
+    machine-readable source code for the Library including whatever
+    changes were used in the work (which must be distributed under
+    Sections 1 and 2 above); and, if the work is an executable linked
+    with the Library, with the complete machine-readable "work that
+    uses the Library", as object code and/or source code, so that the
+    user can modify the Library and then relink to produce a modified
+    executable containing the modified Library.  (It is understood
+    that the user who changes the contents of definitions files in the
+    Library will not necessarily be able to recompile the application
+    to use the modified definitions.)
+
+    b) Accompany the work with a written offer, valid for at
+    least three years, to give the same user the materials
+    specified in Subsection 6a, above, for a charge no more
+    than the cost of performing this distribution.
+
+    c) If distribution of the work is made by offering access to copy
+    from a designated place, offer equivalent access to copy the above
+    specified materials from the same place.
+
+    d) Verify that the user has already received a copy of these
+    materials or that you have already sent this user a copy.
+
+  For an executable, the required form of the "work that uses the
+Library" must include any data and utility programs needed for
+reproducing the executable from it.  However, as a special exception,
+the source code distributed need not include anything that is normally
+distributed (in either source or binary form) with the major
+components (compiler, kernel, and so on) of the operating system on
+which the executable runs, unless that component itself accompanies
+the executable.
+
+  It may happen that this requirement contradicts the license
+restrictions of other proprietary libraries that do not normally
+accompany the operating system.  Such a contradiction means you cannot
+use both them and the Library together in an executable that you
+distribute.
+
+  7. You may place library facilities that are a work based on the
+Library side-by-side in a single library together with other library
+facilities not covered by this License, and distribute such a combined
+library, provided that the separate distribution of the work based on
+the Library and of the other library facilities is otherwise
+permitted, and provided that you do these two things:
+
+    a) Accompany the combined library with a copy of the same work
+    based on the Library, uncombined with any other library
+    facilities.  This must be distributed under the terms of the
+    Sections above.
+
+    b) Give prominent notice with the combined library of the fact
+    that part of it is a work based on the Library, and explaining
+    where to find the accompanying uncombined form of the same work.
+
+  8. You may not copy, modify, sublicense, link with, or distribute
+the Library except as expressly provided under this License.  Any
+attempt otherwise to copy, modify, sublicense, link with, or
+distribute the Library is void, and will automatically terminate your
+rights under this License.  However, parties who have received copies,
+or rights, from you under this License will not have their licenses
+terminated so long as such parties remain in full compliance.
+
+  9. You are not required to accept this License, since you have not
+signed it.  However, nothing else grants you permission to modify or
+distribute the Library or its derivative works.  These actions are
+prohibited by law if you do not accept this License.  Therefore, by
+modifying or distributing the Library (or any work based on the
+Library), you indicate your acceptance of this License to do so, and
+all its terms and conditions for copying, distributing or modifying
+the Library or works based on it.
+
+  10. Each time you redistribute the Library (or any work based on the
+Library), the recipient automatically receives a license from the
+original licensor to copy, distribute, link with or modify the Library
+subject to these terms and conditions.  You may not impose any further
+restrictions on the recipients' exercise of the rights granted herein.
+You are not responsible for enforcing compliance by third parties to
+this License.
+
+  11. If, as a consequence of a court judgment or allegation of patent
+infringement or for any other reason (not limited to patent issues),
+conditions are imposed on you (whether by court order, agreement or
+otherwise) that contradict the conditions of this License, they do not
+excuse you from the conditions of this License.  If you cannot
+distribute so as to satisfy simultaneously your obligations under this
+License and any other pertinent obligations, then as a consequence you
+may not distribute the Library at all.  For example, if a patent
+license would not permit royalty-free redistribution of the Library by
+all those who receive copies directly or indirectly through you, then
+the only way you could satisfy both it and this License would be to
+refrain entirely from distribution of the Library.
+
+If any portion of this section is held invalid or unenforceable under any
+particular circumstance, the balance of the section is intended to apply,
+and the section as a whole is intended to apply in other circumstances.
+
+It is not the purpose of this section to induce you to infringe any
+patents or other property right claims or to contest validity of any
+such claims; this section has the sole purpose of protecting the
+integrity of the free software distribution system which is
+implemented by public license practices.  Many people have made
+generous contributions to the wide range of software distributed
+through that system in reliance on consistent application of that
+system; it is up to the author/donor to decide if he or she is willing
+to distribute software through any other system and a licensee cannot
+impose that choice.
+
+This section is intended to make thoroughly clear what is believed to
+be a consequence of the rest of this License.
+
+  12. If the distribution and/or use of the Library is restricted in
+certain countries either by patents or by copyrighted interfaces, the
+original copyright holder who places the Library under this License may add
+an explicit geographical distribution limitation excluding those countries,
+so that distribution is permitted only in or among countries not thus
+excluded.  In such case, this License incorporates the limitation as if
+written in the body of this License.
+
+  13. The Free Software Foundation may publish revised and/or new
+versions of the Library General Public License from time to time.
+Such new versions will be similar in spirit to the present version,
+but may differ in detail to address new problems or concerns.
+
+Each version is given a distinguishing version number.  If the Library
+specifies a version number of this License which applies to it and
+"any later version", you have the option of following the terms and
+conditions either of that version or of any later version published by
+the Free Software Foundation.  If the Library does not specify a
+license version number, you may choose any version ever published by
+the Free Software Foundation.
+
+  14. If you wish to incorporate parts of the Library into other free
+programs whose distribution conditions are incompatible with these,
+write to the author to ask for permission.  For software which is
+copyrighted by the Free Software Foundation, write to the Free
+Software Foundation; we sometimes make exceptions for this.  Our
+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
+
+  15. BECAUSE THE LIBRARY IS LICENSED FREE OF CHARGE, THERE IS NO
+WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW.
+EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR
+OTHER PARTIES PROVIDE THE LIBRARY "AS IS" WITHOUT WARRANTY OF ANY
+KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+PURPOSE.  THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE
+LIBRARY IS WITH YOU.  SHOULD THE LIBRARY PROVE DEFECTIVE, YOU ASSUME
+THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
+
+  16. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN
+WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY
+AND/OR REDISTRIBUTE THE LIBRARY AS PERMITTED ABOVE, BE LIABLE TO YOU
+FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR
+CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE
+LIBRARY (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING
+RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A
+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
+
+           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
+everyone can redistribute and change.  You can do so by permitting
+redistribution under these terms (or, alternatively, under the terms of the
+ordinary General Public License).
+
+  To apply these terms, attach the following notices to the library.  It is
+safest to attach them to the start of each source file to most effectively
+convey the exclusion of warranty; and each file should have at least the
+"copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the library's name and a brief idea of what it does.>
+    Copyright (C) <year>  <name of author>
+
+    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
+
+Also add information on how to contact you by electronic and paper mail.
+
+You should also get your employer (if you work as a programmer) or your
+school, if any, to sign a "copyright disclaimer" for the library, if
+necessary.  Here is a sample; alter the names:
+
+  Yoyodyne, Inc., hereby disclaims all copyright interest in the
+  library `Frob' (a library for tweaking knobs) written by James Random Hacker.
+
+  <signature of Ty Coon>, 1 April 1990
+  Ty Coon, President of Vice
+
+That's all there is to it!

ChangeLog

@@ -0,0 +1,305 @@
+openal-soft-1.18.2:
+
+    Fixed resetting the FPU rounding mode after certain function calls on
+    Windows.
+
+    Fixed use of SSE intrinsics when building with Clang on Windows.
+
+    Fixed a crash with the JACK backend when using JACK1.
+
+    Fixed use of pthread_setnane_np on NetBSD.
+
+    Fixed building on FreeBSD with an older freebsd-lib.
+
+    OSS now links with libossaudio if found at build time (for NetBSD).
+
+openal-soft-1.18.1:
+
+    Fixed an issue where resuming a source might not restart playing it.
+
+    Fixed PulseAudio playback when the configured stream length is much less
+    than the requested length.
+
+    Fixed MMDevAPI capture with sample rates not matching the backing device.
+
+    Fixed int32 output for the Wave Writer.
+
+    Fixed enumeration of OSS devices that are missing device files.
+
+    Added correct retrieval of the executable's path on FreeBSD.
+
+    Added a config option to specify the dithering depth.
+
+    Added a 5.1 decoder preset that excludes front-center output.
+
+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

@@ -0,0 +1,190 @@
+#ifndef _AL_AUXEFFECTSLOT_H_
+#define _AL_AUXEFFECTSLOT_H_
+
+#include "alMain.h"
+#include "alEffect.h"
+
+#include "atomic.h"
+#include "align.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct ALeffectStateVtable;
+struct ALeffectslot;
+
+typedef struct ALeffectState {
+    RefCount Ref;
+    const struct ALeffectStateVtable *vtbl;
+
+    ALfloat (*OutBuffer)[BUFFERSIZE];
+    ALsizei OutChannels;
+} ALeffectState;
+
+void ALeffectState_Construct(ALeffectState *state);
+void ALeffectState_Destruct(ALeffectState *state);
+
+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;
+
+    ALenum Type;
+    ALeffectProps Props;
+
+    ALeffectState *State;
+
+    ATOMIC(struct ALeffectslotProps*) 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;
+
+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); }
+
+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);
+
+
+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);
+
+ALeffectStateFactory *ALdedicatedStateFactory_getFactory(void);
+
+
+ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect);
+
+void InitEffectFactoryMap(void);
+void DeinitEffectFactoryMap(void);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alBuffer.h

@@ -0,0 +1,128 @@
+#ifndef _AL_BUFFER_H_
+#define _AL_BUFFER_H_
+
+#include "alMain.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* 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 */
+};
+
+ALsizei BytesFromUserFmt(enum UserFmtType type);
+ALsizei ChannelsFromUserFmt(enum UserFmtChannels chans);
+inline ALsizei FrameSizeFromUserFmt(enum UserFmtChannels chans, enum UserFmtType type)
+{
+    return ChannelsFromUserFmt(chans) * BytesFromUserFmt(type);
+}
+
+
+/* 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)
+
+ALsizei BytesFromFmt(enum FmtType type);
+ALsizei ChannelsFromFmt(enum FmtChannels chans);
+inline ALsizei FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type)
+{
+    return ChannelsFromFmt(chans) * BytesFromFmt(type);
+}
+
+
+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;
+
+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);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alEffect.h

@@ -0,0 +1,196 @@
+#ifndef _AL_EFFECT_H_
+#define _AL_EFFECT_H_
+
+#include "alMain.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct ALeffect;
+
+enum {
+    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];
+
+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;
+        ALfloat Diffusion;
+        ALfloat Gain;
+        ALfloat GainHF;
+        ALfloat DecayTime;
+        ALfloat DecayHFRatio;
+        ALfloat ReflectionsGain;
+        ALfloat ReflectionsDelay;
+        ALfloat LateReverbGain;
+        ALfloat LateReverbDelay;
+        ALfloat AirAbsorptionGainHF;
+        ALfloat RoomRolloffFactor;
+        ALboolean DecayHFLimit;
+
+        // Additional EAX Reverb Properties
+        ALfloat GainLF;
+        ALfloat DecayLFRatio;
+        ALfloat ReflectionsPan[3];
+        ALfloat LateReverbPan[3];
+        ALfloat EchoTime;
+        ALfloat EchoDepth;
+        ALfloat ModulationTime;
+        ALfloat ModulationDepth;
+        ALfloat HFReference;
+        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;
+
+        ALfloat Damping;
+        ALfloat Feedback;
+
+        ALfloat Spread;
+    } Echo;
+
+    struct {
+        ALfloat LowCutoff;
+        ALfloat LowGain;
+        ALfloat Mid1Center;
+        ALfloat Mid1Gain;
+        ALfloat Mid1Width;
+        ALfloat Mid2Center;
+        ALfloat Mid2Gain;
+        ALfloat Mid2Width;
+        ALfloat HighCutoff;
+        ALfloat HighGain;
+    } Equalizer;
+
+    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;
+
+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); }
+
+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); }
+
+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
+
+#endif

OpenAL32/Include/alError.h

@@ -0,0 +1,33 @@
+#ifndef _AL_ERROR_H_
+#define _AL_ERROR_H_
+
+#include "alMain.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+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
+}
+#endif
+
+#endif

OpenAL32/Include/alFilter.h

@@ -0,0 +1,164 @@
+#ifndef _AL_FILTER_H_
+#define _AL_FILTER_H_
+
+#include "alMain.h"
+
+#include "math_defs.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define LOWPASSFREQREF  (5000.0f)
+#define HIGHPASSFREQREF  (250.0f)
+
+
+/* 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)
+{
+    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));
+}
+
+inline void ALfilterState_clear(ALfilterState *filter)
+{
+    filter->x[0] = 0.0f;
+    filter->x[1] = 0.0f;
+    filter->y[0] = 0.0f;
+    filter->y[1] = 0.0f;
+}
+
+void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat rcpQ);
+
+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];
+    }
+}
+
+
+typedef struct ALfilter {
+    // Filter type (AL_FILTER_NULL, ...)
+    ALenum type;
+
+    ALfloat Gain;
+    ALfloat GainHF;
+    ALfloat HFReference;
+    ALfloat GainLF;
+    ALfloat LFReference;
+
+    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);
+
+    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;
+
+#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)))
+
+#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)))
+
+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);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alListener.h

@@ -0,0 +1,66 @@
+#ifndef _AL_LISTENER_H_
+#define _AL_LISTENER_H_
+
+#include "alMain.h"
+#include "alu.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+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 {
+    alignas(16) 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
+
+#endif

OpenAL32/Include/alSource.h

@@ -0,0 +1,128 @@
+#ifndef _AL_SOURCE_H_
+#define _AL_SOURCE_H_
+
+#include "bool.h"
+#include "alMain.h"
+#include "alu.h"
+#include "hrtf.h"
+#include "atomic.h"
+
+#define MAX_SENDS      16
+#define DEFAULT_SENDS  2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct ALbuffer;
+struct ALsource;
+
+
+typedef struct ALbufferlistitem {
+    struct ALbuffer *buffer;
+    ATOMIC(struct ALbufferlistitem*) next;
+} ALbufferlistitem;
+
+
+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;
+    ALboolean WetGainHFAuto;
+    ALfloat   OuterGainHF;
+
+    ALfloat AirAbsorptionFactor;
+    ALfloat RoomRolloffFactor;
+    ALfloat DopplerFactor;
+
+    /* NOTE: Stereo pan angles are specified in radians, counter-clockwise
+     * rather than clockwise.
+     */
+    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;
+
+    /**
+     * Last user-specified offset, and the offset type (bytes, samples, or
+     * seconds).
+     */
+    ALdouble Offset;
+    ALenum   OffsetType;
+
+    /** Source type (static, streaming, or undetermined) */
+    ALint SourceType;
+
+    /** Source state (initial, playing, paused, or stopped) */
+    ATOMIC(ALenum) state;
+
+    /** 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
+}
+#endif
+
+#endif

