Don't change the format tag in MakeExtensible

Backported fix from d1a86075 on master.

.gitignore

@@ -1,9 +1,5 @@
 build*/
-winbuild
-win64build
-
-## kdevelop
-*.kdev4
-
-## qt-creator
-CMakeLists.txt.user*
+winbuild/
+win64build/
+openal-soft.kdev4
+.kdev4/

.travis.yml

@@ -1,19 +1,13 @@
-language: cpp
+language: c
 matrix:
   include:
     - os: linux
-      dist: xenial
+      dist: trusty
     - os: linux
       dist: trusty
       env:
         - BUILD_ANDROID=true
-    - os: freebsd
-      compiler: clang
     - os: osx
-    - os: osx
-      osx_image: xcode11
-      env:
-        - BUILD_IOS=true
 sudo: required
 install:
   - >
@@ -30,44 +24,19 @@ install:
     fi
   - >
     if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
-      curl -o ~/android-ndk.zip https://dl.google.com/android/repository/android-ndk-r21-linux-x86_64.zip
+      curl -o ~/android-ndk.zip https://dl.google.com/android/repository/android-ndk-r15-linux-x86_64.zip
       unzip -q ~/android-ndk.zip -d ~ \
-        'android-ndk-r21/build/cmake/*' \
-        'android-ndk-r21/build/core/toolchains/arm-linux-androideabi-*/*' \
-        'android-ndk-r21/platforms/android-16/arch-arm/*' \
-        'android-ndk-r21/source.properties' \
-        'android-ndk-r21/sources/android/support/include/*' \
-        'android-ndk-r21/sources/cxx-stl/llvm-libc++/libs/armeabi-v7a/*' \
-        'android-ndk-r21/sources/cxx-stl/llvm-libc++/include/*' \
-        'android-ndk-r21/sysroot/*' \
-        'android-ndk-r21/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/*' \
-        'android-ndk-r21/toolchains/llvm/prebuilt/linux-x86_64/*'
-      export OBOE_LOC=~/oboe
-      git clone --depth 1 -b 1.3-stable https://github.com/google/oboe "$OBOE_LOC"
-    fi
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "freebsd" ]]; then
-      # Install Ninja as it's used downstream.
-      # Install dependencies for all supported backends.
-      # Install Qt5 dependency for alsoft-config.
-      # Install ffmpeg for examples.
-      sudo pkg install -y \
-        alsa-lib \
-        ffmpeg \
-        jackit \
-        libmysofa \
-        ninja \
-        portaudio \
-        pulseaudio \
-        qt5-buildtools \
-        qt5-qmake \
-        qt5-widgets \
-        sdl2 \
-        sndio \
-        $NULL
+        'android-ndk-r15/build/cmake/*' \
+        'android-ndk-r15/build/core/toolchains/arm-linux-androideabi-*/*' \
+        'android-ndk-r15/platforms/android-14/arch-arm/*' \
+        'android-ndk-r15/source.properties' \
+        'android-ndk-r15/sources/cxx-stl/gnu-libstdc++/4.9/libs/armeabi-v7a/*' \
+        'android-ndk-r15/sources/cxx-stl/gnu-libstdc++/4.9/include/*' \
+        'android-ndk-r15/sysroot/*' \
+        'android-ndk-r15/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/*' \
+        'android-ndk-r15/toolchains/llvm/prebuilt/linux-x86_64/*'
     fi
 script:
-  - cmake --version
   - >
     if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then
       cmake \
@@ -82,43 +51,16 @@ script:
   - >
     if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
       cmake \
-        -DANDROID_STL=c++_shared \
-        -DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r21/build/cmake/android.toolchain.cmake \
-        -DOBOE_SOURCE="$OBOE_LOC" \
-        -DALSOFT_REQUIRE_OBOE=ON \
+        -DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r15/build/cmake/android.toolchain.cmake \
         -DALSOFT_REQUIRE_OPENSL=ON \
         -DALSOFT_EMBED_HRTF_DATA=YES \
         .
     fi
   - >
-    if [[ "${TRAVIS_OS_NAME}" == "freebsd" ]]; then
-      cmake -GNinja \
-        -DALSOFT_REQUIRE_ALSA=ON \
-        -DALSOFT_REQUIRE_JACK=ON \
-        -DALSOFT_REQUIRE_OSS=ON \
-        -DALSOFT_REQUIRE_PORTAUDIO=ON \
-        -DALSOFT_REQUIRE_PULSEAUDIO=ON \
-        -DALSOFT_REQUIRE_SDL2=ON \
-        -DALSOFT_REQUIRE_SNDIO=ON \
-        -DALSOFT_EMBED_HRTF_DATA=YES \
-        .
-    fi
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "osx" && -z "${BUILD_IOS}" ]]; then
+    if [[ "${TRAVIS_OS_NAME}" == "osx" ]]; then
       cmake \
         -DALSOFT_REQUIRE_COREAUDIO=ON \
         -DALSOFT_EMBED_HRTF_DATA=YES \
         .
     fi
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "osx" && "${BUILD_IOS}" == "true" ]]; then
-      cmake \
-        -GXcode \
-        -DCMAKE_SYSTEM_NAME=iOS \
-        -DALSOFT_OSX_FRAMEWORK=ON \
-        -DALSOFT_REQUIRE_COREAUDIO=ON \
-        -DALSOFT_EMBED_HRTF_DATA=YES \
-        "-DCMAKE_OSX_ARCHITECTURES=armv7;arm64" \
-        .
-    fi
-  - cmake --build . --clean-first
+  - make -j2

Alc/alconfig.c

@@ -0,0 +1,698 @@
+/**
+ * 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
+#ifdef __APPLE__
+#include <CoreFoundation/CoreFoundation.h>
+#endif
+
+#include "alMain.h"
+#include "alconfig.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)
+{
+    al_string ppath = AL_STRING_INIT_STATIC();
+    WCHAR buffer[MAX_PATH];
+    const WCHAR *str;
+    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);
+    }
+
+    GetProcBinary(&ppath, NULL);
+    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)
+{
+    al_string confpaths = AL_STRING_INIT_STATIC();
+    al_string fname = AL_STRING_INIT_STATIC();
+    const char *str;
+    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";
+    alstr_copy_cstr(&confpaths, str);
+    /* 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(!alstr_empty(confpaths))
+    {
+        char *next = strrchr(alstr_get_cstr(confpaths), ':');
+        if(next)
+        {
+            size_t len = next - alstr_get_cstr(confpaths);
+            alstr_copy_cstr(&fname, next+1);
+            VECTOR_RESIZE(confpaths, len, len+1);
+            VECTOR_ELEM(confpaths, len) = 0;
+        }
+        else
+        {
+            alstr_reset(&fname);
+            fname = confpaths;
+            AL_STRING_INIT(confpaths);
+        }
+
+        if(alstr_empty(fname) || VECTOR_FRONT(fname) != '/')
+            WARN("Ignoring XDG config dir: %s\n", alstr_get_cstr(fname));
+        else
+        {
+            if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf");
+            else alstr_append_cstr(&fname, "alsoft.conf");
+
+            TRACE("Loading config %s...\n", alstr_get_cstr(fname));
+            f = al_fopen(alstr_get_cstr(fname), "r");
+            if(f)
+            {
+                LoadConfigFromFile(f);
+                fclose(f);
+            }
+        }
+        alstr_clear(&fname);
+    }
+
+#ifdef __APPLE__
+    CFBundleRef mainBundle = CFBundleGetMainBundle();
+    if(mainBundle)
+    {
+        unsigned char fileName[PATH_MAX];
+        CFURLRef configURL;
+
+        if((configURL=CFBundleCopyResourceURL(mainBundle, CFSTR(".alsoftrc"), CFSTR(""), NULL)) &&
+           CFURLGetFileSystemRepresentation(configURL, true, fileName, sizeof(fileName)))
+        {
+            f = al_fopen((const char*)fileName, "r");
+            if(f)
+            {
+                LoadConfigFromFile(f);
+                fclose(f);
+            }
+        }
+    }
+#endif
+
+    if((str=getenv("HOME")) != NULL && *str)
+    {
+        alstr_copy_cstr(&fname, str);
+        if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/.alsoftrc");
+        else alstr_append_cstr(&fname, ".alsoftrc");
+
+        TRACE("Loading config %s...\n", alstr_get_cstr(fname));
+        f = al_fopen(alstr_get_cstr(fname), "r");
+        if(f)
+        {
+            LoadConfigFromFile(f);
+            fclose(f);
+        }
+    }
+
+    if((str=getenv("XDG_CONFIG_HOME")) != NULL && str[0] != 0)
+    {
+        alstr_copy_cstr(&fname, str);
+        if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf");
+        else alstr_append_cstr(&fname, "alsoft.conf");
+    }
+    else
+    {
+        alstr_clear(&fname);
+        if((str=getenv("HOME")) != NULL && str[0] != 0)
+        {
+            alstr_copy_cstr(&fname, str);
+            if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/.config/alsoft.conf");
+            else alstr_append_cstr(&fname, ".config/alsoft.conf");
+        }
+    }
+    if(!alstr_empty(fname))
+    {
+        TRACE("Loading config %s...\n", alstr_get_cstr(fname));
+        f = al_fopen(alstr_get_cstr(fname), "r");
+        if(f)
+        {
+            LoadConfigFromFile(f);
+            fclose(f);
+        }
+    }
+
+    alstr_clear(&fname);
+    GetProcBinary(&fname, NULL);
+    if(!alstr_empty(fname))
+    {
+        if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf");
+        else alstr_append_cstr(&fname, "alsoft.conf");
+
+        TRACE("Loading config %s...\n", alstr_get_cstr(fname));
+        f = al_fopen(alstr_get_cstr(fname), "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(&fname);
+    alstr_reset(&confpaths);
+}
+#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/alconfig.h

@@ -0,0 +1,17 @@
+#ifndef ALCONFIG_H
+#define ALCONFIG_H
+
+void ReadALConfig(void);
+void FreeALConfig(void);
+
+int ConfigValueExists(const char *devName, const char *blockName, const char *keyName);
+const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def);
+int GetConfigValueBool(const char *devName, const char *blockName, const char *keyName, int def);
+
+int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret);
+int ConfigValueInt(const char *devName, const char *blockName, const char *keyName, int *ret);
+int ConfigValueUInt(const char *devName, const char *blockName, const char *keyName, unsigned int *ret);
+int ConfigValueFloat(const char *devName, const char *blockName, const char *keyName, float *ret);
+int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret);
+
+#endif /* ALCONFIG_H */

Alc/alstring.h

@@ -0,0 +1,58 @@
+#ifndef ALSTRING_H
+#define ALSTRING_H
+
+#include <string.h>
+
+#include "vector.h"
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+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_copy_wrange(al_string *str, const wchar_t *from, const wchar_t *to);
+void alstr_append_wrange(al_string *str, const wchar_t *from, const wchar_t *to);
+#endif
+
+#ifdef __cplusplus
+} /* extern "C" */
+#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 +1,84 @@
+
+#include "config.h"
+
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alu.h"
+
+#include "backends/base.h"
+
+
+extern inline ALuint64 GetDeviceClockTime(ALCdevice *device);
+extern inline void ALCdevice_Lock(ALCdevice *device);
+extern inline void ALCdevice_Unlock(ALCdevice *device);
+extern inline ClockLatency GetClockLatency(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 +1,178 @@
+#ifndef AL_BACKENDS_BASE_H
+#define AL_BACKENDS_BASE_H
+
+#include "alMain.h"
+#include "threads.h"
+#include "alstring.h"
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+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*);
+
+    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, 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_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, al_string *outnames);
+
+    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_THUNK2(T, ALCbackendFactory, void, probe, enum DevProbe, al_string*)  \
+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 *SndioBackendFactory_getFactory(void);
+ALCbackendFactory *ALCqsaBackendFactory_getFactory(void);
+ALCbackendFactory *ALCwasapiBackendFactory_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 *ALCsdl2BackendFactory_getFactory(void);
+ALCbackendFactory *ALCloopbackFactory_getFactory(void);
+
+
+inline void ALCdevice_Lock(ALCdevice *device)
+{ V0(device->Backend,lock)(); }
+
+inline void ALCdevice_Unlock(ALCdevice *device)
+{ V0(device->Backend,unlock)(); }
+
+
+inline ClockLatency GetClockLatency(ALCdevice *device)
+{
+    ClockLatency ret = V0(device->Backend,getClockLatency)();
+    ret.Latency += device->FixedLatency;
+    return ret;
+}
+
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif /* AL_BACKENDS_BASE_H */

Alc/backends/coreaudio.c

@@ -0,0 +1,816 @@
+/**
+ * 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 "ringbuffer.h"
+
+#include <unistd.h>
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+#include "backends/base.h"
+
+
+static const ALCchar ca_device[] = "CoreAudio Default";
+
+
+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 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)
+{
+    AudioUnitUninitialize(self->audioUnit);
+    AudioComponentInstanceDispose(self->audioUnit);
+
+    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;
+
+    ALCcoreAudioPlayback_lock(self);
+    aluMixData(device, ioData->mBuffers[0].mData,
+               ioData->mBuffers[0].mDataByteSize / self->frameSize);
+    ALCcoreAudioPlayback_unlock(self);
+
+    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;
+#if TARGET_OS_IOS
+    desc.componentSubType = kAudioUnitSubType_RemoteIO;
+#else
+    desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+#endif
+    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 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 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 AudioBufferList *allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
+{
+    AudioBufferList *list;
+
+    list = calloc(1, FAM_SIZE(AudioBufferList, mBuffers, 1) + byteSize);
+    if(list)
+    {
+        list->mNumberBuffers = 1;
+
+        list->mBuffers[0].mNumberChannels = channelCount;
+        list->mBuffers[0].mDataByteSize = byteSize;
+        list->mBuffers[0].mData = &list->mBuffers[1];
+    }
+    return list;
+}
+
+static void destroy_buffer_list(AudioBufferList *list)
+{
+    free(list);
+}
+
+
+static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self);
+
+    self->audioUnit = 0;
+    self->audioConverter = NULL;
+    self->bufferList = NULL;
+    self->resampleBuffer = NULL;
+    self->ring = NULL;
+}
+
+static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self)
+{
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+
+    free(self->resampleBuffer);
+    self->resampleBuffer = NULL;
+
+    destroy_buffer_list(self->bufferList);
+    self->bufferList = NULL;
+
+    if(self->audioConverter)
+        AudioConverterDispose(self->audioConverter);
+    self->audioConverter = NULL;
+
+    if(self->audioUnit)
+        AudioComponentInstanceDispose(self->audioUnit);
+    self->audioUnit = 0;
+
+    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;
+    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;
+#if TARGET_OS_IOS
+    desc.componentSubType = kAudioUnitSubType_RemoteIO;
+#else
+    desc.componentSubType = kAudioUnitSubType_HALOutput;
+#endif
+    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;
+    }
+
+#if !TARGET_OS_IOS
+    // Get the default input device
+    AudioDeviceID inputDevice = kAudioDeviceUnknown;
+
+    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;
+    }
+#endif
+
+    // 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(
+        (size_t)ceil(device->UpdateSize*self->sampleRateRatio*device->NumUpdates),
+        self->frameSize, false
+    );
+    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);
+    self->resampleBuffer = NULL;
+    destroy_buffer_list(self->bufferList);
+    self->bufferList = NULL;
+
+    if(self->audioConverter)
+        AudioConverterDispose(self->audioConverter);
+    self->audioConverter = NULL;
+    if(self->audioUnit)
+        AudioComponentInstanceDispose(self->audioUnit);
+    self->audioUnit = 0;
+
+    return ALC_INVALID_VALUE;
+}
+
+
+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)
+{
+    union {
+        ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)];
+        AudioBufferList list;
+    } audiobuf = { { 0 } };
+    UInt32 frameCount;
+    OSStatus err;
+
+    // If no samples are requested, just return
+    if(samples == 0) return ALC_NO_ERROR;
+
+    // Point the resampling buffer to the capture buffer
+    audiobuf.list.mNumberBuffers = 1;
+    audiobuf.list.mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
+    audiobuf.list.mBuffers[0].mDataByteSize = samples * self->frameSize;
+    audiobuf.list.mBuffers[0].mData = buffer;
+
+    // Resample into another AudioBufferList
+    frameCount = samples;
+    err = AudioConverterFillComplexBuffer(self->audioConverter,
+        ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.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, al_string *outnames);
+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, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+        case CAPTURE_DEVICE_PROBE:
+            alstr_append_range(outnames, ca_device, ca_device+sizeof(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 +1,607 @@
+/**
+ * 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 "alconfig.h"
+#include "ringbuffer.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;
+    alsem_t Sem;
+
+    ATOMIC(ALenum) 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 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);
+
+    alsem_init(&self->Sem, 0);
+
+    self->Client = NULL;
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        self->Port[i] = NULL;
+    self->Ring = NULL;
+
+    ATOMIC_INIT(&self->killNow, AL_TRUE);
+}
+
+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;
+    }
+
+    alsem_destroy(&self->Sem);
+
+    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);
+    device->NumUpdates = (bufsize+device->UpdateSize) / 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),
+        true
+    );
+    if(!self->Ring)
+    {
+        ERR("Failed to reallocate ringbuffer\n");
+        aluHandleDisconnect(device, "Failed to reallocate %u-sample buffer", bufsize);
+    }
+    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);
+    alsem_post(&self->Sem);
+
+    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(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
+    {
+        ALuint todo, len1, len2;
+
+        if(ll_ringbuffer_write_space(self->Ring) < device->UpdateSize)
+        {
+            ALCjackPlayback_unlock(self);
+            alsem_wait(&self->Sem);
+            ALCjackPlayback_lock(self);
+            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 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.
+     */
+    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);
+    device->NumUpdates = (bufsize+device->UpdateSize) / 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),
+        true
+    );
+    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);
+
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    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(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+
+    alsem_post(&self->Sem);
+    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, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            alstr_append_range(outnames, jackDevice, jackDevice+sizeof(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 +1,128 @@
+/**
+ * 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 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 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, al_string *outnames);
+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), al_string* UNUSED(outnames))
+{
+}
+
+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 +1,220 @@
+/**
+ * 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);
+
+    ATOMIC(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 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);
+
+    ATOMIC_INIT(&self->killNow, AL_TRUE);
+}
+
+
+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(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
+    {
+        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 ALCboolean ALCnullBackend_reset(ALCnullBackend *self)
+{
+    SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCnullBackend_start(ALCnullBackend *self)
+{
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    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(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+    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, al_string *outnames);
+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, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+        case CAPTURE_DEVICE_PROBE:
+            alstr_append_range(outnames, nullDevice, nullDevice+sizeof(nullDevice));
+            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 +1,869 @@
+/**
+ * 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 "alconfig.h"
+#include "ringbuffer.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 void ALCplaybackOSS_Destruct(ALCplaybackOSS *self);
+static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name);
+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(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
+    {
+        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, "Failed waiting for playback buffer: %s", strerror(errno));
+            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, "Failed writing playback samples: %s",
+                                    strerror(errno));
+                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);
+
+    self->fd = -1;
+    ATOMIC_INIT(&self->killNow, AL_FALSE);
+}
+
+static void ALCplaybackOSS_Destruct(ALCplaybackOSS *self)
+{
+    if(self->fd != -1)
+        close(self->fd);
+    self->fd = -1;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+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 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 void ALCcaptureOSS_Destruct(ALCcaptureOSS *self);
+static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name);
+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, "Failed to check capture samples: %s", strerror(errno));
+            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, "Failed reading capture samples: %s", strerror(errno));
+                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);
+
+    self->fd = -1;
+    self->ring = NULL;
+    ATOMIC_INIT(&self->killNow, AL_FALSE);
+}
+
+static void ALCcaptureOSS_Destruct(ALCcaptureOSS *self)
+{
+    if(self->fd != -1)
+        close(self->fd);
+    self->fd = -1;
+
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+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, frameSize, false);
+    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 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, al_string *outnames);
+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, al_string *outnames)
+{
+    struct oss_device *cur = NULL;
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            ALCossListFree(&oss_playback);
+            ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT);
+            cur = &oss_playback;
+            break;
+
+        case CAPTURE_DEVICE_PROBE:
+            ALCossListFree(&oss_capture);
+            ALCossListPopulate(&oss_capture, DSP_CAP_INPUT);
+            cur = &oss_capture;
+            break;
+    }
+    while(cur != NULL)
+    {
+#ifdef HAVE_STAT
+        struct stat buf;
+        if(stat(cur->path, &buf) == 0)
+#endif
+            alstr_append_range(outnames, cur->handle, cur->handle+strlen(cur->handle)+1);
+        cur = cur->next;
+    }
+}
+
+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 +1,555 @@
+/**
+ * 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 "alconfig.h"
+#include "ringbuffer.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 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)
+{
+    PaError err = self->stream ? Pa_CloseStream(self->stream) : paNoError;
+    if(err != paNoError)
+        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
+    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 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 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;
+    self->ring = NULL;
+}
+
+static void ALCportCapture_Destruct(ALCportCapture *self)
+{
+    PaError err = self->stream ? Pa_CloseStream(self->stream) : paNoError;
+    if(err != paNoError)
+        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
+    self->stream = NULL;
+
+    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, false);
+    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 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, al_string *outnames);
+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, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+        case CAPTURE_DEVICE_PROBE:
+            alstr_append_range(outnames, pa_device, pa_device+sizeof(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/sdl2.c

@@ -0,0 +1,288 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2018 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 <SDL2/SDL.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "threads.h"
+#include "compat.h"
+
+#include "backends/base.h"
+
+
+#ifdef _WIN32
+#define DEVNAME_PREFIX "OpenAL Soft on "
+#else
+#define DEVNAME_PREFIX ""
+#endif
+
+typedef struct ALCsdl2Backend {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    SDL_AudioDeviceID deviceID;
+    ALsizei frameSize;
+
+    ALuint Frequency;
+    enum DevFmtChannels FmtChans;
+    enum DevFmtType     FmtType;
+    ALuint UpdateSize;
+} ALCsdl2Backend;
+
+static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device);
+static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self);
+static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name);
+static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self);
+static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self);
+static void ALCsdl2Backend_stop(ALCsdl2Backend *self);
+static DECLARE_FORWARD2(ALCsdl2Backend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ClockLatency, getClockLatency)
+static void ALCsdl2Backend_lock(ALCsdl2Backend *self);
+static void ALCsdl2Backend_unlock(ALCsdl2Backend *self);
+DECLARE_DEFAULT_ALLOCATORS(ALCsdl2Backend)
+
+DEFINE_ALCBACKEND_VTABLE(ALCsdl2Backend);
+
+static const ALCchar defaultDeviceName[] = DEVNAME_PREFIX "Default Device";
+
+static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCsdl2Backend, ALCbackend, self);
+
+    self->deviceID = 0;
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    self->Frequency = device->Frequency;
+    self->FmtChans = device->FmtChans;
+    self->FmtType = device->FmtType;
+    self->UpdateSize = device->UpdateSize;
+}
+
+static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self)
+{
+    if(self->deviceID)
+        SDL_CloseAudioDevice(self->deviceID);
+    self->deviceID = 0;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static void ALCsdl2Backend_audioCallback(void *ptr, Uint8 *stream, int len)
+{
+    ALCsdl2Backend *self = (ALCsdl2Backend*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    assert((len % self->frameSize) == 0);
+    aluMixData(device, stream, len / self->frameSize);
+}
+
+static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    SDL_AudioSpec want, have;
+
+    SDL_zero(want);
+    SDL_zero(have);
+
+    want.freq = device->Frequency;
+    switch(device->FmtType)
+    {
+        case DevFmtUByte: want.format = AUDIO_U8; break;
+        case DevFmtByte: want.format = AUDIO_S8; break;
+        case DevFmtUShort: want.format = AUDIO_U16SYS; break;
+        case DevFmtShort: want.format = AUDIO_S16SYS; break;
+        case DevFmtUInt: /* fall-through */
+        case DevFmtInt: want.format = AUDIO_S32SYS; break;
+        case DevFmtFloat: want.format = AUDIO_F32; break;
+    }
+    want.channels = (device->FmtChans == DevFmtMono) ? 1 : 2;
+    want.samples = device->UpdateSize;
+    want.callback = ALCsdl2Backend_audioCallback;
+    want.userdata = self;
+
+    /* Passing NULL to SDL_OpenAudioDevice opens a default, which isn't
+     * necessarily the first in the list.
+     */
+    if(!name || strcmp(name, defaultDeviceName) == 0)
+        self->deviceID = SDL_OpenAudioDevice(NULL, SDL_FALSE, &want, &have,
+                                             SDL_AUDIO_ALLOW_ANY_CHANGE);
+    else
+    {
+        const size_t prefix_len = strlen(DEVNAME_PREFIX);
+        if(strncmp(name, DEVNAME_PREFIX, prefix_len) == 0)
+            self->deviceID = SDL_OpenAudioDevice(name+prefix_len, SDL_FALSE, &want, &have,
+                                                 SDL_AUDIO_ALLOW_ANY_CHANGE);
+        else
+            self->deviceID = SDL_OpenAudioDevice(name, SDL_FALSE, &want, &have,
+                                                 SDL_AUDIO_ALLOW_ANY_CHANGE);
+    }
+    if(self->deviceID == 0)
+        return ALC_INVALID_VALUE;
+
+    device->Frequency = have.freq;
+    if(have.channels == 1)
+        device->FmtChans = DevFmtMono;
+    else if(have.channels == 2)
+        device->FmtChans = DevFmtStereo;
+    else
+    {
+        ERR("Got unhandled SDL channel count: %d\n", (int)have.channels);
+        return ALC_INVALID_VALUE;
+    }
+    switch(have.format)
+    {
+        case AUDIO_U8:     device->FmtType = DevFmtUByte;  break;
+        case AUDIO_S8:     device->FmtType = DevFmtByte;   break;
+        case AUDIO_U16SYS: device->FmtType = DevFmtUShort; break;
+        case AUDIO_S16SYS: device->FmtType = DevFmtShort;  break;
+        case AUDIO_S32SYS: device->FmtType = DevFmtInt;    break;
+        case AUDIO_F32SYS: device->FmtType = DevFmtFloat;  break;
+        default:
+            ERR("Got unsupported SDL format: 0x%04x\n", have.format);
+            return ALC_INVALID_VALUE;
+    }
+    device->UpdateSize = have.samples;
+    device->NumUpdates = 2; /* SDL always (tries to) use two periods. */
+
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    self->Frequency = device->Frequency;
+    self->FmtChans = device->FmtChans;
+    self->FmtType = device->FmtType;
+    self->UpdateSize = device->UpdateSize;
+
+    alstr_copy_cstr(&device->DeviceName, name ? name : defaultDeviceName);
+
+    return ALC_NO_ERROR;
+}
+
+static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    device->Frequency = self->Frequency;
+    device->FmtChans = self->FmtChans;
+    device->FmtType = self->FmtType;
+    device->UpdateSize = self->UpdateSize;
+    device->NumUpdates = 2;
+    SetDefaultWFXChannelOrder(device);
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self)
+{
+    SDL_PauseAudioDevice(self->deviceID, 0);
+    return ALC_TRUE;
+}
+
+static void ALCsdl2Backend_stop(ALCsdl2Backend *self)
+{
+    SDL_PauseAudioDevice(self->deviceID, 1);
+}
+
+static void ALCsdl2Backend_lock(ALCsdl2Backend *self)
+{
+    SDL_LockAudioDevice(self->deviceID);
+}
+
+static void ALCsdl2Backend_unlock(ALCsdl2Backend *self)
+{
+    SDL_UnlockAudioDevice(self->deviceID);
+}
+
+
+typedef struct ALCsdl2BackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCsdl2BackendFactory;
+#define ALCsdl2BACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsdl2BackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void);
+
+static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory *self);
+static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory *self);
+static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory *self, ALCbackend_Type type);
+static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory *self, enum DevProbe type, al_string *outnames);
+static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsdl2BackendFactory);
+
+
+ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void)
+{
+    static ALCsdl2BackendFactory factory = ALCsdl2BACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory* UNUSED(self))
+{
+    if(SDL_InitSubSystem(SDL_INIT_AUDIO) == 0)
+        return AL_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory* UNUSED(self))
+{
+    SDL_QuitSubSystem(SDL_INIT_AUDIO);
+}
+
+static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
+{
+    int num_devices, i;
+    al_string name;
+
+    if(type != ALL_DEVICE_PROBE)
+        return;
+
+    AL_STRING_INIT(name);
+    num_devices = SDL_GetNumAudioDevices(SDL_FALSE);
+
+    alstr_append_range(outnames, defaultDeviceName, defaultDeviceName+sizeof(defaultDeviceName));
+    for(i = 0;i < num_devices;++i)
+    {
+        alstr_copy_cstr(&name, DEVNAME_PREFIX);
+        alstr_append_cstr(&name, SDL_GetAudioDeviceName(i, SDL_FALSE));
+        if(!alstr_empty(name))
+            alstr_append_range(outnames, VECTOR_BEGIN(name), VECTOR_END(name)+1);
+    }
+    alstr_reset(&name);
+}
+
+static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCsdl2Backend *backend;
+        NEW_OBJ(backend, ALCsdl2Backend)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/sndio.c

@@ -0,0 +1,600 @@
+/**
+ * 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 "ringbuffer.h"
+
+#include "backends/base.h"
+
+#include <sndio.h>
+
+
+static const ALCchar sndio_device[] = "SndIO Default";
+
+
+typedef struct SndioPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    struct sio_hdl *sndHandle;
+
+    ALvoid *mix_data;
+    ALsizei data_size;
+
+    ATOMIC(int) killNow;
+    althrd_t thread;
+} SndioPlayback;
+
+static int SndioPlayback_mixerProc(void *ptr);
+
+static void SndioPlayback_Construct(SndioPlayback *self, ALCdevice *device);
+static void SndioPlayback_Destruct(SndioPlayback *self);
+static ALCenum SndioPlayback_open(SndioPlayback *self, const ALCchar *name);
+static ALCboolean SndioPlayback_reset(SndioPlayback *self);
+static ALCboolean SndioPlayback_start(SndioPlayback *self);
+static void SndioPlayback_stop(SndioPlayback *self);
+static DECLARE_FORWARD2(SndioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(SndioPlayback, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(SndioPlayback, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(SndioPlayback, ALCbackend, void, lock)
+static DECLARE_FORWARD(SndioPlayback, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(SndioPlayback)
+
+DEFINE_ALCBACKEND_VTABLE(SndioPlayback);
+
+
+static void SndioPlayback_Construct(SndioPlayback *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(SndioPlayback, ALCbackend, self);
+
+    self->sndHandle = NULL;
+    self->mix_data = NULL;
+    ATOMIC_INIT(&self->killNow, AL_TRUE);
+}
+
+static void SndioPlayback_Destruct(SndioPlayback *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 SndioPlayback_mixerProc(void *ptr)
+{
+    SndioPlayback *self = (SndioPlayback*)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(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
+    {
+        ALsizei len = self->data_size;
+        ALubyte *WritePtr = self->mix_data;
+
+        SndioPlayback_lock(self);
+        aluMixData(device, WritePtr, len/frameSize);
+        SndioPlayback_unlock(self);
+        while(len > 0 && !ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
+        {
+            wrote = sio_write(self->sndHandle, WritePtr, len);
+            if(wrote == 0)
+            {
+                ERR("sio_write failed\n");
+                ALCdevice_Lock(device);
+                aluHandleDisconnect(device, "Failed to write playback samples");
+                ALCdevice_Unlock(device);
+                break;
+            }
+
+            len -= wrote;
+            WritePtr += wrote;
+        }
+    }
+
+    return 0;
+}
+
+
+static ALCenum SndioPlayback_open(SndioPlayback *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 ALCboolean SndioPlayback_reset(SndioPlayback *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 SndioPlayback_start(SndioPlayback *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;
+    }
+
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    if(althrd_create(&self->thread, SndioPlayback_mixerProc, self) != althrd_success)
+    {
+        sio_stop(self->sndHandle);
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void SndioPlayback_stop(SndioPlayback *self)
+{
+    int res;
+
+    if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+    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 SndioCapture {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    struct sio_hdl *sndHandle;
+
+    ll_ringbuffer_t *ring;
+
+    ATOMIC(int) killNow;
+    althrd_t thread;
+} SndioCapture;
+
+static int SndioCapture_recordProc(void *ptr);
+
+static void SndioCapture_Construct(SndioCapture *self, ALCdevice *device);
+static void SndioCapture_Destruct(SndioCapture *self);
+static ALCenum SndioCapture_open(SndioCapture *self, const ALCchar *name);
+static DECLARE_FORWARD(SndioCapture, ALCbackend, ALCboolean, reset)
+static ALCboolean SndioCapture_start(SndioCapture *self);
+static void SndioCapture_stop(SndioCapture *self);
+static ALCenum SndioCapture_captureSamples(SndioCapture *self, void *buffer, ALCuint samples);
+static ALCuint SndioCapture_availableSamples(SndioCapture *self);
+static DECLARE_FORWARD(SndioCapture, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(SndioCapture, ALCbackend, void, lock)
+static DECLARE_FORWARD(SndioCapture, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(SndioCapture)
+
+DEFINE_ALCBACKEND_VTABLE(SndioCapture);
+
+
+static void SndioCapture_Construct(SndioCapture *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(SndioCapture, ALCbackend, self);
+
+    self->sndHandle = NULL;
+    self->ring = NULL;
+    ATOMIC_INIT(&self->killNow, AL_TRUE);
+}
+
+static void SndioCapture_Destruct(SndioCapture *self)
+{
+    if(self->sndHandle)
+        sio_close(self->sndHandle);
+    self->sndHandle = NULL;
+
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static int SndioCapture_recordProc(void* ptr)
+{
+    SndioCapture *self = (SndioCapture*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ALsizei frameSize;
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), RECORD_THREAD_NAME);
+
+    frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+
+    while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
+    {
+        ll_ringbuffer_data_t data[2];
+        size_t total, todo;
+
+        ll_ringbuffer_get_write_vector(self->ring, data);
+        todo = data[0].len + data[1].len;
+        if(todo == 0)
+        {
+            static char junk[4096];
+            sio_read(self->sndHandle, junk, minz(sizeof(junk)/frameSize, device->UpdateSize)*frameSize);
+            continue;
+        }
+
+        total = 0;
+        data[0].len *= frameSize;
+        data[1].len *= frameSize;
+        todo = minz(todo, device->UpdateSize) * frameSize;
+        while(total < todo)
+        {
+            size_t got;
+
+            if(!data[0].len)
+                data[0] = data[1];
+
+            got = sio_read(self->sndHandle, data[0].buf, minz(todo-total, data[0].len));
+            if(!got)
+            {
+                SndioCapture_lock(self);
+                aluHandleDisconnect(device, "Failed to read capture samples");
+                SndioCapture_unlock(self);
+                break;
+            }
+
+            data[0].buf += got;
+            data[0].len -= got;
+            total += got;
+        }
+        ll_ringbuffer_write_advance(self->ring, total / frameSize);
+    }
+
+    return 0;
+}
+
+
+static ALCenum SndioCapture_open(SndioCapture *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    struct sio_par par;
+
+    if(!name)
+        name = sndio_device;
+    else if(strcmp(name, sndio_device) != 0)
+        return ALC_INVALID_VALUE;
+
+    self->sndHandle = sio_open(NULL, SIO_REC, 0);
+    if(self->sndHandle == NULL)
+    {
+        ERR("Could not open device\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    sio_initpar(&par);
+
+    switch(device->FmtType)
+    {
+        case DevFmtByte:
+            par.bps = 1;
+            par.sig = 1;
+            break;
+        case DevFmtUByte:
+            par.bps = 1;
+            par.sig = 0;
+            break;
+        case DevFmtShort:
+            par.bps = 2;
+            par.sig = 1;
+            break;
+        case DevFmtUShort:
+            par.bps = 2;
+            par.sig = 0;
+            break;
+        case DevFmtInt:
+            par.bps = 4;
+            par.sig = 1;
+            break;
+        case DevFmtUInt:
+            par.bps = 4;
+            par.sig = 0;
+            break;
+        case DevFmtFloat:
+            ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
+            return ALC_INVALID_VALUE;
+    }
+    par.bits = par.bps * 8;
+    par.le = SIO_LE_NATIVE;
+    par.msb = SIO_LE_NATIVE ? 0 : 1;
+    par.rchan = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
+    par.rate = device->Frequency;
+
+    par.appbufsz = maxu(device->UpdateSize*device->NumUpdates, (device->Frequency+9)/10);
+    par.round = clampu(par.appbufsz/device->NumUpdates, (device->Frequency+99)/100,
+                       (device->Frequency+19)/20);
+
+    device->UpdateSize = par.round;
+    device->NumUpdates = maxu(par.appbufsz/par.round, 1);
+
+    if(!sio_setpar(self->sndHandle, &par) || !sio_getpar(self->sndHandle, &par))
+    {
+        ERR("Failed to set device parameters\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    if(par.bits != par.bps*8)
+    {
+        ERR("Padded samples not supported (%u of %u bits)\n", par.bits, par.bps*8);
+        return ALC_INVALID_VALUE;
+    }
+
+    if(!((device->FmtType == DevFmtByte && par.bits == 8 && par.sig != 0) ||
+         (device->FmtType == DevFmtUByte && par.bits == 8 && par.sig == 0) ||
+         (device->FmtType == DevFmtShort && par.bits == 16 && par.sig != 0) ||
+         (device->FmtType == DevFmtUShort && par.bits == 16 && par.sig == 0) ||
+         (device->FmtType == DevFmtInt && par.bits == 32 && par.sig != 0) ||
+         (device->FmtType == DevFmtUInt && par.bits == 32 && par.sig == 0)) ||
+       ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)par.rchan ||
+       device->Frequency != par.rate)
+    {
+        ERR("Failed to set format %s %s %uhz, got %c%u %u-channel %uhz instead\n",
+            DevFmtTypeString(device->FmtType), DevFmtChannelsString(device->FmtChans),
+            device->Frequency, par.sig?'s':'u', par.bits, par.rchan, par.rate);
+        return ALC_INVALID_VALUE;
+    }
+
+    self->ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates, par.bps*par.rchan, 0);
+    if(!self->ring)
+    {
+        ERR("Failed to allocate %u-byte ringbuffer\n",
+            device->UpdateSize*device->NumUpdates*par.bps*par.rchan);
+        return ALC_OUT_OF_MEMORY;
+    }
+
+    SetDefaultChannelOrder(device);
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+}
+
+static ALCboolean SndioCapture_start(SndioCapture *self)
+{
+    if(!sio_start(self->sndHandle))
+    {
+        ERR("Error starting playback\n");
+        return ALC_FALSE;
+    }
+
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    if(althrd_create(&self->thread, SndioCapture_recordProc, self) != althrd_success)
+    {
+        sio_stop(self->sndHandle);
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void SndioCapture_stop(SndioCapture *self)
+{
+    int res;
+
+    if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+    althrd_join(self->thread, &res);
+
+    if(!sio_stop(self->sndHandle))
+        ERR("Error stopping device\n");
+}
+
+static ALCenum SndioCapture_captureSamples(SndioCapture *self, void *buffer, ALCuint samples)
+{
+    ll_ringbuffer_read(self->ring, buffer, samples);
+    return ALC_NO_ERROR;
+}
+
+static ALCuint SndioCapture_availableSamples(SndioCapture *self)
+{
+    return ll_ringbuffer_read_space(self->ring);
+}
+
+
+typedef struct SndioBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} SndioBackendFactory;
+#define SNDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(SndioBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *SndioBackendFactory_getFactory(void);
+
+static ALCboolean SndioBackendFactory_init(SndioBackendFactory *self);
+static DECLARE_FORWARD(SndioBackendFactory, ALCbackendFactory, void, deinit)
+static ALCboolean SndioBackendFactory_querySupport(SndioBackendFactory *self, ALCbackend_Type type);
+static void SndioBackendFactory_probe(SndioBackendFactory *self, enum DevProbe type, al_string *outnames);
+static ALCbackend* SndioBackendFactory_createBackend(SndioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(SndioBackendFactory);
+
+ALCbackendFactory *SndioBackendFactory_getFactory(void)
+{
+    static SndioBackendFactory factory = SNDIOBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+static ALCboolean SndioBackendFactory_init(SndioBackendFactory* UNUSED(self))
+{
+    /* No dynamic loading */
+    return ALC_TRUE;
+}
+
+static ALCboolean SndioBackendFactory_querySupport(SndioBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback || type == ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void SndioBackendFactory_probe(SndioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+        case CAPTURE_DEVICE_PROBE:
+            alstr_append_range(outnames, sndio_device, sndio_device+sizeof(sndio_device));
+            break;
+    }
+}
+
+static ALCbackend* SndioBackendFactory_createBackend(SndioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        SndioPlayback *backend;
+        NEW_OBJ(backend, SndioPlayback)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+    if(type == ALCbackend_Capture)
+    {
+        SndioCapture *backend;
+        NEW_OBJ(backend, SndioCapture)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}