OpenAL32/Include/alThunk.h

@@ -0,0 +1,20 @@
+#ifndef ALTHUNK_H
+#define ALTHUNK_H
+
+#include "alMain.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+void ThunkInit(void);
+void ThunkExit(void);
+ALenum NewThunkEntry(ALuint *index);
+void FreeThunkEntry(ALuint index);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif //ALTHUNK_H
+

OpenAL32/Include/alu.h

@@ -0,0 +1,516 @@
+#ifndef _ALU_H_
+#define _ALU_H_
+
+#include <limits.h>
+#include <math.h>
+#ifdef HAVE_FLOAT_H
+#include <float.h>
+#endif
+#ifdef HAVE_IEEEFP_H
+#include <ieeefp.h>
+#endif
+
+#include "alMain.h"
+#include "alBuffer.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+
+#include "hrtf.h"
+#include "align.h"
+#include "nfcfilter.h"
+#include "math_defs.h"
+
+
+#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
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct ALsource;
+struct ALbufferlistitem;
+struct ALvoice;
+struct ALeffectslot;
+
+
+#define DITHER_RNG_SEED 22222
+
+
+enum SpatializeMode {
+    SpatializeOff = AL_FALSE,
+    SpatializeOn = AL_TRUE,
+    SpatializeAuto = AL_AUTO_SOFT
+};
+
+enum Resampler {
+    PointResampler,
+    LinearResampler,
+    FIR4Resampler,
+    BSincResampler,
+
+    ResamplerMax = BSincResampler
+};
+extern enum Resampler ResamplerDefault;
+
+/* 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;
+
+ALboolean BsincPrepare(const ALuint increment, BsincState *state);
+
+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)
+{
+    vector->v[0] = x;
+    vector->v[1] = y;
+    vector->v[2] = z;
+    vector->v[3] = w;
+}
+
+
+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 *OutBuffer, ALsizei NumSamples);
+/* Caller must lock the device. */
+void aluHandleDisconnect(ALCdevice *device);
+
+extern ALfloat ConeScale;
+extern ALfloat ZScale;
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/bs2b.h

@@ -0,0 +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 */
+    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/alAuxEffectSlot.c

@@ -0,0 +1,727 @@
+/**
+ * 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 <math.h>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "alMain.h"
+#include "alAuxEffectSlot.h"
+#include "alThunk.h"
+#include "alError.h"
+#include "alListener.h"
+#include "alSource.h"
+
+#include "almalloc.h"
+
+
+extern inline void LockEffectSlotsRead(ALCcontext *context);
+extern inline void UnlockEffectSlotsRead(ALCcontext *context);
+extern inline void LockEffectSlotsWrite(ALCcontext *context);
+extern inline void UnlockEffectSlotsWrite(ALCcontext *context);
+extern inline struct ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id);
+extern inline struct ALeffectslot *RemoveEffectSlot(ALCcontext *context, ALuint id);
+
+static UIntMap EffectStateFactoryMap;
+static inline ALeffectStateFactory *getFactoryByType(ALenum type)
+{
+    ALeffectStateFactory* (*getFactory)(void) = LookupUIntMapKey(&EffectStateFactoryMap, type);
+    if(getFactory != NULL)
+        return getFactory();
+    return NULL;
+}
+
+static void ALeffectState_IncRef(ALeffectState *state);
+static void ALeffectState_DecRef(ALeffectState *state);
+
+#define DO_UPDATEPROPS() do {                                                 \
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))          \
+        UpdateEffectSlotProps(slot);                                          \
+    else                                                                      \
+        ATOMIC_FLAG_CLEAR(&slot->PropsClean, almemory_order_release);         \
+} while(0)
+
+
+AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots)
+{
+    ALCcontext *context;
+    ALeffectslot **tmpslots = NULL;
+    ALsizei cur;
+    ALenum err;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    tmpslots = al_malloc(DEF_ALIGN, sizeof(ALeffectslot*)*n);
+
+    LockEffectSlotsWrite(context);
+    for(cur = 0;cur < n;cur++)
+    {
+        ALeffectslot *slot = al_calloc(16, sizeof(ALeffectslot));
+        err = AL_OUT_OF_MEMORY;
+        if(!slot || (err=InitEffectSlot(slot)) != AL_NO_ERROR)
+        {
+            al_free(slot);
+            UnlockEffectSlotsWrite(context);
+
+            alDeleteAuxiliaryEffectSlots(cur, effectslots);
+            SET_ERROR_AND_GOTO(context, err, done);
+        }
+
+        err = NewThunkEntry(&slot->id);
+        if(err == AL_NO_ERROR)
+            err = InsertUIntMapEntryNoLock(&context->EffectSlotMap, slot->id, slot);
+        if(err != AL_NO_ERROR)
+        {
+            FreeThunkEntry(slot->id);
+            ALeffectState_DecRef(slot->Effect.State);
+            if(slot->Params.EffectState)
+                ALeffectState_DecRef(slot->Params.EffectState);
+            al_free(slot);
+            UnlockEffectSlotsWrite(context);
+
+            alDeleteAuxiliaryEffectSlots(cur, effectslots);
+            SET_ERROR_AND_GOTO(context, err, done);
+        }
+
+        aluInitEffectPanning(slot);
+
+        tmpslots[cur] = slot;
+        effectslots[cur] = slot->id;
+    }
+    if(n > 0)
+    {
+        struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire);
+        struct ALeffectslotArray *newarray = NULL;
+        ALsizei newcount = curarray->count + n;
+        ALCdevice *device;
+
+        newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
+        newarray->count = newcount;
+        memcpy(newarray->slot, tmpslots, sizeof(ALeffectslot*)*n);
+        if(curarray)
+            memcpy(newarray->slot+n, curarray->slot, sizeof(ALeffectslot*)*curarray->count);
+
+        newarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray,
+                                       almemory_order_acq_rel);
+        device = context->Device;
+        while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
+            althrd_yield();
+        al_free(newarray);
+    }
+    UnlockEffectSlotsWrite(context);
+
+done:
+    al_free(tmpslots);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots)
+{
+    ALCcontext *context;
+    ALeffectslot *slot;
+    ALsizei i;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    LockEffectSlotsWrite(context);
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    for(i = 0;i < n;i++)
+    {
+        if((slot=LookupEffectSlot(context, effectslots[i])) == NULL)
+            SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+        if(ReadRef(&slot->ref) != 0)
+            SET_ERROR_AND_GOTO(context, AL_INVALID_OPERATION, done);
+    }
+
+    // All effectslots are valid
+    if(n > 0)
+    {
+        struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire);
+        struct ALeffectslotArray *newarray = NULL;
+        ALsizei newcount = curarray->count - n;
+        ALCdevice *device;
+        ALsizei j, k;
+
+        assert(newcount >= 0);
+        newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
+        newarray->count = newcount;
+        for(i = j = 0;i < newarray->count;)
+        {
+            slot = curarray->slot[j++];
+            for(k = 0;k < n;k++)
+            {
+                if(slot->id == effectslots[k])
+                    break;
+            }
+            if(k == n)
+                newarray->slot[i++] = slot;
+        }
+
+        newarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray,
+                                       almemory_order_acq_rel);
+        device = context->Device;
+        while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
+            althrd_yield();
+        al_free(newarray);
+    }
+
+    for(i = 0;i < n;i++)
+    {
+        if((slot=RemoveEffectSlot(context, effectslots[i])) == NULL)
+            continue;
+        FreeThunkEntry(slot->id);
+
+        DeinitEffectSlot(slot);
+
+        memset(slot, 0, sizeof(*slot));
+        al_free(slot);
+    }
+
+done:
+    UnlockEffectSlotsWrite(context);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot)
+{
+    ALCcontext *context;
+    ALboolean  ret;
+
+    context = GetContextRef();
+    if(!context) return AL_FALSE;
+
+    LockEffectSlotsRead(context);
+    ret = (LookupEffectSlot(context, effectslot) ? AL_TRUE : AL_FALSE);
+    UnlockEffectSlotsRead(context);
+
+    ALCcontext_DecRef(context);
+
+    return ret;
+}
+
+AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint value)
+{
+    ALCdevice *device;
+    ALCcontext *context;
+    ALeffectslot *slot;
+    ALeffect *effect = NULL;
+    ALenum err;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    LockEffectSlotsRead(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_EFFECT:
+        device = context->Device;
+
+        LockEffectsRead(device);
+        effect = (value ? LookupEffect(device, value) : NULL);
+        if(!(value == 0 || effect != NULL))
+        {
+            UnlockEffectsRead(device);
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        }
+        err = InitializeEffect(device, slot, effect);
+        UnlockEffectsRead(device);
+
+        if(err != AL_NO_ERROR)
+            SET_ERROR_AND_GOTO(context, err, done);
+        break;
+
+    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
+        if(!(value == AL_TRUE || value == AL_FALSE))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        slot->AuxSendAuto = value;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+    DO_UPDATEPROPS();
+
+done:
+    UnlockEffectSlotsRead(context);
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *values)
+{
+    ALCcontext *context;
+
+    switch(param)
+    {
+    case AL_EFFECTSLOT_EFFECT:
+    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
+        alAuxiliaryEffectSloti(effectslot, param, values[0]);
+        return;
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    LockEffectSlotsRead(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    UnlockEffectSlotsRead(context);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat value)
+{
+    ALCcontext *context;
+    ALeffectslot *slot;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    LockEffectSlotsRead(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_GAIN:
+        if(!(value >= 0.0f && value <= 1.0f))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        slot->Gain = value;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+    DO_UPDATEPROPS();
+
+done:
+    UnlockEffectSlotsRead(context);
+    WriteUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *values)
+{
+    ALCcontext *context;
+
+    switch(param)
+    {
+    case AL_EFFECTSLOT_GAIN:
+        alAuxiliaryEffectSlotf(effectslot, param, values[0]);
+        return;
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    LockEffectSlotsRead(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    UnlockEffectSlotsRead(context);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *value)
+{
+    ALCcontext *context;
+    ALeffectslot *slot;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    LockEffectSlotsRead(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
+        *value = slot->AuxSendAuto;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    UnlockEffectSlotsRead(context);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *values)
+{
+    ALCcontext *context;
+
+    switch(param)
+    {
+    case AL_EFFECTSLOT_EFFECT:
+    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
+        alGetAuxiliaryEffectSloti(effectslot, param, values);
+        return;
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    LockEffectSlotsRead(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    UnlockEffectSlotsRead(context);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *value)
+{
+    ALCcontext *context;
+    ALeffectslot *slot;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    LockEffectSlotsRead(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_GAIN:
+        *value = slot->Gain;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    UnlockEffectSlotsRead(context);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *values)
+{
+    ALCcontext *context;
+
+    switch(param)
+    {
+    case AL_EFFECTSLOT_GAIN:
+        alGetAuxiliaryEffectSlotf(effectslot, param, values);
+        return;
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    LockEffectSlotsRead(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    switch(param)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    UnlockEffectSlotsRead(context);
+    ALCcontext_DecRef(context);
+}
+
+
+void InitEffectFactoryMap(void)
+{
+    InitUIntMap(&EffectStateFactoryMap, INT_MAX);
+
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_NULL, ALnullStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_EAXREVERB, ALreverbStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_REVERB, ALreverbStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_CHORUS, ALchorusStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_COMPRESSOR, ALcompressorStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_DISTORTION, ALdistortionStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_ECHO, ALechoStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_EQUALIZER, ALequalizerStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_FLANGER, ALflangerStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_RING_MODULATOR, ALmodulatorStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_DEDICATED_DIALOGUE, ALdedicatedStateFactory_getFactory);
+    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT, ALdedicatedStateFactory_getFactory);
+}
+
+void DeinitEffectFactoryMap(void)
+{
+    ResetUIntMap(&EffectStateFactoryMap);
+}
+
+
+ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect)
+{
+    ALenum newtype = (effect ? effect->type : AL_EFFECT_NULL);
+    struct ALeffectslotProps *props;
+    ALeffectState *State;
+
+    if(newtype != EffectSlot->Effect.Type)
+    {
+        ALeffectStateFactory *factory;
+
+        factory = getFactoryByType(newtype);
+        if(!factory)
+        {
+            ERR("Failed to find factory for effect type 0x%04x\n", newtype);
+            return AL_INVALID_ENUM;
+        }
+        State = V0(factory,create)();
+        if(!State) return AL_OUT_OF_MEMORY;
+
+        START_MIXER_MODE();
+        almtx_lock(&Device->BackendLock);
+        State->OutBuffer = Device->Dry.Buffer;
+        State->OutChannels = Device->Dry.NumChannels;
+        if(V(State,deviceUpdate)(Device) == AL_FALSE)
+        {
+            almtx_unlock(&Device->BackendLock);
+            LEAVE_MIXER_MODE();
+            ALeffectState_DecRef(State);
+            return AL_OUT_OF_MEMORY;
+        }
+        almtx_unlock(&Device->BackendLock);
+        END_MIXER_MODE();
+
+        if(!effect)
+        {
+            EffectSlot->Effect.Type = AL_EFFECT_NULL;
+            memset(&EffectSlot->Effect.Props, 0, sizeof(EffectSlot->Effect.Props));
+        }
+        else
+        {
+            EffectSlot->Effect.Type = effect->type;
+            EffectSlot->Effect.Props = effect->Props;
+        }
+
+        ALeffectState_DecRef(EffectSlot->Effect.State);
+        EffectSlot->Effect.State = State;
+    }
+    else if(effect)
+        EffectSlot->Effect.Props = effect->Props;
+
+    /* Remove state references from old effect slot property updates. */
+    props = ATOMIC_LOAD_SEQ(&EffectSlot->FreeList);
+    while(props)
+    {
+        if(props->State)
+            ALeffectState_DecRef(props->State);
+        props->State = NULL;
+        props = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
+    }
+
+    return AL_NO_ERROR;
+}
+
+
+static void ALeffectState_IncRef(ALeffectState *state)
+{
+    uint ref;
+    ref = IncrementRef(&state->Ref);
+    TRACEREF("%p increasing refcount to %u\n", state, ref);
+}
+
+static void ALeffectState_DecRef(ALeffectState *state)
+{
+    uint ref;
+    ref = DecrementRef(&state->Ref);
+    TRACEREF("%p decreasing refcount to %u\n", state, ref);
+    if(ref == 0) DELETE_OBJ(state);
+}
+
+
+void ALeffectState_Construct(ALeffectState *state)
+{
+    InitRef(&state->Ref, 1);
+
+    state->OutBuffer = NULL;
+    state->OutChannels = 0;
+}
+
+void ALeffectState_Destruct(ALeffectState *UNUSED(state))
+{
+}
+
+
+ALenum InitEffectSlot(ALeffectslot *slot)
+{
+    ALeffectStateFactory *factory;
+
+    slot->Effect.Type = AL_EFFECT_NULL;
+
+    factory = getFactoryByType(AL_EFFECT_NULL);
+    if(!(slot->Effect.State=V0(factory,create)()))
+        return AL_OUT_OF_MEMORY;
+
+    slot->Gain = 1.0;
+    slot->AuxSendAuto = AL_TRUE;
+    ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_relaxed);
+    InitRef(&slot->ref, 0);
+
+    ATOMIC_INIT(&slot->Update, NULL);
+    ATOMIC_INIT(&slot->FreeList, NULL);
+
+    slot->Params.Gain = 1.0f;
+    slot->Params.AuxSendAuto = AL_TRUE;
+    ALeffectState_IncRef(slot->Effect.State);
+    slot->Params.EffectState = slot->Effect.State;
+    slot->Params.RoomRolloff = 0.0f;
+    slot->Params.DecayTime = 0.0f;
+    slot->Params.DecayHFRatio = 0.0f;
+    slot->Params.DecayHFLimit = AL_FALSE;
+    slot->Params.AirAbsorptionGainHF = 1.0f;
+
+    return AL_NO_ERROR;
+}
+
+void DeinitEffectSlot(ALeffectslot *slot)
+{
+    struct ALeffectslotProps *props;
+    size_t count = 0;
+
+    props = ATOMIC_LOAD_SEQ(&slot->Update);
+    if(props)
+    {
+        if(props->State) ALeffectState_DecRef(props->State);
+        TRACE("Freed unapplied AuxiliaryEffectSlot update %p\n", props);
+        al_free(props);
+    }
+    props = ATOMIC_LOAD(&slot->FreeList, almemory_order_relaxed);
+    while(props)
+    {
+        struct ALeffectslotProps *next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
+        if(props->State) ALeffectState_DecRef(props->State);
+        al_free(props);
+        props = next;
+        ++count;
+    }
+    TRACE("Freed "SZFMT" AuxiliaryEffectSlot property object%s\n", count, (count==1)?"":"s");
+
+    ALeffectState_DecRef(slot->Effect.State);
+    if(slot->Params.EffectState)
+        ALeffectState_DecRef(slot->Params.EffectState);
+}
+
+void UpdateEffectSlotProps(ALeffectslot *slot)
+{
+    struct ALeffectslotProps *props;
+    ALeffectState *oldstate;
+
+    /* Get an unused property container, or allocate a new one as needed. */
+    props = ATOMIC_LOAD(&slot->FreeList, almemory_order_relaxed);
+    if(!props)
+        props = al_calloc(16, sizeof(*props));
+    else
+    {
+        struct ALeffectslotProps *next;
+        do {
+            next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
+        } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&slot->FreeList, &props, next,
+                almemory_order_seq_cst, almemory_order_acquire) == 0);
+    }
+
+    /* Copy in current property values. */
+    props->Gain = slot->Gain;
+    props->AuxSendAuto = slot->AuxSendAuto;
+
+    props->Type = slot->Effect.Type;
+    props->Props = slot->Effect.Props;
+    /* Swap out any stale effect state object there may be in the container, to
+     * delete it.
+     */
+    ALeffectState_IncRef(slot->Effect.State);
+    oldstate = props->State;
+    props->State = slot->Effect.State;
+
+    /* Set the new container for updating internal parameters. */
+    props = ATOMIC_EXCHANGE_PTR(&slot->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 ALeffectslotProps*, &slot->FreeList, props);
+    }
+
+    if(oldstate)
+        ALeffectState_DecRef(oldstate);
+}
+
+void UpdateAllEffectSlotProps(ALCcontext *context)
+{
+    struct ALeffectslotArray *auxslots;
+    ALsizei i;
+
+    LockEffectSlotsRead(context);
+    auxslots = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire);
+    for(i = 0;i < auxslots->count;i++)
+    {
+        ALeffectslot *slot = auxslots->slot[i];
+        if(!ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_acq_rel))
+            UpdateEffectSlotProps(slot);
+    }
+    UnlockEffectSlotsRead(context);
+}
+
+ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context)
+{
+    ALsizei pos;
+    for(pos = 0;pos < Context->EffectSlotMap.size;pos++)
+    {
+        ALeffectslot *temp = Context->EffectSlotMap.values[pos];
+        Context->EffectSlotMap.values[pos] = NULL;
+
+        DeinitEffectSlot(temp);
+
+        FreeThunkEntry(temp->id);
+        memset(temp, 0, sizeof(ALeffectslot));
+        al_free(temp);
+    }
+}

OpenAL32/alEffect.c

@@ -0,0 +1,716 @@
+/**
+ * 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 <math.h>
+#include <float.h>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "alMain.h"
+#include "alEffect.h"
+#include "alThunk.h"
+#include "alError.h"
+
+
+ALboolean DisabledEffects[MAX_EFFECTS];
+
+extern inline void LockEffectsRead(ALCdevice *device);
+extern inline void UnlockEffectsRead(ALCdevice *device);
+extern inline void LockEffectsWrite(ALCdevice *device);
+extern inline void UnlockEffectsWrite(ALCdevice *device);
+extern inline struct ALeffect *LookupEffect(ALCdevice *device, ALuint id);
+extern inline struct ALeffect *RemoveEffect(ALCdevice *device, ALuint id);
+extern inline ALboolean IsReverbEffect(ALenum type);
+
+static void InitEffectParams(ALeffect *effect, ALenum type);
+
+
+AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects)
+{
+    ALCdevice *device;
+    ALCcontext *context;
+    ALsizei cur;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    device = context->Device;
+    for(cur = 0;cur < n;cur++)
+    {
+        ALeffect *effect = al_calloc(16, sizeof(ALeffect));
+        ALenum err = AL_OUT_OF_MEMORY;
+        if(!effect || (err=InitEffect(effect)) != AL_NO_ERROR)
+        {
+            al_free(effect);
+            alDeleteEffects(cur, effects);
+            SET_ERROR_AND_GOTO(context, err, done);
+        }
+
+        err = NewThunkEntry(&effect->id);
+        if(err == AL_NO_ERROR)
+            err = InsertUIntMapEntry(&device->EffectMap, effect->id, effect);
+        if(err != AL_NO_ERROR)
+        {
+            FreeThunkEntry(effect->id);
+            memset(effect, 0, sizeof(ALeffect));
+            al_free(effect);
+
+            alDeleteEffects(cur, effects);
+            SET_ERROR_AND_GOTO(context, err, done);
+        }
+
+        effects[cur] = effect->id;
+    }
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects)
+{
+    ALCdevice *device;
+    ALCcontext *context;
+    ALeffect *effect;
+    ALsizei i;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    device = context->Device;
+    LockEffectsWrite(device);
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    for(i = 0;i < n;i++)
+    {
+        if(effects[i] && LookupEffect(device, effects[i]) == NULL)
+            SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    }
+    for(i = 0;i < n;i++)
+    {
+        if((effect=RemoveEffect(device, effects[i])) == NULL)
+            continue;
+        FreeThunkEntry(effect->id);
+
+        memset(effect, 0, sizeof(*effect));
+        al_free(effect);
+    }
+
+done:
+    UnlockEffectsWrite(device);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect)
+{
+    ALCcontext *Context;
+    ALboolean  result;
+
+    Context = GetContextRef();
+    if(!Context) return AL_FALSE;
+
+    LockEffectsRead(Context->Device);
+    result = ((!effect || LookupEffect(Context->Device, effect)) ?
+              AL_TRUE : AL_FALSE);
+    UnlockEffectsRead(Context->Device);
+
+    ALCcontext_DecRef(Context);
+
+    return result;
+}
+
+AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint value)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsWrite(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        if(param == AL_EFFECT_TYPE)
+        {
+            ALboolean isOk = (value == AL_EFFECT_NULL);
+            ALint i;
+            for(i = 0;!isOk && EffectList[i].val;i++)
+            {
+                if(value == EffectList[i].val &&
+                   !DisabledEffects[EffectList[i].type])
+                    isOk = AL_TRUE;
+            }
+
+            if(isOk)
+                InitEffectParams(ALEffect, value);
+            else
+                alSetError(Context, AL_INVALID_VALUE);
+        }
+        else
+        {
+            /* Call the appropriate handler */
+            V(ALEffect,setParami)(Context, param, value);
+        }
+    }
+    UnlockEffectsWrite(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *values)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    switch(param)
+    {
+        case AL_EFFECT_TYPE:
+            alEffecti(effect, param, values[0]);
+            return;
+    }
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsWrite(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        V(ALEffect,setParamiv)(Context, param, values);
+    }
+    UnlockEffectsWrite(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat value)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsWrite(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        V(ALEffect,setParamf)(Context, param, value);
+    }
+    UnlockEffectsWrite(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *values)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsWrite(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        V(ALEffect,setParamfv)(Context, param, values);
+    }
+    UnlockEffectsWrite(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *value)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsRead(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        if(param == AL_EFFECT_TYPE)
+            *value = ALEffect->type;
+        else
+        {
+            /* Call the appropriate handler */
+            V(ALEffect,getParami)(Context, param, value);
+        }
+    }
+    UnlockEffectsRead(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *values)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    switch(param)
+    {
+        case AL_EFFECT_TYPE:
+            alGetEffecti(effect, param, values);
+            return;
+    }
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsRead(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        V(ALEffect,getParamiv)(Context, param, values);
+    }
+    UnlockEffectsRead(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *value)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsRead(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        V(ALEffect,getParamf)(Context, param, value);
+    }
+    UnlockEffectsRead(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *values)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALeffect   *ALEffect;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockEffectsRead(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        V(ALEffect,getParamfv)(Context, param, values);
+    }
+    UnlockEffectsRead(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+
+ALenum InitEffect(ALeffect *effect)
+{
+    InitEffectParams(effect, AL_EFFECT_NULL);
+    return AL_NO_ERROR;
+}
+
+ALvoid ReleaseALEffects(ALCdevice *device)
+{
+    ALsizei i;
+    for(i = 0;i < device->EffectMap.size;i++)
+    {
+        ALeffect *temp = device->EffectMap.values[i];
+        device->EffectMap.values[i] = NULL;
+
+        // Release effect structure
+        FreeThunkEntry(temp->id);
+        memset(temp, 0, sizeof(ALeffect));
+        al_free(temp);
+    }
+}
+
+
+static void InitEffectParams(ALeffect *effect, ALenum type)
+{
+    switch(type)
+    {
+    case AL_EFFECT_EAXREVERB:
+        effect->Props.Reverb.Density   = AL_EAXREVERB_DEFAULT_DENSITY;
+        effect->Props.Reverb.Diffusion = AL_EAXREVERB_DEFAULT_DIFFUSION;
+        effect->Props.Reverb.Gain   = AL_EAXREVERB_DEFAULT_GAIN;
+        effect->Props.Reverb.GainHF = AL_EAXREVERB_DEFAULT_GAINHF;
+        effect->Props.Reverb.GainLF = AL_EAXREVERB_DEFAULT_GAINLF;
+        effect->Props.Reverb.DecayTime    = AL_EAXREVERB_DEFAULT_DECAY_TIME;
+        effect->Props.Reverb.DecayHFRatio = AL_EAXREVERB_DEFAULT_DECAY_HFRATIO;
+        effect->Props.Reverb.DecayLFRatio = AL_EAXREVERB_DEFAULT_DECAY_LFRATIO;
+        effect->Props.Reverb.ReflectionsGain   = AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN;
+        effect->Props.Reverb.ReflectionsDelay  = AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY;
+        effect->Props.Reverb.ReflectionsPan[0] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
+        effect->Props.Reverb.ReflectionsPan[1] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
+        effect->Props.Reverb.ReflectionsPan[2] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
+        effect->Props.Reverb.LateReverbGain   = AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN;
+        effect->Props.Reverb.LateReverbDelay  = AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY;
+        effect->Props.Reverb.LateReverbPan[0] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
+        effect->Props.Reverb.LateReverbPan[1] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
+        effect->Props.Reverb.LateReverbPan[2] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
+        effect->Props.Reverb.EchoTime  = AL_EAXREVERB_DEFAULT_ECHO_TIME;
+        effect->Props.Reverb.EchoDepth = AL_EAXREVERB_DEFAULT_ECHO_DEPTH;
+        effect->Props.Reverb.ModulationTime  = AL_EAXREVERB_DEFAULT_MODULATION_TIME;
+        effect->Props.Reverb.ModulationDepth = AL_EAXREVERB_DEFAULT_MODULATION_DEPTH;
+        effect->Props.Reverb.AirAbsorptionGainHF = AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF;
+        effect->Props.Reverb.HFReference = AL_EAXREVERB_DEFAULT_HFREFERENCE;
+        effect->Props.Reverb.LFReference = AL_EAXREVERB_DEFAULT_LFREFERENCE;
+        effect->Props.Reverb.RoomRolloffFactor = AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR;
+        effect->Props.Reverb.DecayHFLimit = AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT;
+        SET_VTABLE1(ALeaxreverb, effect);
+        break;
+    case AL_EFFECT_REVERB:
+        effect->Props.Reverb.Density   = AL_REVERB_DEFAULT_DENSITY;
+        effect->Props.Reverb.Diffusion = AL_REVERB_DEFAULT_DIFFUSION;
+        effect->Props.Reverb.Gain   = AL_REVERB_DEFAULT_GAIN;
+        effect->Props.Reverb.GainHF = AL_REVERB_DEFAULT_GAINHF;
+        effect->Props.Reverb.GainLF = 1.0f;
+        effect->Props.Reverb.DecayTime    = AL_REVERB_DEFAULT_DECAY_TIME;
+        effect->Props.Reverb.DecayHFRatio = AL_REVERB_DEFAULT_DECAY_HFRATIO;
+        effect->Props.Reverb.DecayLFRatio = 1.0f;
+        effect->Props.Reverb.ReflectionsGain   = AL_REVERB_DEFAULT_REFLECTIONS_GAIN;
+        effect->Props.Reverb.ReflectionsDelay  = AL_REVERB_DEFAULT_REFLECTIONS_DELAY;
+        effect->Props.Reverb.ReflectionsPan[0] = 0.0f;
+        effect->Props.Reverb.ReflectionsPan[1] = 0.0f;
+        effect->Props.Reverb.ReflectionsPan[2] = 0.0f;
+        effect->Props.Reverb.LateReverbGain   = AL_REVERB_DEFAULT_LATE_REVERB_GAIN;
+        effect->Props.Reverb.LateReverbDelay  = AL_REVERB_DEFAULT_LATE_REVERB_DELAY;
+        effect->Props.Reverb.LateReverbPan[0] = 0.0f;
+        effect->Props.Reverb.LateReverbPan[1] = 0.0f;
+        effect->Props.Reverb.LateReverbPan[2] = 0.0f;
+        effect->Props.Reverb.EchoTime  = 0.25f;
+        effect->Props.Reverb.EchoDepth = 0.0f;
+        effect->Props.Reverb.ModulationTime  = 0.25f;
+        effect->Props.Reverb.ModulationDepth = 0.0f;
+        effect->Props.Reverb.AirAbsorptionGainHF = AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF;
+        effect->Props.Reverb.HFReference = 5000.0f;
+        effect->Props.Reverb.LFReference = 250.0f;
+        effect->Props.Reverb.RoomRolloffFactor = AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR;
+        effect->Props.Reverb.DecayHFLimit = AL_REVERB_DEFAULT_DECAY_HFLIMIT;
+        SET_VTABLE1(ALreverb, effect);
+        break;
+    case AL_EFFECT_CHORUS:
+        effect->Props.Chorus.Waveform = AL_CHORUS_DEFAULT_WAVEFORM;