Alc/backends/solaris.c

@@ -0,0 +1,360 @@
+/**
+ * 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 "alconfig.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 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;
+    self->mix_data = NULL;
+    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(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
+    {
+        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, "Failed to wait for playback buffer: %s", strerror(errno));
+            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, "Failed to write playback samples: %s",
+                                    strerror(errno));
+                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 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, al_string *outnames);
+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, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+        {
+#ifdef HAVE_STAT
+            struct stat buf;
+            if(stat(solaris_driver, &buf) == 0)
+#endif
+                alstr_append_range(outnames, solaris_device, solaris_device+sizeof(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 "alconfig.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;
+
+    ATOMIC(ALenum) killNow;
+    althrd_t thread;
+} ALCwaveBackend;
+
+static int ALCwaveBackend_mixerProc(void *ptr);
+
+static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device);
+static void ALCwaveBackend_Destruct(ALCwaveBackend *self);
+static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name);
+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;
+
+    ATOMIC_INIT(&self->killNow, AL_TRUE);
+}
+
+static void ALCwaveBackend_Destruct(ALCwaveBackend *self)
+{
+    if(self->mFile)
+        fclose(self->mFile);
+    self->mFile = NULL;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+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(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
+    {
+        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, "Failed to write playback samples");
+                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 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;
+    }
+
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    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(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+    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, al_string *outnames);
+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, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+            alstr_append_range(outnames, waveDevice, waveDevice+sizeof(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 +1,786 @@
+/**
+ * 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 "ringbuffer.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;
+
+    ATOMIC(ALenum) 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 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;
+
+    ATOMIC_INIT(&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(ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
+        {
+            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 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;
+
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    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(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+    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;
+
+    ATOMIC(ALenum) 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 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;
+
+    ATOMIC_INIT(&self->killNow, AL_TRUE);
+}
+
+static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self)
+{
+    void *buffer = NULL;
+    int i;
+
+    /* Tell the processing thread to quit and wait for it to do so. */
+    if(!ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+    {
+        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
+    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(ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
+            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, self->Format.nBlockAlign, false);
+    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);
+    }
+
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    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 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 (ALCuint)ll_ringbuffer_read_space(self->Ring);
+}
+
+
+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, al_string *outnames);
+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, al_string *outnames)
+{
+    switch(type)
+    {
+#define APPEND_OUTNAME(n) do {                                                \
+    if(!alstr_empty(*(n)))                                                    \
+        alstr_append_range(outnames, VECTOR_BEGIN(*(n)), VECTOR_END(*(n))+1); \
+} while(0)
+        case ALL_DEVICE_PROBE:
+            ProbePlaybackDevices();
+            VECTOR_FOR_EACH(const al_string, PlaybackDevices, APPEND_OUTNAME);
+            break;
+
+        case CAPTURE_DEVICE_PROBE:
+            ProbeCaptureDevices();
+            VECTOR_FOR_EACH(const al_string, CaptureDevices, APPEND_OUTNAME);
+            break;
+#undef APPEND_OUTNAME
+    }
+}
+
+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 +1,492 @@
+
+#include "config.h"
+
+#include "bformatdec.h"
+#include "ambdec.h"
+#include "filters/splitter.h"
+#include "alu.h"
+
+#include "bool.h"
+#include "threads.h"
+#include "almalloc.h"
+
+
+/* NOTE: These are scale factors as applied to Ambisonics content. Decoder
+ * coefficients should be divided by these values to get proper N3D scalings.
+ */
+const ALfloat N3D2N3DScale[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
+};
+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) */
+};
+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) */
+};
+
+
+#define HF_BAND 0
+#define LF_BAND 1
+#define NUM_BANDS 2
+
+/* 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][MAX_AMBI_COEFFS] = {
+    { 0.125f,  0.125f,  0.125f,  0.125f },
+    { 0.125f, -0.125f,  0.125f,  0.125f },
+    { 0.125f,  0.125f,  0.125f, -0.125f },
+    { 0.125f, -0.125f,  0.125f, -0.125f },
+    { 0.125f,  0.125f, -0.125f,  0.125f },
+    { 0.125f, -0.125f, -0.125f,  0.125f },
+    { 0.125f,  0.125f, -0.125f, -0.125f },
+    { 0.125f, -0.125f, -0.125f, -0.125f },
+};
+static const ALfloat Ambi3DDecoderHFScale[MAX_AMBI_COEFFS] = {
+    2.0f,
+    1.15470054f, 1.15470054f, 1.15470054f
+};
+
+
+/* NOTE: BandSplitter filters are unused with single-band decoding */
+typedef struct BFormatDec {
+    ALuint Enabled; /* Bitfield of enabled channels. */
+
+    union {
+        alignas(16) ALfloat Dual[MAX_OUTPUT_CHANNELS][NUM_BANDS][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[NUM_BANDS];
+    } UpSampler[4];
+
+    ALsizei NumChannels;
+    ALboolean DualBand;
+} BFormatDec;
+
+BFormatDec *bformatdec_alloc()
+{
+    return al_calloc(16, sizeof(BFormatDec));
+}
+
+void bformatdec_free(BFormatDec **dec)
+{
+    if(dec && *dec)
+    {
+        al_free((*dec)->Samples);
+        (*dec)->Samples = NULL;
+        (*dec)->SamplesHF = NULL;
+        (*dec)->SamplesLF = NULL;
+
+        al_free(*dec);
+        *dec = NULL;
+    }
+}
+
+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 = N3D2N3DScale;
+    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;
+
+    dec->Enabled = 0;
+    for(i = 0;i < conf->NumSpeakers;i++)
+        dec->Enabled |= 1 << chanmap[i];
+
+    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[HF_BAND] = (conf->ChanMask > 0x1ff) ? W_SCALE_3H3P :
+                                           (conf->ChanMask > 0xf) ? W_SCALE_2H2P : 1.0f;
+        dec->UpSampler[0].Gains[LF_BAND] = 1.0f;
+        for(i = 1;i < 4;i++)
+        {
+            dec->UpSampler[i].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? XYZ_SCALE_3H3P :
+                                               (conf->ChanMask > 0xf) ? XYZ_SCALE_2H2P : 1.0f;
+            dec->UpSampler[i].Gains[LF_BAND] = 1.0f;
+        }
+    }
+    else
+    {
+        periphonic = false;
+
+        dec->UpSampler[0].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? W_SCALE_3H0P :
+                                           (conf->ChanMask > 0xf) ? W_SCALE_2H0P : 1.0f;
+        dec->UpSampler[0].Gains[LF_BAND] = 1.0f;
+        for(i = 1;i < 3;i++)
+        {
+            dec->UpSampler[i].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? XYZ_SCALE_3H0P :
+                                               (conf->ChanMask > 0xf) ? XYZ_SCALE_2H0P : 1.0f;
+            dec->UpSampler[i].Gains[LF_BAND] = 1.0f;
+        }
+        dec->UpSampler[3].Gains[HF_BAND] = 0.0f;
+        dec->UpSampler[3].Gains[LF_BAND] = 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][HF_BAND][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][LF_BAND][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][HF_BAND][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][LF_BAND][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&(1<<chan)))
+                continue;
+
+            memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
+            MixRowSamples(dec->ChannelMix, dec->Matrix.Dual[chan][HF_BAND],
+                dec->SamplesHF, dec->NumChannels, 0, SamplesToDo
+            );
+            MixRowSamples(dec->ChannelMix, dec->Matrix.Dual[chan][LF_BAND],
+                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&(1<<chan)))
+                continue;
+
+            memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
+            MixRowSamples(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[HF_BAND], dec->Samples[LF_BAND],
+            InSamples[i], SamplesToDo
+        );
+
+        /* Now write each band to the output. */
+        MixRowSamples(OutBuffer[i], dec->UpSampler[i].Gains,
+            dec->Samples, NUM_BANDS, 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[NUM_BANDS][BUFFERSIZE];
+
+    BandSplitter XOver[4];
+
+    ALfloat Gains[4][MAX_OUTPUT_CHANNELS][NUM_BANDS];
+} AmbiUpsampler;
+
+AmbiUpsampler *ambiup_alloc()
+{
+    return al_calloc(16, sizeof(AmbiUpsampler));
+}
+
+void ambiup_free(struct AmbiUpsampler **ambiup)
+{
+    if(ambiup)
+    {
+        al_free(*ambiup);
+        *ambiup = NULL;
+    }
+}
+
+void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device, ALfloat w_scale, ALfloat xyz_scale)
+{
+    ALfloat ratio;
+    ALsizei 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(k = 0;k < COUNTOF(Ambi3DPoints);k++)
+        {
+            ALfloat coeffs[MAX_AMBI_COEFFS] = { 0.0f };
+            CalcDirectionCoeffs(Ambi3DPoints[k], 0.0f, coeffs);
+            ComputePanGains(&device->Dry, coeffs, 1.0f, encgains[k]);
+        }
+
+        /* 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++)
+            {
+                ALdouble gain = 0.0;
+                for(k = 0;k < COUNTOF(Ambi3DDecoder);k++)
+                    gain += (ALdouble)Ambi3DDecoder[k][i] * encgains[k][j];
+                ambiup->Gains[i][j][HF_BAND] = (ALfloat)(gain * Ambi3DDecoderHFScale[i]);
+                ambiup->Gains[i][j][LF_BAND] = (ALfloat)gain;
+            }
+        }
+    }
+    else
+    {
+        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][HF_BAND] = scale * ((i==0) ? w_scale : xyz_scale);
+                ambiup->Gains[i][index][LF_BAND] = 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[HF_BAND], ambiup->Samples[LF_BAND],
+            InSamples[i], SamplesToDo
+        );
+
+        for(j = 0;j < OutChannels;j++)
+            MixRowSamples(OutBuffer[j], ambiup->Gains[i][j],
+                ambiup->Samples, NUM_BANDS, 0, SamplesToDo
+            );
+    }
+}

Alc/bformatdec.h

@@ -0,0 +1,57 @@
+#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_SCALE_2H0P   1.224744871f /* sqrt(1.5) */
+#define XYZ_SCALE_2H0P 1.0f
+#define W_SCALE_3H0P   1.414213562f /* sqrt(2) */
+#define XYZ_SCALE_3H0P 1.082392196f
+
+/* These are the necessary scales for first-order HF responses to play over
+ * higher-order 3D (periphonic) decoders.
+ */
+#define W_SCALE_2H2P   1.341640787f /* sqrt(1.8) */
+#define XYZ_SCALE_2H2P 1.0f
+#define W_SCALE_3H3P   1.695486018f
+#define XYZ_SCALE_3H3P 1.136697713f
+
+
+/* NOTE: These are scale factors as applied to Ambisonics content. Decoder
+ * coefficients should be divided by these values to get proper N3D scalings.
+ */
+const ALfloat N3D2N3DScale[MAX_AMBI_COEFFS];
+const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS];
+const ALfloat FuMa2N3DScale[MAX_AMBI_COEFFS];
+
+
+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. Assumes a periphonic (4-channel) input mix!
+ */
+struct AmbiUpsampler *ambiup_alloc();
+void ambiup_free(struct AmbiUpsampler **ambiup);
+void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device, ALfloat w_scale, ALfloat xyz_scale);
+
+void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
+
+#endif /* BFORMATDEC_H */

Alc/bs2b.c

@@ -23,19 +23,18 @@
 
 #include "config.h"
 
-#include <algorithm>
-#include <cmath>
-#include <iterator>
+#include <math.h>
+#include <string.h>
 
 #include "bs2b.h"
-#include "math_defs.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 G_lo,  G_hi;
     float x, g;
 
     switch(bs2b->level)
@@ -91,11 +90,11 @@ static void init(struct bs2b *bs2b)
      * $d  = 1 / 2 / pi / $fc;
      * $x  = exp(-1 / $d);
      */
-    x           = std::exp(-al::MathDefs<float>::Tau() * Fc_lo / static_cast<float>(bs2b->srate));
+    x           = expf(-2.0f * F_PI * Fc_lo / bs2b->srate);
     bs2b->b1_lo = x;
     bs2b->a0_lo = G_lo * (1.0f - x) * g;
 
-    x           = std::exp(-al::MathDefs<float>::Tau() * Fc_hi / static_cast<float>(bs2b->srate));
+    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;
@@ -127,55 +126,60 @@ int bs2b_get_srate(struct bs2b *bs2b)
 
 void bs2b_clear(struct bs2b *bs2b)
 {
-    std::fill(std::begin(bs2b->history), std::end(bs2b->history), bs2b::t_last_sample{});
+    memset(&bs2b->last_sample, 0, sizeof(bs2b->last_sample));
 } /* bs2b_clear */
 
-void bs2b_cross_feed(struct bs2b *bs2b, float *Left, float *Right, size_t SamplesToDo)
+void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo)
 {
-    const float a0_lo{bs2b->a0_lo};
-    const float b1_lo{bs2b->b1_lo};
-    const float a0_hi{bs2b->a0_hi};
-    const float a1_hi{bs2b->a1_hi};
-    const float b1_hi{bs2b->b1_hi};
     float lsamples[128][2];
     float rsamples[128][2];
+    int base;
 
-    for(size_t base{0};base < SamplesToDo;)
+    for(base = 0;base < SamplesToDo;)
     {
-        const size_t todo{std::min<size_t>(128, SamplesToDo-base)};
+        int todo = mini(128, SamplesToDo-base);
+        int i;
 
         /* Process left input */
-        float z_lo{bs2b->history[0].lo};
-        float z_hi{bs2b->history[0].hi};
-        for(size_t i{0};i < todo;i++)
+        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] = a0_lo*Left[i] + z_lo;
-            z_lo = b1_lo*lsamples[i][0];
-
-            lsamples[i][1] = a0_hi*Left[i] + z_hi;
-            z_hi = a1_hi*Left[i] + b1_hi*lsamples[i][1];
+            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->history[0].lo = z_lo;
-        bs2b->history[0].hi = z_hi;
+        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 */
-        z_lo = bs2b->history[1].lo;
-        z_hi = bs2b->history[1].hi;
-        for(size_t i{0};i < todo;i++)
+        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] = a0_lo*Right[i] + z_lo;
-            z_lo = b1_lo*rsamples[i][0];
-
-            rsamples[i][1] = a0_hi*Right[i] + z_hi;
-            z_hi = a1_hi*Right[i] + b1_hi*rsamples[i][1];
+            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->history[1].lo = z_lo;
-        bs2b->history[1].hi = z_hi;
+        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(size_t i{0};i < todo;i++)
+        for(i = 0;i < todo;i++)
             *(Left++) = lsamples[i][1] + rsamples[i][0];
-        for(size_t i{0};i < todo;i++)
+        for(i = 0;i < todo;i++)
             *(Right++) = rsamples[i][1] + lsamples[i][0];
 
         base += todo;

Alc/compat.h

@@ -0,0 +1,57 @@
+#ifndef AL_COMPAT_H
+#define AL_COMPAT_H
+
+#include "alstring.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#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);
+
+void GetProcBinary(al_string *path, al_string *fname);
+
+#ifdef HAVE_DYNLOAD
+void *LoadLib(const char *name);
+void CloseLib(void *handle);
+void *GetSymbol(void *handle, const char *name);
+#endif
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif /* AL_COMPAT_H */

Alc/converter.c

@@ -0,0 +1,468 @@
+
+#include "config.h"
+
+#include "converter.h"
+
+#include "fpu_modes.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 = (ALsizei)mind(((ALdouble)srcRate/dstRate*FRACTIONONE) + 0.5,
+                         MAX_PITCH * FRACTIONONE);
+    converter->mIncrement = maxi(step, 1);
+    if(converter->mIncrement == FRACTIONONE)
+        converter->mResample = Resample_copy_C;
+    else
+    {
+        /* TODO: Allow other resamplers. */
+        BsincPrepare(converter->mIncrement, &converter->mState.bsinc, &bsinc12);
+        converter->mResample = SelectResampler(BSinc12Resampler);
+    }
+    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_RESAMPLE_PADDING*2 &&
+       MAX_RESAMPLE_PADDING*2 - prepcount >= srcframes)
+    {
+        /* Not enough input samples to generate an output sample. */
+        return 0;
+    }
+
+    DataSize64  = prepcount;
+    DataSize64 += srcframes;
+    DataSize64 -= MAX_RESAMPLE_PADDING*2;
+    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_RESAMPLE_PADDING*2);
+
+        if(prepcount < MAX_RESAMPLE_PADDING*2 &&
+           MAX_RESAMPLE_PADDING*2 - 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_RESAMPLE_PADDING*2;
+        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_RESAMPLE_PADDING*2, 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_RESAMPLE_PADDING, 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(prepcount + toread - (DataPosFrac>>FRACTIONBITS),
+                                        MAX_RESAMPLE_PADDING*2);
+        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_RESAMPLE_PADDING*2];
+    } 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/cpu_caps.h

@@ -1,7 +1,6 @@
 #ifndef CPU_CAPS_H
 #define CPU_CAPS_H
 
-
 extern int CPUCapFlags;
 enum {
     CPU_CAP_SSE    = 1<<0,

Alc/effects/autowah.c

@@ -0,0 +1,321 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2018 by Raul Herraiz.
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.h"
+
+#define MIN_FREQ 20.0f
+#define MAX_FREQ 2500.0f
+#define Q_FACTOR 5.0f
+
+typedef struct ALautowahState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    /* Effect parameters */
+    ALfloat AttackRate;
+    ALfloat ReleaseRate;
+    ALfloat ResonanceGain;
+    ALfloat PeakGain;
+    ALfloat FreqMinNorm;
+    ALfloat BandwidthNorm;
+    ALfloat env_delay;
+
+    /* Filter components derived from the envelope. */
+    struct {
+        ALfloat cos_w0;
+        ALfloat alpha;
+    } Env[BUFFERSIZE];
+
+    struct {
+        /* Effect filters' history. */
+        struct {
+            ALfloat z1, z2;
+        } Filter;
+
+        /* Effect gains for each output channel */
+        ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
+        ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
+    } Chans[MAX_EFFECT_CHANNELS];
+
+    /* Effects buffers */
+    alignas(16) ALfloat BufferOut[BUFFERSIZE];
+} ALautowahState;
+
+static ALvoid ALautowahState_Destruct(ALautowahState *state);
+static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *device);
+static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
+static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALautowahState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALautowahState);
+
+static void ALautowahState_Construct(ALautowahState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALautowahState, ALeffectState, state);
+}
+
+static ALvoid ALautowahState_Destruct(ALautowahState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *UNUSED(device))
+{
+    /* (Re-)initializing parameters and clear the buffers. */
+    ALsizei i, j;
+
+    state->AttackRate    = 1.0f;
+    state->ReleaseRate   = 1.0f;
+    state->ResonanceGain = 10.0f;
+    state->PeakGain      = 4.5f;
+    state->FreqMinNorm   = 4.5e-4f;
+    state->BandwidthNorm = 0.05f;
+    state->env_delay     = 0.0f;
+
+    memset(state->Env, 0, sizeof(state->Env));
+
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+    {
+        for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
+            state->Chans[i].CurrentGains[j] = 0.0f;
+        state->Chans[i].Filter.z1 = 0.0f;
+        state->Chans[i].Filter.z2 = 0.0f;
+    }
+
+    return AL_TRUE;
+}
+
+static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALfloat ReleaseTime;
+    ALsizei i;
+
+    ReleaseTime = clampf(props->Autowah.ReleaseTime, 0.001f, 1.0f);
+
+    state->AttackRate    = expf(-1.0f / (props->Autowah.AttackTime*device->Frequency));
+    state->ReleaseRate   = expf(-1.0f / (ReleaseTime*device->Frequency));
+    /* 0-20dB Resonance Peak gain */
+    state->ResonanceGain = sqrtf(log10f(props->Autowah.Resonance)*10.0f / 3.0f);
+    state->PeakGain      = 1.0f - log10f(props->Autowah.PeakGain/AL_AUTOWAH_MAX_PEAK_GAIN);
+    state->FreqMinNorm   = MIN_FREQ / device->Frequency;
+    state->BandwidthNorm = (MAX_FREQ-MIN_FREQ) / device->Frequency;
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+        ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain,
+                        state->Chans[i].TargetGains);
+}
+
+static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    const ALfloat attack_rate = state->AttackRate;
+    const ALfloat release_rate = state->ReleaseRate;
+    const ALfloat res_gain = state->ResonanceGain;
+    const ALfloat peak_gain = state->PeakGain;
+    const ALfloat freq_min = state->FreqMinNorm;
+    const ALfloat bandwidth = state->BandwidthNorm;
+    ALfloat env_delay;
+    ALsizei c, i;
+
+    env_delay = state->env_delay;
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        ALfloat w0, sample, a;
+
+        /* Envelope follower described on the book: Audio Effects, Theory,
+         * Implementation and Application.
+         */
+        sample = peak_gain * fabsf(SamplesIn[0][i]);
+        a = (sample > env_delay) ? attack_rate : release_rate;
+        env_delay = lerp(sample, env_delay, a);
+
+        /* Calculate the cos and alpha components for this sample's filter. */
+        w0 = minf((bandwidth*env_delay + freq_min), 0.46f) * F_TAU;
+        state->Env[i].cos_w0 = cosf(w0);
+        state->Env[i].alpha = sinf(w0)/(2.0f * Q_FACTOR);
+    }
+    state->env_delay = env_delay;
+
+    for(c = 0;c < MAX_EFFECT_CHANNELS; c++)
+    {
+        /* This effectively inlines BiquadFilter_setParams for a peaking
+         * filter and BiquadFilter_processC. The alpha and cosine components
+         * for the filter coefficients were previously calculated with the
+         * envelope. Because the filter changes for each sample, the
+         * coefficients are transient and don't need to be held.
+         */
+        ALfloat z1 = state->Chans[c].Filter.z1;
+        ALfloat z2 = state->Chans[c].Filter.z2;
+
+        for(i = 0;i < SamplesToDo;i++)
+        {
+            const ALfloat alpha = state->Env[i].alpha;
+            const ALfloat cos_w0 = state->Env[i].cos_w0;
+            ALfloat input, output;
+            ALfloat a[3], b[3];
+
+            b[0] =  1.0f + alpha*res_gain;
+            b[1] = -2.0f * cos_w0;
+            b[2] =  1.0f - alpha*res_gain;
+            a[0] =  1.0f + alpha/res_gain;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha/res_gain;
+
+            input = SamplesIn[c][i];
+            output = input*(b[0]/a[0]) + z1;
+            z1 = input*(b[1]/a[0]) - output*(a[1]/a[0]) + z2;
+            z2 = input*(b[2]/a[0]) - output*(a[2]/a[0]);
+            state->BufferOut[i] = output;
+        }
+        state->Chans[c].Filter.z1 = z1;
+        state->Chans[c].Filter.z2 = z2;
+
+        /* Now, mix the processed sound data to the output. */
+        MixSamples(state->BufferOut, NumChannels, SamplesOut, state->Chans[c].CurrentGains,
+                   state->Chans[c].TargetGains, SamplesToDo, 0, SamplesToDo);
+    }
+}
+
+typedef struct AutowahStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} AutowahStateFactory;
+
+static ALeffectState *AutowahStateFactory_create(AutowahStateFactory *UNUSED(factory))
+{
+    ALautowahState *state;
+
+    NEW_OBJ0(state, ALautowahState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(AutowahStateFactory);
+
+EffectStateFactory *AutowahStateFactory_getFactory(void)
+{
+    static AutowahStateFactory AutowahFactory = { { GET_VTABLE2(AutowahStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &AutowahFactory);
+}
+
+void ALautowah_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_AUTOWAH_ATTACK_TIME:
+            if(!(val >= AL_AUTOWAH_MIN_ATTACK_TIME && val <= AL_AUTOWAH_MAX_ATTACK_TIME))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah attack time out of range");
+            props->Autowah.AttackTime = val;
+            break;
+
+        case AL_AUTOWAH_RELEASE_TIME:
+            if(!(val >= AL_AUTOWAH_MIN_RELEASE_TIME && val <= AL_AUTOWAH_MAX_RELEASE_TIME))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah release time out of range");
+            props->Autowah.ReleaseTime = val;
+            break;
+
+        case AL_AUTOWAH_RESONANCE:
+            if(!(val >= AL_AUTOWAH_MIN_RESONANCE && val <= AL_AUTOWAH_MAX_RESONANCE))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah resonance out of range");
+            props->Autowah.Resonance = val;
+            break;
+
+        case AL_AUTOWAH_PEAK_GAIN:
+            if(!(val >= AL_AUTOWAH_MIN_PEAK_GAIN && val <= AL_AUTOWAH_MAX_PEAK_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah peak gain out of range");
+            props->Autowah.PeakGain = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param);
+    }
+}
+
+void ALautowah_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALautowah_setParamf(effect, context, param, vals[0]);
+}
+
+void ALautowah_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{
+    alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param);
+}
+
+void ALautowah_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{
+    alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param);
+}
+
+void ALautowah_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val))
+{
+    alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param);
+}
+void ALautowah_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{
+    alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param);
+}
+
+void ALautowah_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_AUTOWAH_ATTACK_TIME:
+            *val = props->Autowah.AttackTime;
+            break;
+
+        case AL_AUTOWAH_RELEASE_TIME:
+            *val = props->Autowah.ReleaseTime;
+            break;
+
+        case AL_AUTOWAH_RESONANCE:
+            *val = props->Autowah.Resonance;
+            break;
+
+        case AL_AUTOWAH_PEAK_GAIN:
+            *val = props->Autowah.PeakGain;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param);
+    }
+
+}
+
+void ALautowah_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALautowah_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALautowah);

Alc/effects/chorus.c

@@ -0,0 +1,555 @@
+/**
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.h"
+
+
+static_assert(AL_CHORUS_WAVEFORM_SINUSOID == AL_FLANGER_WAVEFORM_SINUSOID, "Chorus/Flanger waveform value mismatch");
+static_assert(AL_CHORUS_WAVEFORM_TRIANGLE == AL_FLANGER_WAVEFORM_TRIANGLE, "Chorus/Flanger waveform value mismatch");
+
+enum WaveForm {
+    WF_Sinusoid,
+    WF_Triangle
+};
+
+typedef struct ALchorusState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    ALfloat *SampleBuffer;
+    ALsizei BufferLength;
+    ALsizei offset;
+
+    ALsizei lfo_offset;
+    ALsizei lfo_range;
+    ALfloat lfo_scale;
+    ALint lfo_disp;
+
+    /* Gains for left and right sides */
+    struct {
+        ALfloat Current[MAX_OUTPUT_CHANNELS];
+        ALfloat Target[MAX_OUTPUT_CHANNELS];
+    } Gains[2];
+
+    /* effect parameters */
+    enum WaveForm 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 ALCcontext *Context, 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 = NULL;
+    state->offset = 0;
+    state->lfo_offset = 0;
+    state->lfo_range = 1;
+    state->waveform = WF_Triangle;
+}
+
+static ALvoid ALchorusState_Destruct(ALchorusState *state)
+{
+    al_free(state->SampleBuffer);
+    state->SampleBuffer = NULL;
+
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device)
+{
+    const ALfloat max_delay = maxf(AL_CHORUS_MAX_DELAY, AL_FLANGER_MAX_DELAY);
+    ALsizei maxlen;
+
+    maxlen = NextPowerOf2(float2int(max_delay*2.0f*Device->Frequency) + 1u);
+    if(maxlen <= 0) return AL_FALSE;
+
+    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;
+    }
+
+    memset(state->SampleBuffer, 0, state->BufferLength*sizeof(ALfloat));
+    memset(state->Gains, 0, sizeof(state->Gains));
+
+    return AL_TRUE;
+}
+
+static ALvoid ALchorusState_update(ALchorusState *state, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props)
+{
+    const ALsizei mindelay = MAX_RESAMPLE_PADDING << FRACTIONBITS;
+    const ALCdevice *device = Context->Device;
+    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 = WF_Triangle;
+            break;
+        case AL_CHORUS_WAVEFORM_SINUSOID:
+            state->waveform = WF_Sinusoid;
+            break;
+    }
+
+    /* The LFO depth is scaled to be relative to the sample delay. Clamp the
+     * delay and depth to allow enough padding for resampling.
+     */
+    state->delay = maxi(float2int(props->Chorus.Delay*frequency*FRACTIONONE + 0.5f),
+                        mindelay);
+    state->depth = minf(props->Chorus.Depth * state->delay,
+                        (ALfloat)(state->delay - mindelay));
+
+    state->feedback = props->Chorus.Feedback;
+
+    /* Gains for left and right sides */
+    CalcAngleCoeffs(-F_PI_2, 0.0f, 0.0f, coeffs);
+    ComputePanGains(&device->Dry, coeffs, Slot->Params.Gain, state->Gains[0].Target);
+    CalcAngleCoeffs( F_PI_2, 0.0f, 0.0f, coeffs);
+    ComputePanGains(&device->Dry, coeffs, Slot->Params.Gain, state->Gains[1].Target);
+
+    phase = props->Chorus.Phase;
+    rate = props->Chorus.Rate;
+    if(!(rate > 0.0f))
+    {
+        state->lfo_offset = 0;
+        state->lfo_range = 1;
+        state->lfo_scale = 0.0f;
+        state->lfo_disp = 0;
+    }
+    else
+    {
+        /* Calculate LFO coefficient (number of samples per cycle). Limit the
+         * max range to avoid overflow when calculating the displacement.
+         */
+        ALsizei lfo_range = float2int(minf(frequency/rate + 0.5f, (ALfloat)(INT_MAX/360 - 180)));
+
+        state->lfo_offset = float2int((ALfloat)state->lfo_offset/state->lfo_range*
+                                      lfo_range + 0.5f) % lfo_range;
+        state->lfo_range = lfo_range;
+        switch(state->waveform)
+        {
+            case WF_Triangle:
+                state->lfo_scale = 4.0f / state->lfo_range;
+                break;
+            case WF_Sinusoid:
+                state->lfo_scale = F_TAU / state->lfo_range;
+                break;
+        }
+
+        /* Calculate lfo phase displacement */
+        if(phase < 0) phase = 360 + phase;
+        state->lfo_disp = (state->lfo_range*phase + 180) / 360;
+    }
+}
+
+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)
+{
+    const ALsizei bufmask = state->BufferLength-1;
+    const ALfloat feedback = state->feedback;
+    const ALsizei avgdelay = (state->delay + (FRACTIONONE>>1)) >> FRACTIONBITS;
+    ALfloat *restrict delaybuf = state->SampleBuffer;
+    ALsizei offset = state->offset;
+    ALsizei i, c;
+    ALsizei base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        const ALsizei todo = mini(256, SamplesToDo-base);
+        ALint moddelays[2][256];
+        alignas(16) ALfloat temps[2][256];
+
+        if(state->waveform == WF_Sinusoid)
+        {
+            GetSinusoidDelays(moddelays[0], state->lfo_offset, state->lfo_range, state->lfo_scale,
+                              state->depth, state->delay, todo);
+            GetSinusoidDelays(moddelays[1], (state->lfo_offset+state->lfo_disp)%state->lfo_range,
+                              state->lfo_range, state->lfo_scale, state->depth, state->delay,
+                              todo);
+        }
+        else /*if(state->waveform == WF_Triangle)*/
+        {
+            GetTriangleDelays(moddelays[0], state->lfo_offset, state->lfo_range, state->lfo_scale,
+                              state->depth, state->delay, todo);
+            GetTriangleDelays(moddelays[1], (state->lfo_offset+state->lfo_disp)%state->lfo_range,
+                              state->lfo_range, state->lfo_scale, state->depth, state->delay,
+                              todo);
+        }
+        state->lfo_offset = (state->lfo_offset+todo) % state->lfo_range;
+
+        for(i = 0;i < todo;i++)
+        {
+            ALint delay;
+            ALfloat mu;
+
+            // Feed the buffer's input first (necessary for delays < 1).
+            delaybuf[offset&bufmask] = SamplesIn[0][base+i];
+
+            // Tap for the left output.
+            delay = offset - (moddelays[0][i]>>FRACTIONBITS);
+            mu = (moddelays[0][i]&FRACTIONMASK) * (1.0f/FRACTIONONE);
+            temps[0][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay  ) & bufmask],
+                                delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask],
+                                mu);
+
+            // Tap for the right output.
+            delay = offset - (moddelays[1][i]>>FRACTIONBITS);
+            mu = (moddelays[1][i]&FRACTIONMASK) * (1.0f/FRACTIONONE);
+            temps[1][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay  ) & bufmask],
+                                delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask],
+                                mu);
+
+            // Accumulate feedback from the average delay of the taps.
+            delaybuf[offset&bufmask] += delaybuf[(offset-avgdelay) & bufmask] * feedback;
+            offset++;
+        }
+
+        for(c = 0;c < 2;c++)
+            MixSamples(temps[c], NumChannels, SamplesOut, state->Gains[c].Current,
+                       state->Gains[c].Target, SamplesToDo-base, base, todo);
+
+        base += todo;
+    }
+
+    state->offset = offset;
+}
+
+
+typedef struct ChorusStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} ChorusStateFactory;
+
+static ALeffectState *ChorusStateFactory_create(ChorusStateFactory *UNUSED(factory))
+{
+    ALchorusState *state;
+
+    NEW_OBJ0(state, ALchorusState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(ChorusStateFactory);
+
+
+EffectStateFactory *ChorusStateFactory_getFactory(void)
+{
+    static ChorusStateFactory ChorusFactory = { { GET_VTABLE2(ChorusStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid chorus waveform");
+            props->Chorus.Waveform = val;
+            break;
+
+        case AL_CHORUS_PHASE:
+            if(!(val >= AL_CHORUS_MIN_PHASE && val <= AL_CHORUS_MAX_PHASE))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus phase out of range");
+            props->Chorus.Phase = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid chorus integer property 0x%04x", param);
+    }
+}
+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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus rate out of range");
+            props->Chorus.Rate = val;
+            break;
+
+        case AL_CHORUS_DEPTH:
+            if(!(val >= AL_CHORUS_MIN_DEPTH && val <= AL_CHORUS_MAX_DEPTH))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus depth out of range");
+            props->Chorus.Depth = val;
+            break;
+
+        case AL_CHORUS_FEEDBACK:
+            if(!(val >= AL_CHORUS_MIN_FEEDBACK && val <= AL_CHORUS_MAX_FEEDBACK))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus feedback out of range");
+            props->Chorus.Feedback = val;
+            break;
+
+        case AL_CHORUS_DELAY:
+            if(!(val >= AL_CHORUS_MIN_DELAY && val <= AL_CHORUS_MAX_DELAY))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus delay out of range");
+            props->Chorus.Delay = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid chorus float property 0x%04x", param);
+    }
+}
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid chorus integer property 0x%04x", param);
+    }
+}
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid chorus float property 0x%04x", param);
+    }
+}
+void ALchorus_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{ ALchorus_getParamf(effect, context, param, vals); }
+
+DEFINE_ALEFFECT_VTABLE(ALchorus);
+
+
+/* Flanger is basically a chorus with a really short delay. They can both use
+ * the same processing functions, so piggyback flanger on the chorus functions.
+ */
+typedef struct FlangerStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} FlangerStateFactory;
+
+ALeffectState *FlangerStateFactory_create(FlangerStateFactory *UNUSED(factory))
+{
+    ALchorusState *state;
+
+    NEW_OBJ0(state, ALchorusState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(FlangerStateFactory);
+
+EffectStateFactory *FlangerStateFactory_getFactory(void)
+{
+    static FlangerStateFactory FlangerFactory = { { GET_VTABLE2(FlangerStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid flanger waveform");
+            props->Chorus.Waveform = val;
+            break;
+
+        case AL_FLANGER_PHASE:
+            if(!(val >= AL_FLANGER_MIN_PHASE && val <= AL_FLANGER_MAX_PHASE))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger phase out of range");
+            props->Chorus.Phase = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid flanger integer property 0x%04x", param);
+    }
+}
+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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger rate out of range");
+            props->Chorus.Rate = val;
+            break;
+
+        case AL_FLANGER_DEPTH:
+            if(!(val >= AL_FLANGER_MIN_DEPTH && val <= AL_FLANGER_MAX_DEPTH))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger depth out of range");
+            props->Chorus.Depth = val;
+            break;
+
+        case AL_FLANGER_FEEDBACK:
+            if(!(val >= AL_FLANGER_MIN_FEEDBACK && val <= AL_FLANGER_MAX_FEEDBACK))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger feedback out of range");
+            props->Chorus.Feedback = val;
+            break;
+
+        case AL_FLANGER_DELAY:
+            if(!(val >= AL_FLANGER_MIN_DELAY && val <= AL_FLANGER_MAX_DELAY))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger delay out of range");
+            props->Chorus.Delay = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid flanger float property 0x%04x", param);
+    }
+}
+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->Chorus.Waveform;
+            break;
+
+        case AL_FLANGER_PHASE:
+            *val = props->Chorus.Phase;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid flanger integer property 0x%04x", param);
+    }
+}
+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->Chorus.Rate;
+            break;
+
+        case AL_FLANGER_DEPTH:
+            *val = props->Chorus.Depth;
+            break;
+
+        case AL_FLANGER_FEEDBACK:
+            *val = props->Chorus.Feedback;
+            break;
+
+        case AL_FLANGER_DELAY:
+            *val = props->Chorus.Delay;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid flanger float property 0x%04x", param);
+    }
+}
+void ALflanger_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{ ALflanger_getParamf(effect, context, param, vals); }
+
+DEFINE_ALEFFECT_VTABLE(ALflanger);