+        effect->Props.Chorus.Phase = AL_CHORUS_DEFAULT_PHASE;
+        effect->Props.Chorus.Rate = AL_CHORUS_DEFAULT_RATE;
+        effect->Props.Chorus.Depth = AL_CHORUS_DEFAULT_DEPTH;
+        effect->Props.Chorus.Feedback = AL_CHORUS_DEFAULT_FEEDBACK;
+        effect->Props.Chorus.Delay = AL_CHORUS_DEFAULT_DELAY;
+        SET_VTABLE1(ALchorus, effect);
+        break;
+    case AL_EFFECT_COMPRESSOR:
+        effect->Props.Compressor.OnOff = AL_COMPRESSOR_DEFAULT_ONOFF;
+        SET_VTABLE1(ALcompressor, effect);
+        break;
+    case AL_EFFECT_DISTORTION:
+        effect->Props.Distortion.Edge = AL_DISTORTION_DEFAULT_EDGE;
+        effect->Props.Distortion.Gain = AL_DISTORTION_DEFAULT_GAIN;
+        effect->Props.Distortion.LowpassCutoff = AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF;
+        effect->Props.Distortion.EQCenter = AL_DISTORTION_DEFAULT_EQCENTER;
+        effect->Props.Distortion.EQBandwidth = AL_DISTORTION_DEFAULT_EQBANDWIDTH;
+        SET_VTABLE1(ALdistortion, effect);
+        break;
+    case AL_EFFECT_ECHO:
+        effect->Props.Echo.Delay    = AL_ECHO_DEFAULT_DELAY;
+        effect->Props.Echo.LRDelay  = AL_ECHO_DEFAULT_LRDELAY;
+        effect->Props.Echo.Damping  = AL_ECHO_DEFAULT_DAMPING;
+        effect->Props.Echo.Feedback = AL_ECHO_DEFAULT_FEEDBACK;
+        effect->Props.Echo.Spread   = AL_ECHO_DEFAULT_SPREAD;
+        SET_VTABLE1(ALecho, effect);
+        break;
+    case AL_EFFECT_EQUALIZER:
+        effect->Props.Equalizer.LowCutoff = AL_EQUALIZER_DEFAULT_LOW_CUTOFF;
+        effect->Props.Equalizer.LowGain = AL_EQUALIZER_DEFAULT_LOW_GAIN;
+        effect->Props.Equalizer.Mid1Center = AL_EQUALIZER_DEFAULT_MID1_CENTER;
+        effect->Props.Equalizer.Mid1Gain = AL_EQUALIZER_DEFAULT_MID1_GAIN;
+        effect->Props.Equalizer.Mid1Width = AL_EQUALIZER_DEFAULT_MID1_WIDTH;
+        effect->Props.Equalizer.Mid2Center = AL_EQUALIZER_DEFAULT_MID2_CENTER;
+        effect->Props.Equalizer.Mid2Gain = AL_EQUALIZER_DEFAULT_MID2_GAIN;
+        effect->Props.Equalizer.Mid2Width = AL_EQUALIZER_DEFAULT_MID2_WIDTH;
+        effect->Props.Equalizer.HighCutoff = AL_EQUALIZER_DEFAULT_HIGH_CUTOFF;
+        effect->Props.Equalizer.HighGain = AL_EQUALIZER_DEFAULT_HIGH_GAIN;
+        SET_VTABLE1(ALequalizer, effect);
+        break;
+    case AL_EFFECT_FLANGER:
+        effect->Props.Flanger.Waveform = AL_FLANGER_DEFAULT_WAVEFORM;
+        effect->Props.Flanger.Phase = AL_FLANGER_DEFAULT_PHASE;
+        effect->Props.Flanger.Rate = AL_FLANGER_DEFAULT_RATE;
+        effect->Props.Flanger.Depth = AL_FLANGER_DEFAULT_DEPTH;
+        effect->Props.Flanger.Feedback = AL_FLANGER_DEFAULT_FEEDBACK;
+        effect->Props.Flanger.Delay = AL_FLANGER_DEFAULT_DELAY;
+        SET_VTABLE1(ALflanger, effect);
+        break;
+    case AL_EFFECT_RING_MODULATOR:
+        effect->Props.Modulator.Frequency      = AL_RING_MODULATOR_DEFAULT_FREQUENCY;
+        effect->Props.Modulator.HighPassCutoff = AL_RING_MODULATOR_DEFAULT_HIGHPASS_CUTOFF;
+        effect->Props.Modulator.Waveform       = AL_RING_MODULATOR_DEFAULT_WAVEFORM;
+        SET_VTABLE1(ALmodulator, effect);
+        break;
+    case AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT:
+    case AL_EFFECT_DEDICATED_DIALOGUE:
+        effect->Props.Dedicated.Gain = 1.0f;
+        SET_VTABLE1(ALdedicated, effect);
+        break;
+    default:
+        SET_VTABLE1(ALnull, effect);
+        break;
+    }
+    effect->type = type;
+}
+
+
+#include "AL/efx-presets.h"
+
+#define DECL(x) { #x, EFX_REVERB_PRESET_##x }
+static const struct {
+    const char name[32];
+    EFXEAXREVERBPROPERTIES props;
+} reverblist[] = {
+    DECL(GENERIC),
+    DECL(PADDEDCELL),
+    DECL(ROOM),
+    DECL(BATHROOM),
+    DECL(LIVINGROOM),
+    DECL(STONEROOM),
+    DECL(AUDITORIUM),
+    DECL(CONCERTHALL),
+    DECL(CAVE),
+    DECL(ARENA),
+    DECL(HANGAR),
+    DECL(CARPETEDHALLWAY),
+    DECL(HALLWAY),
+    DECL(STONECORRIDOR),
+    DECL(ALLEY),
+    DECL(FOREST),
+    DECL(CITY),
+    DECL(MOUNTAINS),
+    DECL(QUARRY),
+    DECL(PLAIN),
+    DECL(PARKINGLOT),
+    DECL(SEWERPIPE),
+    DECL(UNDERWATER),
+    DECL(DRUGGED),
+    DECL(DIZZY),
+    DECL(PSYCHOTIC),
+
+    DECL(CASTLE_SMALLROOM),
+    DECL(CASTLE_SHORTPASSAGE),
+    DECL(CASTLE_MEDIUMROOM),
+    DECL(CASTLE_LARGEROOM),
+    DECL(CASTLE_LONGPASSAGE),
+    DECL(CASTLE_HALL),
+    DECL(CASTLE_CUPBOARD),
+    DECL(CASTLE_COURTYARD),
+    DECL(CASTLE_ALCOVE),
+
+    DECL(FACTORY_SMALLROOM),
+    DECL(FACTORY_SHORTPASSAGE),
+    DECL(FACTORY_MEDIUMROOM),
+    DECL(FACTORY_LARGEROOM),
+    DECL(FACTORY_LONGPASSAGE),
+    DECL(FACTORY_HALL),
+    DECL(FACTORY_CUPBOARD),
+    DECL(FACTORY_COURTYARD),
+    DECL(FACTORY_ALCOVE),
+
+    DECL(ICEPALACE_SMALLROOM),
+    DECL(ICEPALACE_SHORTPASSAGE),
+    DECL(ICEPALACE_MEDIUMROOM),
+    DECL(ICEPALACE_LARGEROOM),
+    DECL(ICEPALACE_LONGPASSAGE),
+    DECL(ICEPALACE_HALL),
+    DECL(ICEPALACE_CUPBOARD),
+    DECL(ICEPALACE_COURTYARD),
+    DECL(ICEPALACE_ALCOVE),
+
+    DECL(SPACESTATION_SMALLROOM),
+    DECL(SPACESTATION_SHORTPASSAGE),
+    DECL(SPACESTATION_MEDIUMROOM),
+    DECL(SPACESTATION_LARGEROOM),
+    DECL(SPACESTATION_LONGPASSAGE),
+    DECL(SPACESTATION_HALL),
+    DECL(SPACESTATION_CUPBOARD),
+    DECL(SPACESTATION_ALCOVE),
+
+    DECL(WOODEN_SMALLROOM),
+    DECL(WOODEN_SHORTPASSAGE),
+    DECL(WOODEN_MEDIUMROOM),
+    DECL(WOODEN_LARGEROOM),
+    DECL(WOODEN_LONGPASSAGE),
+    DECL(WOODEN_HALL),
+    DECL(WOODEN_CUPBOARD),
+    DECL(WOODEN_COURTYARD),
+    DECL(WOODEN_ALCOVE),
+
+    DECL(SPORT_EMPTYSTADIUM),
+    DECL(SPORT_SQUASHCOURT),
+    DECL(SPORT_SMALLSWIMMINGPOOL),
+    DECL(SPORT_LARGESWIMMINGPOOL),
+    DECL(SPORT_GYMNASIUM),
+    DECL(SPORT_FULLSTADIUM),
+    DECL(SPORT_STADIUMTANNOY),
+
+    DECL(PREFAB_WORKSHOP),
+    DECL(PREFAB_SCHOOLROOM),
+    DECL(PREFAB_PRACTISEROOM),
+    DECL(PREFAB_OUTHOUSE),
+    DECL(PREFAB_CARAVAN),
+
+    DECL(DOME_TOMB),
+    DECL(PIPE_SMALL),
+    DECL(DOME_SAINTPAULS),
+    DECL(PIPE_LONGTHIN),
+    DECL(PIPE_LARGE),
+    DECL(PIPE_RESONANT),
+
+    DECL(OUTDOORS_BACKYARD),
+    DECL(OUTDOORS_ROLLINGPLAINS),
+    DECL(OUTDOORS_DEEPCANYON),
+    DECL(OUTDOORS_CREEK),
+    DECL(OUTDOORS_VALLEY),
+
+    DECL(MOOD_HEAVEN),
+    DECL(MOOD_HELL),
+    DECL(MOOD_MEMORY),
+
+    DECL(DRIVING_COMMENTATOR),
+    DECL(DRIVING_PITGARAGE),
+    DECL(DRIVING_INCAR_RACER),
+    DECL(DRIVING_INCAR_SPORTS),
+    DECL(DRIVING_INCAR_LUXURY),
+    DECL(DRIVING_FULLGRANDSTAND),
+    DECL(DRIVING_EMPTYGRANDSTAND),
+    DECL(DRIVING_TUNNEL),
+
+    DECL(CITY_STREETS),
+    DECL(CITY_SUBWAY),
+    DECL(CITY_MUSEUM),
+    DECL(CITY_LIBRARY),
+    DECL(CITY_UNDERPASS),
+    DECL(CITY_ABANDONED),
+
+    DECL(DUSTYROOM),
+    DECL(CHAPEL),
+    DECL(SMALLWATERROOM),
+};
+#undef DECL
+
+ALvoid LoadReverbPreset(const char *name, ALeffect *effect)
+{
+    size_t i;
+
+    if(strcasecmp(name, "NONE") == 0)
+    {
+        InitEffectParams(effect, AL_EFFECT_NULL);
+        TRACE("Loading reverb '%s'\n", "NONE");
+        return;
+    }
+
+    if(!DisabledEffects[AL__EAXREVERB])
+        InitEffectParams(effect, AL_EFFECT_EAXREVERB);
+    else if(!DisabledEffects[AL__REVERB])
+        InitEffectParams(effect, AL_EFFECT_REVERB);
+    else
+        InitEffectParams(effect, AL_EFFECT_NULL);
+    for(i = 0;i < COUNTOF(reverblist);i++)
+    {
+        const EFXEAXREVERBPROPERTIES *props;
+
+        if(strcasecmp(name, reverblist[i].name) != 0)
+            continue;
+
+        TRACE("Loading reverb '%s'\n", reverblist[i].name);
+        props = &reverblist[i].props;
+        effect->Props.Reverb.Density   = props->flDensity;
+        effect->Props.Reverb.Diffusion = props->flDiffusion;
+        effect->Props.Reverb.Gain   = props->flGain;
+        effect->Props.Reverb.GainHF = props->flGainHF;
+        effect->Props.Reverb.GainLF = props->flGainLF;
+        effect->Props.Reverb.DecayTime    = props->flDecayTime;
+        effect->Props.Reverb.DecayHFRatio = props->flDecayHFRatio;
+        effect->Props.Reverb.DecayLFRatio = props->flDecayLFRatio;
+        effect->Props.Reverb.ReflectionsGain   = props->flReflectionsGain;
+        effect->Props.Reverb.ReflectionsDelay  = props->flReflectionsDelay;
+        effect->Props.Reverb.ReflectionsPan[0] = props->flReflectionsPan[0];
+        effect->Props.Reverb.ReflectionsPan[1] = props->flReflectionsPan[1];
+        effect->Props.Reverb.ReflectionsPan[2] = props->flReflectionsPan[2];
+        effect->Props.Reverb.LateReverbGain   = props->flLateReverbGain;
+        effect->Props.Reverb.LateReverbDelay  = props->flLateReverbDelay;
+        effect->Props.Reverb.LateReverbPan[0] = props->flLateReverbPan[0];
+        effect->Props.Reverb.LateReverbPan[1] = props->flLateReverbPan[1];
+        effect->Props.Reverb.LateReverbPan[2] = props->flLateReverbPan[2];
+        effect->Props.Reverb.EchoTime  = props->flEchoTime;
+        effect->Props.Reverb.EchoDepth = props->flEchoDepth;
+        effect->Props.Reverb.ModulationTime  = props->flModulationTime;
+        effect->Props.Reverb.ModulationDepth = props->flModulationDepth;
+        effect->Props.Reverb.AirAbsorptionGainHF = props->flAirAbsorptionGainHF;
+        effect->Props.Reverb.HFReference = props->flHFReference;
+        effect->Props.Reverb.LFReference = props->flLFReference;
+        effect->Props.Reverb.RoomRolloffFactor = props->flRoomRolloffFactor;
+        effect->Props.Reverb.DecayHFLimit = props->iDecayHFLimit;
+        return;
+    }
+
+    WARN("Reverb preset '%s' not found\n", name);
+}

OpenAL32/alError.c

@@ -0,0 +1,82 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2000 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 <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"
+
+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 = 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 = ATOMIC_EXCHANGE_SEQ(&Context->LastError, AL_NO_ERROR);
+
+    ALCcontext_DecRef(Context);
+
+    return errorCode;
+}

OpenAL32/alExtension.c

@@ -0,0 +1,103 @@
+/**
+ * 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 <string.h>
+#include <ctype.h>
+
+#include "alError.h"
+#include "alMain.h"
+#include "alFilter.h"
+#include "alEffect.h"
+#include "alAuxEffectSlot.h"
+#include "alSource.h"
+#include "alBuffer.h"
+#include "AL/al.h"
+#include "AL/alc.h"
+
+
+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 }
+};
+
+
+AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName)
+{
+    ALboolean ret = AL_FALSE;
+    ALCcontext *context;
+    const char *ptr;
+    size_t len;
+
+    context = GetContextRef();
+    if(!context) return AL_FALSE;
+
+    if(!(extName))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    len = strlen(extName);
+    ptr = context->ExtensionList;
+    while(ptr && *ptr)
+    {
+        if(strncasecmp(ptr, extName, len) == 0 &&
+           (ptr[len] == '\0' || isspace(ptr[len])))
+        {
+            ret = AL_TRUE;
+            break;
+        }
+        if((ptr=strchr(ptr, ' ')) != NULL)
+        {
+            do {
+                ++ptr;
+            } while(isspace(*ptr));
+        }
+    }
+
+done:
+    ALCcontext_DecRef(context);
+    return ret;
+}
+
+
+AL_API ALvoid* AL_APIENTRY alGetProcAddress(const ALchar *funcName)
+{
+    if(!funcName)
+        return NULL;
+    return alcGetProcAddress(NULL, funcName);
+}
+
+AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *enumName)
+{
+    if(!enumName)
+        return (ALenum)0;
+    return alcGetEnumValue(NULL, enumName);
+}