Alc/effects/compressor.c

@@ -0,0 +1,243 @@
+/**
+ * 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"
+
+
+#define AMP_ENVELOPE_MIN  0.5f
+#define AMP_ENVELOPE_MAX  2.0f
+
+#define ATTACK_TIME  0.1f /* 100ms to rise from min to max */
+#define RELEASE_TIME 0.2f /* 200ms to drop from max to min */
+
+
+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 AttackMult;
+    ALfloat ReleaseMult;
+    ALfloat EnvFollower;
+} ALcompressorState;
+
+static ALvoid ALcompressorState_Destruct(ALcompressorState *state);
+static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device);
+static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCcontext *context, 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->AttackMult = 1.0f;
+    state->ReleaseMult = 1.0f;
+    state->EnvFollower = 1.0f;
+}
+
+static ALvoid ALcompressorState_Destruct(ALcompressorState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device)
+{
+    /* Number of samples to do a full attack and release (non-integer sample
+     * counts are okay).
+     */
+    const ALfloat attackCount  = (ALfloat)device->Frequency * ATTACK_TIME;
+    const ALfloat releaseCount = (ALfloat)device->Frequency * RELEASE_TIME;
+
+    /* Calculate per-sample multipliers to attack and release at the desired
+     * rates.
+     */
+    state->AttackMult  = powf(AMP_ENVELOPE_MAX/AMP_ENVELOPE_MIN, 1.0f/attackCount);
+    state->ReleaseMult = powf(AMP_ENVELOPE_MIN/AMP_ENVELOPE_MAX, 1.0f/releaseCount);
+
+    return AL_TRUE;
+}
+
+static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    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++)
+        ComputePanGains(&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 gains[256];
+        ALsizei td = mini(256, SamplesToDo-base);
+        ALfloat env = state->EnvFollower;
+
+        /* Generate the per-sample gains from the signal envelope. */
+        if(state->Enabled)
+        {
+            for(i = 0;i < td;++i)
+            {
+                /* Clamp the absolute amplitude to the defined envelope limits,
+                 * then attack or release the envelope to reach it.
+                 */
+                ALfloat amplitude = clampf(fabsf(SamplesIn[0][base+i]),
+                                           AMP_ENVELOPE_MIN, AMP_ENVELOPE_MAX);
+                if(amplitude > env)
+                    env = minf(env*state->AttackMult, amplitude);
+                else if(amplitude < env)
+                    env = maxf(env*state->ReleaseMult, amplitude);
+
+                /* Apply the reciprocal of the envelope to normalize the volume
+                 * (compress the dynamic range).
+                 */
+                gains[i] = 1.0f / env;
+            }
+        }
+        else
+        {
+            /* Same as above, except the amplitude is forced to 1. This helps
+             * ensure smooth gain changes when the compressor is turned on and
+             * off.
+             */
+            for(i = 0;i < td;++i)
+            {
+                ALfloat amplitude = 1.0f;
+                if(amplitude > env)
+                    env = minf(env*state->AttackMult, amplitude);
+                else if(amplitude < env)
+                    env = maxf(env*state->ReleaseMult, amplitude);
+
+                gains[i] = 1.0f / env;
+            }
+        }
+        state->EnvFollower = env;
+
+        /* Now compress the signal amplitude to output. */
+        for(j = 0;j < MAX_EFFECT_CHANNELS;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] += SamplesIn[j][base+i] * gains[i] * gain;
+            }
+        }
+
+        base += td;
+    }
+}
+
+
+typedef struct CompressorStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} CompressorStateFactory;
+
+static ALeffectState *CompressorStateFactory_create(CompressorStateFactory *UNUSED(factory))
+{
+    ALcompressorState *state;
+
+    NEW_OBJ0(state, ALcompressorState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(CompressorStateFactory);
+
+EffectStateFactory *CompressorStateFactory_getFactory(void)
+{
+    static CompressorStateFactory CompressorFactory = { { GET_VTABLE2(CompressorStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Compressor state out of range");
+            props->Compressor.OnOff = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid compressor integer property 0x%04x",
+                       param);
+    }
+}
+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 param, ALfloat UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float property 0x%04x", param); }
+void ALcompressor_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float-vector property 0x%04x", param); }
+
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid compressor integer property 0x%04x",
+                       param);
+    }
+}
+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 param, ALfloat *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float property 0x%04x", param); }
+void ALcompressor_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float-vector property 0x%04x", param); }
+
+DEFINE_ALEFFECT_VTABLE(ALcompressor);

Alc/effects/dedicated.c

@@ -0,0 +1,184 @@
+/**
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.h"
+
+
+typedef struct ALdedicatedState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
+    ALfloat TargetGains[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 ALCcontext *context, 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)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALdedicatedState, ALeffectState, state);
+}
+
+static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *state, ALCdevice *UNUSED(device))
+{
+    ALsizei i;
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        state->CurrentGains[i] = 0.0f;
+    return AL_TRUE;
+}
+
+static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALfloat Gain;
+    ALsizei i;
+
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        state->TargetGains[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->TargetGains[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->TargetGains[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;
+            ComputePanGains(&device->Dry, coeffs, Gain, state->TargetGains);
+        }
+    }
+}
+
+static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    MixSamples(SamplesIn[0], NumChannels, SamplesOut, state->CurrentGains,
+               state->TargetGains, SamplesToDo, 0, SamplesToDo);
+}
+
+
+typedef struct DedicatedStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} DedicatedStateFactory;
+
+ALeffectState *DedicatedStateFactory_create(DedicatedStateFactory *UNUSED(factory))
+{
+    ALdedicatedState *state;
+
+    NEW_OBJ0(state, ALdedicatedState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(DedicatedStateFactory);
+
+
+EffectStateFactory *DedicatedStateFactory_getFactory(void)
+{
+    static DedicatedStateFactory DedicatedFactory = { { GET_VTABLE2(DedicatedStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &DedicatedFactory);
+}
+
+
+void ALdedicated_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer property 0x%04x", param); }
+void ALdedicated_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer-vector property 0x%04x", param); }
+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)))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Dedicated gain out of range");
+            props->Dedicated.Gain = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid dedicated float property 0x%04x", param);
+    }
+}
+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 param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer property 0x%04x", param); }
+void ALdedicated_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer-vector property 0x%04x", param); }
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid dedicated float property 0x%04x", param);
+    }
+}
+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 +1,286 @@
+/**
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.h"
+
+
+typedef struct ALdistortionState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    /* Effect gains for each channel */
+    ALfloat Gain[MAX_OUTPUT_CHANNELS];
+
+    /* Effect parameters */
+    BiquadFilter lowpass;
+    BiquadFilter bandpass;
+    ALfloat attenuation;
+    ALfloat edge_coeff;
+
+    ALfloat Buffer[2][BUFFERSIZE];
+} ALdistortionState;
+
+static ALvoid ALdistortionState_Destruct(ALdistortionState *state);
+static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *device);
+static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCcontext *context, 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);
+}
+
+static ALvoid ALdistortionState_Destruct(ALdistortionState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *UNUSED(device))
+{
+    BiquadFilter_clear(&state->lowpass);
+    BiquadFilter_clear(&state->bandpass);
+    return AL_TRUE;
+}
+
+static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALfloat frequency = (ALfloat)device->Frequency;
+    ALfloat coeffs[MAX_AMBI_COEFFS];
+    ALfloat bandwidth;
+    ALfloat cutoff;
+    ALfloat edge;
+
+    /* 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.
+     */
+    BiquadFilter_setParams(&state->lowpass, BiquadType_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);
+    BiquadFilter_setParams(&state->bandpass, BiquadType_BandPass, 1.0f,
+        cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
+    );
+
+    CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
+    ComputePanGains(&device->Dry, coeffs, slot->Params.Gain*props->Distortion.Gain, state->Gain);
+}
+
+static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALfloat (*restrict buffer)[BUFFERSIZE] = state->Buffer;
+    const ALfloat fc = state->edge_coeff;
+    ALsizei base;
+    ALsizei i, k;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        /* 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.
+         */
+        ALsizei todo = mini(BUFFERSIZE, (SamplesToDo-base) * 4);
+
+        /* Fill oversample buffer using zero stuffing. Multiply the sample by
+         * the amount of oversampling to maintain the signal's power.
+         */
+        for(i = 0;i < todo;i++)
+            buffer[0][i] = !(i&3) ? SamplesIn[0][(i>>2)+base] * 4.0f : 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.
+         */
+        BiquadFilter_process(&state->lowpass, buffer[1], buffer[0], todo);
+
+        /* 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(i = 0;i < todo;i++)
+        {
+            ALfloat smp = buffer[1][i];
+
+            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][i] = smp;
+        }
+
+        /* Third step, do bandpass filtering of distorted signal. */
+        BiquadFilter_process(&state->bandpass, buffer[1], buffer[0], todo);
+
+        todo >>= 2;
+        for(k = 0;k < NumChannels;k++)
+        {
+            /* Fourth step, final, do attenuation and perform decimation,
+             * storing only one sample out of four.
+             */
+            ALfloat gain = state->Gain[k];
+            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+                continue;
+
+            for(i = 0;i < todo;i++)
+                SamplesOut[k][base+i] += gain * buffer[1][i*4];
+        }
+
+        base += todo;
+    }
+}
+
+
+typedef struct DistortionStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} DistortionStateFactory;
+
+static ALeffectState *DistortionStateFactory_create(DistortionStateFactory *UNUSED(factory))
+{
+    ALdistortionState *state;
+
+    NEW_OBJ0(state, ALdistortionState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(DistortionStateFactory);
+
+
+EffectStateFactory *DistortionStateFactory_getFactory(void)
+{
+    static DistortionStateFactory DistortionFactory = { { GET_VTABLE2(DistortionStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &DistortionFactory);
+}
+
+
+void ALdistortion_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); }
+void ALdistortion_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); }
+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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion edge out of range");
+            props->Distortion.Edge = val;
+            break;
+
+        case AL_DISTORTION_GAIN:
+            if(!(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion gain out of range");
+            props->Distortion.Gain = val;
+            break;
+
+        case AL_DISTORTION_LOWPASS_CUTOFF:
+            if(!(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion low-pass cutoff out of range");
+            props->Distortion.LowpassCutoff = val;
+            break;
+
+        case AL_DISTORTION_EQCENTER:
+            if(!(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ center out of range");
+            props->Distortion.EQCenter = val;
+            break;
+
+        case AL_DISTORTION_EQBANDWIDTH:
+            if(!(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ bandwidth out of range");
+            props->Distortion.EQBandwidth = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x",
+                       param);
+    }
+}
+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 param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); }
+void ALdistortion_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); }
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x",
+                       param);
+    }
+}
+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 +1,310 @@
+/**
+ * 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"
+#include "filters/defs.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 */
+    struct {
+        ALfloat Current[MAX_OUTPUT_CHANNELS];
+        ALfloat Target[MAX_OUTPUT_CHANNELS];
+    } Gains[2];
+
+    ALfloat FeedGain;
+
+    BiquadFilter Filter;
+} ALechoState;
+
+static ALvoid ALechoState_Destruct(ALechoState *state);
+static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device);
+static ALvoid ALechoState_update(ALechoState *state, const ALCcontext *context, 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;
+
+    BiquadFilter_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;
+
+    // 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 = float2int(AL_ECHO_MAX_DELAY*Device->Frequency + 0.5f) +
+             float2int(AL_ECHO_MAX_LRDELAY*Device->Frequency + 0.5f);
+    maxlen = NextPowerOf2(maxlen);
+    if(maxlen <= 0) return AL_FALSE;
+
+    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;
+    }
+
+    memset(state->SampleBuffer, 0, state->BufferLength*sizeof(ALfloat));
+    memset(state->Gains, 0, sizeof(state->Gains));
+
+    return AL_TRUE;
+}
+
+static ALvoid ALechoState_update(ALechoState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALuint frequency = device->Frequency;
+    ALfloat coeffs[MAX_AMBI_COEFFS];
+    ALfloat gainhf, lrpan, spread;
+
+    state->Tap[0].delay = maxi(float2int(props->Echo.Delay*frequency + 0.5f), 1);
+    state->Tap[1].delay = float2int(props->Echo.LRDelay*frequency + 0.5f);
+    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;
+
+    gainhf = maxf(1.0f - props->Echo.Damping, 0.0625f); /* Limit -24dB */
+    BiquadFilter_setParams(&state->Filter, BiquadType_HighShelf,
+        gainhf, LOWPASSFREQREF/frequency, calc_rcpQ_from_slope(gainhf, 1.0f)
+    );
+
+    /* First tap panning */
+    CalcAngleCoeffs(-F_PI_2*lrpan, 0.0f, spread, coeffs);
+    ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->Gains[0].Target);
+
+    /* Second tap panning */
+    CalcAngleCoeffs( F_PI_2*lrpan, 0.0f, spread, coeffs);
+    ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->Gains[1].Target);
+}
+
+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;
+    ALfloat *restrict delaybuf = state->SampleBuffer;
+    ALsizei offset = state->Offset;
+    ALfloat z1, z2, in, out;
+    ALsizei base;
+    ALsizei c, i;
+
+    z1 = state->Filter.z1;
+    z2 = state->Filter.z2;
+    for(base = 0;base < SamplesToDo;)
+    {
+        alignas(16) ALfloat temps[2][128];
+        ALsizei td = mini(128, SamplesToDo-base);
+
+        for(i = 0;i < td;i++)
+        {
+            /* Feed the delay buffer's input first. */
+            delaybuf[offset&mask] = SamplesIn[0][i+base];
+
+            /* First tap */
+            temps[0][i] = delaybuf[(offset-tap1) & mask];
+            /* Second tap */
+            temps[1][i] = delaybuf[(offset-tap2) & mask];
+
+            /* Apply damping to the second tap, then add it to the buffer with
+             * feedback attenuation.
+             */
+            in = temps[1][i];
+            out = in*state->Filter.b0 + z1;
+            z1 = in*state->Filter.b1 - out*state->Filter.a1 + z2;
+            z2 = in*state->Filter.b2 - out*state->Filter.a2;
+
+            delaybuf[offset&mask] += out * state->FeedGain;
+            offset++;
+        }
+
+        for(c = 0;c < 2;c++)
+            MixSamples(temps[c], NumChannels, SamplesOut, state->Gains[c].Current,
+                       state->Gains[c].Target, SamplesToDo-base, base, td);
+
+        base += td;
+    }
+    state->Filter.z1 = z1;
+    state->Filter.z2 = z2;
+
+    state->Offset = offset;
+}
+
+
+typedef struct EchoStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} EchoStateFactory;
+
+ALeffectState *EchoStateFactory_create(EchoStateFactory *UNUSED(factory))
+{
+    ALechoState *state;
+
+    NEW_OBJ0(state, ALechoState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(EchoStateFactory);
+
+EffectStateFactory *EchoStateFactory_getFactory(void)
+{
+    static EchoStateFactory EchoFactory = { { GET_VTABLE2(EchoStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &EchoFactory);
+}
+
+
+void ALecho_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer property 0x%04x", param); }
+void ALecho_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer-vector property 0x%04x", param); }
+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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo delay out of range");
+            props->Echo.Delay = val;
+            break;
+
+        case AL_ECHO_LRDELAY:
+            if(!(val >= AL_ECHO_MIN_LRDELAY && val <= AL_ECHO_MAX_LRDELAY))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo LR delay out of range");
+            props->Echo.LRDelay = val;
+            break;
+
+        case AL_ECHO_DAMPING:
+            if(!(val >= AL_ECHO_MIN_DAMPING && val <= AL_ECHO_MAX_DAMPING))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo damping out of range");
+            props->Echo.Damping = val;
+            break;
+
+        case AL_ECHO_FEEDBACK:
+            if(!(val >= AL_ECHO_MIN_FEEDBACK && val <= AL_ECHO_MAX_FEEDBACK))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo feedback out of range");
+            props->Echo.Feedback = val;
+            break;
+
+        case AL_ECHO_SPREAD:
+            if(!(val >= AL_ECHO_MIN_SPREAD && val <= AL_ECHO_MAX_SPREAD))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo spread out of range");
+            props->Echo.Spread = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid echo float property 0x%04x", param);
+    }
+}
+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 param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer property 0x%04x", param); }
+void ALecho_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer-vector property 0x%04x", param); }
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid echo float property 0x%04x", param);
+    }
+}
+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 +1,355 @@
+/**
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.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                   */
+
+
+typedef struct ALequalizerState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    struct {
+        /* Effect parameters */
+        BiquadFilter filter[4];
+
+        /* Effect gains for each channel */
+        ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
+        ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
+    } Chans[MAX_EFFECT_CHANNELS];
+
+    ALfloat SampleBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE];
+} ALequalizerState;
+
+static ALvoid ALequalizerState_Destruct(ALequalizerState *state);
+static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *device);
+static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext *context, 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)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALequalizerState, ALeffectState, state);
+}
+
+static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *UNUSED(device))
+{
+    ALsizei i, j;
+
+    for(i = 0; i < MAX_EFFECT_CHANNELS;i++)
+    {
+        for(j = 0;j < 4;j++)
+            BiquadFilter_clear(&state->Chans[i].filter[j]);
+        for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
+            state->Chans[i].CurrentGains[j] = 0.0f;
+    }
+    return AL_TRUE;
+}
+
+static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALfloat frequency = (ALfloat)device->Frequency;
+    ALfloat gain, f0norm;
+    ALuint 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 */
+    f0norm = props->Equalizer.LowCutoff/frequency;
+    BiquadFilter_setParams(&state->Chans[0].filter[0], BiquadType_LowShelf,
+        gain, f0norm, calc_rcpQ_from_slope(gain, 0.75f)
+    );
+
+    gain = maxf(props->Equalizer.Mid1Gain, 0.0625f);
+    f0norm = props->Equalizer.Mid1Center/frequency;
+    BiquadFilter_setParams(&state->Chans[0].filter[1], BiquadType_Peaking,
+        gain, f0norm, calc_rcpQ_from_bandwidth(
+            f0norm, props->Equalizer.Mid1Width
+        )
+    );
+
+    gain = maxf(props->Equalizer.Mid2Gain, 0.0625f);
+    f0norm = props->Equalizer.Mid2Center/frequency;
+    BiquadFilter_setParams(&state->Chans[0].filter[2], BiquadType_Peaking,
+        gain, f0norm, calc_rcpQ_from_bandwidth(
+            f0norm, props->Equalizer.Mid2Width
+        )
+    );
+
+    gain = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f);
+    f0norm = props->Equalizer.HighCutoff/frequency;
+    BiquadFilter_setParams(&state->Chans[0].filter[3], BiquadType_HighShelf,
+        gain, f0norm, calc_rcpQ_from_slope(gain, 0.75f)
+    );
+
+    /* Copy the filter coefficients for the other input channels. */
+    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
+    {
+        BiquadFilter_copyParams(&state->Chans[i].filter[0], &state->Chans[0].filter[0]);
+        BiquadFilter_copyParams(&state->Chans[i].filter[1], &state->Chans[0].filter[1]);
+        BiquadFilter_copyParams(&state->Chans[i].filter[2], &state->Chans[0].filter[2]);
+        BiquadFilter_copyParams(&state->Chans[i].filter[3], &state->Chans[0].filter[3]);
+    }
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+        ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain,
+                        state->Chans[i].TargetGains);
+}
+
+static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    ALfloat (*restrict temps)[BUFFERSIZE] = state->SampleBuffer;
+    ALsizei c;
+
+    for(c = 0;c < MAX_EFFECT_CHANNELS;c++)
+    {
+        BiquadFilter_process(&state->Chans[c].filter[0], temps[0], SamplesIn[c], SamplesToDo);
+        BiquadFilter_process(&state->Chans[c].filter[1], temps[1], temps[0], SamplesToDo);
+        BiquadFilter_process(&state->Chans[c].filter[2], temps[2], temps[1], SamplesToDo);
+        BiquadFilter_process(&state->Chans[c].filter[3], temps[3], temps[2], SamplesToDo);
+
+        MixSamples(temps[3], NumChannels, SamplesOut,
+            state->Chans[c].CurrentGains, state->Chans[c].TargetGains,
+            SamplesToDo, 0, SamplesToDo
+        );
+    }
+}
+
+
+typedef struct EqualizerStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} EqualizerStateFactory;
+
+ALeffectState *EqualizerStateFactory_create(EqualizerStateFactory *UNUSED(factory))
+{
+    ALequalizerState *state;
+
+    NEW_OBJ0(state, ALequalizerState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(EqualizerStateFactory);
+
+EffectStateFactory *EqualizerStateFactory_getFactory(void)
+{
+    static EqualizerStateFactory EqualizerFactory = { { GET_VTABLE2(EqualizerStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &EqualizerFactory);
+}
+
+
+void ALequalizer_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer property 0x%04x", param); }
+void ALequalizer_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer-vector property 0x%04x", param); }
+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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer low-band gain out of range");
+            props->Equalizer.LowGain = val;
+            break;
+
+        case AL_EQUALIZER_LOW_CUTOFF:
+            if(!(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer low-band cutoff out of range");
+            props->Equalizer.LowCutoff = val;
+            break;
+
+        case AL_EQUALIZER_MID1_GAIN:
+            if(!(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid1-band gain out of range");
+            props->Equalizer.Mid1Gain = val;
+            break;
+
+        case AL_EQUALIZER_MID1_CENTER:
+            if(!(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid1-band center out of range");
+            props->Equalizer.Mid1Center = val;
+            break;
+
+        case AL_EQUALIZER_MID1_WIDTH:
+            if(!(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid1-band width out of range");
+            props->Equalizer.Mid1Width = val;
+            break;
+
+        case AL_EQUALIZER_MID2_GAIN:
+            if(!(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid2-band gain out of range");
+            props->Equalizer.Mid2Gain = val;
+            break;
+
+        case AL_EQUALIZER_MID2_CENTER:
+            if(!(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid2-band center out of range");
+            props->Equalizer.Mid2Center = val;
+            break;
+
+        case AL_EQUALIZER_MID2_WIDTH:
+            if(!(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer mid2-band width out of range");
+            props->Equalizer.Mid2Width = val;
+            break;
+
+        case AL_EQUALIZER_HIGH_GAIN:
+            if(!(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer high-band gain out of range");
+            props->Equalizer.HighGain = val;
+            break;
+
+        case AL_EQUALIZER_HIGH_CUTOFF:
+            if(!(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Equalizer high-band cutoff out of range");
+            props->Equalizer.HighCutoff = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid equalizer float property 0x%04x", param);
+    }
+}
+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 param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer property 0x%04x", param); }
+void ALequalizer_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid equalizer integer-vector property 0x%04x", param); }
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid equalizer float property 0x%04x", param);
+    }
+}
+void ALequalizer_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{ ALequalizer_getParamf(effect, context, param, vals); }
+
+DEFINE_ALEFFECT_VTABLE(ALequalizer);

Alc/effects/fshifter.c

@@ -0,0 +1,329 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2018 by Raul Herraiz.
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.h"
+
+#include "alcomplex.h"
+
+#define HIL_SIZE 1024
+#define OVERSAMP (1<<2)
+
+#define HIL_STEP     (HIL_SIZE / OVERSAMP)
+#define FIFO_LATENCY (HIL_STEP * (OVERSAMP-1))
+
+
+typedef struct ALfshifterState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    /* Effect parameters */
+    ALsizei  count;
+    ALsizei  PhaseStep;
+    ALsizei  Phase;
+    ALdouble ld_sign;
+
+    /*Effects buffers*/ 
+    ALfloat   InFIFO[HIL_SIZE];
+    ALcomplex OutFIFO[HIL_SIZE];
+    ALcomplex OutputAccum[HIL_SIZE];
+    ALcomplex Analytic[HIL_SIZE];
+    ALcomplex Outdata[BUFFERSIZE];
+
+    alignas(16) ALfloat BufferOut[BUFFERSIZE];
+
+    /* Effect gains for each output channel */
+    ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
+    ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
+} ALfshifterState;
+
+static ALvoid ALfshifterState_Destruct(ALfshifterState *state);
+static ALboolean ALfshifterState_deviceUpdate(ALfshifterState *state, ALCdevice *device);
+static ALvoid ALfshifterState_update(ALfshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
+static ALvoid ALfshifterState_process(ALfshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALfshifterState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALfshifterState);
+
+/* Define a Hann window, used to filter the HIL input and output. */
+alignas(16) static ALdouble HannWindow[HIL_SIZE];
+
+static void InitHannWindow(void)
+{
+    ALsizei i;
+
+    /* Create lookup table of the Hann window for the desired size, i.e. HIL_SIZE */
+    for(i = 0;i < HIL_SIZE>>1;i++)
+    {
+        ALdouble val = sin(M_PI * (ALdouble)i / (ALdouble)(HIL_SIZE-1));
+        HannWindow[i] = HannWindow[HIL_SIZE-1-i] = val * val;
+    }
+}
+
+static alonce_flag HannInitOnce = AL_ONCE_FLAG_INIT;
+
+static void ALfshifterState_Construct(ALfshifterState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALfshifterState, ALeffectState, state);
+
+    alcall_once(&HannInitOnce, InitHannWindow);
+}
+
+static ALvoid ALfshifterState_Destruct(ALfshifterState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALfshifterState_deviceUpdate(ALfshifterState *state, ALCdevice *UNUSED(device))
+{
+    /* (Re-)initializing parameters and clear the buffers. */
+    state->count     = FIFO_LATENCY;
+    state->PhaseStep = 0;
+    state->Phase     = 0;
+    state->ld_sign   = 1.0;
+
+    memset(state->InFIFO,      0, sizeof(state->InFIFO));
+    memset(state->OutFIFO,     0, sizeof(state->OutFIFO));
+    memset(state->OutputAccum, 0, sizeof(state->OutputAccum));
+    memset(state->Analytic,    0, sizeof(state->Analytic));
+
+    memset(state->CurrentGains, 0, sizeof(state->CurrentGains));
+    memset(state->TargetGains,  0, sizeof(state->TargetGains));
+
+    return AL_TRUE;
+}
+
+static ALvoid ALfshifterState_update(ALfshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALfloat coeffs[MAX_AMBI_COEFFS];
+    ALfloat step;
+
+    step = props->Fshifter.Frequency / (ALfloat)device->Frequency;
+    state->PhaseStep = fastf2i(minf(step, 0.5f) * FRACTIONONE);
+
+    switch(props->Fshifter.LeftDirection)
+    {
+        case AL_FREQUENCY_SHIFTER_DIRECTION_DOWN:
+            state->ld_sign = -1.0;
+            break;
+
+        case AL_FREQUENCY_SHIFTER_DIRECTION_UP:
+            state->ld_sign = 1.0;
+            break;
+
+        case AL_FREQUENCY_SHIFTER_DIRECTION_OFF:
+            state->Phase = 0;
+            state->PhaseStep = 0;
+            break;
+    }
+
+    CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
+    ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains);
+}
+
+static ALvoid ALfshifterState_process(ALfshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    static const ALcomplex complex_zero = { 0.0, 0.0 };
+    ALfloat *restrict BufferOut = state->BufferOut;
+    ALsizei j, k, base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALsizei todo = mini(HIL_SIZE-state->count, SamplesToDo-base);
+
+        ASSUME(todo > 0);
+
+        /* Fill FIFO buffer with samples data */
+        k = state->count;
+        for(j = 0;j < todo;j++,k++)
+        {
+            state->InFIFO[k] = SamplesIn[0][base+j];
+            state->Outdata[base+j]  = state->OutFIFO[k-FIFO_LATENCY];
+        }
+        state->count += todo;
+        base += todo;
+
+        /* Check whether FIFO buffer is filled */
+        if(state->count < HIL_SIZE) continue;
+
+        state->count = FIFO_LATENCY;
+
+        /* Real signal windowing and store in Analytic buffer */
+        for(k = 0;k < HIL_SIZE;k++)
+        {
+            state->Analytic[k].Real = state->InFIFO[k] * HannWindow[k];
+            state->Analytic[k].Imag = 0.0;
+        }
+
+        /* Processing signal by Discrete Hilbert Transform (analytical signal). */
+        complex_hilbert(state->Analytic, HIL_SIZE);
+
+        /* Windowing and add to output accumulator */
+        for(k = 0;k < HIL_SIZE;k++)
+        {
+            state->OutputAccum[k].Real += 2.0/OVERSAMP*HannWindow[k]*state->Analytic[k].Real;
+            state->OutputAccum[k].Imag += 2.0/OVERSAMP*HannWindow[k]*state->Analytic[k].Imag;
+        }
+
+        /* Shift accumulator, input & output FIFO */
+        for(k = 0;k < HIL_STEP;k++) state->OutFIFO[k] = state->OutputAccum[k];
+        for(j = 0;k < HIL_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k];
+        for(;j < HIL_SIZE;j++) state->OutputAccum[j] = complex_zero;
+        for(k = 0;k < FIFO_LATENCY;k++)
+            state->InFIFO[k] = state->InFIFO[k+HIL_STEP];
+    }
+
+    /* Process frequency shifter using the analytic signal obtained. */
+    for(k = 0;k < SamplesToDo;k++)
+    {
+        ALdouble phase = state->Phase * ((1.0/FRACTIONONE) * 2.0*M_PI);
+        BufferOut[k] = (ALfloat)(state->Outdata[k].Real*cos(phase) +
+                                 state->Outdata[k].Imag*sin(phase)*state->ld_sign);
+
+        state->Phase += state->PhaseStep;
+        state->Phase &= FRACTIONMASK;
+    }
+
+    /* Now, mix the processed sound data to the output. */
+    MixSamples(BufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains,
+               maxi(SamplesToDo, 512), 0, SamplesToDo);
+}
+
+typedef struct FshifterStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} FshifterStateFactory;
+
+static ALeffectState *FshifterStateFactory_create(FshifterStateFactory *UNUSED(factory))
+{
+    ALfshifterState *state;
+
+    NEW_OBJ0(state, ALfshifterState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(FshifterStateFactory);
+
+EffectStateFactory *FshifterStateFactory_getFactory(void)
+{
+    static FshifterStateFactory FshifterFactory = { { GET_VTABLE2(FshifterStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &FshifterFactory);
+}
+
+void ALfshifter_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FREQUENCY_SHIFTER_FREQUENCY:
+            if(!(val >= AL_FREQUENCY_SHIFTER_MIN_FREQUENCY && val <= AL_FREQUENCY_SHIFTER_MAX_FREQUENCY))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter frequency out of range");
+            props->Fshifter.Frequency = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter float property 0x%04x", param);
+    }
+}
+
+void ALfshifter_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    ALfshifter_setParamf(effect, context, param, vals[0]);
+}
+
+void ALfshifter_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FREQUENCY_SHIFTER_LEFT_DIRECTION:
+            if(!(val >= AL_FREQUENCY_SHIFTER_MIN_LEFT_DIRECTION && val <= AL_FREQUENCY_SHIFTER_MAX_LEFT_DIRECTION))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter left direction out of range");
+            props->Fshifter.LeftDirection = val;
+            break;
+
+        case AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION:
+            if(!(val >= AL_FREQUENCY_SHIFTER_MIN_RIGHT_DIRECTION && val <= AL_FREQUENCY_SHIFTER_MAX_RIGHT_DIRECTION))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter right direction out of range");
+            props->Fshifter.RightDirection = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter integer property 0x%04x", param);
+    }
+}
+void ALfshifter_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALfshifter_setParami(effect, context, param, vals[0]);
+}
+
+void ALfshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FREQUENCY_SHIFTER_LEFT_DIRECTION:
+            *val = props->Fshifter.LeftDirection;
+            break;
+        case AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION:
+            *val = props->Fshifter.RightDirection;
+            break;
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter integer property 0x%04x", param);
+    }
+}
+void ALfshifter_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALfshifter_getParami(effect, context, param, vals);
+}
+
+void ALfshifter_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_FREQUENCY_SHIFTER_FREQUENCY:
+            *val = props->Fshifter.Frequency;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter float property 0x%04x", param);
+    }
+
+}
+
+void ALfshifter_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    ALfshifter_getParamf(effect, context, param, vals);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALfshifter);