OpenAL32/alFilter.c

@@ -0,0 +1,719 @@
+/**
+ * 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 "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);
+
+
+AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
+{
+    ALCdevice *device;
+    ALCcontext *context;
+    ALsizei cur = 0;
+    ALenum err;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    device = context->Device;
+    for(cur = 0;cur < n;cur++)
+    {
+        ALfilter *filter = al_calloc(16, sizeof(ALfilter));
+        if(!filter)
+        {
+            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;
+    }
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters)
+{
+    ALCdevice *device;
+    ALCcontext *context;
+    ALfilter *filter;
+    ALsizei i;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    device = context->Device;
+    LockFiltersWrite(device);
+    if(!(n >= 0))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    for(i = 0;i < n;i++)
+    {
+        if(filters[i] && LookupFilter(device, filters[i]) == NULL)
+            SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
+    }
+    for(i = 0;i < n;i++)
+    {
+        if((filter=RemoveFilter(device, filters[i])) == NULL)
+            continue;
+        FreeThunkEntry(filter->id);
+
+        memset(filter, 0, sizeof(*filter));
+        al_free(filter);
+    }
+
+done:
+    UnlockFiltersWrite(device);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter)
+{
+    ALCcontext *Context;
+    ALboolean  result;
+
+    Context = GetContextRef();
+    if(!Context) return AL_FALSE;
+
+    LockFiltersRead(Context->Device);
+    result = ((!filter || LookupFilter(Context->Device, filter)) ?
+              AL_TRUE : AL_FALSE);
+    UnlockFiltersRead(Context->Device);
+
+    ALCcontext_DecRef(Context);
+
+    return result;
+}
+
+AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint value)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        if(param == AL_FILTER_TYPE)
+        {
+            if(value == AL_FILTER_NULL || value == AL_FILTER_LOWPASS ||
+               value == AL_FILTER_HIGHPASS || value == AL_FILTER_BANDPASS)
+                InitFilterParams(ALFilter, value);
+            else
+                alSetError(Context, AL_INVALID_VALUE);
+        }
+        else
+        {
+            /* Call the appropriate handler */
+            ALfilter_SetParami(ALFilter, Context, param, value);
+        }
+    }
+    UnlockFiltersWrite(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *values)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
+
+    switch(param)
+    {
+        case AL_FILTER_TYPE:
+            alFilteri(filter, param, values[0]);
+            return;
+    }
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_SetParamiv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersWrite(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat value)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_SetParamf(ALFilter, Context, param, value);
+    }
+    UnlockFiltersWrite(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *values)
+{
+    ALCcontext *Context;
+    ALCdevice  *Device;
+    ALfilter   *ALFilter;
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockFiltersWrite(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_SetParamfv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersWrite(Device);
+
+    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:
+            alGetFilteri(filter, param, values);
+            return;
+    }
+
+    Context = GetContextRef();
+    if(!Context) return;
+
+    Device = Context->Device;
+    LockFiltersRead(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_GetParamiv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersRead(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *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
+    {
+        /* Call the appropriate handler */
+        ALfilter_GetParamf(ALFilter, Context, param, value);
+    }
+    UnlockFiltersRead(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *values)
+{
+    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
+    {
+        /* Call the appropriate handler */
+        ALfilter_GetParamfv(ALFilter, Context, param, values);
+    }
+    UnlockFiltersRead(Device);
+
+    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)
+{
+    ALsizei i;
+    for(i = 0;i < device->FilterMap.size;i++)
+    {
+        ALfilter *temp = device->FilterMap.values[i];
+        device->FilterMap.values[i] = NULL;
+
+        // Release filter structure
+        FreeThunkEntry(temp->id);
+        memset(temp, 0, sizeof(ALfilter));
+        al_free(temp);
+    }
+}
+
+
+static void InitFilterParams(ALfilter *filter, ALenum 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->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

@@ -0,0 +1,510 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2000 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 "alMain.h"
+#include "AL/alc.h"
+#include "alError.h"
+#include "alListener.h"
+#include "alSource.h"
+
+AL_API ALvoid AL_APIENTRY alListenerf(ALenum param, ALfloat value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    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:
+        case AL_METERS_PER_UNIT:
+            alListenerf(param, values[0]);
+            return;
+
+        case AL_POSITION:
+        case AL_VELOCITY:
+            alListener3f(param, values[0], values[1], values[2]);
+            return;
+        }
+    }
+
+    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);
+}
+
+
+AL_API ALvoid AL_APIENTRY alListeneri(ALenum param, ALint UNUSED(value))
+{
+    ALCcontext *context;
+
+    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 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_POSITION:
+        case AL_VELOCITY:
+            alListener3f(param, (ALfloat)values[0], (ALfloat)values[1], (ALfloat)values[2]);
+            return;
+
+        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;
+        }
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    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 ALvoid AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    ReadLock(&context->PropLock);
+    if(!(value))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
+    {
+    case AL_GAIN:
+        *value = context->Listener->Gain;
+        break;
+
+    case AL_METERS_PER_UNIT:
+        *value = context->Listener->MetersPerUnit;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ReadUnlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alGetListener3f(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *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 = context->Listener->Position[0];
+        *value2 = context->Listener->Position[1];
+        *value3 = context->Listener->Position[2];
+        break;
+
+    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);
+}
+
+
+AL_API ALvoid AL_APIENTRY alGetListenerfv(ALenum param, ALfloat *values)
+{
+    ALCcontext *context;
+
+    switch(param)
+    {
+    case AL_GAIN:
+    case AL_METERS_PER_UNIT:
+        alGetListenerf(param, values);
+        return;
+
+    case AL_POSITION:
+    case AL_VELOCITY:
+        alGetListener3f(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] = 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);
+}
+
+
+AL_API ALvoid AL_APIENTRY alGetListeneri(ALenum param, ALint *value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    ReadLock(&context->PropLock);
+    if(!(value))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(param)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+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);
+    }
+}

OpenAL32/alState.c

@@ -0,0 +1,779 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2000 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 "version.h"
+
+#include <stdlib.h>
+#include "alMain.h"
+#include "AL/alc.h"
+#include "AL/al.h"
+#include "AL/alext.h"
+#include "alError.h"
+#include "alListener.h"
+#include "alSource.h"
+#include "alAuxEffectSlot.h"
+
+#include "backends/base.h"
+
+
+static const ALchar alVendor[] = "OpenAL Community";
+static const ALchar alVersion[] = "1.1 ALSOFT "ALSOFT_VERSION;
+static const ALchar alRenderer[] = "OpenAL Soft";
+
+// Error Messages
+static const ALchar alNoError[] = "No Error";
+static const ALchar alErrInvalidName[] = "Invalid Name";
+static const ALchar alErrInvalidEnum[] = "Invalid Enum";
+static const ALchar alErrInvalidValue[] = "Invalid Value";
+static const ALchar alErrInvalidOp[] = "Invalid Operation";
+static const ALchar alErrOutOfMemory[] = "Out of Memory";
+
+/* Resampler strings */
+static const ALchar alPointResampler[] = "Nearest";
+static const ALchar alLinearResampler[] = "Linear";
+static const ALchar alSinc4Resampler[] = "4-Point Sinc";
+static const ALchar alBSincResampler[] = "Band-limited Sinc (12/24)";
+
+AL_API ALvoid AL_APIENTRY alEnable(ALenum capability)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    switch(capability)
+    {
+    case AL_SOURCE_DISTANCE_MODEL:
+        context->SourceDistanceModel = AL_TRUE;
+        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 alDisable(ALenum capability)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    WriteLock(&context->PropLock);
+    switch(capability)
+    {
+    case AL_SOURCE_DISTANCE_MODEL:
+        context->SourceDistanceModel = AL_FALSE;
+        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 ALboolean AL_APIENTRY alIsEnabled(ALenum capability)
+{
+    ALCcontext *context;
+    ALboolean value=AL_FALSE;
+
+    context = GetContextRef();
+    if(!context) return AL_FALSE;
+
+    switch(capability)
+    {
+    case AL_SOURCE_DISTANCE_MODEL:
+        value = context->SourceDistanceModel;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}
+
+AL_API ALboolean AL_APIENTRY alGetBoolean(ALenum pname)
+{
+    ALCcontext *context;
+    ALboolean value=AL_FALSE;
+
+    context = GetContextRef();
+    if(!context) return AL_FALSE;
+
+    switch(pname)
+    {
+    case AL_DOPPLER_FACTOR:
+        if(context->DopplerFactor != 0.0f)
+            value = AL_TRUE;
+        break;
+
+    case AL_DOPPLER_VELOCITY:
+        if(context->DopplerVelocity != 0.0f)
+            value = AL_TRUE;
+        break;
+
+    case AL_DISTANCE_MODEL:
+        if(context->DistanceModel == AL_INVERSE_DISTANCE_CLAMPED)
+            value = AL_TRUE;
+        break;
+
+    case AL_SPEED_OF_SOUND:
+        if(context->SpeedOfSound != 0.0f)
+            value = AL_TRUE;
+        break;
+
+    case AL_DEFERRED_UPDATES_SOFT:
+        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+            value = AL_TRUE;
+        break;
+
+    case AL_GAIN_LIMIT_SOFT:
+        if(GAIN_MIX_MAX/context->GainBoost != 0.0f)
+            value = AL_TRUE;
+        break;
+
+    case AL_NUM_RESAMPLERS_SOFT:
+        /* Always non-0. */
+        value = AL_TRUE;
+        break;
+
+    case AL_DEFAULT_RESAMPLER_SOFT:
+        value = ResamplerDefault ? AL_TRUE : AL_FALSE;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}
+
+AL_API ALdouble AL_APIENTRY alGetDouble(ALenum pname)
+{
+    ALCcontext *context;
+    ALdouble value = 0.0;
+
+    context = GetContextRef();
+    if(!context) return 0.0;
+
+    switch(pname)
+    {
+    case AL_DOPPLER_FACTOR:
+        value = (ALdouble)context->DopplerFactor;
+        break;
+
+    case AL_DOPPLER_VELOCITY:
+        value = (ALdouble)context->DopplerVelocity;
+        break;
+
+    case AL_DISTANCE_MODEL:
+        value = (ALdouble)context->DistanceModel;
+        break;
+
+    case AL_SPEED_OF_SOUND:
+        value = (ALdouble)context->SpeedOfSound;
+        break;
+
+    case AL_DEFERRED_UPDATES_SOFT:
+        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+            value = (ALdouble)AL_TRUE;
+        break;
+
+    case AL_GAIN_LIMIT_SOFT:
+        value = (ALdouble)GAIN_MIX_MAX/context->GainBoost;
+        break;
+
+    case AL_NUM_RESAMPLERS_SOFT:
+        value = (ALdouble)(ResamplerMax + 1);
+        break;
+
+    case AL_DEFAULT_RESAMPLER_SOFT:
+        value = (ALdouble)ResamplerDefault;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}
+
+AL_API ALfloat AL_APIENTRY alGetFloat(ALenum pname)
+{
+    ALCcontext *context;
+    ALfloat value = 0.0f;
+
+    context = GetContextRef();
+    if(!context) return 0.0f;
+
+    switch(pname)
+    {
+    case AL_DOPPLER_FACTOR:
+        value = context->DopplerFactor;
+        break;
+
+    case AL_DOPPLER_VELOCITY:
+        value = context->DopplerVelocity;
+        break;
+
+    case AL_DISTANCE_MODEL:
+        value = (ALfloat)context->DistanceModel;
+        break;
+
+    case AL_SPEED_OF_SOUND:
+        value = context->SpeedOfSound;
+        break;
+
+    case AL_DEFERRED_UPDATES_SOFT:
+        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+            value = (ALfloat)AL_TRUE;
+        break;
+
+    case AL_GAIN_LIMIT_SOFT:
+        value = GAIN_MIX_MAX/context->GainBoost;
+        break;
+
+    case AL_NUM_RESAMPLERS_SOFT:
+        value = (ALfloat)(ResamplerMax + 1);
+        break;
+
+    case AL_DEFAULT_RESAMPLER_SOFT:
+        value = (ALfloat)ResamplerDefault;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}
+
+AL_API ALint AL_APIENTRY alGetInteger(ALenum pname)
+{
+    ALCcontext *context;
+    ALint value = 0;
+
+    context = GetContextRef();
+    if(!context) return 0;
+
+    switch(pname)
+    {
+    case AL_DOPPLER_FACTOR:
+        value = (ALint)context->DopplerFactor;
+        break;
+
+    case AL_DOPPLER_VELOCITY:
+        value = (ALint)context->DopplerVelocity;
+        break;
+
+    case AL_DISTANCE_MODEL:
+        value = (ALint)context->DistanceModel;
+        break;
+
+    case AL_SPEED_OF_SOUND:
+        value = (ALint)context->SpeedOfSound;
+        break;
+
+    case AL_DEFERRED_UPDATES_SOFT:
+        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+            value = (ALint)AL_TRUE;
+        break;
+
+    case AL_GAIN_LIMIT_SOFT:
+        value = (ALint)(GAIN_MIX_MAX/context->GainBoost);
+        break;
+
+    case AL_NUM_RESAMPLERS_SOFT:
+        value = ResamplerMax + 1;
+        break;
+
+    case AL_DEFAULT_RESAMPLER_SOFT:
+        value = ResamplerDefault;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}
+
+AL_API ALint64SOFT AL_APIENTRY alGetInteger64SOFT(ALenum pname)
+{
+    ALCcontext *context;
+    ALint64SOFT value = 0;
+
+    context = GetContextRef();
+    if(!context) return 0;
+
+    switch(pname)
+    {
+    case AL_DOPPLER_FACTOR:
+        value = (ALint64SOFT)context->DopplerFactor;
+        break;
+
+    case AL_DOPPLER_VELOCITY:
+        value = (ALint64SOFT)context->DopplerVelocity;
+        break;
+
+    case AL_DISTANCE_MODEL:
+        value = (ALint64SOFT)context->DistanceModel;
+        break;
+
+    case AL_SPEED_OF_SOUND:
+        value = (ALint64SOFT)context->SpeedOfSound;
+        break;
+
+    case AL_DEFERRED_UPDATES_SOFT:
+        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+            value = (ALint64SOFT)AL_TRUE;
+        break;
+
+    case AL_GAIN_LIMIT_SOFT:
+        value = (ALint64SOFT)(GAIN_MIX_MAX/context->GainBoost);
+        break;
+
+    case AL_NUM_RESAMPLERS_SOFT:
+        value = (ALint64SOFT)(ResamplerMax + 1);
+        break;
+
+    case AL_DEFAULT_RESAMPLER_SOFT:
+        value = (ALint64SOFT)ResamplerDefault;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}
+
+AL_API ALvoid AL_APIENTRY alGetBooleanv(ALenum pname, ALboolean *values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        switch(pname)
+        {
+            case AL_DOPPLER_FACTOR:
+            case AL_DOPPLER_VELOCITY:
+            case AL_DISTANCE_MODEL:
+            case AL_SPEED_OF_SOUND:
+            case AL_DEFERRED_UPDATES_SOFT:
+            case AL_GAIN_LIMIT_SOFT:
+            case AL_NUM_RESAMPLERS_SOFT:
+            case AL_DEFAULT_RESAMPLER_SOFT:
+                values[0] = alGetBoolean(pname);
+                return;
+        }
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(pname)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetDoublev(ALenum pname, ALdouble *values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        switch(pname)
+        {
+            case AL_DOPPLER_FACTOR:
+            case AL_DOPPLER_VELOCITY:
+            case AL_DISTANCE_MODEL:
+            case AL_SPEED_OF_SOUND:
+            case AL_DEFERRED_UPDATES_SOFT:
+            case AL_GAIN_LIMIT_SOFT:
+            case AL_NUM_RESAMPLERS_SOFT:
+            case AL_DEFAULT_RESAMPLER_SOFT:
+                values[0] = alGetDouble(pname);
+                return;
+        }
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(pname)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetFloatv(ALenum pname, ALfloat *values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        switch(pname)
+        {
+            case AL_DOPPLER_FACTOR:
+            case AL_DOPPLER_VELOCITY:
+            case AL_DISTANCE_MODEL:
+            case AL_SPEED_OF_SOUND:
+            case AL_DEFERRED_UPDATES_SOFT:
+            case AL_GAIN_LIMIT_SOFT:
+            case AL_NUM_RESAMPLERS_SOFT:
+            case AL_DEFAULT_RESAMPLER_SOFT:
+                values[0] = alGetFloat(pname);
+                return;
+        }
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(values))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+    switch(pname)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alGetIntegerv(ALenum pname, ALint *values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        switch(pname)
+        {
+            case AL_DOPPLER_FACTOR:
+            case AL_DOPPLER_VELOCITY:
+            case AL_DISTANCE_MODEL:
+            case AL_SPEED_OF_SOUND:
+            case AL_DEFERRED_UPDATES_SOFT:
+            case AL_GAIN_LIMIT_SOFT:
+            case AL_NUM_RESAMPLERS_SOFT:
+            case AL_DEFAULT_RESAMPLER_SOFT:
+                values[0] = alGetInteger(pname);
+                return;
+        }
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    switch(pname)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API void AL_APIENTRY alGetInteger64vSOFT(ALenum pname, ALint64SOFT *values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        switch(pname)
+        {
+            case AL_DOPPLER_FACTOR:
+            case AL_DOPPLER_VELOCITY:
+            case AL_DISTANCE_MODEL:
+            case AL_SPEED_OF_SOUND:
+            case AL_DEFERRED_UPDATES_SOFT:
+            case AL_GAIN_LIMIT_SOFT:
+            case AL_NUM_RESAMPLERS_SOFT:
+            case AL_DEFAULT_RESAMPLER_SOFT:
+                values[0] = alGetInteger64SOFT(pname);
+                return;
+        }
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    switch(pname)
+    {
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API const ALchar* AL_APIENTRY alGetString(ALenum pname)
+{
+    const ALchar *value = NULL;
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return NULL;
+
+    switch(pname)
+    {
+    case AL_VENDOR:
+        value = alVendor;
+        break;
+
+    case AL_VERSION:
+        value = alVersion;
+        break;
+
+    case AL_RENDERER:
+        value = alRenderer;
+        break;
+
+    case AL_EXTENSIONS:
+        value = context->ExtensionList;
+        break;
+
+    case AL_NO_ERROR:
+        value = alNoError;
+        break;
+
+    case AL_INVALID_NAME:
+        value = alErrInvalidName;
+        break;
+
+    case AL_INVALID_ENUM:
+        value = alErrInvalidEnum;
+        break;
+
+    case AL_INVALID_VALUE:
+        value = alErrInvalidValue;
+        break;
+
+    case AL_INVALID_OPERATION:
+        value = alErrInvalidOp;
+        break;
+
+    case AL_OUT_OF_MEMORY:
+        value = alErrOutOfMemory;
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}
+
+AL_API ALvoid AL_APIENTRY alDopplerFactor(ALfloat value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(value >= 0.0f && isfinite(value)))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    WriteLock(&context->PropLock);
+    context->DopplerFactor = value;
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+    WriteUnlock(&context->PropLock);
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alDopplerVelocity(ALfloat value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(value >= 0.0f && isfinite(value)))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    WriteLock(&context->PropLock);
+    context->DopplerVelocity = value;
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+    WriteUnlock(&context->PropLock);
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alSpeedOfSound(ALfloat value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(value > 0.0f && isfinite(value)))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    WriteLock(&context->PropLock);
+    context->SpeedOfSound = value;
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+        UpdateListenerProps(context);
+    WriteUnlock(&context->PropLock);
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alDistanceModel(ALenum value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(value == AL_INVERSE_DISTANCE || value == AL_INVERSE_DISTANCE_CLAMPED ||
+         value == AL_LINEAR_DISTANCE || value == AL_LINEAR_DISTANCE_CLAMPED ||
+         value == AL_EXPONENT_DISTANCE || value == AL_EXPONENT_DISTANCE_CLAMPED ||
+         value == AL_NONE))
+        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+
+    WriteLock(&context->PropLock);
+    context->DistanceModel = value;
+    if(!context->SourceDistanceModel)
+    {
+        if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+            UpdateListenerProps(context);
+    }
+    WriteUnlock(&context->PropLock);
+
+done:
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alDeferUpdatesSOFT(void)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    ALCcontext_DeferUpdates(context);
+
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alProcessUpdatesSOFT(void)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    ALCcontext_ProcessUpdates(context);
+
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API const ALchar* AL_APIENTRY alGetStringiSOFT(ALenum pname, ALsizei index)
+{
+    const char *ResamplerNames[] = {
+        alPointResampler, alLinearResampler,
+        alSinc4Resampler, alBSincResampler,
+    };
+    const ALchar *value = NULL;
+    ALCcontext *context;
+
+    static_assert(COUNTOF(ResamplerNames) == ResamplerMax+1, "Incorrect ResamplerNames list");
+
+    context = GetContextRef();
+    if(!context) return NULL;
+
+    switch(pname)
+    {
+    case AL_RESAMPLER_NAME_SOFT:
+        if(index < 0 || (size_t)index >= COUNTOF(ResamplerNames))
+            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
+        value = ResamplerNames[index];
+        break;
+
+    default:
+        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
+    }
+
+done:
+    ALCcontext_DecRef(context);
+
+    return value;
+}