Alc/effects/modulator.c

@@ -0,0 +1,307 @@
+/**
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.h"
+
+
+#define MAX_UPDATE_SAMPLES 128
+
+typedef struct ALmodulatorState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    void (*GetSamples)(ALfloat*, ALsizei, const ALsizei, ALsizei);
+
+    ALsizei index;
+    ALsizei step;
+
+    struct {
+        BiquadFilter Filter;
+
+        ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
+        ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
+    } Chans[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 ALCcontext *context, 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((ALfloat)index * (F_TAU / WAVEFORM_FRACONE));
+}
+
+static inline ALfloat Saw(ALsizei index)
+{
+    return (ALfloat)index*(2.0f/WAVEFORM_FRACONE) - 1.0f;
+}
+
+static inline ALfloat Square(ALsizei index)
+{
+    return (ALfloat)(((index>>(WAVEFORM_FRACBITS-2))&2) - 1);
+}
+
+static inline ALfloat One(ALsizei UNUSED(index))
+{
+    return 1.0f;
+}
+
+#define DECL_TEMPLATE(func)                                                   \
+static void Modulate##func(ALfloat *restrict dst, ALsizei index,              \
+                           const ALsizei step, ALsizei todo)                  \
+{                                                                             \
+    ALsizei i;                                                                \
+    for(i = 0;i < todo;i++)                                                   \
+    {                                                                         \
+        index += step;                                                        \
+        index &= WAVEFORM_FRACMASK;                                           \
+        dst[i] = func(index);                                                 \
+    }                                                                         \
+}
+
+DECL_TEMPLATE(Sin)
+DECL_TEMPLATE(Saw)
+DECL_TEMPLATE(Square)
+DECL_TEMPLATE(One)
+
+#undef DECL_TEMPLATE
+
+
+static void ALmodulatorState_Construct(ALmodulatorState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALmodulatorState, ALeffectState, state);
+
+    state->index = 0;
+    state->step = 1;
+}
+
+static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *UNUSED(device))
+{
+    ALsizei i, j;
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+    {
+        BiquadFilter_clear(&state->Chans[i].Filter);
+        for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
+            state->Chans[i].CurrentGains[j] = 0.0f;
+    }
+    return AL_TRUE;
+}
+
+static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALfloat f0norm;
+    ALsizei i;
+
+    state->step = fastf2i(props->Modulator.Frequency / (ALfloat)device->Frequency *
+                          WAVEFORM_FRACONE);
+    state->step = clampi(state->step, 0, WAVEFORM_FRACONE-1);
+
+    if(state->step == 0)
+        state->GetSamples = ModulateOne;
+    else if(props->Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
+        state->GetSamples = ModulateSin;
+    else if(props->Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
+        state->GetSamples = ModulateSaw;
+    else /*if(Slot->Params.EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)*/
+        state->GetSamples = ModulateSquare;
+
+    f0norm = props->Modulator.HighPassCutoff / (ALfloat)device->Frequency;
+    f0norm = clampf(f0norm, 1.0f/512.0f, 0.49f);
+    /* Bandwidth value is constant in octaves. */
+    BiquadFilter_setParams(&state->Chans[0].Filter, BiquadType_HighPass, 1.0f,
+                           f0norm, calc_rcpQ_from_bandwidth(f0norm, 0.75f));
+    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
+        BiquadFilter_copyParams(&state->Chans[i].Filter, &state->Chans[0].Filter);
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
+    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
+        ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain,
+                        state->Chans[i].TargetGains);
+}
+
+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 base;
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        alignas(16) ALfloat modsamples[MAX_UPDATE_SAMPLES];
+        ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base);
+        ALsizei c, i;
+
+        state->GetSamples(modsamples, state->index, step, td);
+        state->index += (step*td) & WAVEFORM_FRACMASK;
+        state->index &= WAVEFORM_FRACMASK;
+
+        for(c = 0;c < MAX_EFFECT_CHANNELS;c++)
+        {
+            alignas(16) ALfloat temps[MAX_UPDATE_SAMPLES];
+
+            BiquadFilter_process(&state->Chans[c].Filter, temps, &SamplesIn[c][base], td);
+            for(i = 0;i < td;i++)
+                temps[i] *= modsamples[i];
+
+            MixSamples(temps, NumChannels, SamplesOut, state->Chans[c].CurrentGains,
+                       state->Chans[c].TargetGains, SamplesToDo-base, base, td);
+        }
+
+        base += td;
+    }
+}
+
+
+typedef struct ModulatorStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} ModulatorStateFactory;
+
+static ALeffectState *ModulatorStateFactory_create(ModulatorStateFactory *UNUSED(factory))
+{
+    ALmodulatorState *state;
+
+    NEW_OBJ0(state, ALmodulatorState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(ModulatorStateFactory);
+
+EffectStateFactory *ModulatorStateFactory_getFactory(void)
+{
+    static ModulatorStateFactory ModulatorFactory = { { GET_VTABLE2(ModulatorStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Modulator frequency out of range");
+            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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Modulator high-pass cutoff out of range");
+            props->Modulator.HighPassCutoff = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid modulator float property 0x%04x", param);
+    }
+}
+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))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid modulator waveform");
+            props->Modulator.Waveform = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid modulator integer property 0x%04x", param);
+    }
+}
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid modulator integer property 0x%04x", param);
+    }
+}
+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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid modulator float property 0x%04x", param);
+    }
+}
+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 ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
+static ALvoid ALnullState_process(ALnullState *state, ALsizei samplesToDo, const ALfloat (*restrict samplesIn)[BUFFERSIZE], ALfloat (*restrict samplesOut)[BUFFERSIZE], ALsizei mumChannels);
+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 ALCcontext* UNUSED(context), 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 NullStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} NullStateFactory;
+
+/* Creates ALeffectState objects of the appropriate type. */
+ALeffectState *NullStateFactory_create(NullStateFactory *UNUSED(factory))
+{
+    ALnullState *state;
+
+    NEW_OBJ0(state, ALnullState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+/* Define the EffectStateFactory vtable for this type. */
+DEFINE_EFFECTSTATEFACTORY_VTABLE(NullStateFactory);
+
+EffectStateFactory *NullStateFactory_getFactory(void)
+{
+    static NullStateFactory NullFactory = { { GET_VTABLE2(NullStateFactory, EffectStateFactory) } };
+    return STATIC_CAST(EffectStateFactory, &NullFactory);
+}
+
+
+void ALnull_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer property 0x%04x", param);
+    }
+}
+void ALnull_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint* UNUSED(vals))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer-vector property 0x%04x", param);
+    }
+}
+void ALnull_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect float property 0x%04x", param);
+    }
+}
+void ALnull_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat* UNUSED(vals))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect float-vector property 0x%04x", param);
+    }
+}
+
+void ALnull_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(val))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer property 0x%04x", param);
+    }
+}
+void ALnull_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(vals))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer-vector property 0x%04x", param);
+    }
+}
+void ALnull_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(val))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect float property 0x%04x", param);
+    }
+}
+void ALnull_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(vals))
+{
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid null effect float-vector property 0x%04x", param);
+    }
+}
+
+DEFINE_ALEFFECT_VTABLE(ALnull);

Alc/effects/pshifter.c

@@ -0,0 +1,441 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2018 by Raul Herraiz.
+ * 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 "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+#include "filters/defs.h"
+
+#include "alcomplex.h"
+
+
+#define STFT_SIZE      1024
+#define STFT_HALF_SIZE (STFT_SIZE>>1)
+#define OVERSAMP       (1<<2)
+
+#define STFT_STEP    (STFT_SIZE / OVERSAMP)
+#define FIFO_LATENCY (STFT_STEP * (OVERSAMP-1))
+
+
+typedef struct ALphasor {
+    ALdouble Amplitude;
+    ALdouble Phase;
+} ALphasor;
+
+typedef struct ALFrequencyDomain {
+    ALdouble Amplitude;
+    ALdouble Frequency;
+} ALfrequencyDomain;
+
+
+typedef struct ALpshifterState {
+    DERIVE_FROM_TYPE(ALeffectState);
+
+    /* Effect parameters */
+    ALsizei count;
+    ALsizei PitchShiftI;
+    ALfloat PitchShift;
+    ALfloat FreqPerBin;
+
+    /*Effects buffers*/
+    ALfloat InFIFO[STFT_SIZE];
+    ALfloat OutFIFO[STFT_STEP];
+    ALdouble LastPhase[STFT_HALF_SIZE+1];
+    ALdouble SumPhase[STFT_HALF_SIZE+1];
+    ALdouble OutputAccum[STFT_SIZE];
+
+    ALcomplex FFTbuffer[STFT_SIZE];
+
+    ALfrequencyDomain Analysis_buffer[STFT_HALF_SIZE+1];
+    ALfrequencyDomain Syntesis_buffer[STFT_HALF_SIZE+1];
+
+    alignas(16) ALfloat BufferOut[BUFFERSIZE];
+
+    /* Effect gains for each output channel */
+    ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
+    ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
+} ALpshifterState;
+
+static ALvoid ALpshifterState_Destruct(ALpshifterState *state);
+static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device);
+static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
+static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
+DECLARE_DEFAULT_ALLOCATORS(ALpshifterState)
+
+DEFINE_ALEFFECTSTATE_VTABLE(ALpshifterState);
+
+
+/* Define a Hann window, used to filter the STFT input and output. */
+alignas(16) static ALdouble HannWindow[STFT_SIZE];
+
+static void InitHannWindow(void)
+{
+    ALsizei i;
+
+    /* Create lookup table of the Hann window for the desired size, i.e. STFT_SIZE */
+    for(i = 0;i < STFT_SIZE>>1;i++)
+    {
+        ALdouble val = sin(M_PI * (ALdouble)i / (ALdouble)(STFT_SIZE-1));
+        HannWindow[i] = HannWindow[STFT_SIZE-1-i] = val * val;
+    }
+}
+static alonce_flag HannInitOnce = AL_ONCE_FLAG_INIT;
+
+
+static inline ALint double2int(ALdouble d)
+{
+#if ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
+     !defined(__SSE2_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP < 2)
+    ALint sign, shift;
+    ALint64 mant;
+    union {
+        ALdouble d;
+        ALint64 i64;
+    } conv;
+
+    conv.d = d;
+    sign = (conv.i64>>63) | 1;
+    shift = ((conv.i64>>52)&0x7ff) - (1023+52);
+
+    /* Over/underflow */
+    if(UNLIKELY(shift >= 63 || shift < -52))
+        return 0;
+
+    mant = (conv.i64&I64(0xfffffffffffff)) | I64(0x10000000000000);
+    if(LIKELY(shift < 0))
+        return (ALint)(mant >> -shift) * sign;
+    return (ALint)(mant << shift) * sign;
+
+#else
+
+    return (ALint)d;
+#endif
+}
+
+
+/* Converts ALcomplex to ALphasor */
+static inline ALphasor rect2polar(ALcomplex number)
+{
+    ALphasor polar;
+
+    polar.Amplitude = sqrt(number.Real*number.Real + number.Imag*number.Imag);
+    polar.Phase     = atan2(number.Imag, number.Real);
+
+    return polar;
+}
+
+/* Converts ALphasor to ALcomplex */
+static inline ALcomplex polar2rect(ALphasor number)
+{
+    ALcomplex cartesian;
+
+    cartesian.Real = number.Amplitude * cos(number.Phase);
+    cartesian.Imag = number.Amplitude * sin(number.Phase);
+
+    return cartesian;
+}
+
+
+static void ALpshifterState_Construct(ALpshifterState *state)
+{
+    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
+    SET_VTABLE2(ALpshifterState, ALeffectState, state);
+
+    alcall_once(&HannInitOnce, InitHannWindow);
+}
+
+static ALvoid ALpshifterState_Destruct(ALpshifterState *state)
+{
+    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
+}
+
+static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device)
+{
+    /* (Re-)initializing parameters and clear the buffers. */
+    state->count       = FIFO_LATENCY;
+    state->PitchShiftI = FRACTIONONE;
+    state->PitchShift  = 1.0f;
+    state->FreqPerBin  = device->Frequency / (ALfloat)STFT_SIZE;
+
+    memset(state->InFIFO,          0, sizeof(state->InFIFO));
+    memset(state->OutFIFO,         0, sizeof(state->OutFIFO));
+    memset(state->FFTbuffer,       0, sizeof(state->FFTbuffer));
+    memset(state->LastPhase,       0, sizeof(state->LastPhase));
+    memset(state->SumPhase,        0, sizeof(state->SumPhase));
+    memset(state->OutputAccum,     0, sizeof(state->OutputAccum));
+    memset(state->Analysis_buffer, 0, sizeof(state->Analysis_buffer));
+    memset(state->Syntesis_buffer, 0, sizeof(state->Syntesis_buffer));
+
+    memset(state->CurrentGains, 0, sizeof(state->CurrentGains));
+    memset(state->TargetGains,  0, sizeof(state->TargetGains));
+
+    return AL_TRUE;
+}
+
+static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
+{
+    const ALCdevice *device = context->Device;
+    ALfloat coeffs[MAX_AMBI_COEFFS];
+    float pitch;
+
+    pitch = powf(2.0f,
+        (ALfloat)(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f
+    );
+    state->PitchShiftI = fastf2i(pitch*FRACTIONONE);
+    state->PitchShift  = state->PitchShiftI * (1.0f/FRACTIONONE);
+
+    CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
+    ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains);
+}
+
+static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+{
+    /* Pitch shifter engine based on the work of Stephan Bernsee.
+     * http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
+     */
+
+    static const ALdouble expected = M_PI*2.0 / OVERSAMP;
+    const ALdouble freq_per_bin = state->FreqPerBin;
+    ALfloat *restrict bufferOut = state->BufferOut;
+    ALsizei count = state->count;
+    ALsizei i, j, k;
+
+    for(i = 0;i < SamplesToDo;)
+    {
+        do {
+            /* Fill FIFO buffer with samples data */
+            state->InFIFO[count] = SamplesIn[0][i];
+            bufferOut[i] = state->OutFIFO[count - FIFO_LATENCY];
+
+            count++;
+        } while(++i < SamplesToDo && count < STFT_SIZE);
+
+        /* Check whether FIFO buffer is filled */
+        if(count < STFT_SIZE) break;
+        count = FIFO_LATENCY;
+
+        /* Real signal windowing and store in FFTbuffer */
+        for(k = 0;k < STFT_SIZE;k++)
+        {
+            state->FFTbuffer[k].Real = state->InFIFO[k] * HannWindow[k];
+            state->FFTbuffer[k].Imag = 0.0;
+        }
+
+        /* ANALYSIS */
+        /* Apply FFT to FFTbuffer data */
+        complex_fft(state->FFTbuffer, STFT_SIZE, -1.0);
+
+        /* Analyze the obtained data. Since the real FFT is symmetric, only
+         * STFT_HALF_SIZE+1 samples are needed.
+         */
+        for(k = 0;k < STFT_HALF_SIZE+1;k++)
+        {
+            ALphasor component;
+            ALdouble tmp;
+            ALint qpd;
+
+            /* Compute amplitude and phase */
+            component = rect2polar(state->FFTbuffer[k]);
+
+            /* Compute phase difference and subtract expected phase difference */
+            tmp = (component.Phase - state->LastPhase[k]) - k*expected;
+
+            /* Map delta phase into +/- Pi interval */
+            qpd = double2int(tmp / M_PI);
+            tmp -= M_PI * (qpd + (qpd%2));
+
+            /* Get deviation from bin frequency from the +/- Pi interval */
+            tmp /= expected;
+
+            /* Compute the k-th partials' true frequency, twice the amplitude
+             * for maintain the gain (because half of bins are used) and store
+             * amplitude and true frequency in analysis buffer.
+             */
+            state->Analysis_buffer[k].Amplitude = 2.0 * component.Amplitude;
+            state->Analysis_buffer[k].Frequency = (k + tmp) * freq_per_bin;
+
+            /* Store actual phase[k] for the calculations in the next frame*/
+            state->LastPhase[k] = component.Phase;
+        }
+
+        /* PROCESSING */
+        /* pitch shifting */
+        for(k = 0;k < STFT_HALF_SIZE+1;k++)
+        {
+            state->Syntesis_buffer[k].Amplitude = 0.0;
+            state->Syntesis_buffer[k].Frequency = 0.0;
+        }
+
+        for(k = 0;k < STFT_HALF_SIZE+1;k++)
+        {
+            j = (k*state->PitchShiftI) >> FRACTIONBITS;
+            if(j >= STFT_HALF_SIZE+1) break;
+
+            state->Syntesis_buffer[j].Amplitude += state->Analysis_buffer[k].Amplitude;
+            state->Syntesis_buffer[j].Frequency  = state->Analysis_buffer[k].Frequency *
+                                                   state->PitchShift;
+        }
+
+        /* SYNTHESIS */
+        /* Synthesis the processing data */
+        for(k = 0;k < STFT_HALF_SIZE+1;k++)
+        {
+            ALphasor component;
+            ALdouble tmp;
+
+            /* Compute bin deviation from scaled freq */
+            tmp = state->Syntesis_buffer[k].Frequency/freq_per_bin - k;
+
+            /* Calculate actual delta phase and accumulate it to get bin phase */
+            state->SumPhase[k] += (k + tmp) * expected;
+
+            component.Amplitude = state->Syntesis_buffer[k].Amplitude;
+            component.Phase     = state->SumPhase[k];
+
+            /* Compute phasor component to cartesian complex number and storage it into FFTbuffer*/
+            state->FFTbuffer[k] = polar2rect(component);
+        }
+        /* zero negative frequencies for recontruct a real signal */
+        for(k = STFT_HALF_SIZE+1;k < STFT_SIZE;k++)
+        {
+            state->FFTbuffer[k].Real = 0.0;
+            state->FFTbuffer[k].Imag = 0.0;
+        }
+
+        /* Apply iFFT to buffer data */
+        complex_fft(state->FFTbuffer, STFT_SIZE, 1.0);
+
+        /* Windowing and add to output */
+        for(k = 0;k < STFT_SIZE;k++)
+            state->OutputAccum[k] += HannWindow[k] * state->FFTbuffer[k].Real /
+                                     (0.5 * STFT_HALF_SIZE * OVERSAMP);
+
+        /* Shift accumulator, input & output FIFO */
+        for(k = 0;k < STFT_STEP;k++) state->OutFIFO[k] = (ALfloat)state->OutputAccum[k];
+        for(j = 0;k < STFT_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k];
+        for(;j < STFT_SIZE;j++) state->OutputAccum[j] = 0.0;
+        for(k = 0;k < FIFO_LATENCY;k++)
+            state->InFIFO[k] = state->InFIFO[k+STFT_STEP];
+    }
+    state->count = count;
+
+    /* Now, mix the processed sound data to the output. */
+    MixSamples(bufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains,
+               maxi(SamplesToDo, 512), 0, SamplesToDo);
+}
+
+typedef struct PshifterStateFactory {
+    DERIVE_FROM_TYPE(EffectStateFactory);
+} PshifterStateFactory;
+
+static ALeffectState *PshifterStateFactory_create(PshifterStateFactory *UNUSED(factory))
+{
+    ALpshifterState *state;
+
+    NEW_OBJ0(state, ALpshifterState)();
+    if(!state) return NULL;
+
+    return STATIC_CAST(ALeffectState, state);
+}
+
+DEFINE_EFFECTSTATEFACTORY_VTABLE(PshifterStateFactory);
+
+EffectStateFactory *PshifterStateFactory_getFactory(void)
+{
+    static PshifterStateFactory PshifterFactory = { { GET_VTABLE2(PshifterStateFactory, EffectStateFactory) } };
+
+    return STATIC_CAST(EffectStateFactory, &PshifterFactory);
+}
+
+
+void ALpshifter_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
+{
+    alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param );
+}
+
+void ALpshifter_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals))
+{
+    alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float-vector property 0x%04x", param );
+}
+
+void ALpshifter_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+    ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_PITCH_SHIFTER_COARSE_TUNE:
+            if(!(val >= AL_PITCH_SHIFTER_MIN_COARSE_TUNE && val <= AL_PITCH_SHIFTER_MAX_COARSE_TUNE))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter coarse tune out of range");
+            props->Pshifter.CoarseTune = val;
+            break;
+
+        case AL_PITCH_SHIFTER_FINE_TUNE:
+            if(!(val >= AL_PITCH_SHIFTER_MIN_FINE_TUNE && val <= AL_PITCH_SHIFTER_MAX_FINE_TUNE))
+                SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter fine tune out of range");
+            props->Pshifter.FineTune = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
+    }
+}
+void ALpshifter_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    ALpshifter_setParami(effect, context, param, vals[0]);
+}
+
+void ALpshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{
+    const ALeffectProps *props = &effect->Props;
+    switch(param)
+    {
+        case AL_PITCH_SHIFTER_COARSE_TUNE:
+            *val = (ALint)props->Pshifter.CoarseTune;
+            break;
+        case AL_PITCH_SHIFTER_FINE_TUNE:
+            *val = (ALint)props->Pshifter.FineTune;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
+    }
+}
+void ALpshifter_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    ALpshifter_getParami(effect, context, param, vals);
+}
+
+void ALpshifter_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(val))
+{
+    alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param);
+}
+
+void ALpshifter_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals))
+{
+    alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float vector-property 0x%04x", param);
+}
+
+DEFINE_ALEFFECT_VTABLE(ALpshifter);

Alc/filters/defs.h

@@ -0,0 +1,112 @@
+#ifndef ALC_FILTER_H
+#define ALC_FILTER_H
+
+#include "AL/al.h"
+#include "math_defs.h"
+
+/* 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 BiquadType {
+    /** EFX-style low-pass filter, specifying a gain and reference frequency. */
+    BiquadType_HighShelf,
+    /** EFX-style high-pass filter, specifying a gain and reference frequency. */
+    BiquadType_LowShelf,
+    /** Peaking filter, specifying a gain and reference frequency. */
+    BiquadType_Peaking,
+
+    /** Low-pass cut-off filter, specifying a cut-off frequency. */
+    BiquadType_LowPass,
+    /** High-pass cut-off filter, specifying a cut-off frequency. */
+    BiquadType_HighPass,
+    /** Band-pass filter, specifying a center frequency. */
+    BiquadType_BandPass,
+} BiquadType;
+
+typedef struct BiquadFilter {
+    ALfloat z1, z2; /* Last two delayed components for direct form II. */
+    ALfloat b0, b1, b2; /* Transfer function coefficients "b" (numerator) */
+    ALfloat a1, a2; /* Transfer function coefficients "a" (denominator; a0 is
+                     * pre-applied). */
+} BiquadFilter;
+/* Currently only a C-based filter process method is implemented. */
+#define BiquadFilter_process BiquadFilter_processC
+
+/**
+ * Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
+ * reference gain and shelf slope parameter.
+ * \param gain 0 < gain
+ * \param slope 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 normalized
+ * reference frequency and bandwidth.
+ * \param f0norm 0 < f0norm < 0.5.
+ * \param bandwidth 0 < bandwidth
+ */
+inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth)
+{
+    ALfloat w0 = F_TAU * f0norm;
+    return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
+}
+
+inline void BiquadFilter_clear(BiquadFilter *filter)
+{
+    filter->z1 = 0.0f;
+    filter->z2 = 0.0f;
+}
+
+/**
+ * Sets up the filter state for the specified filter type and its parameters.
+ *
+ * \param filter The filter object to prepare.
+ * \param type The type of filter for the object to apply.
+ * \param gain The gain for the reference frequency response. Only used by the
+ *             Shelf and Peaking filter types.
+ * \param f0norm The normalized reference frequency (ref_freq / sample_rate).
+ *               This is the center point for the Shelf, Peaking, and BandPass
+ *               filter types, or the cutoff frequency for the LowPass and
+ *               HighPass filter types.
+ * \param rcpQ The reciprocal of the Q coefficient for the filter's transition
+ *             band. Can be generated from calc_rcpQ_from_slope or
+ *             calc_rcpQ_from_bandwidth depending on the available data.
+ */
+void BiquadFilter_setParams(BiquadFilter *filter, BiquadType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ);
+
+inline void BiquadFilter_copyParams(BiquadFilter *restrict dst, const BiquadFilter *restrict src)
+{
+    dst->b0 = src->b0;
+    dst->b1 = src->b1;
+    dst->b2 = src->b2;
+    dst->a1 = src->a1;
+    dst->a2 = src->a2;
+}
+
+void BiquadFilter_processC(BiquadFilter *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples);
+
+inline void BiquadFilter_passthru(BiquadFilter *filter, ALsizei numsamples)
+{
+    if(LIKELY(numsamples >= 2))
+    {
+        filter->z1 = 0.0f;
+        filter->z2 = 0.0f;
+    }
+    else if(numsamples == 1)
+    {
+        filter->z1 = filter->z2;
+        filter->z2 = 0.0f;
+    }
+}
+
+#endif /* ALC_FILTER_H */

Alc/filters/filter.c

@@ -1,35 +1,39 @@
 
 #include "config.h"
 
-#include "biquad.h"
+#include "AL/alc.h"
+#include "AL/al.h"
 
-#include <algorithm>
-#include <cassert>
-#include <cmath>
+#include "alMain.h"
+#include "defs.h"
 
-#include "opthelpers.h"
+extern inline void BiquadFilter_clear(BiquadFilter *filter);
+extern inline void BiquadFilter_copyParams(BiquadFilter *restrict dst, const BiquadFilter *restrict src);
+extern inline void BiquadFilter_passthru(BiquadFilter *filter, ALsizei numsamples);
+extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope);
+extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth);
 
 
-template<typename Real>
-void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Real rcpQ)
+void BiquadFilter_setParams(BiquadFilter *filter, BiquadType type, ALfloat gain, ALfloat f0norm, 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);
 
-    const Real w0{al::MathDefs<Real>::Tau() * f0norm};
-    const Real sin_w0{std::sin(w0)};
-    const Real cos_w0{std::cos(w0)};
-    const Real alpha{sin_w0/2.0f * rcpQ};
-
-    Real sqrtgain_alpha_2;
-    Real a[3]{ 1.0f, 0.0f, 0.0f };
-    Real b[3]{ 1.0f, 0.0f, 0.0f };
+    w0 = F_TAU * f0norm;
+    sin_w0 = sinf(w0);
+    cos_w0 = cosf(w0);
+    alpha = sin_w0/2.0f * rcpQ;
 
     /* Calculate filter coefficients depending on filter type */
     switch(type)
     {
-        case BiquadType::HighShelf:
-            sqrtgain_alpha_2 = 2.0f * std::sqrt(gain) * alpha;
+        case BiquadType_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);
@@ -37,8 +41,8 @@ void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Rea
             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 BiquadType::LowShelf:
-            sqrtgain_alpha_2 = 2.0f * std::sqrt(gain) * alpha;
+        case BiquadType_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);
@@ -46,7 +50,8 @@ void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Rea
             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 BiquadType::Peaking:
+        case BiquadType_Peaking:
+            gain = sqrtf(gain);
             b[0] =  1.0f + alpha * gain;
             b[1] = -2.0f * cos_w0;
             b[2] =  1.0f - alpha * gain;
@@ -55,7 +60,7 @@ void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Rea
             a[2] =  1.0f - alpha / gain;
             break;
 
-        case BiquadType::LowPass:
+        case BiquadType_LowPass:
             b[0] = (1.0f - cos_w0) / 2.0f;
             b[1] =  1.0f - cos_w0;
             b[2] = (1.0f - cos_w0) / 2.0f;
@@ -63,7 +68,7 @@ void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Rea
             a[1] = -2.0f * cos_w0;
             a[2] =  1.0f - alpha;
             break;
-        case BiquadType::HighPass:
+        case BiquadType_HighPass:
             b[0] =  (1.0f + cos_w0) / 2.0f;
             b[1] = -(1.0f + cos_w0);
             b[2] =  (1.0f + cos_w0) / 2.0f;
@@ -71,9 +76,9 @@ void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Rea
             a[1] =  -2.0f * cos_w0;
             a[2] =   1.0f - alpha;
             break;
-        case BiquadType::BandPass:
+        case BiquadType_BandPass:
             b[0] =  alpha;
-            b[1] =  0.0f;
+            b[1] =  0;
             b[2] = -alpha;
             a[0] =  1.0f + alpha;
             a[1] = -2.0f * cos_w0;
@@ -81,23 +86,26 @@ void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Rea
             break;
     }
 
-    mA1 = a[1] / a[0];
-    mA2 = a[2] / a[0];
-    mB0 = b[0] / a[0];
-    mB1 = b[1] / a[0];
-    mB2 = b[2] / a[0];
+    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];
 }
 
-template<typename Real>
-void BiquadFilterR<Real>::process(const al::span<const Real> src, Real *dst)
+
+void BiquadFilter_processC(BiquadFilter *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples)
 {
-    const Real b0{mB0};
-    const Real b1{mB1};
-    const Real b2{mB2};
-    const Real a1{mA1};
-    const Real a2{mA2};
-    Real z1{mZ1};
-    Real z2{mZ2};
+    const ALfloat a1 = filter->a1;
+    const ALfloat a2 = filter->a2;
+    const ALfloat b0 = filter->b0;
+    const ALfloat b1 = filter->b1;
+    const ALfloat b2 = filter->b2;
+    ALfloat z1 = filter->z1;
+    ALfloat z2 = filter->z2;
+    ALsizei i;
+
+    ASSUME(numsamples > 0);
 
     /* Processing loop is Transposed Direct Form II. This requires less storage
      * compared to Direct Form I (only two delay components, instead of a four-
@@ -107,57 +115,15 @@ void BiquadFilterR<Real>::process(const al::span<const Real> src, Real *dst)
      *
      * See: http://www.earlevel.com/main/2003/02/28/biquads/
      */
-    auto proc_sample = [b0,b1,b2,a1,a2,&z1,&z2](Real input) noexcept -> Real
+    for(i = 0;i < numsamples;i++)
     {
-        const Real output{input*b0 + z1};
+        ALfloat input = src[i];
+        ALfloat output = input*b0 + z1;
         z1 = input*b1 - output*a1 + z2;
         z2 = input*b2 - output*a2;
-        return output;
-    };
-    std::transform(src.cbegin(), src.cend(), dst, proc_sample);
+        dst[i] = output;
+    }
 
-    mZ1 = z1;
-    mZ2 = z2;
+    filter->z1 = z1;
+    filter->z2 = z2;
 }
-
-template<typename Real>
-void BiquadFilterR<Real>::dualProcess(BiquadFilterR &other, const al::span<const Real> src,
-    Real *dst)
-{
-    const Real b00{mB0};
-    const Real b01{mB1};
-    const Real b02{mB2};
-    const Real a01{mA1};
-    const Real a02{mA2};
-    const Real b10{other.mB0};
-    const Real b11{other.mB1};
-    const Real b12{other.mB2};
-    const Real a11{other.mA1};
-    const Real a12{other.mA2};
-    Real z01{mZ1};
-    Real z02{mZ2};
-    Real z11{other.mZ1};
-    Real z12{other.mZ2};
-
-    auto proc_sample = [b00,b01,b02,a01,a02,b10,b11,b12,a11,a12,&z01,&z02,&z11,&z12](Real input) noexcept -> Real
-    {
-        const Real tmpout{input*b00 + z01};
-        z01 = input*b01 - tmpout*a01 + z02;
-        z02 = input*b02 - tmpout*a02;
-        input = tmpout;
-
-        const Real output{input*b10 + z11};
-        z11 = input*b11 - output*a11 + z12;
-        z12 = input*b12 - output*a12;
-        return output;
-    };
-    std::transform(src.cbegin(), src.cend(), dst, proc_sample);
-
-    mZ1 = z01;
-    mZ2 = z02;
-    other.mZ1 = z11;
-    other.mZ2 = z12;
-}
-
-template class BiquadFilterR<float>;
-template class BiquadFilterR<double>;

Alc/filters/nfc.c

@@ -0,0 +1,426 @@
+
+#include "config.h"
+
+#include "nfc.h"
+#include "alMain.h"
+
+#include <string.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 }*/
+};
+
+static void NfcFilterCreate1(struct NfcFilter1 *nfc, const float w0, const float w1)
+{
+    float b_00, g_0;
+    float r;
+
+    nfc->base_gain = 1.0f;
+    nfc->gain = 1.0f;
+
+    /* Calculate bass-boost coefficients. */
+    r = 0.5f * w0;
+    b_00 = B[1][0] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->gain *= g_0;
+    nfc->b1 = 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->base_gain /= g_0;
+    nfc->gain /= g_0;
+    nfc->a1 = 2.0f * b_00 / g_0;
+}
+
+static void NfcFilterAdjust1(struct NfcFilter1 *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->gain = nfc->base_gain * g_0;
+    nfc->b1 = 2.0f * b_00 / g_0;
+}
+
+
+static void NfcFilterCreate2(struct NfcFilter2 *nfc, const float w0, const float w1)
+{
+    float b_10, b_11, g_1;
+    float r;
+
+    nfc->base_gain = 1.0f;
+    nfc->gain = 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->gain *= g_1;
+    nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
+    nfc->b2 = 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->base_gain /= g_1;
+    nfc->gain /= g_1;
+    nfc->a1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
+    nfc->a2 = 4.0f * b_11 / g_1;
+}
+
+static void NfcFilterAdjust2(struct NfcFilter2 *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->gain = nfc->base_gain * g_1;
+    nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
+    nfc->b2 = 4.0f * b_11 / g_1;
+}
+
+
+static void NfcFilterCreate3(struct NfcFilter3 *nfc, const float w0, const float w1)
+{
+    float b_10, b_11, g_1;
+    float b_00, g_0;
+    float r;
+
+    nfc->base_gain = 1.0f;
+    nfc->gain = 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->gain *= g_1;
+    nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
+    nfc->b2 = 4.0f * b_11 / g_1;
+
+    b_00 = B[3][2] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->gain *= g_0;
+    nfc->b3 = 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->base_gain /= g_1;
+    nfc->gain /= g_1;
+    nfc->a1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
+    nfc->a2 = 4.0f * b_11 / g_1;
+
+    b_00 = B[3][2] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->base_gain /= g_0;
+    nfc->gain /= g_0;
+    nfc->a3 = 2.0f * b_00 / g_0;
+}
+
+static void NfcFilterAdjust3(struct NfcFilter3 *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->gain = nfc->base_gain * g_1;
+    nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
+    nfc->b2 = 4.0f * b_11 / g_1;
+
+    b_00 = B[3][2] * r;
+    g_0 = 1.0f + b_00;
+
+    nfc->gain *= g_0;
+    nfc->b3 = 2.0f * b_00 / g_0;
+}
+
+
+void NfcFilterCreate(NfcFilter *nfc, const float w0, const float w1)
+{
+    memset(nfc, 0, sizeof(*nfc));
+    NfcFilterCreate1(&nfc->first, w0, w1);
+    NfcFilterCreate2(&nfc->second, w0, w1);
+    NfcFilterCreate3(&nfc->third, w0, w1);
+}
+
+void NfcFilterAdjust(NfcFilter *nfc, const float w0)
+{
+    NfcFilterAdjust1(&nfc->first, w0);
+    NfcFilterAdjust2(&nfc->second, w0);
+    NfcFilterAdjust3(&nfc->third, w0);
+}
+
+
+void NfcFilterProcess1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count)
+{
+    const float gain = nfc->first.gain;
+    const float b1 = nfc->first.b1;
+    const float a1 = nfc->first.a1;
+    float z1 = nfc->first.z[0];
+    int i;
+
+    ASSUME(count > 0);
+
+    for(i = 0;i < count;i++)
+    {
+        float y = src[i]*gain - a1*z1;
+        float out = y + b1*z1;
+        z1 += y;
+
+        dst[i] = out;
+    }
+    nfc->first.z[0] = z1;
+}
+
+void NfcFilterProcess2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count)
+{
+    const float gain = nfc->second.gain;
+    const float b1 = nfc->second.b1;
+    const float b2 = nfc->second.b2;
+    const float a1 = nfc->second.a1;
+    const float a2 = nfc->second.a2;
+    float z1 = nfc->second.z[0];
+    float z2 = nfc->second.z[1];
+    int i;
+
+    ASSUME(count > 0);
+
+    for(i = 0;i < count;i++)
+    {
+        float y = src[i]*gain - a1*z1 - a2*z2;
+        float out = y + b1*z1 + b2*z2;
+        z2 += z1;
+        z1 += y;
+
+        dst[i] = out;
+    }
+    nfc->second.z[0] = z1;
+    nfc->second.z[1] = z2;
+}
+
+void NfcFilterProcess3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count)
+{
+    const float gain = nfc->third.gain;
+    const float b1 = nfc->third.b1;
+    const float b2 = nfc->third.b2;
+    const float b3 = nfc->third.b3;
+    const float a1 = nfc->third.a1;
+    const float a2 = nfc->third.a2;
+    const float a3 = nfc->third.a3;
+    float z1 = nfc->third.z[0];
+    float z2 = nfc->third.z[1];
+    float z3 = nfc->third.z[2];
+    int i;
+
+    ASSUME(count > 0);
+
+    for(i = 0;i < count;i++)
+    {
+        float y = src[i]*gain - a1*z1 - a2*z2;
+        float out = y + b1*z1 + b2*z2;
+        z2 += z1;
+        z1 += y;
+
+        y = out - a3*z3;
+        out = y + b3*z3;
+        z3 += y;
+
+        dst[i] = out;
+    }
+    nfc->third.z[0] = z1;
+    nfc->third.z[1] = z2;
+    nfc->third.z[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 NfcFilterProcess(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/filters/nfc.h

@@ -0,0 +1,49 @@
+#ifndef FILTER_NFC_H
+#define FILTER_NFC_H
+
+struct NfcFilter1 {
+    float base_gain, gain;
+    float b1, a1;
+    float z[1];
+};
+struct NfcFilter2 {
+    float base_gain, gain;
+    float b1, b2, a1, a2;
+    float z[2];
+};
+struct NfcFilter3 {
+    float base_gain, gain;
+    float b1, b2, b3, a1, a2, a3;
+    float z[3];
+};
+
+typedef struct NfcFilter {
+    struct NfcFilter1 first;
+    struct NfcFilter2 second;
+    struct NfcFilter3 third;
+} 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.
+ */
+
+void NfcFilterCreate(NfcFilter *nfc, const float w0, const float w1);
+void NfcFilterAdjust(NfcFilter *nfc, const float w0);
+
+/* Near-field control filter for first-order ambisonic channels (1-3). */
+void NfcFilterProcess1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
+
+/* Near-field control filter for second-order ambisonic channels (4-8). */
+void NfcFilterProcess2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
+
+/* Near-field control filter for third-order ambisonic channels (9-15). */
+void NfcFilterProcess3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
+
+#endif /* FILTER_NFC_H */

Alc/filters/splitter.c

@@ -0,0 +1,109 @@
+
+#include "config.h"
+
+#include "splitter.h"
+
+#include "math_defs.h"
+
+
+void bandsplit_init(BandSplitter *splitter, ALfloat f0norm)
+{
+    ALfloat w = f0norm * 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 lp_coeff, hp_coeff, lp_y, hp_y, d;
+    ALfloat lp_z1, lp_z2, hp_z1;
+    ALsizei i;
+
+    ASSUME(count > 0);
+
+    hp_coeff = splitter->coeff;
+    lp_coeff = splitter->coeff*0.5f + 0.5f;
+    lp_z1 = splitter->lp_z1;
+    lp_z2 = splitter->lp_z2;
+    hp_z1 = splitter->hp_z1;
+    for(i = 0;i < count;i++)
+    {
+        ALfloat in = input[i];
+
+        /* Low-pass sample processing. */
+        d = (in - lp_z1) * lp_coeff;
+        lp_y = lp_z1 + d;
+        lp_z1 = lp_y + d;
+
+        d = (lp_y - lp_z2) * lp_coeff;
+        lp_y = lp_z2 + d;
+        lp_z2 = lp_y + d;
+
+        lpout[i] = lp_y;
+
+        /* All-pass sample processing. */
+        hp_y = in*hp_coeff + hp_z1;
+        hp_z1 = in - hp_y*hp_coeff;
+
+        /* High-pass generated from removing low-passed output. */
+        hpout[i] = hp_y - lp_y;
+    }
+    splitter->lp_z1 = lp_z1;
+    splitter->lp_z2 = lp_z2;
+    splitter->hp_z1 = hp_z1;
+}
+
+
+void splitterap_init(SplitterAllpass *splitter, ALfloat f0norm)
+{
+    ALfloat w = f0norm * 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, in, out;
+    ALfloat z1;
+    ALsizei i;
+
+    ASSUME(count > 0);
+
+    coeff = splitter->coeff;
+    z1 = splitter->z1;
+    for(i = 0;i < count;i++)
+    {
+        in = samples[i];
+
+        out = in*coeff + z1;
+        z1 = in - out*coeff;
+
+        samples[i] = out;
+    }
+    splitter->z1 = z1;
+}

Alc/filters/splitter.h

@@ -0,0 +1,40 @@
+#ifndef FILTER_SPLITTER_H
+#define FILTER_SPLITTER_H
+
+#include "alMain.h"
+
+
+/* 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 f0norm);
+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 f0norm);
+void splitterap_clear(SplitterAllpass *splitter);
+void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count);
+
+
+typedef struct FrontStablizer {
+    SplitterAllpass APFilter[MAX_OUTPUT_CHANNELS];
+    BandSplitter LFilter, RFilter;
+    alignas(16) ALfloat LSplit[2][BUFFERSIZE];
+    alignas(16) ALfloat RSplit[2][BUFFERSIZE];
+} FrontStablizer;
+
+#endif /* FILTER_SPLITTER_H */

Alc/fpu_modes.h

@@ -0,0 +1,34 @@
+#ifndef FPU_MODES_H
+#define FPU_MODES_H
+
+#ifdef HAVE_FENV_H
+#include <fenv.h>
+#endif
+
+
+typedef struct FPUCtl {
+#if defined(__GNUC__) && defined(HAVE_SSE)
+    unsigned int sse_state;
+#elif defined(HAVE___CONTROL87_2)
+    unsigned int state;
+    unsigned int sse_state;
+#elif defined(HAVE__CONTROLFP)
+    unsigned int state;
+#endif
+} FPUCtl;
+void SetMixerFPUMode(FPUCtl *ctl);
+void RestoreFPUMode(const FPUCtl *ctl);
+
+#ifdef __GNUC__
+/* Use an alternate macro set with GCC to avoid accidental continue or break
+ * statements within the mixer mode.
+ */
+#define START_MIXER_MODE() __extension__({ FPUCtl _oldMode; SetMixerFPUMode(&_oldMode)
+#define END_MIXER_MODE() RestoreFPUMode(&_oldMode); })
+#else
+#define START_MIXER_MODE() do { FPUCtl _oldMode; SetMixerFPUMode(&_oldMode)
+#define END_MIXER_MODE() RestoreFPUMode(&_oldMode); } while(0)
+#endif
+#define LEAVE_MIXER_MODE() RestoreFPUMode(&_oldMode)
+
+#endif /* FPU_MODES_H */

Alc/hrtf.h

@@ -0,0 +1,84 @@
+#ifndef ALC_HRTF_H
+#define ALC_HRTF_H
+
+#include "AL/al.h"
+#include "AL/alc.h"
+
+#include "alMain.h"
+#include "alstring.h"
+#include "atomic.h"
+
+
+#define HRTF_HISTORY_BITS   (6)
+#define HRTF_HISTORY_LENGTH (1<<HRTF_HISTORY_BITS)
+#define HRTF_HISTORY_MASK   (HRTF_HISTORY_LENGTH-1)
+
+#define HRIR_BITS        (7)
+#define HRIR_LENGTH      (1<<HRIR_BITS)
+#define HRIR_MASK        (HRIR_LENGTH-1)
+
+
+struct HrtfEntry;
+
+struct Hrtf {
+    RefCount ref;
+
+    ALuint sampleRate;
+    ALsizei irSize;
+
+    ALfloat distance;
+    ALubyte evCount;
+
+    const ALubyte *azCount;
+    const ALushort *evOffset;
+    const ALfloat (*coeffs)[2];
+    const ALubyte (*delays)[2];
+};
+
+
+typedef struct HrtfState {
+    alignas(16) ALfloat History[HRTF_HISTORY_LENGTH];
+    alignas(16) ALfloat Values[HRIR_LENGTH][2];
+} HrtfState;
+
+typedef struct HrtfParams {
+    alignas(16) ALfloat Coeffs[HRIR_LENGTH][2];
+    ALsizei Delay[2];
+    ALfloat Gain;
+} HrtfParams;
+
+typedef struct DirectHrtfState {
+    /* HRTF filter state for dry buffer content */
+    ALsizei Offset;
+    ALsizei IrSize;
+    struct {
+        alignas(16) ALfloat Values[HRIR_LENGTH][2];
+        alignas(16) ALfloat Coeffs[HRIR_LENGTH][2];
+    } Chan[];
+} DirectHrtfState;
+
+struct AngularPoint {
+    ALfloat Elev;
+    ALfloat Azim;
+};
+
+
+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, a matching decoding matrix, and per-order high-
+ * frequency gains for the decoder. The calculated impulse responses are
+ * ordered and scaled according to the matrix input.
+ */
+void BuildBFormatHrtf(const struct Hrtf *Hrtf, DirectHrtfState *state, ALsizei NumChannels, const struct AngularPoint *AmbiPoints, const ALfloat (*restrict AmbiMatrix)[MAX_AMBI_COEFFS], ALsizei AmbiCount, const ALfloat *restrict AmbiOrderHFGain);
+
+#endif /* ALC_HRTF_H */

Alc/inprogext.h

@@ -0,0 +1,87 @@
+#ifndef INPROGEXT_H
+#define INPROGEXT_H
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "AL/alext.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifndef ALC_SOFT_loopback2
+#define ALC_SOFT_loopback2 1
+#define ALC_AMBISONIC_LAYOUT_SOFT                0xfff0
+#define ALC_AMBISONIC_SCALING_SOFT               0xfff1
+#define ALC_AMBISONIC_ORDER_SOFT                 0xfff2
+#define ALC_MAX_AMBISONIC_ORDER_SOFT             0xfff3
+
+#define ALC_BFORMAT3D_SOFT                       0x1508
+
+/* Ambisonic layouts */
+#define ALC_ACN_SOFT                             0xfff4
+#define ALC_FUMA_SOFT                            0xfff5
+
+/* Ambisonic scalings (normalization) */
+/*#define ALC_FUMA_SOFT*/
+#define ALC_SN3D_SOFT                            0xfff6
+#define ALC_N3D_SOFT                             0xfff7
+#endif
+
+#ifndef AL_SOFT_map_buffer
+#define AL_SOFT_map_buffer 1
+typedef unsigned int ALbitfieldSOFT;
+#define AL_MAP_READ_BIT_SOFT                     0x00000001
+#define AL_MAP_WRITE_BIT_SOFT                    0x00000002
+#define AL_MAP_PERSISTENT_BIT_SOFT               0x00000004
+#define AL_PRESERVE_DATA_BIT_SOFT                0x00000008
+typedef void (AL_APIENTRY*LPALBUFFERSTORAGESOFT)(ALuint buffer, ALenum format, const ALvoid *data, ALsizei size, ALsizei freq, ALbitfieldSOFT flags);
+typedef void* (AL_APIENTRY*LPALMAPBUFFERSOFT)(ALuint buffer, ALsizei offset, ALsizei length, ALbitfieldSOFT access);
+typedef void (AL_APIENTRY*LPALUNMAPBUFFERSOFT)(ALuint buffer);
+typedef void (AL_APIENTRY*LPALFLUSHMAPPEDBUFFERSOFT)(ALuint buffer, ALsizei offset, ALsizei length);
+#ifdef AL_ALEXT_PROTOTYPES
+AL_API void AL_APIENTRY alBufferStorageSOFT(ALuint buffer, ALenum format, const ALvoid *data, ALsizei size, ALsizei freq, ALbitfieldSOFT flags);
+AL_API void* AL_APIENTRY alMapBufferSOFT(ALuint buffer, ALsizei offset, ALsizei length, ALbitfieldSOFT access);
+AL_API void AL_APIENTRY alUnmapBufferSOFT(ALuint buffer);
+AL_API void AL_APIENTRY alFlushMappedBufferSOFT(ALuint buffer, ALsizei offset, ALsizei length);
+#endif
+#endif
+
+#ifndef AL_SOFT_events
+#define AL_SOFT_events 1
+#define AL_EVENT_CALLBACK_FUNCTION_SOFT          0x1220
+#define AL_EVENT_CALLBACK_USER_PARAM_SOFT        0x1221
+#define AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT      0x1222
+#define AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT  0x1223
+#define AL_EVENT_TYPE_ERROR_SOFT                 0x1224
+#define AL_EVENT_TYPE_PERFORMANCE_SOFT           0x1225
+#define AL_EVENT_TYPE_DEPRECATED_SOFT            0x1226
+#define AL_EVENT_TYPE_DISCONNECTED_SOFT          0x1227
+typedef void (AL_APIENTRY*ALEVENTPROCSOFT)(ALenum eventType, ALuint object, ALuint param,
+                                           ALsizei length, const ALchar *message,
+                                           void *userParam);
+typedef void (AL_APIENTRY*LPALEVENTCONTROLSOFT)(ALsizei count, const ALenum *types, ALboolean enable);
+typedef void (AL_APIENTRY*LPALEVENTCALLBACKSOFT)(ALEVENTPROCSOFT callback, void *userParam);
+typedef void* (AL_APIENTRY*LPALGETPOINTERSOFT)(ALenum pname);
+typedef void (AL_APIENTRY*LPALGETPOINTERVSOFT)(ALenum pname, void **values);
+#ifdef AL_ALEXT_PROTOTYPES
+AL_API void AL_APIENTRY alEventControlSOFT(ALsizei count, const ALenum *types, ALboolean enable);
+AL_API void AL_APIENTRY alEventCallbackSOFT(ALEVENTPROCSOFT callback, void *userParam);
+AL_API void* AL_APIENTRY alGetPointerSOFT(ALenum pname);
+AL_API void AL_APIENTRY alGetPointervSOFT(ALenum pname, void **values);
+#endif
+#endif
+
+#ifndef AL_SOFT_buffer_layers
+#define AL_SOFT_buffer_layers
+typedef void (AL_APIENTRY*LPALSOURCEQUEUEBUFFERLAYERSSOFT)(ALuint src, ALsizei nb, const ALuint *buffers);
+#ifdef AL_ALEXT_PROTOTYPES
+AL_API void AL_APIENTRY alSourceQueueBufferLayersSOFT(ALuint src, ALsizei nb, const ALuint *buffers);
+#endif
+#endif
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif /* INPROGEXT_H */

Alc/logging.h

@@ -0,0 +1,69 @@
+#ifndef LOGGING_H
+#define LOGGING_H
+
+#include <stdio.h>
+
+
+#ifdef __GNUC__
+#define DECL_FORMAT(x, y, z) __attribute__((format(x, (y), (z))))
+#else
+#define DECL_FORMAT(x, y, z)
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+extern FILE *LogFile;
+
+#if defined(__GNUC__) && !defined(_WIN32)
+#define AL_PRINT(T, MSG, ...) fprintf(LogFile, "AL lib: %s %s: "MSG, T, __FUNCTION__ , ## __VA_ARGS__)
+#else
+void al_print(const char *type, const char *func, const char *fmt, ...) DECL_FORMAT(printf, 3,4);
+#define AL_PRINT(T, ...) al_print((T), __FUNCTION__, __VA_ARGS__)
+#endif
+
+#ifdef __ANDROID__
+#include <android/log.h>
+#define LOG_ANDROID(T, MSG, ...) __android_log_print(T, "openal", "AL lib: %s: "MSG, __FUNCTION__ , ## __VA_ARGS__)
+#else
+#define LOG_ANDROID(T, MSG, ...) ((void)0)
+#endif
+
+enum LogLevel {
+    NoLog,
+    LogError,
+    LogWarning,
+    LogTrace,
+    LogRef
+};
+extern enum LogLevel LogLevel;
+
+#define TRACEREF(...) do {                                                    \
+    if(LogLevel >= LogRef)                                                    \
+        AL_PRINT("(--)", __VA_ARGS__);                                        \
+} while(0)
+
+#define TRACE(...) do {                                                       \
+    if(LogLevel >= LogTrace)                                                  \
+        AL_PRINT("(II)", __VA_ARGS__);                                        \
+    LOG_ANDROID(ANDROID_LOG_DEBUG, __VA_ARGS__);                              \
+} while(0)
+
+#define WARN(...) do {                                                        \
+    if(LogLevel >= LogWarning)                                                \
+        AL_PRINT("(WW)", __VA_ARGS__);                                        \
+    LOG_ANDROID(ANDROID_LOG_WARN, __VA_ARGS__);                               \
+} while(0)
+
+#define ERR(...) do {                                                         \
+    if(LogLevel >= LogError)                                                  \
+        AL_PRINT("(EE)", __VA_ARGS__);                                        \
+    LOG_ANDROID(ANDROID_LOG_ERROR, __VA_ARGS__);                              \
+} while(0)
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif /* LOGGING_H */

Alc/mastering.c

@@ -0,0 +1,543 @@
+#include "config.h"
+
+#include <math.h>
+
+#include "mastering.h"
+#include "alu.h"
+#include "almalloc.h"
+#include "static_assert.h"
+#include "math_defs.h"
+
+
+/* Early MSVC lacks round/roundf */
+#if defined(_MSC_VER) && _MSC_VER < 1800
+static double round(double val)
+{
+    if(val < 0.0)
+        return ceil(val-0.5);
+    return floor(val+0.5);
+}
+#define roundf(f) ((float)round((float)(f)))
+#endif
+
+
+/* These structures assume BUFFERSIZE is a power of 2. */
+static_assert((BUFFERSIZE & (BUFFERSIZE-1)) == 0, "BUFFERSIZE is not a power of 2");
+
+typedef struct SlidingHold {
+    ALfloat Values[BUFFERSIZE];
+    ALsizei Expiries[BUFFERSIZE];
+    ALsizei LowerIndex;
+    ALsizei UpperIndex;
+    ALsizei Length;
+} SlidingHold;
+
+/* General topology and basic automation was based on the following paper:
+ *
+ *   D. Giannoulis, M. Massberg and J. D. Reiss,
+ *   "Parameter Automation in a Dynamic Range Compressor,"
+ *   Journal of the Audio Engineering Society, v61 (10), Oct. 2013
+ *
+ * Available (along with supplemental reading) at:
+ *
+ *   http://c4dm.eecs.qmul.ac.uk/audioengineering/compressors/
+ */
+typedef struct Compressor {
+    ALsizei NumChans;
+    ALuint SampleRate;
+
+    struct {
+        ALuint Knee : 1;
+        ALuint Attack : 1;
+        ALuint Release : 1;
+        ALuint PostGain : 1;
+        ALuint Declip : 1;
+    } Auto;
+
+    ALsizei LookAhead;
+
+    ALfloat PreGain;
+    ALfloat PostGain;
+
+    ALfloat Threshold;
+    ALfloat Slope;
+    ALfloat Knee;
+
+    ALfloat Attack;
+    ALfloat Release;
+
+    alignas(16) ALfloat SideChain[2*BUFFERSIZE];
+    alignas(16) ALfloat CrestFactor[BUFFERSIZE];
+
+    SlidingHold *Hold;
+    ALfloat (*Delay)[BUFFERSIZE];
+    ALsizei DelayIndex;
+
+    ALfloat CrestCoeff;
+    ALfloat GainEstimate;
+    ALfloat AdaptCoeff;
+
+    ALfloat LastPeakSq;
+    ALfloat LastRmsSq;
+    ALfloat LastRelease;
+    ALfloat LastAttack;
+    ALfloat LastGainDev;
+} Compressor;
+
+
+/* This sliding hold follows the input level with an instant attack and a
+ * fixed duration hold before an instant release to the next highest level.
+ * It is a sliding window maximum (descending maxima) implementation based on
+ * Richard Harter's ascending minima algorithm available at:
+ *
+ *   http://www.richardhartersworld.com/cri/2001/slidingmin.html
+ */
+static ALfloat UpdateSlidingHold(SlidingHold *Hold, const ALsizei i, const ALfloat in)
+{
+    const ALsizei mask = BUFFERSIZE - 1;
+    const ALsizei length = Hold->Length;
+    ALfloat *restrict values = Hold->Values;
+    ALsizei *restrict expiries = Hold->Expiries;
+    ALsizei lowerIndex = Hold->LowerIndex;
+    ALsizei upperIndex = Hold->UpperIndex;
+
+    if(i >= expiries[upperIndex])
+        upperIndex = (upperIndex + 1) & mask;
+
+    if(in >= values[upperIndex])
+    {
+        values[upperIndex] = in;
+        expiries[upperIndex] = i + length;
+        lowerIndex = upperIndex;
+    }
+    else
+    {
+        do {
+            do {
+                if(!(in >= values[lowerIndex]))
+                    goto found_place;
+            } while(lowerIndex--);
+            lowerIndex = mask;
+        } while(1);
+    found_place:
+
+        lowerIndex = (lowerIndex + 1) & mask;
+        values[lowerIndex] = in;
+        expiries[lowerIndex] = i + length;
+    }
+
+    Hold->LowerIndex = lowerIndex;
+    Hold->UpperIndex = upperIndex;
+
+    return values[upperIndex];
+}
+
+static void ShiftSlidingHold(SlidingHold *Hold, const ALsizei n)
+{
+    const ALsizei lowerIndex = Hold->LowerIndex;
+    ALsizei *restrict expiries = Hold->Expiries;
+    ALsizei i = Hold->UpperIndex;
+
+    if(lowerIndex < i)
+    {
+        for(;i < BUFFERSIZE;i++)
+            expiries[i] -= n;
+        i = 0;
+    }
+    for(;i < lowerIndex;i++)
+        expiries[i] -= n;
+
+    expiries[i] -= n;
+}
+
+/* Multichannel compression is linked via the absolute maximum of all
+ * channels.
+ */
+static void LinkChannels(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE])
+{
+    const ALsizei index = Comp->LookAhead;
+    const ALsizei numChans = Comp->NumChans;
+    ALfloat *restrict sideChain = Comp->SideChain;
+    ALsizei c, i;
+
+    ASSUME(SamplesToDo > 0);
+    ASSUME(numChans > 0);
+
+    for(i = 0;i < SamplesToDo;i++)
+        sideChain[index + i] = 0.0f;
+
+    for(c = 0;c < numChans;c++)
+    {
+        ALsizei offset = index;
+        for(i = 0;i < SamplesToDo;i++)
+        {
+            sideChain[offset] = maxf(sideChain[offset], fabsf(OutBuffer[c][i]));
+            ++offset;
+        }
+    }
+}
+
+/* This calculates the squared crest factor of the control signal for the
+ * basic automation of the attack/release times.  As suggested by the paper,
+ * it uses an instantaneous squared peak detector and a squared RMS detector
+ * both with 200ms release times.
+ */
+static void CrestDetector(Compressor *Comp, const ALsizei SamplesToDo)
+{
+    const ALfloat a_crest = Comp->CrestCoeff;
+    const ALsizei index = Comp->LookAhead;
+    const ALfloat *restrict sideChain = Comp->SideChain;
+    ALfloat *restrict crestFactor = Comp->CrestFactor;
+    ALfloat y2_peak = Comp->LastPeakSq;
+    ALfloat y2_rms = Comp->LastRmsSq;
+    ALsizei i;
+
+    ASSUME(SamplesToDo > 0);
+
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        ALfloat x_abs = sideChain[index + i];
+        ALfloat x2 = maxf(0.000001f, x_abs * x_abs);
+
+        y2_peak = maxf(x2, lerp(x2, y2_peak, a_crest));
+        y2_rms = lerp(x2, y2_rms, a_crest);
+        crestFactor[i] = y2_peak / y2_rms;
+    }
+
+    Comp->LastPeakSq = y2_peak;
+    Comp->LastRmsSq = y2_rms;
+}
+
+/* The side-chain starts with a simple peak detector (based on the absolute
+ * value of the incoming signal) and performs most of its operations in the
+ * log domain.
+ */
+static void PeakDetector(Compressor *Comp, const ALsizei SamplesToDo)
+{
+    const ALsizei index = Comp->LookAhead;
+    ALfloat *restrict sideChain = Comp->SideChain;
+    ALsizei i;
+
+    ASSUME(SamplesToDo > 0);
+
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        const ALuint offset = index + i;
+        const ALfloat x_abs = sideChain[offset];
+
+        sideChain[offset] = logf(maxf(0.000001f, x_abs));
+    }
+}
+
+/* An optional hold can be used to extend the peak detector so it can more
+ * solidly detect fast transients.  This is best used when operating as a
+ * limiter.
+ */
+static void PeakHoldDetector(Compressor *Comp, const ALsizei SamplesToDo)
+{
+    const ALsizei index = Comp->LookAhead;
+    ALfloat *restrict sideChain = Comp->SideChain;
+    SlidingHold *hold = Comp->Hold;
+    ALsizei i;
+
+    ASSUME(SamplesToDo > 0);
+
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        const ALsizei offset = index + i;
+        const ALfloat x_abs = sideChain[offset];
+        const ALfloat x_G = logf(maxf(0.000001f, x_abs));
+
+        sideChain[offset] = UpdateSlidingHold(hold, i, x_G);
+    }
+
+    ShiftSlidingHold(hold, SamplesToDo);
+}
+
+/* This is the heart of the feed-forward compressor.  It operates in the log
+ * domain (to better match human hearing) and can apply some basic automation
+ * to knee width, attack/release times, make-up/post gain, and clipping
+ * reduction.
+ */
+static void GainCompressor(Compressor *Comp, const ALsizei SamplesToDo)
+{
+    const bool autoKnee = Comp->Auto.Knee;
+    const bool autoAttack = Comp->Auto.Attack;
+    const bool autoRelease = Comp->Auto.Release;
+    const bool autoPostGain = Comp->Auto.PostGain;
+    const bool autoDeclip = Comp->Auto.Declip;
+    const ALsizei lookAhead = Comp->LookAhead;
+    const ALfloat threshold = Comp->Threshold;
+    const ALfloat slope = Comp->Slope;
+    const ALfloat attack = Comp->Attack;
+    const ALfloat release = Comp->Release;
+    const ALfloat c_est = Comp->GainEstimate;
+    const ALfloat a_adp = Comp->AdaptCoeff;
+    const ALfloat *restrict crestFactor = Comp->CrestFactor;
+    ALfloat *restrict sideChain = Comp->SideChain;
+    ALfloat postGain = Comp->PostGain;
+    ALfloat knee = Comp->Knee;
+    ALfloat t_att = attack;
+    ALfloat t_rel = release - attack;
+    ALfloat a_att = expf(-1.0f / t_att);
+    ALfloat a_rel = expf(-1.0f / t_rel);
+    ALfloat y_1 = Comp->LastRelease;
+    ALfloat y_L = Comp->LastAttack;
+    ALfloat c_dev = Comp->LastGainDev;
+    ALsizei i;
+
+    ASSUME(SamplesToDo > 0);
+
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        const ALfloat y2_crest = crestFactor[i];
+        const ALfloat x_G = sideChain[lookAhead + i];
+        const ALfloat x_over = x_G - threshold;
+        ALfloat knee_h;
+        ALfloat y_G;
+        ALfloat x_L;
+
+        if(autoKnee)
+            knee = maxf(0.0f, 2.5f * (c_dev + c_est));
+        knee_h = 0.5f * knee;
+
+        /* This is the gain computer.  It applies a static compression curve
+         * to the control signal.
+         */
+        if(x_over <= -knee_h)
+            y_G = 0.0f;
+        else if(fabsf(x_over) < knee_h)
+            y_G = (x_over + knee_h) * (x_over + knee_h) / (2.0f * knee);
+        else
+            y_G = x_over;
+
+        x_L = -slope * y_G;
+
+        if(autoAttack)
+        {
+            t_att = 2.0f * attack / y2_crest;
+            a_att = expf(-1.0f / t_att);
+        }
+
+        if(autoRelease)
+        {
+            t_rel = 2.0f * release / y2_crest - t_att;
+            a_rel = expf(-1.0f / t_rel);
+        }
+
+        /* Gain smoothing (ballistics) is done via a smooth decoupled peak
+         * detector.  The attack time is subtracted from the release time
+         * above to compensate for the chained operating mode.
+         */
+        y_1 = maxf(x_L, lerp(x_L, y_1, a_rel));
+        y_L = lerp(y_1, y_L, a_att);
+
+        /* Knee width and make-up gain automation make use of a smoothed
+         * measurement of deviation between the control signal and estimate.
+         * The estimate is also used to bias the measurement to hot-start its
+         * average.
+         */
+        c_dev = lerp(-y_L - c_est, c_dev, a_adp);
+
+        if(autoPostGain)
+        {
+            /* Clipping reduction is only viable when make-up gain is being
+             * automated.  It modifies the deviation to further attenuate the
+             * control signal when clipping is detected.  The adaptation
+             * time is sufficiently long enough to suppress further clipping
+             * at the same output level.
+             */
+            if(autoDeclip)
+                c_dev = maxf(c_dev, sideChain[i] - y_L - threshold - c_est);
+
+            postGain = -(c_dev + c_est);
+        }
+
+        sideChain[i] = expf(postGain - y_L);
+    }
+
+    Comp->LastRelease = y_1;
+    Comp->LastAttack = y_L;
+    Comp->LastGainDev = c_dev;
+}
+
+/* Combined with the hold time, a look-ahead delay can improve handling of
+ * fast transients by allowing the envelope time to converge prior to
+ * reaching the offending impulse.  This is best used when operating as a
+ * limiter.
+ */
+static void SignalDelay(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE])
+{
+    const ALsizei mask = BUFFERSIZE - 1;
+    const ALsizei numChans = Comp->NumChans;
+    const ALsizei indexIn = Comp->DelayIndex;
+    const ALsizei indexOut = Comp->DelayIndex - Comp->LookAhead;
+    ALfloat (*restrict delay)[BUFFERSIZE] = Comp->Delay;
+    ALsizei c, i;
+
+    ASSUME(SamplesToDo > 0);
+    ASSUME(numChans > 0);
+
+    for(c = 0;c < numChans;c++)
+    {
+        for(i = 0;i < SamplesToDo;i++)
+        {
+            ALfloat sig = OutBuffer[c][i];
+
+            OutBuffer[c][i] = delay[c][(indexOut + i) & mask];
+            delay[c][(indexIn + i) & mask] = sig;
+        }
+    }
+
+    Comp->DelayIndex = (indexIn + SamplesToDo) & mask;
+}
+
+/* The compressor is initialized with the following settings:
+ *
+ *   NumChans       - Number of channels to process.
+ *   SampleRate     - Sample rate to process.
+ *   AutoKnee       - Whether to automate the knee width parameter.
+ *   AutoAttack     - Whether to automate the attack time parameter.
+ *   AutoRelease    - Whether to automate the release time parameter.
+ *   AutoPostGain   - Whether to automate the make-up (post) gain parameter.
+ *   AutoDeclip     - Whether to automate clipping reduction.  Ignored when
+ *                    not automating make-up gain.
+ *   LookAheadTime  - Look-ahead time (in seconds).
+ *   HoldTime       - Peak hold-time (in seconds).
+ *   PreGainDb      - Gain applied before detection (in dB).
+ *   PostGainDb     - Make-up gain applied after compression (in dB).
+ *   ThresholdDb    - Triggering threshold (in dB).
+ *   Ratio          - Compression ratio (x:1).  Set to INFINITY for true
+ *                    limiting.  Ignored when automating knee width.
+ *   KneeDb         - Knee width (in dB).  Ignored when automating knee
+ *                    width.
+ *   AttackTimeMin  - Attack time (in seconds).  Acts as a maximum when
+ *                    automating attack time.
+ *   ReleaseTimeMin - Release time (in seconds).  Acts as a maximum when
+ *                    automating release time.
+ */
+Compressor* CompressorInit(const ALsizei NumChans, const ALuint SampleRate,
+                           const ALboolean AutoKnee, const ALboolean AutoAttack,
+                           const ALboolean AutoRelease, const ALboolean AutoPostGain,
+                           const ALboolean AutoDeclip, const ALfloat LookAheadTime,
+                           const ALfloat HoldTime, const ALfloat PreGainDb,
+                           const ALfloat PostGainDb, const ALfloat ThresholdDb,
+                           const ALfloat Ratio, const ALfloat KneeDb,
+                           const ALfloat AttackTime, const ALfloat ReleaseTime)
+{
+    Compressor *Comp;
+    ALsizei lookAhead;
+    ALsizei hold;
+    size_t size;
+
+    lookAhead = (ALsizei)clampf(roundf(LookAheadTime*SampleRate), 0.0f, BUFFERSIZE-1);
+    hold = (ALsizei)clampf(roundf(HoldTime*SampleRate), 0.0f, BUFFERSIZE-1);
+    /* The sliding hold implementation doesn't handle a length of 1. A 1-sample
+     * hold is useless anyway, it would only ever give back what was just given
+     * to it.
+     */
+    if(hold == 1)
+        hold = 0;
+
+    size = sizeof(*Comp);
+    if(lookAhead > 0)
+    {
+        size += sizeof(*Comp->Delay) * NumChans;
+        if(hold > 0)
+            size += sizeof(*Comp->Hold);
+    }
+
+    Comp = al_calloc(16, size);
+    Comp->NumChans = NumChans;
+    Comp->SampleRate = SampleRate;
+    Comp->Auto.Knee = AutoKnee;
+    Comp->Auto.Attack = AutoAttack;
+    Comp->Auto.Release = AutoRelease;
+    Comp->Auto.PostGain = AutoPostGain;
+    Comp->Auto.Declip = AutoPostGain && AutoDeclip;
+    Comp->LookAhead = lookAhead;
+    Comp->PreGain = powf(10.0f, PreGainDb / 20.0f);
+    Comp->PostGain = PostGainDb * logf(10.0f) / 20.0f;
+    Comp->Threshold = ThresholdDb * logf(10.0f) / 20.0f;
+    Comp->Slope = 1.0f / maxf(1.0f, Ratio) - 1.0f;
+    Comp->Knee = maxf(0.0f, KneeDb * logf(10.0f) / 20.0f);
+    Comp->Attack = maxf(1.0f, AttackTime * SampleRate);
+    Comp->Release = maxf(1.0f, ReleaseTime * SampleRate);
+
+    /* Knee width automation actually treats the compressor as a limiter.  By
+     * varying the knee width, it can effectively be seen as applying
+     * compression over a wide range of ratios.
+     */
+    if(AutoKnee)
+        Comp->Slope = -1.0f;
+
+    if(lookAhead > 0)
+    {
+        if(hold > 0)
+        {
+            Comp->Hold = (SlidingHold*)(Comp + 1);
+            Comp->Hold->Values[0] = -HUGE_VALF;
+            Comp->Hold->Expiries[0] = hold;
+            Comp->Hold->Length = hold;
+            Comp->Delay = (ALfloat(*)[BUFFERSIZE])(Comp->Hold + 1);
+        }
+        else
+        {
+            Comp->Delay = (ALfloat(*)[BUFFERSIZE])(Comp + 1);
+        }
+    }
+
+    Comp->CrestCoeff = expf(-1.0f / (0.200f * SampleRate)); // 200ms
+    Comp->GainEstimate = Comp->Threshold * -0.5f * Comp->Slope;
+    Comp->AdaptCoeff = expf(-1.0f / (2.0f * SampleRate)); // 2s
+
+    return Comp;
+}
+
+void ApplyCompression(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE])
+{
+    const ALsizei numChans = Comp->NumChans;
+    const ALfloat preGain = Comp->PreGain;
+    ALfloat *restrict sideChain;
+    ALsizei c, i;
+
+    ASSUME(SamplesToDo > 0);
+    ASSUME(numChans > 0);
+
+    if(preGain != 1.0f)
+    {
+        for(c = 0;c < numChans;c++)
+        {
+            for(i = 0;i < SamplesToDo;i++)
+                OutBuffer[c][i] *= preGain;
+        }
+    }
+
+    LinkChannels(Comp, SamplesToDo, OutBuffer);
+
+    if(Comp->Auto.Attack || Comp->Auto.Release)
+        CrestDetector(Comp, SamplesToDo);
+
+    if(Comp->Hold)
+        PeakHoldDetector(Comp, SamplesToDo);
+    else
+        PeakDetector(Comp, SamplesToDo);
+
+    GainCompressor(Comp, SamplesToDo);
+
+    if(Comp->Delay)
+        SignalDelay(Comp, SamplesToDo, OutBuffer);
+
+    sideChain = Comp->SideChain;
+    for(c = 0;c < numChans;c++)
+    {
+        for(i = 0;i < SamplesToDo;i++)
+            OutBuffer[c][i] *= sideChain[i];
+    }
+
+    memmove(sideChain, sideChain+SamplesToDo, Comp->LookAhead*sizeof(ALfloat));
+}
+
+
+ALsizei GetCompressorLookAhead(const Compressor *Comp)
+{ return Comp->LookAhead; }

Alc/mastering.h

@@ -0,0 +1,49 @@
+#ifndef MASTERING_H
+#define MASTERING_H
+
+#include "AL/al.h"
+
+/* For BUFFERSIZE. */
+#include "alMain.h"
+
+struct Compressor;
+
+/* The compressor is initialized with the following settings:
+ *
+ *   NumChans       - Number of channels to process.
+ *   SampleRate     - Sample rate to process.
+ *   AutoKnee       - Whether to automate the knee width parameter.
+ *   AutoAttack     - Whether to automate the attack time parameter.
+ *   AutoRelease    - Whether to automate the release time parameter.
+ *   AutoPostGain   - Whether to automate the make-up (post) gain parameter.
+ *   AutoDeclip     - Whether to automate clipping reduction.  Ignored when
+ *                    not automating make-up gain.
+ *   LookAheadTime  - Look-ahead time (in seconds).
+ *   HoldTime       - Peak hold-time (in seconds).
+ *   PreGainDb      - Gain applied before detection (in dB).
+ *   PostGainDb     - Make-up gain applied after compression (in dB).
+ *   ThresholdDb    - Triggering threshold (in dB).
+ *   Ratio          - Compression ratio (x:1).  Set to INFINIFTY for true
+ *                    limiting.  Ignored when automating knee width.
+ *   KneeDb         - Knee width (in dB).  Ignored when automating knee
+ *                    width.
+ *   AttackTimeMin  - Attack time (in seconds).  Acts as a maximum when
+ *                    automating attack time.
+ *   ReleaseTimeMin - Release time (in seconds).  Acts as a maximum when
+ *                    automating release time.
+ */
+struct Compressor* CompressorInit(const ALsizei NumChans, const ALuint SampleRate,
+    const ALboolean AutoKnee, const ALboolean AutoAttack,
+    const ALboolean AutoRelease, const ALboolean AutoPostGain,
+    const ALboolean AutoDeclip, const ALfloat LookAheadTime,
+    const ALfloat HoldTime, const ALfloat PreGainDb,
+    const ALfloat PostGainDb, const ALfloat ThresholdDb,
+    const ALfloat Ratio, const ALfloat KneeDb,
+    const ALfloat AttackTime, const ALfloat ReleaseTime);
+
+void ApplyCompression(struct Compressor *Comp, const ALsizei SamplesToDo,
+                      ALfloat (*restrict OutBuffer)[BUFFERSIZE]);
+
+ALsizei GetCompressorLookAhead(const struct Compressor *Comp);
+
+#endif /* MASTERING_H */

Alc/mixer/defs.h

@@ -0,0 +1,119 @@
+#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_copy_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_point_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_lerp_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_cubic_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
+const ALfloat *Resample_bsinc_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, ALsizei *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_lerp_SSE2(const InterpState *state, const ALfloat *restrict src,
+                                  ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                  ALsizei numsamples);
+const ALfloat *Resample_lerp_SSE41(const InterpState *state, const ALfloat *restrict src,
+                                   ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                   ALsizei numsamples);
+
+const ALfloat *Resample_bsinc_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_lerp_Neon(const InterpState *state, const ALfloat *restrict src,
+                                  ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                  ALsizei numsamples);
+const ALfloat *Resample_bsinc_Neon(const InterpState *state, const ALfloat *restrict src,
+                                   ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                   ALsizei dstlen);
+
+#endif /* MIXER_DEFS_H */