OpenAL32/alThunk.c

@@ -0,0 +1,108 @@
+/**
+ * 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 "alMain.h"
+#include "alThunk.h"
+
+#include "almalloc.h"
+
+
+static ATOMIC_FLAG *ThunkArray;
+static ALsizei      ThunkArraySize;
+static RWLock ThunkLock;
+
+void ThunkInit(void)
+{
+    RWLockInit(&ThunkLock);
+    ThunkArraySize = 1024;
+    ThunkArray = al_calloc(16, ThunkArraySize * sizeof(*ThunkArray));
+}
+
+void ThunkExit(void)
+{
+    al_free(ThunkArray);
+    ThunkArray = NULL;
+    ThunkArraySize = 0;
+}
+
+ALenum NewThunkEntry(ALuint *index)
+{
+    void *NewList;
+    ALsizei i;
+
+    ReadLock(&ThunkLock);
+    for(i = 0;i < ThunkArraySize;i++)
+    {
+        if(!ATOMIC_FLAG_TEST_AND_SET(&ThunkArray[i], almemory_order_acq_rel))
+        {
+            ReadUnlock(&ThunkLock);
+            *index = i+1;
+            return AL_NO_ERROR;
+        }
+    }
+    ReadUnlock(&ThunkLock);
+
+    WriteLock(&ThunkLock);
+    /* Double-check that there's still no free entries, in case another
+     * invocation just came through and increased the size of the array.
+     */
+    for(;i < ThunkArraySize;i++)
+    {
+        if(!ATOMIC_FLAG_TEST_AND_SET(&ThunkArray[i], almemory_order_acq_rel))
+        {
+            WriteUnlock(&ThunkLock);
+            *index = i+1;
+            return AL_NO_ERROR;
+        }
+    }
+
+    NewList = al_calloc(16, ThunkArraySize*2 * sizeof(*ThunkArray));
+    if(!NewList)
+    {
+        WriteUnlock(&ThunkLock);
+        ERR("Realloc failed to increase to %u entries!\n", ThunkArraySize*2);
+        return AL_OUT_OF_MEMORY;
+    }
+    memcpy(NewList, ThunkArray, ThunkArraySize*sizeof(*ThunkArray));
+    al_free(ThunkArray);
+    ThunkArray = NewList;
+    ThunkArraySize *= 2;
+
+    ATOMIC_FLAG_TEST_AND_SET(&ThunkArray[i], almemory_order_seq_cst);
+    *index = ++i;
+
+    for(;i < ThunkArraySize;i++)
+        ATOMIC_FLAG_CLEAR(&ThunkArray[i], almemory_order_relaxed);
+    WriteUnlock(&ThunkLock);
+
+    return AL_NO_ERROR;
+}
+
+void FreeThunkEntry(ALuint index)
+{
+    ReadLock(&ThunkLock);
+    if(index > 0 && (ALsizei)index <= ThunkArraySize)
+        ATOMIC_FLAG_CLEAR(&ThunkArray[index-1], almemory_order_release);
+    ReadUnlock(&ThunkLock);
+}

README

@@ -0,0 +1,55 @@
+Source Install
+==============
+
+To install OpenAL Soft, use your favorite shell to go into the build/
+directory, and run:
+
+cmake ..
+
+Assuming configuration went well, you can then build it, typically using GNU
+Make (KDevelop, MSVC, and others are possible depending on your system setup
+and CMake configuration).
+
+Please Note: Double check that the appropriate backends were detected. Often,
+complaints of no sound, crashing, and missing devices can be solved by making
+sure the correct backends are being used. CMake's output will identify which
+backends were enabled.
+
+For most systems, you will likely want to make sure ALSA, OSS, and PulseAudio
+were detected (if your target system uses them). For Windows, make sure
+DirectSound was detected.
+
+
+Utilities
+=========
+
+The source package comes with an informational utility, openal-info, and is
+built by default. It prints out information provided by the ALC and AL sub-
+systems, including discovered devices, version information, and extensions.
+
+
+Configuration
+=============
+
+OpenAL Soft can be configured on a per-user and per-system basis. This allows
+users and sysadmins to control information provided to applications, as well
+as application-agnostic behavior of the library. See alsoftrc.sample for
+available settings.
+
+
+Acknowledgements
+================
+
+Special thanks go to:
+
+Creative Labs for the original source code this is based off of.
+
+Christopher Fitzgerald for the current reverb effect implementation, and
+helping with the low-pass and HRTF filters.
+
+Christian Borss for the 3D panning code previous versions used as a base.
+
+Ben Davis for the idea behind a previous version of the click-removal code.
+
+Richard Furse for helping with my understanding of Ambisonics that is used by
+the various parts of the library.

README.md

@@ -1,16 +0,0 @@
-# openal-soft-vita
-
-This repo contains patchsets for vita [openal-soft](https://openal-soft.org/) port.
-
-Format is openal-soft-<openal_version>-vita-<patchset_version>.patch
-
-For full source see vita-* branches
-
-
-# building
-
-```
-mkdir build && cd build
-cmake -DCMAKE_TOOLCHAIN_FILE=${VITASDK}/share/vita.toolchain.cmake -DCMAKE_BUILD_TYPE=Release ..
-make install
-```

XCompile-Android.txt

@@ -0,0 +1,39 @@
+# Cross-compiling requires CMake 2.6 or newer. Example:
+# cmake .. -DCMAKE_TOOLCHAIN_FILE=../XCompile-Android.txt -DHOST=arm-linux-androideabi
+# Where 'arm-linux-androideabi' is the host prefix for the cross-compiler. If
+# you already have a toolchain file setup, you may use that instead of this
+# file. Make sure to set CMAKE_FIND_ROOT_PATH to where the NDK toolchain was
+# installed (e.g. "$ENV{HOME}/toolchains/arm-linux-androideabi-r10c-21").
+
+# the name of the target operating system
+SET(CMAKE_SYSTEM_NAME Linux)
+
+# which compilers to use for C and C++
+SET(CMAKE_C_COMPILER "${HOST}-gcc")
+SET(CMAKE_CXX_COMPILER "${HOST}-g++")
+SET(CMAKE_RC_COMPILER "${HOST}-windres")
+
+# here is the target environment located
+SET(CMAKE_FIND_ROOT_PATH "SET THIS TO THE NDK TOOLCHAIN'S INSTALL PATH")
+
+# here is where stuff gets installed to
+SET(CMAKE_INSTALL_PREFIX "${CMAKE_FIND_ROOT_PATH}" CACHE STRING "Install path prefix, prepended onto install directories." FORCE)
+
+# adjust the default behaviour of the FIND_XXX() commands:
+# search headers and libraries in the target environment, search 
+# programs in the host environment
+set(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER)
+set(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY)
+set(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY)
+
+# set env vars so that pkg-config will look in the appropriate directory for
+# .pc files (as there seems to be no way to force using ${HOST}-pkg-config)
+set(ENV{PKG_CONFIG_LIBDIR} "${CMAKE_INSTALL_PREFIX}/lib/pkgconfig")
+set(ENV{PKG_CONFIG_PATH} "")
+
+# Qt4 tools
+SET(QT_QMAKE_EXECUTABLE ${HOST}-qmake)
+SET(QT_MOC_EXECUTABLE ${HOST}-moc)
+SET(QT_RCC_EXECUTABLE ${HOST}-rcc)
+SET(QT_UIC_EXECUTABLE ${HOST}-uic)
+SET(QT_LRELEASE_EXECUTABLE ${HOST}-lrelease)