Alc/mixer/hrtf_inc.c

@@ -0,0 +1,128 @@
+#include "config.h"
+
+#include "alMain.h"
+#include "alSource.h"
+
+#include "hrtf.h"
+#include "align.h"
+#include "alu.h"
+#include "defs.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] };
+    const ALfloat gainstep = hrtfparams->GainStep;
+    const ALfloat gain = hrtfparams->Gain;
+    ALfloat g, stepcount = 0.0f;
+    ALfloat left, right;
+    ALsizei i;
+
+    ASSUME(IrSize >= 4);
+    ASSUME(BufferSize > 0);
+
+    LeftOut  += OutPos;
+    RightOut += OutPos;
+    for(i = 0;i < BufferSize;i++)
+    {
+        hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++);
+
+        g = gain + gainstep*stepcount;
+        left = hrtfstate->History[(Offset-Delay[0])&HRTF_HISTORY_MASK]*g;
+        right = hrtfstate->History[(Offset-Delay[1])&HRTF_HISTORY_MASK]*g;
+
+        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];
+
+        stepcount += 1.0f;
+        Offset++;
+    }
+    hrtfparams->Gain = gain + gainstep*stepcount;
+}
+
+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] };
+    const ALfloat oldGain = oldparams->Gain;
+    const ALfloat oldGainStep = -oldGain / (ALfloat)BufferSize;
+    const ALfloat (*NewCoeffs)[2] = ASSUME_ALIGNED(newparams->Coeffs, 16);
+    const ALsizei NewDelay[2] = { newparams->Delay[0], newparams->Delay[1] };
+    const ALfloat newGain = newparams->Gain;
+    const ALfloat newGainStep = newparams->GainStep;
+    ALfloat g, stepcount = 0.0f;
+    ALfloat left, right;
+    ALsizei i;
+
+    ASSUME(IrSize >= 4);
+    ASSUME(BufferSize > 0);
+
+    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++);
+
+        g = oldGain + oldGainStep*stepcount;
+        left = hrtfstate->History[(Offset-OldDelay[0])&HRTF_HISTORY_MASK]*g;
+        right = hrtfstate->History[(Offset-OldDelay[1])&HRTF_HISTORY_MASK]*g;
+        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, OldCoeffs, left, right);
+
+        g = newGain + newGainStep*stepcount;
+        left = hrtfstate->History[(Offset-NewDelay[0])&HRTF_HISTORY_MASK]*g;
+        right = hrtfstate->History[(Offset-NewDelay[1])&HRTF_HISTORY_MASK]*g;
+        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, NewCoeffs, left, right);
+
+        *(LeftOut++)  += hrtfstate->Values[Offset&HRIR_MASK][0];
+        *(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1];
+
+        stepcount += 1.0f;
+        Offset++;
+    }
+    newparams->Gain = newGain + newGainStep*stepcount;
+}
+
+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;
+
+    ASSUME(IrSize >= 4);
+    ASSUME(BufferSize > 0);
+
+    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/mixer_c.c

@@ -0,0 +1,169 @@
+#include "config.h"
+
+#include <assert.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "alSource.h"
+#include "alAuxEffectSlot.h"
+#include "defs.h"
+
+
+static inline ALfloat do_point(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei UNUSED(frac))
+{ return vals[0]; }
+static inline ALfloat do_lerp(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei frac)
+{ return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); }
+static inline ALfloat do_cubic(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei frac)
+{ return cubic(vals[0], vals[1], vals[2], vals[3], frac * (1.0f/FRACTIONONE)); }
+static inline ALfloat do_bsinc(const InterpState *state, const ALfloat *restrict vals, ALsizei frac)
+{
+    const ALfloat *fil, *scd, *phd, *spd;
+    ALsizei j_f, pi;
+    ALfloat pf, r;
+
+    ASSUME(state->bsinc.m > 0);
+
+    // 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.filter + state->bsinc.m*pi*4, 16);
+    scd = ASSUME_ALIGNED(fil + state->bsinc.m, 16);
+    phd = ASSUME_ALIGNED(scd + state->bsinc.m, 16);
+    spd = ASSUME_ALIGNED(phd + state->bsinc.m, 16);
+
+    // Apply the scale and phase interpolated filter.
+    r = 0.0f;
+    for(j_f = 0;j_f < state->bsinc.m;j_f++)
+        r += (fil[j_f] + state->bsinc.sf*scd[j_f] + pf*(phd[j_f] + state->bsinc.sf*spd[j_f])) * vals[j_f];
+    return r;
+}
+
+const ALfloat *Resample_copy_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(Tag, Sampler, O)                                        \
+const ALfloat *Resample_##Tag##_C(const InterpState *state,                   \
+  const ALfloat *restrict src, ALsizei frac, ALint increment,                 \
+  ALfloat *restrict dst, ALsizei numsamples)                                  \
+{                                                                             \
+    const InterpState istate = *state;                                        \
+    ALsizei i;                                                                \
+                                                                              \
+    ASSUME(numsamples > 0);                                                   \
+                                                                              \
+    src -= O;                                                                 \
+    for(i = 0;i < numsamples;i++)                                             \
+    {                                                                         \
+        dst[i] = Sampler(&istate, src, frac);                                 \
+                                                                              \
+        frac += increment;                                                    \
+        src  += frac>>FRACTIONBITS;                                           \
+        frac &= FRACTIONMASK;                                                 \
+    }                                                                         \
+    return dst;                                                               \
+}
+
+DECL_TEMPLATE(point, do_point, 0)
+DECL_TEMPLATE(lerp, do_lerp, 0)
+DECL_TEMPLATE(cubic, do_cubic, 1)
+DECL_TEMPLATE(bsinc, do_bsinc, istate.bsinc.l)
+
+#undef DECL_TEMPLATE
+
+
+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 "hrtf_inc.c"
+
+
+void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+           ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+           ALsizei BufferSize)
+{
+    const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
+    ALsizei c;
+
+    ASSUME(OutChans > 0);
+    ASSUME(BufferSize > 0);
+
+    for(c = 0;c < OutChans;c++)
+    {
+        ALsizei pos = 0;
+        ALfloat gain = CurrentGains[c];
+        const ALfloat diff = TargetGains[c] - gain;
+
+        if(fabsf(diff) > FLT_EPSILON)
+        {
+            ALsizei minsize = mini(BufferSize, Counter);
+            const ALfloat step = diff * delta;
+            ALfloat step_count = 0.0f;
+            for(;pos < minsize;pos++)
+            {
+                OutBuffer[c][OutPos+pos] += data[pos] * (gain + step*step_count);
+                step_count += 1.0f;
+            }
+            if(pos == Counter)
+                gain = TargetGains[c];
+            else
+                gain += step*step_count;
+            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;
+
+    ASSUME(InChans > 0);
+    ASSUME(BufferSize > 0);
+
+    for(c = 0;c < InChans;c++)
+    {
+        const 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/mixer_neon.c

@@ -0,0 +1,283 @@
+#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 "defs.h"
+
+
+const ALfloat *Resample_lerp_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) ALsizei pos_[4], frac_[4];
+    int32x4_t pos4, frac4;
+    ALsizei todo, pos, i;
+
+    ASSUME(numsamples > 0);
+
+    InitiatePositionArrays(frac, increment, frac_, pos_, 4);
+    frac4 = vld1q_s32(frac_);
+    pos4 = vld1q_s32(pos_);
+
+    todo = numsamples & ~3;
+    for(i = 0;i < todo;i += 4)
+    {
+        const int pos0 = vgetq_lane_s32(pos4, 0);
+        const int pos1 = vgetq_lane_s32(pos4, 1);
+        const int pos2 = vgetq_lane_s32(pos4, 2);
+        const int pos3 = vgetq_lane_s32(pos4, 3);
+        const float32x4_t val1 = (float32x4_t){src[pos0], src[pos1], src[pos2], src[pos3]};
+        const float32x4_t val2 = (float32x4_t){src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+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);
+    }
+
+    /* NOTE: These four elements represent the position *after* the last four
+     * samples, so the lowest element is the next position to resample.
+     */
+    pos = vgetq_lane_s32(pos4, 0);
+    frac = vgetq_lane_s32(frac4, 0);
+
+    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_bsinc_Neon(const InterpState *state,
+  const ALfloat *restrict src, ALsizei frac, ALint increment,
+  ALfloat *restrict dst, ALsizei dstlen)
+{
+    const ALfloat *const filter = state->bsinc.filter;
+    const float32x4_t sf4 = vdupq_n_f32(state->bsinc.sf);
+    const ALsizei m = state->bsinc.m;
+    const float32x4_t *fil, *scd, *phd, *spd;
+    ALsizei pi, i, j, offset;
+    float32x4_t r4;
+    ALfloat pf;
+
+    ASSUME(m > 0);
+    ASSUME(dstlen > 0);
+
+    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
+
+        offset = m*pi*4;
+        fil = ASSUME_ALIGNED(filter + offset, 16); offset += m;
+        scd = ASSUME_ALIGNED(filter + offset, 16); offset += m;
+        phd = ASSUME_ALIGNED(filter + offset, 16); offset += m;
+        spd = ASSUME_ALIGNED(filter + offset, 16);
+
+        // Apply the scale and phase interpolated filter.
+        r4 = vdupq_n_f32(0.0f);
+        {
+            const ALsizei count = m >> 2;
+            const float32x4_t pf4 = vdupq_n_f32(pf);
+
+            ASSUME(count > 0);
+
+            for(j = 0;j < count;j++)
+            {
+                /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
+                const float32x4_t f4 = vmlaq_f32(
+                    vmlaq_f32(fil[j], sf4, scd[j]),
+                    pf4, vmlaq_f32(phd[j], sf4, spd[j])
+                );
+                /* r += f*src */
+                r4 = vmlaq_f32(r4, f4, vld1q_f32(&src[j*4]));
+            }
+        }
+        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 "hrtf_inc.c"
+
+
+void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+              ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+              ALsizei BufferSize)
+{
+    const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
+    ALsizei c;
+
+    ASSUME(OutChans > 0);
+    ASSUME(BufferSize > 0);
+    data = ASSUME_ALIGNED(data, 16);
+    OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
+
+    for(c = 0;c < OutChans;c++)
+    {
+        ALsizei pos = 0;
+        ALfloat gain = CurrentGains[c];
+        const ALfloat diff = TargetGains[c] - gain;
+
+        if(fabsf(diff) > FLT_EPSILON)
+        {
+            ALsizei minsize = mini(BufferSize, Counter);
+            const ALfloat step = diff * delta;
+            ALfloat step_count = 0.0f;
+            /* Mix with applying gain steps in aligned multiples of 4. */
+            if(LIKELY(minsize > 3))
+            {
+                const float32x4_t four4 = vdupq_n_f32(4.0f);
+                const float32x4_t step4 = vdupq_n_f32(step);
+                const float32x4_t gain4 = vdupq_n_f32(gain);
+                float32x4_t step_count4 = vsetq_lane_f32(0.0f,
+                    vsetq_lane_f32(1.0f,
+                    vsetq_lane_f32(2.0f,
+                    vsetq_lane_f32(3.0f, vdupq_n_f32(0.0f), 3),
+                    2), 1), 0
+                );
+                ALsizei todo = minsize >> 2;
+
+                do {
+                    const float32x4_t val4 = vld1q_f32(&data[pos]);
+                    float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]);
+                    dry4 = vmlaq_f32(dry4, val4, vmlaq_f32(gain4, step4, step_count4));
+                    step_count4 = vaddq_f32(step_count4, four4);
+                    vst1q_f32(&OutBuffer[c][OutPos+pos], dry4);
+                    pos += 4;
+                } while(--todo);
+                /* NOTE: step_count4 now represents the next four counts after
+                 * the last four mixed samples, so the lowest element
+                 * represents the next step count to apply.
+                 */
+                step_count = vgetq_lane_f32(step_count4, 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 + step*step_count);
+                step_count += 1.0f;
+            }
+            if(pos == Counter)
+                gain = TargetGains[c];
+            else
+                gain += step*step_count;
+            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;
+        if(LIKELY(BufferSize-pos > 3))
+        {
+            ALsizei todo = (BufferSize-pos) >> 2;
+            const float32x4_t gain4 = vdupq_n_f32(gain);
+            do {
+                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);
+                pos += 4;
+            } while(--todo);
+        }
+        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)
+{
+    ALsizei c;
+
+    ASSUME(InChans > 0);
+    ASSUME(BufferSize > 0);
+
+    for(c = 0;c < InChans;c++)
+    {
+        ALsizei pos = 0;
+        const ALfloat gain = Gains[c];
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+
+        if(LIKELY(BufferSize > 3))
+        {
+            ALsizei todo = BufferSize >> 2;
+            float32x4_t gain4 = vdupq_n_f32(gain);
+            do {
+                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);
+                pos += 4;
+            } while(--todo);
+        }
+        for(;pos < BufferSize;pos++)
+            OutBuffer[pos] += data[c][InPos+pos]*gain;
+    }
+}

Alc/mixer/mixer_sse.c

@@ -0,0 +1,250 @@
+#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 "defs.h"
+
+
+const ALfloat *Resample_bsinc_SSE(const InterpState *state, const ALfloat *restrict src,
+                                  ALsizei frac, ALint increment, ALfloat *restrict dst,
+                                  ALsizei dstlen)
+{
+    const ALfloat *const filter = state->bsinc.filter;
+    const __m128 sf4 = _mm_set1_ps(state->bsinc.sf);
+    const ALsizei m = state->bsinc.m;
+    const __m128 *fil, *scd, *phd, *spd;
+    ALsizei pi, i, j, offset;
+    ALfloat pf;
+    __m128 r4;
+
+    ASSUME(m > 0);
+    ASSUME(dstlen > 0);
+
+    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
+
+        offset = m*pi*4;
+        fil = (const __m128*)ASSUME_ALIGNED(filter + offset, 16); offset += m;
+        scd = (const __m128*)ASSUME_ALIGNED(filter + offset, 16); offset += m;
+        phd = (const __m128*)ASSUME_ALIGNED(filter + offset, 16); offset += m;
+        spd = (const __m128*)ASSUME_ALIGNED(filter + offset, 16);
+
+        // Apply the scale and phase interpolated filter.
+        r4 = _mm_setzero_ps();
+        {
+            const ALsizei count = m >> 2;
+            const __m128 pf4 = _mm_set1_ps(pf);
+
+            ASSUME(count > 0);
+
+#define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z))
+            for(j = 0;j < count;j++)
+            {
+                /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
+                const __m128 f4 = MLA4(
+                    MLA4(fil[j], sf4, scd[j]),
+                    pf4, MLA4(phd[j], sf4, spd[j])
+                );
+                /* r += f*src */
+                r4 = MLA4(r4, f4, _mm_loadu_ps(&src[j*4]));
+            }
+#undef MLA4
+        }
+        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 "hrtf_inc.c"
+
+
+void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
+             ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
+             ALsizei BufferSize)
+{
+    const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
+    ALsizei c;
+
+    ASSUME(OutChans > 0);
+    ASSUME(BufferSize > 0);
+
+    for(c = 0;c < OutChans;c++)
+    {
+        ALsizei pos = 0;
+        ALfloat gain = CurrentGains[c];
+        const ALfloat diff = TargetGains[c] - gain;
+
+        if(fabsf(diff) > FLT_EPSILON)
+        {
+            ALsizei minsize = mini(BufferSize, Counter);
+            const ALfloat step = diff * delta;
+            ALfloat step_count = 0.0f;
+            /* Mix with applying gain steps in aligned multiples of 4. */
+            if(LIKELY(minsize > 3))
+            {
+                const __m128 four4 = _mm_set1_ps(4.0f);
+                const __m128 step4 = _mm_set1_ps(step);
+                const __m128 gain4 = _mm_set1_ps(gain);
+                __m128 step_count4 = _mm_setr_ps(0.0f, 1.0f, 2.0f, 3.0f);
+                ALsizei todo = minsize >> 2;
+                do {
+                    const __m128 val4 = _mm_load_ps(&data[pos]);
+                    __m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]);
+#define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z))
+                    /* dry += val * (gain + step*step_count) */
+                    dry4 = MLA4(dry4, val4, MLA4(gain4, step4, step_count4));
+#undef MLA4
+                    _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
+                    step_count4 = _mm_add_ps(step_count4, four4);
+                    pos += 4;
+                } while(--todo);
+                /* NOTE: step_count4 now represents the next four counts after
+                 * the last four mixed samples, so the lowest element
+                 * represents the next step count to apply.
+                 */
+                step_count = _mm_cvtss_f32(step_count4);
+            }
+            /* 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 + step*step_count);
+                step_count += 1.0f;
+            }
+            if(pos == Counter)
+                gain = TargetGains[c];
+            else
+                gain += step*step_count;
+            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;
+        if(LIKELY(BufferSize-pos > 3))
+        {
+            ALsizei todo = (BufferSize-pos) >> 2;
+            const __m128 gain4 = _mm_set1_ps(gain);
+            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));
+                _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
+                pos += 4;
+            } while(--todo);
+        }
+        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)
+{
+    ALsizei c;
+
+    ASSUME(InChans > 0);
+    ASSUME(BufferSize > 0);
+
+    for(c = 0;c < InChans;c++)
+    {
+        ALsizei pos = 0;
+        const ALfloat gain = Gains[c];
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+            continue;
+
+        if(LIKELY(BufferSize > 3))
+        {
+            ALsizei todo = BufferSize >> 2;
+            const __m128 gain4 = _mm_set1_ps(gain);
+            do {
+                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);
+                pos += 4;
+            } while(--todo);
+        }
+        for(;pos < BufferSize;pos++)
+            OutBuffer[pos] += data[c][InPos+pos]*gain;
+    }
+}

Alc/mixer/mixer_sse2.c

@@ -0,0 +1,84 @@
+/**
+ * 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 "defs.h"
+
+
+const ALfloat *Resample_lerp_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);
+    alignas(16) ALsizei pos_[4], frac_[4];
+    __m128i frac4, pos4;
+    ALsizei todo, pos, i;
+
+    ASSUME(numsamples > 0);
+
+    InitiatePositionArrays(frac, increment, frac_, pos_, 4);
+    frac4 = _mm_setr_epi32(frac_[0], frac_[1], frac_[2], frac_[3]);
+    pos4 = _mm_setr_epi32(pos_[0], pos_[1], pos_[2], pos_[3]);
+
+    todo = numsamples & ~3;
+    for(i = 0;i < todo;i += 4)
+    {
+        const int pos0 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(0, 0, 0, 0)));
+        const int pos1 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(1, 1, 1, 1)));
+        const int pos2 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(2, 2, 2, 2)));
+        const int pos3 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(3, 3, 3, 3)));
+        const __m128 val1 = _mm_setr_ps(src[pos0  ], src[pos1  ], src[pos2  ], src[pos3  ]);
+        const __m128 val2 = _mm_setr_ps(src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+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);
+    }
+
+    /* NOTE: These four elements represent the position *after* the last four
+     * samples, so the lowest element is the next position to resample.
+     */
+    pos = _mm_cvtsi128_si32(pos4);
+    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/mixer_sse41.c

@@ -0,0 +1,85 @@
+/**
+ * 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 "defs.h"
+
+
+const ALfloat *Resample_lerp_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);
+    alignas(16) ALsizei pos_[4], frac_[4];
+    __m128i frac4, pos4;
+    ALsizei todo, pos, i;
+
+    ASSUME(numsamples > 0);
+
+    InitiatePositionArrays(frac, increment, frac_, pos_, 4);
+    frac4 = _mm_setr_epi32(frac_[0], frac_[1], frac_[2], frac_[3]);
+    pos4 = _mm_setr_epi32(pos_[0], pos_[1], pos_[2], pos_[3]);
+
+    todo = numsamples & ~3;
+    for(i = 0;i < todo;i += 4)
+    {
+        const int pos0 = _mm_extract_epi32(pos4, 0);
+        const int pos1 = _mm_extract_epi32(pos4, 1);
+        const int pos2 = _mm_extract_epi32(pos4, 2);
+        const int pos3 = _mm_extract_epi32(pos4, 3);
+        const __m128 val1 = _mm_setr_ps(src[pos0  ], src[pos1  ], src[pos2  ], src[pos3  ]);
+        const __m128 val2 = _mm_setr_ps(src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+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);
+    }
+
+    /* NOTE: These four elements represent the position *after* the last four
+     * samples, so the lowest element is the next position to resample.
+     */
+    pos = _mm_cvtsi128_si32(pos4);
+    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/mixvoice.c

@@ -0,0 +1,762 @@
+/**
+ * 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 "sample_cvt.h"
+#include "alu.h"
+#include "alconfig.h"
+#include "ringbuffer.h"
+
+#include "cpu_caps.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, ALsizei *restrict pos_arr, ALsizei size);
+
+
+/* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */
+static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!");
+
+
+enum Resampler ResamplerDefault = LinearResampler;
+
+MixerFunc MixSamples = Mix_C;
+RowMixerFunc MixRowSamples = MixRow_C;
+static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
+static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
+
+static 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;
+}
+
+static 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_point_C;
+        case LinearResampler:
+#ifdef HAVE_NEON
+            if((CPUCapFlags&CPU_CAP_NEON))
+                return Resample_lerp_Neon;
+#endif
+#ifdef HAVE_SSE4_1
+            if((CPUCapFlags&CPU_CAP_SSE4_1))
+                return Resample_lerp_SSE41;
+#endif
+#ifdef HAVE_SSE2
+            if((CPUCapFlags&CPU_CAP_SSE2))
+                return Resample_lerp_SSE2;
+#endif
+            return Resample_lerp_C;
+        case FIR4Resampler:
+            return Resample_cubic_C;
+        case BSinc12Resampler:
+        case BSinc24Resampler:
+#ifdef HAVE_NEON
+            if((CPUCapFlags&CPU_CAP_NEON))
+                return Resample_bsinc_Neon;
+#endif
+#ifdef HAVE_SSE
+            if((CPUCapFlags&CPU_CAP_SSE))
+                return Resample_bsinc_SSE;
+#endif
+            return Resample_bsinc_C;
+    }
+
+    return Resample_point_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, "cubic") == 0)
+            ResamplerDefault = FIR4Resampler;
+        else if(strcasecmp(str, "bsinc12") == 0)
+            ResamplerDefault = BSinc12Resampler;
+        else if(strcasecmp(str, "bsinc24") == 0)
+            ResamplerDefault = BSinc24Resampler;
+        else if(strcasecmp(str, "bsinc") == 0)
+        {
+            WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
+            ResamplerDefault = BSinc12Resampler;
+        }
+        else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0)
+        {
+            WARN("Resampler option \"%s\" is deprecated, using cubic\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();
+    MixRowSamples = SelectRowMixer();
+}
+
+
+static void SendAsyncEvent(ALCcontext *context, ALuint enumtype, ALenum type,
+                           ALuint objid, ALuint param, const char *msg)
+{
+    AsyncEvent evt = ASYNC_EVENT(enumtype);
+    evt.u.user.type = type;
+    evt.u.user.id = objid;
+    evt.u.user.param = param;
+    strcpy(evt.u.user.msg, msg);
+    if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1)
+        alsem_post(&context->EventSem);
+}
+
+
+static inline ALfloat Sample_ALubyte(ALubyte val)
+{ return (val-128) * (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; }
+
+static inline ALfloat Sample_ALdouble(ALdouble val)
+{ return (ALfloat)val; }
+
+typedef ALubyte ALmulaw;
+static inline ALfloat Sample_ALmulaw(ALmulaw val)
+{ return muLawDecompressionTable[val] * (1.0f/32768.0f); }
+
+typedef ALubyte ALalaw;
+static inline ALfloat Sample_ALalaw(ALalaw val)
+{ return aLawDecompressionTable[val] * (1.0f/32768.0f); }
+
+#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(ALubyte)
+DECL_TEMPLATE(ALshort)
+DECL_TEMPLATE(ALfloat)
+DECL_TEMPLATE(ALdouble)
+DECL_TEMPLATE(ALmulaw)
+DECL_TEMPLATE(ALalaw)
+
+#undef DECL_TEMPLATE
+
+static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
+                        enum FmtType srctype, ALsizei samples)
+{
+#define HANDLE_FMT(ET, ST) case ET: Load_##ST(dst, src, srcstep, samples); break
+    switch(srctype)
+    {
+        HANDLE_FMT(FmtUByte, ALubyte);
+        HANDLE_FMT(FmtShort, ALshort);
+        HANDLE_FMT(FmtFloat, ALfloat);
+        HANDLE_FMT(FmtDouble, ALdouble);
+        HANDLE_FMT(FmtMulaw, ALmulaw);
+        HANDLE_FMT(FmtAlaw, ALalaw);
+    }
+#undef HANDLE_FMT
+}
+
+
+static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
+                                ALfloat *restrict dst, const ALfloat *restrict src,
+                                ALsizei numsamples, enum ActiveFilters type)
+{
+    ALsizei i;
+    switch(type)
+    {
+        case AF_None:
+            BiquadFilter_passthru(lpfilter, numsamples);
+            BiquadFilter_passthru(hpfilter, numsamples);
+            break;
+
+        case AF_LowPass:
+            BiquadFilter_process(lpfilter, dst, src, numsamples);
+            BiquadFilter_passthru(hpfilter, numsamples);
+            return dst;
+        case AF_HighPass:
+            BiquadFilter_passthru(lpfilter, numsamples);
+            BiquadFilter_process(hpfilter, dst, src, numsamples);
+            return dst;
+
+        case AF_BandPass:
+            for(i = 0;i < numsamples;)
+            {
+                ALfloat temp[256];
+                ALsizei todo = mini(256, numsamples-i);
+
+                BiquadFilter_process(lpfilter, temp, src+i, todo);
+                BiquadFilter_process(hpfilter, dst+i, temp, todo);
+                i += todo;
+            }
+            return dst;
+    }
+    return src;
+}
+
+
+/* This function uses these device temp buffers. */
+#define SOURCE_DATA_BUF 0
+#define RESAMPLED_BUF 1
+#define FILTERED_BUF 2
+#define NFC_DATA_BUF 3
+ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo)
+{
+    ALCdevice *Device = Context->Device;
+    ALbufferlistitem *BufferListItem;
+    ALbufferlistitem *BufferLoopItem;
+    ALsizei NumChannels, SampleSize;
+    ALbitfieldSOFT enabledevt;
+    ALsizei buffers_done = 0;
+    ResamplerFunc Resample;
+    ALsizei DataPosInt;
+    ALsizei DataPosFrac;
+    ALint64 DataSize64;
+    ALint increment;
+    ALsizei Counter;
+    ALsizei OutPos;
+    ALsizei IrSize;
+    bool isplaying;
+    bool firstpass;
+    bool isstatic;
+    ALsizei chan;
+    ALsizei send;
+
+    /* Get source info */
+    isplaying      = true; /* Will only be called while playing. */
+    isstatic       = !!(voice->Flags&VOICE_IS_STATIC);
+    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_copy_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_RESAMPLE_PADDING*2;
+        SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
+
+        /* Figure out how many samples we can actually mix from this. */
+        DataSize64  = SrcBufferSize;
+        DataSize64 -= MAX_RESAMPLE_PADDING*2;
+        DataSize64 <<= FRACTIONBITS;
+        DataSize64 -= DataPosFrac;
+        DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment,
+                                        SamplesToDo - OutPos);
+
+        /* Some mixers like having a multiple of 4, so try to give that unless
+         * this is the last update. */
+        if(DstBufferSize < SamplesToDo-OutPos)
+            DstBufferSize &= ~3;
+
+        /* It's impossible to have a buffer list item with no entries. */
+        assert(BufferListItem->num_buffers > 0);
+
+        for(chan = 0;chan < NumChannels;chan++)
+        {
+            const ALfloat *ResampledData;
+            ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF];
+            ALsizei FilledAmt;
+
+            /* Load the previous samples into the source data first, and clear the rest. */
+            memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat));
+            memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)*
+                                                    sizeof(ALfloat));
+            FilledAmt = MAX_RESAMPLE_PADDING;
+
+            if(isstatic)
+            {
+                /* TODO: For static sources, loop points are taken from the
+                 * first buffer (should be adjusted by any buffer offset, to
+                 * possibly be added later).
+                 */
+                const ALbuffer *Buffer0 = BufferListItem->buffers[0];
+                const ALsizei LoopStart = Buffer0->LoopStart;
+                const ALsizei LoopEnd   = Buffer0->LoopEnd;
+                const ALsizei LoopSize  = LoopEnd - LoopStart;
+
+                /* If current pos is beyond the loop range, do not loop */
+                if(!BufferLoopItem || DataPosInt >= LoopEnd)
+                {
+                    ALsizei SizeToDo = SrcBufferSize - FilledAmt;
+                    ALsizei CompLen = 0;
+                    ALsizei i;
+
+                    BufferLoopItem = NULL;
+
+                    for(i = 0;i < BufferListItem->num_buffers;i++)
+                    {
+                        const ALbuffer *buffer = BufferListItem->buffers[i];
+                        const ALubyte *Data = buffer->data;
+                        ALsizei DataSize;
+
+                        if(DataPosInt >= buffer->SampleLen)
+                            continue;
+
+                        /* Load what's left to play from the buffer */
+                        DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
+                        CompLen = maxi(CompLen, DataSize);
+
+                        LoadSamples(&SrcData[FilledAmt],
+                            &Data[(DataPosInt*NumChannels + chan)*SampleSize],
+                            NumChannels, buffer->FmtType, DataSize
+                        );
+                    }
+                    FilledAmt += CompLen;
+                }
+                else
+                {
+                    ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt);
+                    ALsizei CompLen = 0;
+                    ALsizei i;
+
+                    for(i = 0;i < BufferListItem->num_buffers;i++)
+                    {
+                        const ALbuffer *buffer = BufferListItem->buffers[i];
+                        const ALubyte *Data = buffer->data;
+                        ALsizei DataSize;
+
+                        if(DataPosInt >= buffer->SampleLen)
+                            continue;
+
+                        /* Load what's left of this loop iteration */
+                        DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
+                        CompLen = maxi(CompLen, DataSize);
+
+                        LoadSamples(&SrcData[FilledAmt],
+                            &Data[(DataPosInt*NumChannels + chan)*SampleSize],
+                            NumChannels, buffer->FmtType, DataSize
+                        );
+                    }
+                    FilledAmt += CompLen;
+
+                    while(SrcBufferSize > FilledAmt)
+                    {
+                        const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize);
+
+                        CompLen = 0;
+                        for(i = 0;i < BufferListItem->num_buffers;i++)
+                        {
+                            const ALbuffer *buffer = BufferListItem->buffers[i];
+                            const ALubyte *Data = buffer->data;
+                            ALsizei DataSize;
+
+                            if(LoopStart >= buffer->SampleLen)
+                                continue;
+
+                            DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
+                            CompLen = maxi(CompLen, DataSize);
+
+                            LoadSamples(&SrcData[FilledAmt],
+                                &Data[(LoopStart*NumChannels + chan)*SampleSize],
+                                NumChannels, buffer->FmtType, DataSize
+                            );
+                        }
+                        FilledAmt += CompLen;
+                    }
+                }
+            }
+            else
+            {
+                /* Crawl the buffer queue to fill in the temp buffer */
+                ALbufferlistitem *tmpiter = BufferListItem;
+                ALsizei pos = DataPosInt;
+
+                while(tmpiter && SrcBufferSize > FilledAmt)
+                {
+                    ALsizei SizeToDo = SrcBufferSize - FilledAmt;
+                    ALsizei CompLen = 0;
+                    ALsizei i;
+
+                    for(i = 0;i < tmpiter->num_buffers;i++)
+                    {
+                        const ALbuffer *ALBuffer = tmpiter->buffers[i];
+                        ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0;
+
+                        if(DataSize > pos)
+                        {
+                            const ALubyte *Data = ALBuffer->data;
+                            Data += (pos*NumChannels + chan)*SampleSize;
+
+                            DataSize = mini(SizeToDo, DataSize - pos);
+                            CompLen = maxi(CompLen, DataSize);
+
+                            LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
+                                        ALBuffer->FmtType, DataSize);
+                        }
+                    }
+                    if(UNLIKELY(!CompLen))
+                        pos -= tmpiter->max_samples;
+                    else
+                    {
+                        FilledAmt += CompLen;
+                        if(SrcBufferSize <= FilledAmt)
+                            break;
+                        pos = 0;
+                    }
+                    tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
+                    if(!tmpiter) tmpiter = BufferLoopItem;
+                }
+            }
+
+            /* Store the last source samples used for next time. */
+            memcpy(voice->PrevSamples[chan],
+                &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
+                MAX_RESAMPLE_PADDING*sizeof(ALfloat)
+            );
+
+            /* Now resample, then filter and mix to the appropriate outputs. */
+            ResampledData = Resample(&voice->ResampleState,
+                &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
+                Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
+            );
+            {
+                DirectParams *parms = &voice->Direct.Params[chan];
+                const ALfloat *samples;
+
+                samples = DoFilters(
+                    &parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
+                    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->TempBuffer[NFC_DATA_BUF];
+                        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)                             \
+    {                                                                         \
+        NfcFilterProcess##order(&parms->NFCtrlFilter, 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 = 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 = 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->TempBuffer[FILTERED_BUF],
+                    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;
+
+        if(isstatic)
+        {
+            if(BufferLoopItem)
+            {
+                /* Handle looping static source */
+                const ALbuffer *Buffer = BufferListItem->buffers[0];
+                ALsizei LoopStart = Buffer->LoopStart;
+                ALsizei LoopEnd = Buffer->LoopEnd;
+                if(DataPosInt >= LoopEnd)
+                {
+                    assert(LoopEnd > LoopStart);
+                    DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
+                }
+            }
+            else
+            {
+                /* Handle non-looping static source */
+                if(DataPosInt >= BufferListItem->max_samples)
+                {
+                    isplaying = false;
+                    BufferListItem = NULL;
+                    DataPosInt = 0;
+                    DataPosFrac = 0;
+                    break;
+                }
+            }
+        }
+        else while(1)
+        {
+            /* Handle streaming source */
+            if(BufferListItem->max_samples > DataPosInt)
+                break;
+
+            DataPosInt -= BufferListItem->max_samples;
+
+            buffers_done += BufferListItem->num_buffers;
+            BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_relaxed);
+            if(!BufferListItem && !(BufferListItem=BufferLoopItem))
+            {
+                isplaying = false;
+                DataPosInt = 0;
+                DataPosFrac = 0;
+                break;
+            }
+        }
+    } 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);
+
+    /* Send any events now, after the position/buffer info was updated. */
+    enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire);
+    if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
+        SendAsyncEvent(Context, EventType_BufferCompleted,
+            AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed"
+        );
+
+    return isplaying;
+}

Alc/polymorphism.h

@@ -0,0 +1,105 @@
+#ifndef POLYMORPHISM_H
+#define POLYMORPHISM_H
+
+/* Macros to declare inheriting types, and to (down-)cast and up-cast. */
+#define DERIVE_FROM_TYPE(t)          t t##_parent
+#define STATIC_CAST(to, obj)         (&(obj)->to##_parent)
+#ifdef __GNUC__
+#define STATIC_UPCAST(to, from, obj) __extension__({                          \
+    static_assert(__builtin_types_compatible_p(from, __typeof(*(obj))),       \
+                  "Invalid upcast object from type");                         \
+    (to*)((char*)(obj) - offsetof(to, from##_parent));                        \
+})
+#else
+#define STATIC_UPCAST(to, from, obj) ((to*)((char*)(obj) - offsetof(to, from##_parent)))
+#endif
+
+/* Defines method forwards, which call the given parent's (T2's) implementation. */
+#define DECLARE_FORWARD(T1, T2, rettype, func)                                \
+rettype T1##_##func(T1 *obj)                                                  \
+{ return T2##_##func(STATIC_CAST(T2, obj)); }
+
+#define DECLARE_FORWARD1(T1, T2, rettype, func, argtype1)                     \
+rettype T1##_##func(T1 *obj, argtype1 a)                                      \
+{ return T2##_##func(STATIC_CAST(T2, obj), a); }
+
+#define DECLARE_FORWARD2(T1, T2, rettype, func, argtype1, argtype2)           \
+rettype T1##_##func(T1 *obj, argtype1 a, argtype2 b)                          \
+{ return T2##_##func(STATIC_CAST(T2, obj), a, b); }
+
+#define DECLARE_FORWARD3(T1, T2, rettype, func, argtype1, argtype2, argtype3) \
+rettype T1##_##func(T1 *obj, argtype1 a, argtype2 b, argtype3 c)              \
+{ return T2##_##func(STATIC_CAST(T2, obj), a, b, c); }
+
+/* Defines method thunks, functions that call to the child's method. */
+#define DECLARE_THUNK(T1, T2, rettype, func)                                  \
+static rettype T1##_##T2##_##func(T2 *obj)                                    \
+{ return T1##_##func(STATIC_UPCAST(T1, T2, obj)); }
+
+#define DECLARE_THUNK1(T1, T2, rettype, func, argtype1)                       \
+static rettype T1##_##T2##_##func(T2 *obj, argtype1 a)                        \
+{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a); }
+
+#define DECLARE_THUNK2(T1, T2, rettype, func, argtype1, argtype2)             \
+static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b)            \
+{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b); }
+
+#define DECLARE_THUNK3(T1, T2, rettype, func, argtype1, argtype2, argtype3)   \
+static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b, argtype3 c) \
+{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b, c); }
+
+#define DECLARE_THUNK4(T1, T2, rettype, func, argtype1, argtype2, argtype3, argtype4) \
+static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b, argtype3 c, argtype4 d) \
+{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b, c, d); }
+
+/* Defines the default functions used to (de)allocate a polymorphic object. */
+#define DECLARE_DEFAULT_ALLOCATORS(T)                                         \
+static void* T##_New(size_t size) { return al_malloc(16, size); }             \
+static void T##_Delete(void *ptr) { al_free(ptr); }
+
+
+/* Helper to extract an argument list for virtual method calls. */
+#define EXTRACT_VCALL_ARGS(...)  __VA_ARGS__))
+
+/* Call a "virtual" method on an object, with arguments. */
+#define V(obj, func)  ((obj)->vtbl->func((obj), EXTRACT_VCALL_ARGS
+/* Call a "virtual" method on an object, with no arguments. */
+#define V0(obj, func) ((obj)->vtbl->func((obj) EXTRACT_VCALL_ARGS
+
+
+/* Helper to extract an argument list for NEW_OBJ calls. */
+#define EXTRACT_NEW_ARGS(...)  __VA_ARGS__);                                  \
+    }                                                                         \
+} while(0)
+
+/* Allocate and construct an object, with arguments. */
+#define NEW_OBJ(_res, T) do {                                                 \
+    _res = T##_New(sizeof(T));                                                \
+    if(_res)                                                                  \
+    {                                                                         \
+        memset(_res, 0, sizeof(T));                                           \
+        T##_Construct(_res, EXTRACT_NEW_ARGS
+/* Allocate and construct an object, with no arguments. */
+#define NEW_OBJ0(_res, T) do {                                                \
+    _res = T##_New(sizeof(T));                                                \
+    if(_res)                                                                  \
+    {                                                                         \
+        memset(_res, 0, sizeof(T));                                           \
+        T##_Construct(_res EXTRACT_NEW_ARGS
+
+/* Destructs and deallocate an object. */
+#define DELETE_OBJ(obj) do {                                                  \
+    if((obj) != NULL)                                                         \
+    {                                                                         \
+        V0((obj),Destruct)();                                                 \
+        V0((obj),Delete)();                                                   \
+    }                                                                         \
+} while(0)
+
+
+/* Helper to get a type's vtable thunk for a child type. */
+#define GET_VTABLE2(T1, T2) (&(T1##_##T2##_vtable))
+/* Helper to set an object's vtable thunk for a child type. Used when constructing an object. */
+#define SET_VTABLE2(T1, T2, obj) (STATIC_CAST(T2, obj)->vtbl = GET_VTABLE2(T1, T2))
+
+#endif /* POLYMORPHISM_H */

Alc/ringbuffer.c

@@ -0,0 +1,295 @@
+/**
+ * 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 <limits.h>
+
+#include "ringbuffer.h"
+#include "align.h"
+#include "atomic.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;
+
+    alignas(16) char buf[];
+};
+
+ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz, int limit_writes)
+{
+    ll_ringbuffer_t *rb;
+    size_t power_of_two = 0;
+
+    if(sz > 0)
+    {
+        power_of_two = sz;
+        power_of_two |= power_of_two>>1;
+        power_of_two |= power_of_two>>2;
+        power_of_two |= power_of_two>>4;
+        power_of_two |= power_of_two>>8;
+        power_of_two |= power_of_two>>16;
+#if SIZE_MAX > UINT_MAX
+        power_of_two |= power_of_two>>32;
+#endif
+    }
+    power_of_two++;
+    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 = limit_writes ? sz : power_of_two;
+    rb->size_mask = power_of_two - 1;
+    rb->elem_size = elem_sz;
+    return rb;
+}
+
+void ll_ringbuffer_free(ll_ringbuffer_t *rb)
+{
+    al_free(rb);
+}
+
+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_mask+1)*rb->elem_size);
+}
+
+
+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;
+}
+
+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);
+    w = (r-w-1) & rb->size_mask;
+    return (w > rb->size) ? rb->size : w;
+}
+
+
+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_mask+1)
+    {
+        n1 = rb->size_mask+1 - 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;
+}
+
+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_mask+1)
+    {
+        n1 = rb->size_mask+1 - 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;
+}
+
+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_mask+1)
+    {
+        n1 = rb->size_mask+1 - 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;
+}
+
+
+void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt)
+{
+    ATOMIC_ADD(&rb->read_ptr, cnt, almemory_order_acq_rel);
+}
+
+void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt)
+{
+    ATOMIC_ADD(&rb->write_ptr, cnt, almemory_order_acq_rel);
+}
+
+
+void ll_ringbuffer_get_read_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2])
+{
+    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_mask+1)
+    {
+        /* 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_mask+1 - 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;
+    }
+}
+
+void ll_ringbuffer_get_write_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2])
+{
+    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;
+    if(free_cnt > rb->size) free_cnt = rb->size;
+
+    cnt2 = w + free_cnt;
+    if(cnt2 > rb->size_mask+1)
+    {
+        /* 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_mask+1 - 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/ringbuffer.h

@@ -0,0 +1,77 @@
+#ifndef RINGBUFFER_H
+#define RINGBUFFER_H
+
+#include <stddef.h>
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef struct ll_ringbuffer ll_ringbuffer_t;
+typedef struct ll_ringbuffer_data {
+    char *buf;
+    size_t len;
+} ll_ringbuffer_data_t;
+
+
+/**
+ * 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 (even if it is
+ * already a power of two, to ensure the requested amount can be written).
+ */
+ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz, int limit_writes);
+/** Free all data associated with the ringbuffer `rb'. */
+void ll_ringbuffer_free(ll_ringbuffer_t *rb);
+/** Reset the read and write pointers to zero. This is not thread safe. */
+void ll_ringbuffer_reset(ll_ringbuffer_t *rb);
+
+/**
+ * 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[2]);
+/**
+ * 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[2]);
+
+/**
+ * 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);
+/**
+ * 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);
+/**
+ * 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);
+/** Advance the read pointer `cnt' places. */
+void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt);
+
+/**
+ * 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);
+/**
+ * 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);
+/** Advance the write pointer `cnt' places. */
+void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt);
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif /* RINGBUFFER_H */

Alc/uhjfilter.c

@@ -0,0 +1,120 @@
+
+#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 Filter1CoeffSqr[4] = {
+    0.479400865589f, 0.876218493539f, 0.976597589508f, 0.997499255936f
+};
+static const ALfloat Filter2CoeffSqr[4] = {
+    0.161758498368f, 0.733028932341f, 0.945349700329f, 0.990599156685f
+};
+
+static void allpass_process(AllPassState *state, ALfloat *restrict dst, const ALfloat *restrict src, const ALfloat aa, ALsizei todo)
+{
+    ALfloat z1 = state->z[0];
+    ALfloat z2 = state->z[1];
+    ALsizei i;
+
+    for(i = 0;i < todo;i++)
+    {
+        ALfloat input = src[i];
+        ALfloat output = input*aa + z1;
+        z1 = z2; z2 = output*aa - input;
+        dst[i] = output;
+    }
+
+    state->z[0] = z1;
+    state->z[1] = z2;
+}
+
+
+/* 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;
+
+    ASSUME(SamplesToDo > 0);
+
+    for(base = 0;base < SamplesToDo;)
+    {
+        ALsizei todo = mini(SamplesToDo - base, MAX_UPDATE_SAMPLES);
+        ASSUME(todo > 0);
+
+        /* 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], Filter1CoeffSqr[0], todo);
+        allpass_process(&enc->Filter1_Y[1], temp[0], temp[1], Filter1CoeffSqr[1], todo);
+        allpass_process(&enc->Filter1_Y[2], temp[1], temp[0], Filter1CoeffSqr[2], todo);
+        allpass_process(&enc->Filter1_Y[3], temp[0], temp[1], Filter1CoeffSqr[3], 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->LastY;
+        for(i = 1;i < todo;i++)
+            D[i] = temp[0][i-1];
+        enc->LastY = 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], Filter2CoeffSqr[0], todo);
+        allpass_process(&enc->Filter2_WX[1], temp[0], temp[1], Filter2CoeffSqr[1], todo);
+        allpass_process(&enc->Filter2_WX[2], temp[1], temp[0], Filter2CoeffSqr[2], todo);
+        allpass_process(&enc->Filter2_WX[3], temp[0], temp[1], Filter2CoeffSqr[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], Filter1CoeffSqr[0], todo);
+        allpass_process(&enc->Filter1_WX[1], temp[0], temp[1], Filter1CoeffSqr[1], todo);
+        allpass_process(&enc->Filter1_WX[2], temp[1], temp[0], Filter1CoeffSqr[2], todo);
+        allpass_process(&enc->Filter1_WX[3], temp[0], temp[1], Filter1CoeffSqr[3], todo);
+        S[0] = enc->LastWX;
+        for(i = 1;i < todo;i++)
+            S[i] = temp[0][i-1];
+        enc->LastWX = 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 z[2];
+} 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_Y[4];
+    AllPassState Filter2_WX[4];
+    AllPassState Filter1_WX[4];
+    ALfloat LastY, LastWX;
+} 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 +1,94 @@
+#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 = (_x) ? (char*)((_x)->Data) - (char*)(_x) :    \
+                                sizeof(*(_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 */

BSD-3Clause

@@ -1,30 +0,0 @@
-Portions of this software are licensed under the BSD 3-Clause license.
-
-Copyright (c) 2015, Archontis Politis
-Copyright (c) 2019, Christopher Robinson
-All rights reserved.
-
-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.
-
-* Neither the name of Spherical-Harmonic-Transform nor the names of its
-  contributors may be used to endorse or promote products derived from
-  this software without specific prior written permission.
-
-THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS 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 COPYRIGHT HOLDER 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.

ChangeLog

@@ -1,177 +1,3 @@
-openal-soft-1.21.0:
-
-    Updated library codebase to C++14.
-
-    Implemented the AL_SOFT_effect_target extension.
-
-    Implemented the AL_SOFT_events extension.
-
-    Implemented the ALC_SOFT_loopback_bformat extension.
-
-    Improved memory use for mixing voices.
-
-    Improved detection of NEON capabilities.
-
-    Improved handling of PulseAudio devices that lack manual start control.
-
-    Improved mixing performance with PulseAudio.
-
-    Improved high-frequency scaling quality for the HRTF B-Format decoder.
-
-    Improved makemhr's HRIR delay calculation.
-
-    Improved WASAPI capture of mono formats with multichannel input.
-
-    Reimplemented the modulation stage for reverb.
-
-    Enabled real-time mixing priority by default, for backends that use the
-    setting. It can still be disabled in the config file.
-
-    Enabled dual-band processing for the built-in quad and 7.1 output decoders.
-
-    Fixed a potential crash when deleting an effect slot immediately after the
-    last source using it stops.
-
-    Fixed building with the static runtime on MSVC.
-
-    Fixed using source stereo angles outside of -pi...+pi.
-
-    Fixed the buffer processed event count for sources that start with empty
-    buffers.
-
-    Fixed trying to open an unopenable WASAPI device causing all devices to
-    stop working.
-
-    Fixed stale devices when re-enumerating WASAPI devices.
-
-    Fixed using unicode paths with the log file on Windows.
-
-    Fixed DirectSound capture reporting bad sample counts or erroring when
-    reading samples.
-
-    Added an in-progress extension for a callback-driven buffer type.
-
-    Added an in-progress extension for higher-order B-Format buffers.
-
-    Added an in-progress extension for convolution reverb.
-
-    Added an experimental Oboe backend for Android playback. This requires the
-    Oboe sources at build time, so that it's built as a static library included
-    in libopenal.
-
-    Added an option for auto-connecting JACK ports.
-
-    Added greater-than-stereo support to the SoundIO backend.
-
-    Modified the mixer to be fully asynchronous with the external API, and
-    should now be real-time safe. Although alcRenderSamplesSOFT is not due to
-    locking to check the device handle validity.
-
-    Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
-    to non-filtered signal phase.
-
-    Converted examples from SDL_sound to libsndfile. To avoid issues when
-    combining SDL2 and SDL_sound.
-
-    Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
-    https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
-
-    Reduced the maximum number of source sends from 16 to 6.
-
-    Removed the QSA backend. It's been broken for who knows how long.
-
-    Got rid of the compile-time native-tools targets, using cmake and global
-    initialization instead. This should make cross-compiling less troublesome.
-
-openal-soft-1.20.1:
-
-    Implemented the AL_SOFT_direct_channels_remix extension. This extends
-    AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
-    a matching output channel.
-
-    Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
-    support for N3D or SN3D scaling, or ACN channel ordering.
-
-    Fixed a potential voice leak when a source is started and stopped or
-    restarted in quick succession.
-
-    Fixed a potential device reset failure with JACK.
-
-    Improved handling of unsupported channel configurations with WASAPI. Such
-    setups will now try to output at least a stereo mix.
-
-    Improved clarity a bit for the HRTF second-order ambisonic decoder.
-
-    Improved detection of compatible layouts for SOFA files in makemhr and
-    sofa-info.
-
-    Added the ability to resample HRTFs on load. MHR files no longer need to
-    match the device sample rate to be usable.
-
-    Added an option to limit the HRTF's filter length.
-
-openal-soft-1.20.0:
-
-    Converted the library codebase to C++11. A lot of hacks and custom
-    structures have been replaced with standard or cleaner implementations.
-
-    Partially implemented the Vocal Morpher effect.
-
-    Fixed the bsinc SSE resamplers on non-GCC compilers.
-
-    Fixed OpenSL capture.
-
-    Fixed support for extended capture formats with OpenSL.
-
-    Fixed handling of WASAPI not reporting a default device.
-
-    Fixed performance problems relating to semaphores on macOS.
-
-    Modified the bsinc12 resampler's transition band to better avoid aliasing
-    noise.
-
-    Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
-
-    Modified the virtual speaker layout for HRTF B-Format decoding.
-
-    Modified the PulseAudio backend to use a custom processing loop.
-
-    Renamed the makehrtf utility to makemhr.
-
-    Improved the efficiency of the bsinc resamplers when up-sampling.
-
-    Improved the quality of the bsinc resamplers slightly.
-
-    Improved the efficiency of the HRTF filters.
-
-    Improved the HRTF B-Format decoder coefficient generation.
-
-    Improved reverb feedback fading to be more consistent with pan fading.
-
-    Improved handling of sources that end prematurely, avoiding loud clicks.
-
-    Improved the performance of some reverb processing loops.
-
-    Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
-    some quality. Notably, down-sampling has less smooth pitch ramping.
-
-    Added support for SOFA input files with makemhr.
-
-    Added a build option to use pre-built native tools. For cross-compiling,
-    use with caution and ensure the native tools' binaries are kept up-to-date.
-
-    Added an adjust-latency config option for the PulseAudio backend.
-
-    Added basic support for multi-field HRTFs.
-
-    Added an option for mixing first- or second-order B-Format with HRTF
-    output. This can improve HRTF performance given a number of sources.
-
-    Added an RC file for proper DLL version information.
-
-    Disabled some old KDE workarounds by default. Specifically, PulseAudio
-    streams can now be moved (KDE may try to move them after opening).
-
 openal-soft-1.19.1:
 
     Implemented capture support for the SoundIO backend.

OpenAL32/Include/alAuxEffectSlot.h

@@ -0,0 +1,185 @@
+#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 ALCcontext *context, 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);
+};
+
+/* Small hack to use a pointer-to-array types as a normal argument type.
+ * Shouldn't be used directly.
+ */
+typedef ALfloat ALfloatBUFFERSIZE[BUFFERSIZE];
+
+#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 ALCcontext*, 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 EffectStateFactoryVtable;
+
+typedef struct EffectStateFactory {
+    const struct EffectStateFactoryVtable *vtab;
+} EffectStateFactory;
+
+struct EffectStateFactoryVtable {
+    ALeffectState *(*const create)(EffectStateFactory *factory);
+};
+#define EffectStateFactory_create(x) ((x)->vtab->create((x)))
+
+#define DEFINE_EFFECTSTATEFACTORY_VTABLE(T)                                   \
+DECLARE_THUNK(T, EffectStateFactory, ALeffectState*, create)                  \
+                                                                              \
+static const struct EffectStateFactoryVtable T##_EffectStateFactory_vtable = { \
+    T##_EffectStateFactory_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;
+
+    struct {
+        ALfloat   Gain;
+        ALboolean AuxSendAuto;
+
+        ALenum EffectType;
+        ALeffectProps EffectProps;
+        ALeffectState *EffectState;
+
+        ALfloat RoomRolloff; /* Added to the source's room rolloff, not multiplied. */
+        ALfloat DecayTime;
+        ALfloat DecayLFRatio;
+        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 * sqrt(3)
+     * * Channel 2 is OpenAL Y * sqrt(3)
+     * * Channel 3 is OpenAL -Z * sqrt(3)
+     * 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 source pan for first-order device
+     * output (FOAOut).
+     */
+    alignas(16) ALfloat WetBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE];
+} ALeffectslot;
+
+ALenum InitEffectSlot(ALeffectslot *slot);
+void DeinitEffectSlot(ALeffectslot *slot);
+void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context);
+void UpdateAllEffectSlotProps(ALCcontext *context);
+ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context);
+
+
+EffectStateFactory *NullStateFactory_getFactory(void);
+EffectStateFactory *ReverbStateFactory_getFactory(void);
+EffectStateFactory *AutowahStateFactory_getFactory(void);
+EffectStateFactory *ChorusStateFactory_getFactory(void);
+EffectStateFactory *CompressorStateFactory_getFactory(void);
+EffectStateFactory *DistortionStateFactory_getFactory(void);
+EffectStateFactory *EchoStateFactory_getFactory(void);
+EffectStateFactory *EqualizerStateFactory_getFactory(void);
+EffectStateFactory *FlangerStateFactory_getFactory(void);
+EffectStateFactory *FshifterStateFactory_getFactory(void);
+EffectStateFactory *ModulatorStateFactory_getFactory(void);
+EffectStateFactory *PshifterStateFactory_getFactory(void);
+
+EffectStateFactory *DedicatedStateFactory_getFactory(void);
+
+
+ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect);
+
+void ALeffectState_DecRef(ALeffectState *state);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alBuffer.h

@@ -0,0 +1,115 @@
+#ifndef _AL_BUFFER_H_
+#define _AL_BUFFER_H_
+
+#include "AL/alc.h"
+#include "AL/al.h"
+#include "AL/alext.h"
+
+#include "inprogext.h"
+#include "atomic.h"
+#include "rwlock.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* User formats */
+enum UserFmtType {
+    UserFmtUByte,
+    UserFmtShort,
+    UserFmtFloat,
+    UserFmtDouble,
+    UserFmtMulaw,
+    UserFmtAlaw,
+    UserFmtIMA4,
+    UserFmtMSADPCM,
+};
+enum UserFmtChannels {
+    UserFmtMono,
+    UserFmtStereo,
+    UserFmtRear,
+    UserFmtQuad,
+    UserFmtX51, /* (WFX order) */
+    UserFmtX61, /* (WFX order) */
+    UserFmtX71, /* (WFX order) */
+    UserFmtBFormat2D, /* WXY */
+    UserFmtBFormat3D, /* 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 {
+    FmtUByte  = UserFmtUByte,
+    FmtShort  = UserFmtShort,
+    FmtFloat  = UserFmtFloat,
+    FmtDouble = UserFmtDouble,
+    FmtMulaw  = UserFmtMulaw,
+    FmtAlaw   = UserFmtAlaw,
+};
+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;
+    ALbitfieldSOFT Access;
+    ALsizei SampleLen;
+
+    enum FmtChannels FmtChannels;
+    enum FmtType     FmtType;
+    ALsizei BytesAlloc;
+
+    enum UserFmtType OriginalType;
+    ALsizei OriginalSize;
+    ALsizei OriginalAlign;
+
+    ALsizei LoopStart;
+    ALsizei LoopEnd;
+
+    ATOMIC(ALsizei) UnpackAlign;
+    ATOMIC(ALsizei) PackAlign;
+
+    ALbitfieldSOFT MappedAccess;
+    ALsizei MappedOffset;
+    ALsizei MappedSize;
+
+    /* Number of times buffer was attached to a source (deletion can only occur when 0) */
+    RefCount ref;
+
+    /* Self ID */
+    ALuint id;
+} ALbuffer;
+
+ALvoid ReleaseALBuffers(ALCdevice *device);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alEffect.h

@@ -0,0 +1,213 @@
+#ifndef _AL_EFFECT_H_
+#define _AL_EFFECT_H_
+
+#include "alMain.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct ALeffect;
+
+enum {
+    EAXREVERB_EFFECT = 0,
+    REVERB_EFFECT,
+    AUTOWAH_EFFECT,
+    CHORUS_EFFECT,
+    COMPRESSOR_EFFECT,
+    DISTORTION_EFFECT,
+    ECHO_EFFECT,
+    EQUALIZER_EFFECT,
+    FLANGER_EFFECT,
+    FSHIFTER_EFFECT,
+    MODULATOR_EFFECT,
+    PSHIFTER_EFFECT,
+    DEDICATED_EFFECT,
+
+    MAX_EFFECTS
+};
+extern ALboolean DisabledEffects[MAX_EFFECTS];
+
+extern ALfloat ReverbBoost;
+
+struct EffectList {
+    const char name[16];
+    int type;
+    ALenum val;
+};
+#define EFFECTLIST_SIZE 14
+extern const struct EffectList EffectList[EFFECTLIST_SIZE];
+
+
+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 ALautowah_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 ALfshifter_vtable;
+extern const struct ALeffectVtable ALmodulator_vtable;
+extern const struct ALeffectVtable ALnull_vtable;
+extern const struct ALeffectVtable ALpshifter_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 {
+        ALfloat AttackTime;
+        ALfloat ReleaseTime;
+        ALfloat Resonance;
+        ALfloat PeakGain;
+    } Autowah;
+
+    struct {
+        ALint Waveform;
+        ALint Phase;
+        ALfloat Rate;
+        ALfloat Depth;
+        ALfloat Feedback;
+        ALfloat Delay;
+    } Chorus; /* Also Flanger */
+
+    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 {
+        ALfloat Frequency;
+        ALint LeftDirection;
+        ALint RightDirection;
+    } Fshifter;
+
+    struct {
+        ALfloat Frequency;
+        ALfloat HighPassCutoff;
+        ALint Waveform;
+    } Modulator;
+
+    struct {
+        ALint CoarseTune;
+        ALint FineTune;
+    } Pshifter;
+
+    struct {
+        ALfloat Gain;
+    } Dedicated;
+} ALeffectProps;
+
+typedef struct ALeffect {
+    // Effect type (AL_EFFECT_NULL, ...)
+    ALenum type;
+
+    ALeffectProps Props;
+
+    const struct ALeffectVtable *vtab;
+
+    /* Self ID */
+    ALuint id;
+} ALeffect;
+#define ALeffect_setParami(o, c, p, v)   ((o)->vtab->setParami(o, c, p, v))
+#define ALeffect_setParamf(o, c, p, v)   ((o)->vtab->setParamf(o, c, p, v))
+#define ALeffect_setParamiv(o, c, p, v)  ((o)->vtab->setParamiv(o, c, p, v))
+#define ALeffect_setParamfv(o, c, p, v)  ((o)->vtab->setParamfv(o, c, p, v))
+#define ALeffect_getParami(o, c, p, v)   ((o)->vtab->getParami(o, c, p, v))
+#define ALeffect_getParamf(o, c, p, v)   ((o)->vtab->getParamf(o, c, p, v))
+#define ALeffect_getParamiv(o, c, p, v)  ((o)->vtab->getParamiv(o, c, p, v))
+#define ALeffect_getParamfv(o, c, p, v)  ((o)->vtab->getParamfv(o, c, p, v))
+
+inline ALboolean IsReverbEffect(ALenum type)
+{ return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; }
+
+void InitEffect(ALeffect *effect);
+void ReleaseALEffects(ALCdevice *device);
+
+void LoadReverbPreset(const char *name, ALeffect *effect);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alError.h

@@ -0,0 +1,29 @@
+#ifndef _AL_ERROR_H_
+#define _AL_ERROR_H_
+
+#include "alMain.h"
+#include "logging.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+extern ALboolean TrapALError;
+
+void alSetError(ALCcontext *context, ALenum errorCode, const char *msg, ...) DECL_FORMAT(printf, 3, 4);
+
+#define SETERR_GOTO(ctx, err, lbl, ...) do {                                   \
+    alSetError((ctx), (err), __VA_ARGS__);                                     \
+    goto lbl;                                                                  \
+} while(0)
+
+#define SETERR_RETURN(ctx, err, retval, ...) do {                              \
+    alSetError((ctx), (err), __VA_ARGS__);                                     \
+    return retval;                                                             \
+} while(0)
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alFilter.h

@@ -0,0 +1,67 @@
+#ifndef _AL_FILTER_H_
+#define _AL_FILTER_H_
+
+#include "AL/alc.h"
+#include "AL/al.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define LOWPASSFREQREF  (5000.0f)
+#define HIGHPASSFREQREF  (250.0f)
+
+
+struct ALfilter;
+
+typedef struct ALfilterVtable {
+    void (*const setParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint val);
+    void (*const setParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals);
+    void (*const setParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val);
+    void (*const setParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals);
+
+    void (*const getParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *val);
+    void (*const getParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals);
+    void (*const getParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val);
+    void (*const getParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals);
+} ALfilterVtable;
+
+#define DEFINE_ALFILTER_VTABLE(T)           \
+const struct ALfilterVtable T##_vtable = {  \
+    T##_setParami, T##_setParamiv,          \
+    T##_setParamf, T##_setParamfv,          \
+    T##_getParami, T##_getParamiv,          \
+    T##_getParamf, T##_getParamfv,          \
+}
+
+typedef struct ALfilter {
+    // Filter type (AL_FILTER_NULL, ...)
+    ALenum type;
+
+    ALfloat Gain;
+    ALfloat GainHF;
+    ALfloat HFReference;
+    ALfloat GainLF;
+    ALfloat LFReference;
+
+    const struct ALfilterVtable *vtab;
+
+    /* Self ID */
+    ALuint id;
+} ALfilter;
+#define ALfilter_setParami(o, c, p, v)   ((o)->vtab->setParami(o, c, p, v))
+#define ALfilter_setParamf(o, c, p, v)   ((o)->vtab->setParamf(o, c, p, v))
+#define ALfilter_setParamiv(o, c, p, v)  ((o)->vtab->setParamiv(o, c, p, v))
+#define ALfilter_setParamfv(o, c, p, v)  ((o)->vtab->setParamfv(o, c, p, v))
+#define ALfilter_getParami(o, c, p, v)   ((o)->vtab->getParami(o, c, p, v))
+#define ALfilter_getParamf(o, c, p, v)   ((o)->vtab->getParamf(o, c, p, v))
+#define ALfilter_getParamiv(o, c, p, v)  ((o)->vtab->getParamiv(o, c, p, v))
+#define ALfilter_getParamfv(o, c, p, v)  ((o)->vtab->getParamfv(o, c, p, v))
+
+void ReleaseALFilters(ALCdevice *device);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alListener.h

@@ -0,0 +1,67 @@
+#ifndef _AL_LISTENER_H_
+#define _AL_LISTENER_H_
+
+#include "alMain.h"
+#include "alu.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct ALcontextProps {
+    ALfloat DopplerFactor;
+    ALfloat DopplerVelocity;
+    ALfloat SpeedOfSound;
+    ALboolean SourceDistanceModel;
+    enum DistanceModel DistanceModel;
+    ALfloat MetersPerUnit;
+
+    ATOMIC(struct ALcontextProps*) next;
+};
+
+struct ALlistenerProps {
+    ALfloat Position[3];
+    ALfloat Velocity[3];
+    ALfloat Forward[3];
+    ALfloat Up[3];
+    ALfloat Gain;
+
+    ATOMIC(struct ALlistenerProps*) next;
+};
+
+typedef struct ALlistener {
+    alignas(16) ALfloat Position[3];
+    ALfloat Velocity[3];
+    ALfloat Forward[3];
+    ALfloat Up[3];
+    ALfloat Gain;
+
+    ATOMIC_FLAG PropsClean;
+
+    /* Pointer to the most recent property values that are awaiting an update.
+     */
+    ATOMIC(struct ALlistenerProps*) Update;
+
+    struct {
+        aluMatrixf Matrix;
+        aluVector  Velocity;
+
+        ALfloat Gain;
+        ALfloat MetersPerUnit;
+
+        ALfloat DopplerFactor;
+        ALfloat SpeedOfSound; /* in units per sec! */
+        ALfloat ReverbSpeedOfSound; /* in meters per sec! */
+
+        ALboolean SourceDistanceModel;
+        enum DistanceModel DistanceModel;
+    } Params;
+} ALlistener;
+
+void UpdateListenerProps(ALCcontext *context);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alMain.h