XCompile.txt

@@ -0,0 +1,37 @@
+# Cross-compiling requires CMake 2.6 or newer. Example:
+# cmake .. -DCMAKE_TOOLCHAIN_FILE=../XCompile.txt -DHOST=i686-w64-mingw32
+# Where 'i686-w64-mingw32' is the host prefix for your cross-compiler. If you
+# already have a toolchain file setup, you may use that instead of this file.
+
+# the name of the target operating system
+SET(CMAKE_SYSTEM_NAME Windows)
+
+# which compilers to use for C and C++
+SET(CMAKE_C_COMPILER "${HOST}-gcc")
+SET(CMAKE_CXX_COMPILER "${HOST}-g++")
+SET(CMAKE_RC_COMPILER "${HOST}-windres")
+
+# here is the target environment located
+SET(CMAKE_FIND_ROOT_PATH "/usr/${HOST}")
+
+# here is where stuff gets installed to
+SET(CMAKE_INSTALL_PREFIX "${CMAKE_FIND_ROOT_PATH}" CACHE STRING "Install path prefix, prepended onto install directories." FORCE)
+
+# adjust the default behaviour of the FIND_XXX() commands:
+# search headers and libraries in the target environment, search 
+# programs in the host environment
+set(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER)
+set(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY)
+set(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY)
+
+# set env vars so that pkg-config will look in the appropriate directory for
+# .pc files (as there seems to be no way to force using ${HOST}-pkg-config)
+set(ENV{PKG_CONFIG_LIBDIR} "${CMAKE_INSTALL_PREFIX}/lib/pkgconfig")
+set(ENV{PKG_CONFIG_PATH} "")
+
+# Qt4 tools
+SET(QT_QMAKE_EXECUTABLE ${HOST}-qmake)
+SET(QT_MOC_EXECUTABLE ${HOST}-moc)
+SET(QT_RCC_EXECUTABLE ${HOST}-rcc)
+SET(QT_UIC_EXECUTABLE ${HOST}-uic)
+SET(QT_LRELEASE_EXECUTABLE ${HOST}-lrelease)

alsoftrc.sample

@@ -0,0 +1,487 @@
+# OpenAL config file.
+#
+# Option blocks may appear multiple times, and duplicated options will take the
+# last value specified. Environment variables may be specified within option
+# values, and are automatically substituted when the config file is loaded.
+# Environment variable names may only contain alpha-numeric characters (a-z,
+# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,
+# specifying "$HOME/file.ext" would typically result in something like
+# "/home/user/file.ext". To specify an actual "$" character, use "$$".
+#
+# Device-specific values may be specified by including the device name in the
+# block name, with "general" replaced by the device name. That is, general
+# options for the device "Name of Device" would be in the [Name of Device]
+# block, while ALSA options would be in the [alsa/Name of Device] block.
+# Options marked as "(global)" are not influenced by the device.
+#
+# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
+# specific override settings in $HOME/.alsoftrc.
+# For Windows, these settings should go into $AppData\alsoft.ini
+#
+# Option and block names are case-senstive. The supplied values are only hints
+# and may not be honored (though generally it'll try to get as close as
+# possible). Note: options that are left unset may default to app- or system-
+# specified values. These are the current available settings:
+
+##
+## General stuff
+##
+[general]
+
+## disable-cpu-exts: (global)
+#  Disables use of specialized methods that use specific CPU intrinsics.
+#  Certain methods may utilize CPU extensions for improved performance, and
+#  this option is useful for preventing some or all of those methods from being
+#  used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
+#  Specifying 'all' disables use of all such specialized methods.
+#disable-cpu-exts =
+
+## drivers: (global)
+#  Sets the backend driver list order, comma-seperated. Unknown backends and
+#  duplicated names are ignored. Unlisted backends won't be considered for use
+#  unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
+#  other backends, while 'oss' will try OSS only). Backends prepended with -
+#  won't be considered for use (e.g. '-oss,' will try all available backends
+#  except OSS). An empty list means to try all backends.
+#drivers =
+
+## channels:
+#  Sets the output channel configuration. If left unspecified, one will try to
+#  be detected from the system, and defaulting to stereo. The available values
+#  are: mono, stereo, quad, surround51, surround51rear, surround61, surround71,
+#  ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic
+#  channels of the given order (using ACN ordering and SN3D normalization by
+#  default), which need to be decoded to play correctly on speakers.
+#channels =
+
+## sample-type:
+#  Sets the output sample type. Currently, all mixing is done with 32-bit float
+#  and converted to the output sample type as needed. Available values are:
+#  int8    - signed 8-bit int
+#  uint8   - unsigned 8-bit int
+#  int16   - signed 16-bit int
+#  uint16  - unsigned 16-bit int
+#  int32   - signed 32-bit int
+#  uint32  - unsigned 32-bit int
+#  float32 - 32-bit float
+#sample-type = float32
+
+## frequency:
+#  Sets the output frequency. If left unspecified it will try to detect a
+#  default from the system, otherwise it will default to 44100.
+#frequency =
+
+## period_size:
+#  Sets the update period size, in frames. This is the number of frames needed
+#  for each mixing update. Acceptable values range between 64 and 8192.
+#period_size = 1024
+
+## periods:
+#  Sets the number of update periods. Higher values create a larger mix ahead,
+#  which helps protect against skips when the CPU is under load, but increases
+#  the delay between a sound getting mixed and being heard. Acceptable values
+#  range between 2 and 16.
+#periods = 3
+
+## stereo-mode:
+#  Specifies if stereo output is treated as being headphones or speakers. With
+#  headphones, HRTF or crossfeed filters may be used for better audio quality.
+#  Valid settings are auto, speakers, and headphones.
+#stereo-mode = auto
+
+## stereo-encoding:
+#  Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)
+#  uses standard amplitude panning (aka pair-wise, stereo pair, etc) between
+#  -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ
+#  output, which encodes some surround sound information into stereo output
+#  that can be decoded with a surround sound receiver. If crossfeed filters are
+#  used, UHJ is disabled.
+#stereo-encoding = panpot
+
+## ambi-format:
+#  Specifies the channel order and normalization for the "ambi*" set of channel
+#  configurations. Valid settings are: fuma, acn+sn3d, acn+n3d
+#ambi-format = acn+sn3d
+
+## hrtf:
+#  Controls HRTF processing. These filters provide better spatialization of
+#  sounds while using headphones, but do require a bit more CPU power. The
+#  default filters will only work with 44100hz or 48000hz stereo output. While
+#  HRTF is used, the cf_level option is ignored. Setting this to auto (default)
+#  will allow HRTF to be used when headphones are detected or the app requests
+#  it, while setting true or false will forcefully enable or disable HRTF
+#  respectively.
+#hrtf = auto
+
+## default-hrtf:
+#  Specifies the default HRTF to use. When multiple HRTFs are available, this
+#  determines the preferred one to use if none are specifically requested. Note
+#  that this is the enumerated HRTF name, not necessarily the filename.
+#default-hrtf =
+
+## hrtf-paths:
+#  Specifies a comma-separated list of paths containing HRTF data sets. The
+#  format of the files are described in docs/hrtf.txt. The files within the
+#  directories must have the .mhr file extension to be recognized. By default,
+#  OS-dependent data paths will be used. They will also be used if the list
+#  ends with a comma. On Windows this is:
+#  $AppData\openal\hrtf
+#  And on other systems, it's (in order):
+#  $XDG_DATA_HOME/openal/hrtf  (defaults to $HOME/.local/share/openal/hrtf)
+#  $XDG_DATA_DIRS/openal/hrtf  (defaults to /usr/local/share/openal/hrtf and
+#                               /usr/share/openal/hrtf)
+#hrtf-paths =
+
+## cf_level:
+#  Sets the crossfeed level for stereo output. Valid values are:
+#  0 - No crossfeed
+#  1 - Low crossfeed
+#  2 - Middle crossfeed
+#  3 - High crossfeed (virtual speakers are closer to itself)
+#  4 - Low easy crossfeed
+#  5 - Middle easy crossfeed
+#  6 - High easy crossfeed
+#  Users of headphones may want to try various settings. Has no effect on non-
+#  stereo modes.
+#cf_level = 0
+
+## resampler: (global)
+#  Selects the resampler used when mixing sources. Valid values are:
+#  point - nearest sample, no interpolation
+#  linear - extrapolates samples using a linear slope between samples
+#  sinc4 - extrapolates samples using a 4-point Sinc filter
+#  bsinc - extrapolates samples using a band-limited Sinc filter (varying
+#          between 12 and 24 points, with anti-aliasing)
+#  Specifying other values will result in using the default (linear).
+#resampler = linear
+
+## rt-prio: (global)
+#  Sets real-time priority for the mixing thread. Not all drivers may use this
+#  (eg. PortAudio) as they already control the priority of the mixing thread.
+#  0 and negative values will disable it. Note that this may constitute a
+#  security risk since a real-time priority thread can indefinitely block
+#  normal-priority threads if it fails to wait. As such, the default is
+#  disabled.
+#rt-prio = 0
+
+## sources:
+#  Sets the maximum number of allocatable sources. Lower values may help for
+#  systems with apps that try to play more sounds than the CPU can handle.
+#sources = 256
+
+## slots:
+#  Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
+#  can use a non-negligible amount of CPU time if an effect is set on it even
+#  if no sources are feeding it, so this may help when apps use more than the
+#  system can handle.
+#slots = 64
+
+## sends:
+#  Limits the number of auxiliary sends allowed per source. Setting this higher
+#  than the default has no effect.
+#sends = 16
+
+## output-limiter:
+#  Applies a gain limiter on the final mixed output. This reduces the volume
+#  when the output samples would otherwise clamp, avoiding excessive clipping
+#  noise.
+#output-limiter = true
+
+## dither:
+#  Applies dithering on the final mix, for 8- and 16-bit output by default.
+#  This replaces the distortion created by nearest-value quantization with low-
+#  level whitenoise.
+#dither = true
+
+## dither-depth:
+#  Quantization bit-depth for dithered output. A value of 0 (or less) will
+#  match the output sample depth. For int32, uint32, and float32 output, 0 will
+#  disable dithering because they're at or beyond the rendered precision. The
+#  maximum dither depth is 24.
+#dither-depth = 0
+
+## volume-adjust:
+#  A global volume adjustment for source output, expressed in decibels. The
+#  value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
+#  be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
+#  value of 0 means no change.
+#volume-adjust = 0
+
+## excludefx: (global)
+#  Sets which effects to exclude, preventing apps from using them. This can
+#  help for apps that try to use effects which are too CPU intensive for the
+#  system to handle. Available effects are: eaxreverb,reverb,chorus,compressor,
+#  distortion,echo,equalizer,flanger,modulator,dedicated
+#excludefx =
+
+## default-reverb: (global)
+#  A reverb preset that applies by default to all sources on send 0
+#  (applications that set their own slots on send 0 will override this).
+#  Available presets are: None, Generic, PaddedCell, Room, Bathroom,
+#  Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
+#  CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
+#  Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
+#default-reverb =
+
+## trap-alc-error: (global)
+#  Generates a SIGTRAP signal when an ALC device error is generated, on systems
+#  that support it. This helps when debugging, while trying to find the cause
+#  of a device error. On Windows, a breakpoint exception is generated.
+#trap-alc-error = false
+
+## trap-al-error: (global)
+#  Generates a SIGTRAP signal when an AL context error is generated, on systems
+#  that support it. This helps when debugging, while trying to find the cause
+#  of a context error. On Windows, a breakpoint exception is generated.
+#trap-al-error = false
+
+##
+## Ambisonic decoder stuff
+##
+[decoder]
+
+## hq-mode:
+#  Enables a high-quality ambisonic decoder. This mode is capable of frequency-
+#  dependent processing, creating a better reproduction of 3D sound rendering
+#  over surround sound speakers. Enabling this also requires specifying decoder
+#  configuration files for the appropriate speaker configuration you intend to
+#  use (see the quad, surround51, etc options below). Currently, up to third-
+#  order decoding is supported.
+hq-mode = false
+
+## distance-comp:
+#  Enables compensation for the speakers' relative distances to the listener.
+#  This applies the necessary delays and attenuation to make the speakers
+#  behave as though they are all equidistant, which is important for proper
+#  playback of 3D sound rendering. Requires the proper distances to be
+#  specified in the decoder configuration file.
+distance-comp = true
+
+## nfc:
+#  Enables near-field control filters. This simulates and compensates for low-
+#  frequency effects caused by the curvature of nearby sound-waves, which
+#  creates a more realistic perception of sound distance. Note that the effect
+#  may be stronger or weaker than intended if the application doesn't use or
+#  specify an appropriate unit scale, or if incorrect speaker distances are set
+#  in the decoder configuration file. Requires hq-mode to be enabled.
+nfc = true
+
+## nfc-ref-delay
+#  Specifies the reference delay value for ambisonic output. When channels is
+#  set to one of the ambi* formats, this option enables NFC-HOA output with the
+#  specified Reference Delay parameter. The specified value can then be shared
+#  with an appropriate NFC-HOA decoder to reproduce correct near-field effects.
+#  Keep in mind that despite being designed for higher-order ambisonics, this
+#  applies to first-order output all the same. When left unset, normal output
+#  is created with no near-field simulation.
+nfc-ref-delay =
+
+## quad:
+#  Decoder configuration file for Quadrophonic channel output. See
+#  docs/ambdec.txt for a description of the file format.
+quad =
+
+## surround51:
+#  Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
+#  See docs/ambdec.txt for a description of the file format.
+surround51 =
+
+## surround61:
+#  Decoder configuration file for 6.1 Surround channel output. See
+#  docs/ambdec.txt for a description of the file format.
+surround61 =
+
+## surround71:
+#  Decoder configuration file for 7.1 Surround channel output. See
+#  docs/ambdec.txt for a description of the file format. Note: This can be used
+#  to enable 3D7.1 with the appropriate configuration and speaker placement,
+#  see docs/3D7.1.txt.
+surround71 =
+
+##
+## Reverb effect stuff (includes EAX reverb)
+##
+[reverb]
+
+## boost: (global)
+#  A global amplification for reverb output, expressed in decibels. The value
+#  is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
+#  scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
+#  value of 0 means no change.
+#boost = 0
+
+## emulate-eax: (global)
+#  Allows the standard reverb effect to be used in place of EAX reverb. EAX
+#  reverb processing is a bit more CPU intensive than standard, so this option
+#  allows a simpler effect to be used at the loss of some quality.
+#emulate-eax = false
+
+##
+## PulseAudio backend stuff
+##
+[pulse]
+
+## spawn-server: (global)
+#  Attempts to autospawn a PulseAudio server whenever needed (initializing the
+#  backend, enumerating devices, etc). Setting autospawn to false in Pulse's
+#  client.conf will still prevent autospawning even if this is set to true.
+#spawn-server = true
+
+## allow-moves: (global)
+#  Allows PulseAudio to move active streams to different devices. Note that the
+#  device specifier (seen by applications) will not be updated when this
+#  occurs, and neither will the AL device configuration (sample rate, format,
+#  etc).
+#allow-moves = false
+
+## fix-rate:
+#  Specifies whether to match the playback stream's sample rate to the device's
+#  sample rate. Enabling this forces OpenAL Soft to mix sources and effects
+#  directly to the actual output rate, avoiding a second resample pass by the
+#  PulseAudio server.
+#fix-rate = false
+
+##
+## ALSA backend stuff
+##
+[alsa]
+
+## device: (global)
+#  Sets the device name for the default playback device.
+#device = default
+
+## device-prefix: (global)
+#  Sets the prefix used by the discovered (non-default) playback devices. This
+#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the
+#  device index for the requested device name.
+#device-prefix = plughw:
+
+## device-prefix-*: (global)
+#  Card- and device-specific prefixes may be used to override the device-prefix
+#  option. The option may specify the card id (eg, device-prefix-NVidia), or
+#  the card id and device index (eg, device-prefix-NVidia-0). The card id is
+#  case-sensitive.
+#device-prefix- =
+
+## capture: (global)
+#  Sets the device name for the default capture device.
+#capture = default
+
+## capture-prefix: (global)
+#  Sets the prefix used by the discovered (non-default) capture devices. This
+#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the
+#  device number for the requested device name.
+#capture-prefix = plughw:
+
+## capture-prefix-*: (global)
+#  Card- and device-specific prefixes may be used to override the
+#  capture-prefix option. The option may specify the card id (eg,
+#  capture-prefix-NVidia), or the card id and device index (eg,
+#  capture-prefix-NVidia-0). The card id is case-sensitive.
+#capture-prefix- =
+
+## mmap:
+#  Sets whether to try using mmap mode (helps reduce latencies and CPU
+#  consumption). If mmap isn't available, it will automatically fall back to
+#  non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
+#  and anything else will force mmap off.
+#mmap = true
+
+## allow-resampler:
+#  Specifies whether to allow ALSA's built-in resampler. Enabling this will
+#  allow the playback device to be set to a different sample rate than the
+#  actual output, causing ALSA to apply its own resampling pass after OpenAL
+#  Soft resamples and mixes the sources and effects for output.
+#allow-resampler = false
+
+##
+## OSS backend stuff
+##
+[oss]
+
+## device: (global)
+#  Sets the device name for OSS output.
+#device = /dev/dsp
+
+## capture: (global)
+#  Sets the device name for OSS capture.
+#capture = /dev/dsp
+
+##
+## Solaris backend stuff
+##
+[solaris]
+
+## device: (global)
+#  Sets the device name for Solaris output.
+#device = /dev/audio
+
+##
+## QSA backend stuff
+##
+[qsa]
+
+##
+## JACK backend stuff
+##
+[jack]
+
+## spawn-server: (global)
+#  Attempts to autospawn a JACK server whenever needed (initializing the
+#  backend, opening devices, etc).
+#spawn-server = false
+
+## buffer-size:
+#  Sets the update buffer size, in samples, that the backend will keep buffered
+#  to handle the server's real-time processing requests. This value must be a
+#  power of 2, or else it will be rounded up to the next power of 2. If it is
+#  less than JACK's buffer update size, it will be clamped. This option may
+#  be useful in case the server's update size is too small and doesn't give the
+#  mixer time to keep enough audio available for the processing requests.
+#buffer-size = 0
+
+##
+## MMDevApi backend stuff
+##
+[mmdevapi]
+
+##
+## DirectSound backend stuff
+##
+[dsound]
+
+##
+## Windows Multimedia backend stuff
+##
+[winmm]
+
+##
+## PortAudio backend stuff
+##
+[port]
+
+## device: (global)
+#  Sets the device index for output. Negative values will use the default as
+#  given by PortAudio itself.
+#device = -1
+
+## capture: (global)
+#  Sets the device index for capture. Negative values will use the default as
+#  given by PortAudio itself.
+#capture = -1
+
+##
+## Wave File Writer stuff
+##
+[wave]
+
+## file: (global)
+#  Sets the filename of the wave file to write to. An empty name prevents the
+#  backend from opening, even when explicitly requested.
+#  THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
+#file =
+
+## bformat: (global)
+#  Creates AMB format files using first-order ambisonics instead of a standard
+#  single- or multi-channel .wav file.
+#bformat = false

appveyor.yml

@@ -0,0 +1,19 @@
+version: 1.18.2.{build}
+
+environment:
+    matrix:
+      - GEN: "Visual Studio 14 2015"
+        CFG: Release
+      - GEN: "Visual Studio 14 2015 Win64"
+        CFG: Release
+
+install:
+    # Remove the VS Xamarin targets to reduce AppVeyor specific noise in build
+    # logs.  See also http://help.appveyor.com/discussions/problems/4569
+    - del "C:\Program Files (x86)\MSBuild\14.0\Microsoft.Common.targets\ImportAfter\Xamarin.Common.targets"
+
+build_script:
+    - cd build
+    - cmake -G"%GEN%" -DALSOFT_REQUIRE_WINMM=ON -DALSOFT_REQUIRE_DSOUND=ON -DALSOFT_REQUIRE_MMDEVAPI=ON -DALSOFT_EMBED_HRTF_DATA=YES ..
+    - cmake --build . --config %CFG% --clean-first
+

cmake/CheckFileOffsetBits.c

@@ -0,0 +1,9 @@
+#include <sys/types.h>
+
+#define KB ((off_t)(1024))
+#define MB ((off_t)(KB*1024))
+#define GB ((off_t)(MB*1024))
+int tb[((GB+GB+GB) > GB) ? 1 : -1];
+
+int main()
+{ return 0; }

cmake/CheckFileOffsetBits.cmake

@@ -0,0 +1,39 @@
+# - Check if the _FILE_OFFSET_BITS macro is needed for large files
+# CHECK_FILE_OFFSET_BITS()
+#
+# The following variables may be set before calling this macro to
+# modify the way the check is run:
+#
+#  CMAKE_REQUIRED_FLAGS = string of compile command line flags
+#  CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar)
+#  CMAKE_REQUIRED_INCLUDES = list of include directories
+# Copyright (c) 2009, Chris Robinson
+#
+# Redistribution and use is allowed according to the terms of the LGPL license.
+
+
+MACRO(CHECK_FILE_OFFSET_BITS)
+
+  IF(NOT DEFINED _FILE_OFFSET_BITS)
+    MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files")
+    TRY_COMPILE(__WITHOUT_FILE_OFFSET_BITS_64
+      ${CMAKE_CURRENT_BINARY_DIR}
+      ${CMAKE_CURRENT_SOURCE_DIR}/cmake/CheckFileOffsetBits.c
+      COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS})
+    IF(NOT __WITHOUT_FILE_OFFSET_BITS_64)
+      TRY_COMPILE(__WITH_FILE_OFFSET_BITS_64
+        ${CMAKE_CURRENT_BINARY_DIR}
+        ${CMAKE_CURRENT_SOURCE_DIR}/cmake/CheckFileOffsetBits.c
+        COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS} -D_FILE_OFFSET_BITS=64)
+    ENDIF(NOT __WITHOUT_FILE_OFFSET_BITS_64)
+
+    IF(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64)
+      SET(_FILE_OFFSET_BITS 64 CACHE INTERNAL "_FILE_OFFSET_BITS macro needed for large files")
+      MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files - 64")
+    ELSE(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64)
+      SET(_FILE_OFFSET_BITS "" CACHE INTERNAL "_FILE_OFFSET_BITS macro needed for large files")
+      MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files - not needed")
+    ENDIF(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64)
+  ENDIF(NOT DEFINED _FILE_OFFSET_BITS)
+
+ENDMACRO(CHECK_FILE_OFFSET_BITS)

cmake/CheckSharedFunctionExists.cmake

@@ -0,0 +1,92 @@
+# - Check if a symbol exists as a function, variable, or macro
+# CHECK_SYMBOL_EXISTS(<symbol> <files> <variable>)
+#
+# Check that the <symbol> is available after including given header
+# <files> and store the result in a <variable>.  Specify the list
+# of files in one argument as a semicolon-separated list.
+#
+# If the header files define the symbol as a macro it is considered
+# available and assumed to work.  If the header files declare the
+# symbol as a function or variable then the symbol must also be
+# available for linking.  If the symbol is a type or enum value
+# it will not be recognized (consider using CheckTypeSize or
+# CheckCSourceCompiles).
+#
+# The following variables may be set before calling this macro to
+# modify the way the check is run:
+#
+#  CMAKE_REQUIRED_FLAGS = string of compile command line flags
+#  CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar)
+#  CMAKE_REQUIRED_INCLUDES = list of include directories
+#  CMAKE_REQUIRED_LIBRARIES = list of libraries to link
+
+#=============================================================================
+# Copyright 2003-2011 Kitware, Inc.
+#
+# Distributed under the OSI-approved BSD License (the "License");
+# see accompanying file Copyright.txt for details.
+#
+# This software is distributed WITHOUT ANY WARRANTY; without even the
+# implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+# See the License for more information.
+#=============================================================================
+# (To distribute this file outside of CMake, substitute the full
+#  License text for the above reference.)
+
+MACRO(CHECK_SHARED_FUNCTION_EXISTS SYMBOL FILES LIBRARY LOCATION VARIABLE)
+  IF(NOT DEFINED "${VARIABLE}" OR "x${${VARIABLE}}" STREQUAL "x${VARIABLE}")
+    SET(CMAKE_CONFIGURABLE_FILE_CONTENT "/* */\n")
+    SET(MACRO_CHECK_SYMBOL_EXISTS_FLAGS ${CMAKE_REQUIRED_FLAGS})
+    IF(CMAKE_REQUIRED_LIBRARIES)
+      SET(CHECK_SYMBOL_EXISTS_LIBS 
+        "-DLINK_LIBRARIES:STRING=${CMAKE_REQUIRED_LIBRARIES};${LIBRARY}")
+    ELSE(CMAKE_REQUIRED_LIBRARIES)
+      SET(CHECK_SYMBOL_EXISTS_LIBS
+        "-DLINK_LIBRARIES:STRING=${LIBRARY}")
+    ENDIF(CMAKE_REQUIRED_LIBRARIES)
+    IF(CMAKE_REQUIRED_INCLUDES)
+      SET(CMAKE_SYMBOL_EXISTS_INCLUDES
+        "-DINCLUDE_DIRECTORIES:STRING=${CMAKE_REQUIRED_INCLUDES}")
+    ELSE(CMAKE_REQUIRED_INCLUDES)
+      SET(CMAKE_SYMBOL_EXISTS_INCLUDES)
+    ENDIF(CMAKE_REQUIRED_INCLUDES)
+    FOREACH(FILE ${FILES})
+      SET(CMAKE_CONFIGURABLE_FILE_CONTENT
+        "${CMAKE_CONFIGURABLE_FILE_CONTENT}#include <${FILE}>\n")
+    ENDFOREACH(FILE)
+    SET(CMAKE_CONFIGURABLE_FILE_CONTENT
+      "${CMAKE_CONFIGURABLE_FILE_CONTENT}\nvoid cmakeRequireSymbol(int dummy,...){(void)dummy;}\nint main()\n{\n  cmakeRequireSymbol(0,&${SYMBOL});\n  return 0;\n}\n")
+
+    CONFIGURE_FILE("${CMAKE_ROOT}/Modules/CMakeConfigurableFile.in"
+      "${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c" @ONLY)
+
+    MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY}")
+    TRY_COMPILE(${VARIABLE}
+      ${CMAKE_CURRENT_BINARY_DIR}
+      ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c
+      COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS}
+      CMAKE_FLAGS 
+      -DCOMPILE_DEFINITIONS:STRING=${MACRO_CHECK_SYMBOL_EXISTS_FLAGS}
+      -DLINK_DIRECTORIES:STRING=${LOCATION}
+      "${CHECK_SYMBOL_EXISTS_LIBS}"
+      "${CMAKE_SYMBOL_EXISTS_INCLUDES}"
+      OUTPUT_VARIABLE OUTPUT)
+    IF(${VARIABLE})
+      MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - found")
+      SET(${VARIABLE} 1 CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}")
+      FILE(APPEND ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeOutput.log
+        "Determining if the ${SYMBOL} "
+        "exist in ${LIBRARY} passed with the following output:\n"
+        "${OUTPUT}\nFile ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n"
+        "${CMAKE_CONFIGURABLE_FILE_CONTENT}\n")
+    ELSE(${VARIABLE})
+      MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - not found.")
+      SET(${VARIABLE} "" CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}")
+      FILE(APPEND ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeError.log
+        "Determining if the ${SYMBOL} "
+        "exist in ${LIBRARY} failed with the following output:\n"
+        "${OUTPUT}\nFile ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n"
+        "${CMAKE_CONFIGURABLE_FILE_CONTENT}\n")
+    ENDIF(${VARIABLE})
+  ENDIF(NOT DEFINED "${VARIABLE}" OR "x${${VARIABLE}}" STREQUAL "x${VARIABLE}")
+ENDMACRO(CHECK_SHARED_FUNCTION_EXISTS)

cmake/FindALSA.cmake

@@ -0,0 +1,73 @@
+# - Find alsa
+# Find the alsa libraries (asound)
+#
+#  This module defines the following variables:
+#     ALSA_FOUND       - True if ALSA_INCLUDE_DIR & ALSA_LIBRARY are found
+#     ALSA_LIBRARIES   - Set when ALSA_LIBRARY is found
+#     ALSA_INCLUDE_DIRS - Set when ALSA_INCLUDE_DIR is found
+#
+#     ALSA_INCLUDE_DIR - where to find asoundlib.h, etc.
+#     ALSA_LIBRARY     - the asound library
+#     ALSA_VERSION_STRING - the version of alsa found (since CMake 2.8.8)
+#
+
+#=============================================================================
+# Copyright 2009-2011 Kitware, Inc.
+# Copyright 2009-2011 Philip Lowman <philip@yhbt.com>
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions are
+# met:
+#
+#  * Redistributions of source code must retain the above copyright notice,
+#    this list of conditions and the following disclaimer.
+#
+#  * Redistributions in binary form must reproduce the above copyright notice,
+#    this list of conditions and the following disclaimer in the documentation
+#    and/or other materials provided with the distribution.
+#
+#  * The names of Kitware, Inc., the Insight Consortium, or the names of
+#    any consortium members, or of any contributors, may not be used to
+#    endorse or promote products derived from this software without
+#    specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDER AND CONTRIBUTORS ``AS IS''
+# AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+# ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR
+# ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+# DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+# SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+# CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+# OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+#=============================================================================
+
+find_path(ALSA_INCLUDE_DIR NAMES alsa/asoundlib.h
+          DOC "The ALSA (asound) include directory"
+)
+
+find_library(ALSA_LIBRARY NAMES asound
+             DOC "The ALSA (asound) library"
+)
+
+if(ALSA_INCLUDE_DIR AND EXISTS "${ALSA_INCLUDE_DIR}/alsa/version.h")
+  file(STRINGS "${ALSA_INCLUDE_DIR}/alsa/version.h" alsa_version_str REGEX "^#define[\t ]+SND_LIB_VERSION_STR[\t ]+\".*\"")
+
+  string(REGEX REPLACE "^.*SND_LIB_VERSION_STR[\t ]+\"([^\"]*)\".*$" "\\1" ALSA_VERSION_STRING "${alsa_version_str}")
+  unset(alsa_version_str)
+endif()
+
+# handle the QUIETLY and REQUIRED arguments and set ALSA_FOUND to TRUE if
+# all listed variables are TRUE
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(ALSA
+                                  REQUIRED_VARS ALSA_LIBRARY ALSA_INCLUDE_DIR
+                                  VERSION_VAR ALSA_VERSION_STRING)
+
+if(ALSA_FOUND)
+  set( ALSA_LIBRARIES ${ALSA_LIBRARY} )
+  set( ALSA_INCLUDE_DIRS ${ALSA_INCLUDE_DIR} )
+endif()
+
+mark_as_advanced(ALSA_INCLUDE_DIR ALSA_LIBRARY)