@@ -0,0 +1,920 @@
+#ifndef AL_MAIN_H
+#define AL_MAIN_H
+
+#include <string.h>
+#include <stdio.h>
+#include <stddef.h>
+#include <stdarg.h>
+#include <assert.h>
+#include <math.h>
+#include <limits.h>
+
+#ifdef HAVE_STRINGS_H
+#include <strings.h>
+#endif
+#ifdef HAVE_INTRIN_H
+#include <intrin.h>
+#endif
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "AL/alext.h"
+
+#include "inprogext.h"
+#include "logging.h"
+#include "polymorphism.h"
+#include "static_assert.h"
+#include "align.h"
+#include "atomic.h"
+#include "vector.h"
+#include "alstring.h"
+#include "almalloc.h"
+#include "threads.h"
+
+
+#if defined(_WIN64)
+#define SZFMT "%I64u"
+#elif defined(_WIN32)
+#define SZFMT "%u"
+#else
+#define SZFMT "%zu"
+#endif
+
+#ifdef __has_builtin
+#define HAS_BUILTIN __has_builtin
+#else
+#define HAS_BUILTIN(x) (0)
+#endif
+
+#ifdef __GNUC__
+/* LIKELY optimizes the case where the condition is true. The condition is not
+ * required to be true, but it can result in more optimal code for the true
+ * path at the expense of a less optimal false path.
+ */
+#define LIKELY(x) __builtin_expect(!!(x), !0)
+/* The opposite of LIKELY, optimizing the case where the condition is false. */
+#define UNLIKELY(x) __builtin_expect(!!(x), 0)
+/* Unlike LIKELY, ASSUME requires the condition to be true or else it invokes
+ * undefined behavior. It's essentially an assert without actually checking the
+ * condition at run-time, allowing for stronger optimizations than LIKELY.
+ */
+#if HAS_BUILTIN(__builtin_assume)
+#define ASSUME __builtin_assume
+#else
+#define ASSUME(x) do { if(!(x)) __builtin_unreachable(); } while(0)
+#endif
+
+#else
+
+#define LIKELY(x) (!!(x))
+#define UNLIKELY(x) (!!(x))
+#ifdef _MSC_VER
+#define ASSUME __assume
+#else
+#define ASSUME(x) ((void)0)
+#endif
+#endif
+
+#ifndef UINT64_MAX
+#define UINT64_MAX U64(18446744073709551615)
+#endif
+
+#ifndef UNUSED
+#if defined(__cplusplus)
+#define UNUSED(x)
+#elif defined(__GNUC__)
+#define UNUSED(x) UNUSED_##x __attribute__((unused))
+#elif defined(__LCLINT__)
+#define UNUSED(x) /*@unused@*/ x
+#else
+#define UNUSED(x) x
+#endif
+#endif
+
+/* Calculates the size of a struct with N elements of a flexible array member.
+ * GCC and Clang allow offsetof(Type, fam[N]) for this, but MSVC seems to have
+ * trouble, so a bit more verbose workaround is needed.
+ */
+#define FAM_SIZE(T, M, N)  (offsetof(T, M) + sizeof(((T*)NULL)->M[0])*(N))
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef ALint64SOFT ALint64;
+typedef ALuint64SOFT ALuint64;
+
+#ifndef U64
+#if defined(_MSC_VER)
+#define U64(x) ((ALuint64)(x##ui64))
+#elif SIZEOF_LONG == 8
+#define U64(x) ((ALuint64)(x##ul))
+#elif SIZEOF_LONG_LONG == 8
+#define U64(x) ((ALuint64)(x##ull))
+#endif
+#endif
+
+#ifndef I64
+#if defined(_MSC_VER)
+#define I64(x) ((ALint64)(x##i64))
+#elif SIZEOF_LONG == 8
+#define I64(x) ((ALint64)(x##l))
+#elif SIZEOF_LONG_LONG == 8
+#define I64(x) ((ALint64)(x##ll))
+#endif
+#endif
+
+/* Define a CTZ64 macro (count trailing zeros, for 64-bit integers). The result
+ * is *UNDEFINED* if the value is 0.
+ */
+#ifdef __GNUC__
+
+#if SIZEOF_LONG == 8
+#define CTZ64 __builtin_ctzl
+#else
+#define CTZ64 __builtin_ctzll
+#endif
+
+#elif defined(HAVE_BITSCANFORWARD64_INTRINSIC)
+
+inline int msvc64_ctz64(ALuint64 v)
+{
+    unsigned long idx = 64;
+    _BitScanForward64(&idx, v);
+    return (int)idx;
+}
+#define CTZ64 msvc64_ctz64
+
+#elif defined(HAVE_BITSCANFORWARD_INTRINSIC)
+
+inline int msvc_ctz64(ALuint64 v)
+{
+    unsigned long idx = 64;
+    if(!_BitScanForward(&idx, v&0xffffffff))
+    {
+        if(_BitScanForward(&idx, v>>32))
+            idx += 32;
+    }
+    return (int)idx;
+}
+#define CTZ64 msvc_ctz64
+
+#else
+
+/* There be black magics here. The popcnt64 method is derived from
+ * https://graphics.stanford.edu/~seander/bithacks.html#CountBitsSetParallel
+ * while the ctz-utilizing-popcnt algorithm is shown here
+ * http://www.hackersdelight.org/hdcodetxt/ntz.c.txt
+ * as the ntz2 variant. These likely aren't the most efficient methods, but
+ * they're good enough if the GCC or MSVC intrinsics aren't available.
+ */
+inline int fallback_popcnt64(ALuint64 v)
+{
+    v = v - ((v >> 1) & U64(0x5555555555555555));
+    v = (v & U64(0x3333333333333333)) + ((v >> 2) & U64(0x3333333333333333));
+    v = (v + (v >> 4)) & U64(0x0f0f0f0f0f0f0f0f);
+    return (int)((v * U64(0x0101010101010101)) >> 56);
+}
+
+inline int fallback_ctz64(ALuint64 value)
+{
+    return fallback_popcnt64(~value & (value - 1));
+}
+#define CTZ64 fallback_ctz64
+#endif
+
+#if defined(__BYTE_ORDER__) && defined(__ORDER_LITTLE_ENDIAN__)
+#define IS_LITTLE_ENDIAN (__BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
+#else
+static const union {
+    ALuint u;
+    ALubyte b[sizeof(ALuint)];
+} EndianTest = { 1 };
+#define IS_LITTLE_ENDIAN (EndianTest.b[0] == 1)
+#endif
+
+#define COUNTOF(x) (sizeof(x) / sizeof(0[x]))
+
+
+struct ll_ringbuffer;
+struct Hrtf;
+struct HrtfEntry;
+struct DirectHrtfState;
+struct FrontStablizer;
+struct Compressor;
+struct ALCbackend;
+struct ALbuffer;
+struct ALeffect;
+struct ALfilter;
+struct ALsource;
+struct ALcontextProps;
+struct ALlistenerProps;
+struct ALvoiceProps;
+struct ALeffectslotProps;
+
+
+#define DEFAULT_OUTPUT_RATE  (44100)
+#define MIN_OUTPUT_RATE      (8000)
+
+
+/* Find the next power-of-2 for non-power-of-2 numbers. */
+inline ALuint NextPowerOf2(ALuint value)
+{
+    if(value > 0)
+    {
+        value--;
+        value |= value>>1;
+        value |= value>>2;
+        value |= value>>4;
+        value |= value>>8;
+        value |= value>>16;
+    }
+    return value+1;
+}
+
+/** Round up a value to the next multiple. */
+inline size_t RoundUp(size_t value, size_t r)
+{
+    value += r-1;
+    return value - (value%r);
+}
+
+/* Fast float-to-int conversion. No particular rounding mode is assumed; the
+ * IEEE-754 default is round-to-nearest with ties-to-even, though an app could
+ * change it on its own threads. On some systems, a truncating conversion may
+ * always be the fastest method.
+ */
+inline ALint fastf2i(ALfloat f)
+{
+#if defined(HAVE_INTRIN_H) && ((defined(_M_IX86_FP) && (_M_IX86_FP > 0)) || defined(_M_X64))
+    return _mm_cvt_ss2si(_mm_set1_ps(f));
+
+#elif defined(_MSC_VER) && defined(_M_IX86_FP)
+
+    ALint i;
+    __asm fld f
+    __asm fistp i
+    return i;
+
+#elif (defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__))
+
+    ALint i;
+#ifdef __SSE_MATH__
+    __asm__("cvtss2si %1, %0" : "=r"(i) : "x"(f));
+#else
+    __asm__ __volatile__("fistpl %0" : "=m"(i) : "t"(f) : "st");
+#endif
+    return i;
+
+    /* On GCC when compiling with -fno-math-errno, lrintf can be inlined to
+     * some simple instructions. Clang does not inline it, always generating a
+     * libc call, while MSVC's implementation is horribly slow, so always fall
+     * back to a normal integer conversion for them.
+     */
+#elif defined(HAVE_LRINTF) && !defined(_MSC_VER) && !defined(__clang__)
+
+    return lrintf(f);
+
+#else
+
+    return (ALint)f;
+#endif
+}
+
+/* Converts float-to-int using standard behavior (truncation). */
+inline int float2int(float f)
+{
+#if ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
+     !defined(__SSE_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP == 0)
+    ALint sign, shift, mant;
+    union {
+        ALfloat f;
+        ALint i;
+    } conv;
+
+    conv.f = f;
+    sign = (conv.i>>31) | 1;
+    shift = ((conv.i>>23)&0xff) - (127+23);
+
+    /* Over/underflow */
+    if(UNLIKELY(shift >= 31 || shift < -23))
+        return 0;
+
+    mant = (conv.i&0x7fffff) | 0x800000;
+    if(LIKELY(shift < 0))
+        return (mant >> -shift) * sign;
+    return (mant << shift) * sign;
+
+#else
+
+    return (ALint)f;
+#endif
+}
+
+/* Rounds a float to the nearest integral value, according to the current
+ * rounding mode. This is essentially an inlined version of rintf, although
+ * makes fewer promises (e.g. -0 or -0.25 rounded to 0 may result in +0).
+ */
+inline float fast_roundf(float f)
+{
+#if (defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
+    !defined(__SSE_MATH__)
+
+    float out;
+    __asm__ __volatile__("frndint" : "=t"(out) : "0"(f));
+    return out;
+
+#else
+
+    /* Integral limit, where sub-integral precision is not available for
+     * floats.
+     */
+    static const float ilim[2] = {
+         8388608.0f /*  0x1.0p+23 */,
+        -8388608.0f /* -0x1.0p+23 */
+    };
+    ALuint sign, expo;
+    union {
+        ALfloat f;
+        ALuint i;
+    } conv;
+
+    conv.f = f;
+    sign = (conv.i>>31)&0x01;
+    expo = (conv.i>>23)&0xff;
+
+    if(UNLIKELY(expo >= 150/*+23*/))
+    {
+        /* An exponent (base-2) of 23 or higher is incapable of sub-integral
+         * precision, so it's already an integral value. We don't need to worry
+         * about infinity or NaN here.
+         */
+        return f;
+    }
+    /* Adding the integral limit to the value (with a matching sign) forces a
+     * result that has no sub-integral precision, and is consequently forced to
+     * round to an integral value. Removing the integral limit then restores
+     * the initial value rounded to the integral. The compiler should not
+     * optimize this out because of non-associative rules on floating-point
+     * math (as long as you don't use -fassociative-math,
+     * -funsafe-math-optimizations, -ffast-math, or -Ofast, in which case this
+     * may break).
+     */
+    f += ilim[sign];
+    return f - ilim[sign];
+#endif
+}
+
+
+enum DevProbe {
+    ALL_DEVICE_PROBE,
+    CAPTURE_DEVICE_PROBE
+};
+
+
+enum DistanceModel {
+    InverseDistanceClamped  = AL_INVERSE_DISTANCE_CLAMPED,
+    LinearDistanceClamped   = AL_LINEAR_DISTANCE_CLAMPED,
+    ExponentDistanceClamped = AL_EXPONENT_DISTANCE_CLAMPED,
+    InverseDistance  = AL_INVERSE_DISTANCE,
+    LinearDistance   = AL_LINEAR_DISTANCE,
+    ExponentDistance = AL_EXPONENT_DISTANCE,
+    DisableDistance  = AL_NONE,
+
+    DefaultDistanceModel = InverseDistanceClamped
+};
+
+enum Channel {
+    FrontLeft = 0,
+    FrontRight,
+    FrontCenter,
+    LFE,
+    BackLeft,
+    BackRight,
+    BackCenter,
+    SideLeft,
+    SideRight,
+
+    UpperFrontLeft,
+    UpperFrontRight,
+    UpperBackLeft,
+    UpperBackRight,
+    LowerFrontLeft,
+    LowerFrontRight,
+    LowerBackLeft,
+    LowerBackRight,
+
+    Aux0,
+    Aux1,
+    Aux2,
+    Aux3,
+    Aux4,
+    Aux5,
+    Aux6,
+    Aux7,
+    Aux8,
+    Aux9,
+    Aux10,
+    Aux11,
+    Aux12,
+    Aux13,
+    Aux14,
+    Aux15,
+
+    InvalidChannel
+};
+
+
+/* Device formats */
+enum DevFmtType {
+    DevFmtByte   = ALC_BYTE_SOFT,
+    DevFmtUByte  = ALC_UNSIGNED_BYTE_SOFT,
+    DevFmtShort  = ALC_SHORT_SOFT,
+    DevFmtUShort = ALC_UNSIGNED_SHORT_SOFT,
+    DevFmtInt    = ALC_INT_SOFT,
+    DevFmtUInt   = ALC_UNSIGNED_INT_SOFT,
+    DevFmtFloat  = ALC_FLOAT_SOFT,
+
+    DevFmtTypeDefault = DevFmtFloat
+};
+enum DevFmtChannels {
+    DevFmtMono   = ALC_MONO_SOFT,
+    DevFmtStereo = ALC_STEREO_SOFT,
+    DevFmtQuad   = ALC_QUAD_SOFT,
+    DevFmtX51    = ALC_5POINT1_SOFT,
+    DevFmtX61    = ALC_6POINT1_SOFT,
+    DevFmtX71    = ALC_7POINT1_SOFT,
+    DevFmtAmbi3D = ALC_BFORMAT3D_SOFT,
+
+    /* Similar to 5.1, except using rear channels instead of sides */
+    DevFmtX51Rear = 0x80000000,
+
+    DevFmtChannelsDefault = DevFmtStereo
+};
+#define MAX_OUTPUT_CHANNELS  (16)
+
+ALsizei BytesFromDevFmt(enum DevFmtType type);
+ALsizei ChannelsFromDevFmt(enum DevFmtChannels chans, ALsizei ambiorder);
+inline ALsizei FrameSizeFromDevFmt(enum DevFmtChannels chans, enum DevFmtType type, ALsizei ambiorder)
+{
+    return ChannelsFromDevFmt(chans, ambiorder) * BytesFromDevFmt(type);
+}
+
+enum AmbiLayout {
+    AmbiLayout_FuMa = ALC_FUMA_SOFT, /* FuMa channel order */
+    AmbiLayout_ACN = ALC_ACN_SOFT,   /* ACN channel order */
+
+    AmbiLayout_Default = AmbiLayout_ACN
+};
+
+enum AmbiNorm {
+    AmbiNorm_FuMa = ALC_FUMA_SOFT, /* FuMa normalization */
+    AmbiNorm_SN3D = ALC_SN3D_SOFT, /* SN3D normalization */
+    AmbiNorm_N3D = ALC_N3D_SOFT,   /* N3D normalization */
+
+    AmbiNorm_Default = AmbiNorm_SN3D
+};
+
+
+enum DeviceType {
+    Playback,
+    Capture,
+    Loopback
+};
+
+
+enum RenderMode {
+    NormalRender,
+    StereoPair,
+    HrtfRender
+};
+
+
+/* The maximum number of Ambisonics coefficients. For a given order (o), the
+ * size needed will be (o+1)**2, thus zero-order has 1, first-order has 4,
+ * second-order has 9, third-order has 16, and fourth-order has 25.
+ */
+#define MAX_AMBI_ORDER  3
+#define MAX_AMBI_COEFFS ((MAX_AMBI_ORDER+1) * (MAX_AMBI_ORDER+1))
+
+/* A bitmask of ambisonic channels with height information. If none of these
+ * channels are used/needed, there's no height (e.g. with most surround sound
+ * speaker setups). This only specifies up to 4th order, which is the highest
+ * order a 32-bit mask value can specify (a 64-bit mask could handle up to 7th
+ * order). This is ACN ordering, with bit 0 being ACN 0, etc.
+ */
+#define AMBI_PERIPHONIC_MASK (0xfe7ce4)
+
+/* The maximum number of Ambisonic coefficients for 2D (non-periphonic)
+ * representation. This is 2 per each order above zero-order, plus 1 for zero-
+ * order. Or simply, o*2 + 1.
+ */
+#define MAX_AMBI2D_COEFFS (MAX_AMBI_ORDER*2 + 1)
+
+
+typedef ALfloat ChannelConfig[MAX_AMBI_COEFFS];
+typedef struct BFChannelConfig {
+    ALfloat Scale;
+    ALsizei Index;
+} BFChannelConfig;
+
+typedef union AmbiConfig {
+    /* Ambisonic coefficients for mixing to the dry buffer. */
+    ChannelConfig Coeffs[MAX_OUTPUT_CHANNELS];
+    /* Coefficient channel mapping for mixing to the dry buffer. */
+    BFChannelConfig Map[MAX_OUTPUT_CHANNELS];
+} AmbiConfig;
+
+
+typedef struct BufferSubList {
+    ALuint64 FreeMask;
+    struct ALbuffer *Buffers; /* 64 */
+} BufferSubList;
+TYPEDEF_VECTOR(BufferSubList, vector_BufferSubList)
+
+typedef struct EffectSubList {
+    ALuint64 FreeMask;
+    struct ALeffect *Effects; /* 64 */
+} EffectSubList;
+TYPEDEF_VECTOR(EffectSubList, vector_EffectSubList)
+
+typedef struct FilterSubList {
+    ALuint64 FreeMask;
+    struct ALfilter *Filters; /* 64 */
+} FilterSubList;
+TYPEDEF_VECTOR(FilterSubList, vector_FilterSubList)
+
+typedef struct SourceSubList {
+    ALuint64 FreeMask;
+    struct ALsource *Sources; /* 64 */
+} SourceSubList;
+TYPEDEF_VECTOR(SourceSubList, vector_SourceSubList)
+
+/* Effect slots are rather large, and apps aren't likely to have more than one
+ * or two (let alone 64), so hold them individually.
+ */
+typedef struct ALeffectslot *ALeffectslotPtr;
+TYPEDEF_VECTOR(ALeffectslotPtr, vector_ALeffectslotPtr)
+
+
+typedef struct EnumeratedHrtf {
+    al_string name;
+
+    struct HrtfEntry *hrtf;
+} EnumeratedHrtf;
+TYPEDEF_VECTOR(EnumeratedHrtf, vector_EnumeratedHrtf)
+
+
+/* Maximum delay in samples for speaker distance compensation. */
+#define MAX_DELAY_LENGTH 1024
+
+typedef struct DistanceComp {
+    ALfloat Gain;
+    ALsizei Length; /* Valid range is [0...MAX_DELAY_LENGTH). */
+    ALfloat *Buffer;
+} DistanceComp;
+
+/* Size for temporary storage of buffer data, in ALfloats. Larger values need
+ * more memory, while smaller values may need more iterations. The value needs
+ * to be a sensible size, however, as it constrains the max stepping value used
+ * for mixing, as well as the maximum number of samples per mixing iteration.
+ */
+#define BUFFERSIZE 2048
+
+typedef struct MixParams {
+    AmbiConfig Ambi;
+    /* Number of coefficients in each Ambi.Coeffs to mix together (4 for first-
+     * order, 9 for second-order, etc). If the count is 0, Ambi.Map is used
+     * instead to map each output to a coefficient index.
+     */
+    ALsizei CoeffCount;
+
+    ALfloat (*Buffer)[BUFFERSIZE];
+    ALsizei NumChannels;
+} MixParams;
+
+typedef struct RealMixParams {
+    enum Channel ChannelName[MAX_OUTPUT_CHANNELS];
+
+    ALfloat (*Buffer)[BUFFERSIZE];
+    ALsizei NumChannels;
+} RealMixParams;
+
+typedef void (*POSTPROCESS)(ALCdevice *device, ALsizei SamplesToDo);
+
+struct ALCdevice_struct {
+    RefCount ref;
+
+    ATOMIC(ALenum) Connected;
+    enum DeviceType Type;
+
+    ALuint Frequency;
+    ALuint UpdateSize;
+    ALuint NumUpdates;
+    enum DevFmtChannels FmtChans;
+    enum DevFmtType     FmtType;
+    ALboolean IsHeadphones;
+    ALsizei AmbiOrder;
+    /* For DevFmtAmbi* output only, specifies the channel order and
+     * normalization.
+     */
+    enum AmbiLayout AmbiLayout;
+    enum AmbiNorm   AmbiScale;
+
+    ALCenum LimiterState;
+
+    al_string DeviceName;
+
+    ATOMIC(ALCenum) LastError;
+
+    // Maximum number of sources that can be created
+    ALuint SourcesMax;
+    // Maximum number of slots that can be created
+    ALuint AuxiliaryEffectSlotMax;
+
+    ALCuint NumMonoSources;
+    ALCuint NumStereoSources;
+    ALsizei NumAuxSends;
+
+    // Map of Buffers for this device
+    vector_BufferSubList BufferList;
+    almtx_t BufferLock;
+
+    // Map of Effects for this device
+    vector_EffectSubList EffectList;
+    almtx_t EffectLock;
+
+    // Map of Filters for this device
+    vector_FilterSubList FilterList;
+    almtx_t FilterLock;
+
+    POSTPROCESS PostProcess;
+
+    /* HRTF state and info */
+    struct DirectHrtfState *Hrtf;
+    al_string HrtfName;
+    struct Hrtf *HrtfHandle;
+    vector_EnumeratedHrtf HrtfList;
+    ALCenum HrtfStatus;
+
+    /* UHJ encoder state */
+    struct Uhj2Encoder *Uhj_Encoder;
+
+    /* High quality Ambisonic decoder */
+    struct BFormatDec *AmbiDecoder;
+
+    /* Stereo-to-binaural filter */
+    struct bs2b *Bs2b;
+
+    /* First-order ambisonic upsampler for higher-order output */
+    struct AmbiUpsampler *AmbiUp;
+
+    /* Rendering mode. */
+    enum RenderMode Render_Mode;
+
+    // Device flags
+    ALuint Flags;
+
+    ALuint64 ClockBase;
+    ALuint SamplesDone;
+    ALuint FixedLatency;
+
+    /* Temp storage used for mixer processing. */
+    alignas(16) ALfloat TempBuffer[4][BUFFERSIZE];
+
+    /* The "dry" path corresponds to the main output. */
+    MixParams Dry;
+    ALsizei NumChannelsPerOrder[MAX_AMBI_ORDER+1];
+
+    /* First-order ambisonics output, to be upsampled to the dry buffer if different. */
+    MixParams FOAOut;
+
+    /* "Real" output, which will be written to the device buffer. May alias the
+     * dry buffer.
+     */
+    RealMixParams RealOut;
+
+    struct FrontStablizer *Stablizer;
+
+    struct Compressor *Limiter;
+
+    /* The average speaker distance as determined by the ambdec configuration
+     * (or alternatively, by the NFC-HOA reference delay). Only used for NFC.
+     */
+    ALfloat AvgSpeakerDist;
+
+    /* Delay buffers used to compensate for speaker distances. */
+    DistanceComp ChannelDelay[MAX_OUTPUT_CHANNELS];
+
+    /* Dithering control. */
+    ALfloat DitherDepth;
+    ALuint DitherSeed;
+
+    /* Running count of the mixer invocations, in 31.1 fixed point. This
+     * actually increments *twice* when mixing, first at the start and then at
+     * the end, so the bottom bit indicates if the device is currently mixing
+     * and the upper bits indicates how many mixes have been done.
+     */
+    RefCount MixCount;
+
+    // Contexts created on this device
+    ATOMIC(ALCcontext*) ContextList;
+
+    almtx_t BackendLock;
+    struct ALCbackend *Backend;
+
+    ATOMIC(ALCdevice*) next;
+};
+
+// Frequency was requested by the app or config file
+#define DEVICE_FREQUENCY_REQUEST                 (1u<<1)
+// Channel configuration was requested by the config file
+#define DEVICE_CHANNELS_REQUEST                  (1u<<2)
+// Sample type was requested by the config file
+#define DEVICE_SAMPLE_TYPE_REQUEST               (1u<<3)
+
+// Specifies if the DSP is paused at user request
+#define DEVICE_PAUSED                            (1u<<30)
+
+// Specifies if the device is currently running
+#define DEVICE_RUNNING                           (1u<<31)
+
+
+/* Nanosecond resolution for the device clock time. */
+#define DEVICE_CLOCK_RES  U64(1000000000)
+
+
+/* Must be less than 15 characters (16 including terminating null) for
+ * compatibility with pthread_setname_np limitations. */
+#define MIXER_THREAD_NAME "alsoft-mixer"
+
+#define RECORD_THREAD_NAME "alsoft-record"
+
+
+enum {
+    /* End event thread processing. */
+    EventType_KillThread = 0,
+
+    /* User event types. */
+    EventType_SourceStateChange = 1<<0,
+    EventType_BufferCompleted   = 1<<1,
+    EventType_Error             = 1<<2,
+    EventType_Performance       = 1<<3,
+    EventType_Deprecated        = 1<<4,
+    EventType_Disconnected      = 1<<5,
+
+    /* Internal events. */
+    EventType_ReleaseEffectState = 65536,
+};
+
+typedef struct AsyncEvent {
+    unsigned int EnumType;
+    union {
+        char dummy;
+        struct {
+            ALenum type;
+            ALuint id;
+            ALuint param;
+            ALchar msg[1008];
+        } user;
+        struct ALeffectState *EffectState;
+    } u;
+} AsyncEvent;
+#define ASYNC_EVENT(t) { t, { 0 } }
+
+struct ALCcontext_struct {
+    RefCount ref;
+
+    struct ALlistener *Listener;
+
+    vector_SourceSubList SourceList;
+    ALuint NumSources;
+    almtx_t SourceLock;
+
+    vector_ALeffectslotPtr EffectSlotList;
+    almtx_t EffectSlotLock;
+
+    ATOMIC(ALenum) LastError;
+
+    enum DistanceModel DistanceModel;
+    ALboolean SourceDistanceModel;
+
+    ALfloat DopplerFactor;
+    ALfloat DopplerVelocity;
+    ALfloat SpeedOfSound;
+    ALfloat MetersPerUnit;
+
+    ATOMIC_FLAG PropsClean;
+    ATOMIC(ALenum) DeferUpdates;
+
+    almtx_t PropLock;
+
+    /* Counter for the pre-mixing updates, in 31.1 fixed point (lowest bit
+     * indicates if updates are currently happening).
+     */
+    RefCount UpdateCount;
+    ATOMIC(ALenum) HoldUpdates;
+
+    ALfloat GainBoost;
+
+    ATOMIC(struct ALcontextProps*) Update;
+
+    /* Linked lists of unused property containers, free to use for future
+     * updates.
+     */
+    ATOMIC(struct ALcontextProps*) FreeContextProps;
+    ATOMIC(struct ALlistenerProps*) FreeListenerProps;
+    ATOMIC(struct ALvoiceProps*) FreeVoiceProps;
+    ATOMIC(struct ALeffectslotProps*) FreeEffectslotProps;
+
+    struct ALvoice **Voices;
+    ALsizei VoiceCount;
+    ALsizei MaxVoices;
+
+    ATOMIC(struct ALeffectslotArray*) ActiveAuxSlots;
+
+    althrd_t EventThread;
+    alsem_t EventSem;
+    struct ll_ringbuffer *AsyncEvents;
+    ATOMIC(ALbitfieldSOFT) EnabledEvts;
+    almtx_t EventCbLock;
+    ALEVENTPROCSOFT EventCb;
+    void *EventParam;
+
+    /* Default effect slot */
+    struct ALeffectslot *DefaultSlot;
+
+    ALCdevice  *Device;
+    const ALCchar *ExtensionList;
+
+    ATOMIC(ALCcontext*) next;
+
+    /* Memory space used by the listener (and possibly default effect slot) */
+    alignas(16) ALCbyte _listener_mem[];
+};
+
+ALCcontext *GetContextRef(void);
+
+void ALCcontext_DecRef(ALCcontext *context);
+
+void ALCcontext_DeferUpdates(ALCcontext *context);
+void ALCcontext_ProcessUpdates(ALCcontext *context);
+
+void AllocateVoices(ALCcontext *context, ALsizei num_voices, ALsizei old_sends);
+
+
+extern ALint RTPrioLevel;
+void SetRTPriority(void);
+
+void SetDefaultChannelOrder(ALCdevice *device);
+void SetDefaultWFXChannelOrder(ALCdevice *device);
+
+const ALCchar *DevFmtTypeString(enum DevFmtType type);
+const ALCchar *DevFmtChannelsString(enum DevFmtChannels chans);
+
+inline ALint GetChannelIndex(const enum Channel names[MAX_OUTPUT_CHANNELS], enum Channel chan)
+{
+    ALint i;
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+    {
+        if(names[i] == chan)
+            return i;
+    }
+    return -1;
+}
+/**
+ * GetChannelIdxByName
+ *
+ * Returns the index for the given channel name (e.g. FrontCenter), or -1 if it
+ * doesn't exist.
+ */
+inline ALint GetChannelIdxByName(const RealMixParams *real, enum Channel chan)
+{ return GetChannelIndex(real->ChannelName, chan); }
+
+
+inline void LockBufferList(ALCdevice *device) { almtx_lock(&device->BufferLock); }
+inline void UnlockBufferList(ALCdevice *device) { almtx_unlock(&device->BufferLock); }
+
+inline void LockEffectList(ALCdevice *device) { almtx_lock(&device->EffectLock); }
+inline void UnlockEffectList(ALCdevice *device) { almtx_unlock(&device->EffectLock); }
+
+inline void LockFilterList(ALCdevice *device) { almtx_lock(&device->FilterLock); }
+inline void UnlockFilterList(ALCdevice *device) { almtx_unlock(&device->FilterLock); }
+
+inline void LockEffectSlotList(ALCcontext *context)
+{ almtx_lock(&context->EffectSlotLock); }
+inline void UnlockEffectSlotList(ALCcontext *context)
+{ almtx_unlock(&context->EffectSlotLock); }
+
+
+int EventThread(void *arg);
+
+
+vector_al_string SearchDataFiles(const char *match, const char *subdir);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alSource.h

@@ -0,0 +1,120 @@
+#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 {
+    ATOMIC(struct ALbufferlistitem*) next;
+    ALsizei max_samples;
+    ALsizei num_buffers;
+    struct ALbuffer *buffers[];
+} 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) */
+    ALenum state;
+
+    /** Source Buffer Queue head. */
+    ALbufferlistitem *queue;
+
+    ATOMIC_FLAG PropsClean;
+
+    /* Index into the context's Voices array. Lazily updated, only checked and
+     * reset when looking up the voice.
+     */
+    ALint VoiceIdx;
+
+    /** Self ID */
+    ALuint id;
+} ALsource;
+
+void UpdateAllSourceProps(ALCcontext *context);
+
+ALvoid ReleaseALSources(ALCcontext *Context);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/alu.h

@@ -0,0 +1,534 @@
+#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 "hrtf.h"
+#include "align.h"
+#include "math_defs.h"
+#include "filters/defs.h"
+#include "filters/nfc.h"
+
+
+#define MAX_PITCH  (255)
+
+/* Maximum number of samples to pad on either end of a buffer for resampling.
+ * Note that both the beginning and end need padding!
+ */
+#define MAX_RESAMPLE_PADDING 24
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct BSincTable;
+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,
+    BSinc12Resampler,
+    BSinc24Resampler,
+
+    ResamplerMax = BSinc24Resampler
+};
+extern enum Resampler ResamplerDefault;
+
+/* The number of distinct scale and phase intervals within the bsinc 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. */
+    ALsizei m;  /* Coefficient count. */
+    ALsizei l;  /* Left coefficient offset. */
+    /* Filter coefficients, followed by the scale, phase, and scale-phase
+     * delta coefficients. Starting at phase index 0, each subsequent phase
+     * index follows contiguously.
+     */
+    const ALfloat *filter;
+} BsincState;
+
+typedef union InterpState {
+    BsincState bsinc;
+} InterpState;
+
+typedef const ALfloat* (*ResamplerFunc)(const InterpState *state,
+    const ALfloat *restrict src, ALsizei frac, ALint increment,
+    ALfloat *restrict dst, ALsizei dstlen
+);
+
+void BsincPrepare(const ALuint increment, BsincState *state, const struct BSincTable *table);
+
+extern const struct BSincTable bsinc12;
+extern const struct BSincTable bsinc24;
+
+
+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 {
+    BiquadFilter LowPass;
+    BiquadFilter HighPass;
+
+    NfcFilter NFCtrlFilter;
+
+    struct {
+        HrtfParams Old;
+        HrtfParams Target;
+        HrtfState State;
+    } Hrtf;
+
+    struct {
+        ALfloat Current[MAX_OUTPUT_CHANNELS];
+        ALfloat Target[MAX_OUTPUT_CHANNELS];
+    } Gains;
+} DirectParams;
+
+typedef struct SendParams {
+    BiquadFilter LowPass;
+    BiquadFilter 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[];
+};
+
+#define VOICE_IS_STATIC (1<<0)
+#define VOICE_IS_FADING (1<<1) /* Fading sources use gain stepping for smooth transitions. */
+#define VOICE_HAS_HRTF  (1<<2)
+#define VOICE_HAS_NFC   (1<<3)
+
+typedef struct ALvoice {
+    struct ALvoiceProps *Props;
+
+    ATOMIC(struct ALvoiceProps*) Update;
+
+    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_RESAMPLE_PADDING];
+
+    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)); }
+
+inline size_t minz(size_t a, size_t b)
+{ return ((a > b) ? b : a); }
+inline size_t maxz(size_t a, size_t b)
+{ return ((a > b) ? a : b); }
+inline size_t clampz(size_t val, size_t min, size_t max)
+{ return minz(max, maxz(min, val)); }
+
+
+inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu)
+{
+    return val1 + (val2-val1)*mu;
+}
+inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu)
+{
+    ALfloat mu2 = mu*mu, mu3 = mu2*mu;
+    ALfloat a0 = -0.5f*mu3 +       mu2 + -0.5f*mu;
+    ALfloat a1 =  1.5f*mu3 + -2.5f*mu2            + 1.0f;
+    ALfloat a2 = -1.5f*mu3 +  2.0f*mu2 +  0.5f*mu;
+    ALfloat a3 =  0.5f*mu3 + -0.5f*mu2;
+    return val1*a0 + val2*a1 + val3*a2 + val4*a3;
+}
+
+
+enum HrtfRequestMode {
+    Hrtf_Default = 0,
+    Hrtf_Enable = 1,
+    Hrtf_Disable = 2,
+};
+
+void aluInit(void);
+
+void aluInitMixer(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);
+
+void aluSelectPostProcess(ALCdevice *device);
+
+/**
+ * Calculates ambisonic encoder coefficients using the X, Y, and Z direction
+ * components, which must represent a normalized (unit length) vector, and the
+ * spread is the angular width of the sound (0...tau).
+ *
+ * NOTE: The components use ambisonic coordinates. As a result:
+ *
+ * Ambisonic Y = OpenAL -X
+ * Ambisonic Z = OpenAL Y
+ * Ambisonic X = OpenAL -Z
+ *
+ * The components are ordered such that OpenAL's X, Y, and Z are the first,
+ * second, and third parameters respectively -- simply negate X and Z.
+ */
+void CalcAmbiCoeffs(const ALfloat y, const ALfloat z, const ALfloat x, const ALfloat spread,
+                    ALfloat coeffs[MAX_AMBI_COEFFS]);
+
+/**
+ * CalcDirectionCoeffs
+ *
+ * Calculates ambisonic coefficients based on an OpenAL direction vector. The
+ * vector must be normalized (unit length), and the spread is the angular width
+ * of the sound (0...tau).
+ */
+inline void CalcDirectionCoeffs(const ALfloat dir[3], ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS])
+{
+    /* Convert from OpenAL coords to Ambisonics. */
+    CalcAmbiCoeffs(-dir[0], dir[1], -dir[2], spread, 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 x = -sinf(azimuth) * cosf(elevation);
+    ALfloat y = sinf(elevation);
+    ALfloat z = cosf(azimuth) * cosf(elevation);
+
+    CalcAmbiCoeffs(x, y, z, spread, coeffs);
+}
+
+/**
+ * ScaleAzimuthFront
+ *
+ * Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
+ * front.
+ */
+inline float ScaleAzimuthFront(float azimuth, float scale)
+{
+    ALfloat sign = copysignf(1.0f, azimuth);
+    if(!(fabsf(azimuth) > F_PI_2))
+        return minf(fabsf(azimuth) * scale, F_PI_2) * sign;
+    return azimuth;
+}
+
+
+void ComputePanningGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALsizei numcoeffs, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+void ComputePanningGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
+
+/**
+ * ComputePanGains
+ *
+ * Computes panning gains using the given channel decoder coefficients and the
+ * pre-calculated direction or angle coefficients. For B-Format sources, the
+ * coeffs are a 'slice' of a transform matrix for the input channel, used to
+ * scale and orient the sound samples.
+ */
+inline void ComputePanGains(const MixParams *dry, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS])
+{
+    if(dry->CoeffCount > 0)
+        ComputePanningGainsMC(dry->Ambi.Coeffs, dry->NumChannels, dry->CoeffCount,
+                              coeffs, ingain, gains);
+    else
+        ComputePanningGainsBF(dry->Ambi.Map, dry->NumChannels, coeffs, ingain, gains);
+}
+
+
+ALboolean MixSource(struct ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo);
+
+void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples);
+/* Caller must lock the device, and the mixer must not be running. */
+void aluHandleDisconnect(ALCdevice *device, const char *msg, ...) DECL_FORMAT(printf, 2, 3);
+
+void UpdateContextProps(ALCcontext *context);
+
+extern MixerFunc MixSamples;
+extern RowMixerFunc MixRowSamples;
+
+extern ALfloat ConeScale;
+extern ALfloat ZScale;
+extern ALboolean OverrideReverbSpeedOfSound;
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

OpenAL32/Include/bs2b.h

@@ -24,8 +24,6 @@
 #ifndef BS2B_H
 #define BS2B_H
 
-#include "almalloc.h"
-
 /* Number of crossfeed levels */
 #define BS2B_CLEVELS           3
 
@@ -44,6 +42,10 @@
 /* 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) */
@@ -57,15 +59,14 @@ struct bs2b {
     float a1_hi;
     float b1_hi;
 
-    /* Buffer of filter history
+    /* Buffer of last filtered sample.
      * [0] - first channel, [1] - second channel
      */
     struct t_last_sample {
+        float asis;
         float lo;
         float hi;
-    } history[2];
-
-    DEF_NEWDEL(bs2b)
+    } last_sample[2];
 };
 
 /* Clear buffers and set new coefficients with new crossfeed level and sample
@@ -73,17 +74,21 @@ struct bs2b {
  * level - crossfeed level of *LEVEL values.
  * srate - sample rate by Hz.
  */
-void bs2b_set_params(bs2b *bs2b, int level, int srate);
+void bs2b_set_params(struct bs2b *bs2b, int level, int srate);
 
 /* Return current crossfeed level value */
-int bs2b_get_level(bs2b *bs2b);
+int bs2b_get_level(struct bs2b *bs2b);
 
 /* Return current sample rate value */
-int bs2b_get_srate(bs2b *bs2b);
+int bs2b_get_srate(struct bs2b *bs2b);
 
 /* Clear buffer */
-void bs2b_clear(bs2b *bs2b);
+void bs2b_clear(struct bs2b *bs2b);
 
-void bs2b_cross_feed(bs2b *bs2b, float *Left, float *Right, size_t SamplesToDo);
+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 +1,15 @@
+#ifndef SAMPLE_CVT_H
+#define SAMPLE_CVT_H
+
+#include "AL/al.h"
+#include "alBuffer.h"
+
+extern const ALshort muLawDecompressionTable[256];
+extern const ALshort aLawDecompressionTable[256];
+
+void Convert_ALshort_ALima4(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len,
+                            ALsizei align);
+void Convert_ALshort_ALmsadpcm(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len,
+                               ALsizei align);
+
+#endif /* SAMPLE_CVT_H */

OpenAL32/alAuxEffectSlot.c

@@ -0,0 +1,802 @@
+/**
+ * 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 "alError.h"
+#include "alListener.h"
+#include "alSource.h"
+
+#include "fpu_modes.h"
+#include "almalloc.h"
+
+
+extern inline void LockEffectSlotList(ALCcontext *context);
+extern inline void UnlockEffectSlotList(ALCcontext *context);
+
+static void AddActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context);
+static void RemoveActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context);
+
+static const struct {
+    ALenum Type;
+    EffectStateFactory* (*GetFactory)(void);
+} FactoryList[] = {
+    { AL_EFFECT_NULL, NullStateFactory_getFactory },
+    { AL_EFFECT_EAXREVERB, ReverbStateFactory_getFactory },
+    { AL_EFFECT_REVERB, ReverbStateFactory_getFactory },
+    { AL_EFFECT_AUTOWAH, AutowahStateFactory_getFactory },
+    { AL_EFFECT_CHORUS, ChorusStateFactory_getFactory },
+    { AL_EFFECT_COMPRESSOR, CompressorStateFactory_getFactory },
+    { AL_EFFECT_DISTORTION, DistortionStateFactory_getFactory },
+    { AL_EFFECT_ECHO, EchoStateFactory_getFactory },
+    { AL_EFFECT_EQUALIZER, EqualizerStateFactory_getFactory },
+    { AL_EFFECT_FLANGER, FlangerStateFactory_getFactory },
+    { AL_EFFECT_FREQUENCY_SHIFTER, FshifterStateFactory_getFactory },
+    { AL_EFFECT_RING_MODULATOR, ModulatorStateFactory_getFactory },
+    { AL_EFFECT_PITCH_SHIFTER, PshifterStateFactory_getFactory},
+    { AL_EFFECT_DEDICATED_DIALOGUE, DedicatedStateFactory_getFactory },
+    { AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT, DedicatedStateFactory_getFactory }
+};
+
+static inline EffectStateFactory *getFactoryByType(ALenum type)
+{
+    size_t i;
+    for(i = 0;i < COUNTOF(FactoryList);i++)
+    {
+        if(FactoryList[i].Type == type)
+            return FactoryList[i].GetFactory();
+    }
+    return NULL;
+}
+
+static void ALeffectState_IncRef(ALeffectState *state);
+
+
+static inline ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id)
+{
+    id--;
+    if(UNLIKELY(id >= VECTOR_SIZE(context->EffectSlotList)))
+        return NULL;
+    return VECTOR_ELEM(context->EffectSlotList, id);
+}
+
+static inline ALeffect *LookupEffect(ALCdevice *device, ALuint id)
+{
+    EffectSubList *sublist;
+    ALuint lidx = (id-1) >> 6;
+    ALsizei slidx = (id-1) & 0x3f;
+
+    if(UNLIKELY(lidx >= VECTOR_SIZE(device->EffectList)))
+        return NULL;
+    sublist = &VECTOR_ELEM(device->EffectList, lidx);
+    if(UNLIKELY(sublist->FreeMask & (U64(1)<<slidx)))
+        return NULL;
+    return sublist->Effects + slidx;
+}
+
+
+#define DO_UPDATEPROPS() do {                                                 \
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))          \
+        UpdateEffectSlotProps(slot, context);                                 \
+    else                                                                      \
+        ATOMIC_FLAG_CLEAR(&slot->PropsClean, almemory_order_release);         \
+} while(0)
+
+
+AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots)
+{
+    ALCdevice *device;
+    ALCcontext *context;
+    ALsizei cur;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(n >= 0))
+        SETERR_GOTO(context, AL_INVALID_VALUE, done, "Generating %d effect slots", n);
+    if(n == 0) goto done;
+
+    LockEffectSlotList(context);
+    device = context->Device;
+    for(cur = 0;cur < n;cur++)
+    {
+        ALeffectslotPtr *iter = VECTOR_BEGIN(context->EffectSlotList);
+        ALeffectslotPtr *end = VECTOR_END(context->EffectSlotList);
+        ALeffectslot *slot = NULL;
+        ALenum err = AL_OUT_OF_MEMORY;
+
+        for(;iter != end;iter++)
+        {
+            if(!*iter)
+                break;
+        }
+        if(iter == end)
+        {
+            if(device->AuxiliaryEffectSlotMax == VECTOR_SIZE(context->EffectSlotList))
+            {
+                UnlockEffectSlotList(context);
+                alDeleteAuxiliaryEffectSlots(cur, effectslots);
+                SETERR_GOTO(context, AL_OUT_OF_MEMORY, done,
+                    "Exceeding %u auxiliary effect slot limit", device->AuxiliaryEffectSlotMax);
+            }
+            VECTOR_PUSH_BACK(context->EffectSlotList, NULL);
+            iter = &VECTOR_BACK(context->EffectSlotList);
+        }
+        slot = al_calloc(16, sizeof(ALeffectslot));
+        if(!slot || (err=InitEffectSlot(slot)) != AL_NO_ERROR)
+        {
+            al_free(slot);
+            UnlockEffectSlotList(context);
+
+            alDeleteAuxiliaryEffectSlots(cur, effectslots);
+            SETERR_GOTO(context, err, done, "Effect slot object allocation failed");
+        }
+        aluInitEffectPanning(slot);
+
+        slot->id = (iter - VECTOR_BEGIN(context->EffectSlotList)) + 1;
+        *iter = slot;
+
+        effectslots[cur] = slot->id;
+    }
+    AddActiveEffectSlots(effectslots, n, context);
+    UnlockEffectSlotList(context);
+
+done:
+    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;
+
+    LockEffectSlotList(context);
+    if(!(n >= 0))
+        SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d effect slots", n);
+    if(n == 0) goto done;
+
+    for(i = 0;i < n;i++)
+    {
+        if((slot=LookupEffectSlot(context, effectslots[i])) == NULL)
+            SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u",
+                        effectslots[i]);
+        if(ReadRef(&slot->ref) != 0)
+            SETERR_GOTO(context, AL_INVALID_NAME, done, "Deleting in-use effect slot %u",
+                        effectslots[i]);
+    }
+
+    // All effectslots are valid
+    RemoveActiveEffectSlots(effectslots, n, context);
+    for(i = 0;i < n;i++)
+    {
+        if((slot=LookupEffectSlot(context, effectslots[i])) == NULL)
+            continue;
+        VECTOR_ELEM(context->EffectSlotList, effectslots[i]-1) = NULL;
+
+        DeinitEffectSlot(slot);
+
+        memset(slot, 0, sizeof(*slot));
+        al_free(slot);
+    }
+
+done:
+    UnlockEffectSlotList(context);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot)
+{
+    ALCcontext *context;
+    ALboolean  ret;
+
+    context = GetContextRef();
+    if(!context) return AL_FALSE;
+
+    LockEffectSlotList(context);
+    ret = (LookupEffectSlot(context, effectslot) ? AL_TRUE : AL_FALSE);
+    UnlockEffectSlotList(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;
+
+    almtx_lock(&context->PropLock);
+    LockEffectSlotList(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_EFFECT:
+        device = context->Device;
+
+        LockEffectList(device);
+        effect = (value ? LookupEffect(device, value) : NULL);
+        if(!(value == 0 || effect != NULL))
+        {
+            UnlockEffectList(device);
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Invalid effect ID %u", value);
+        }
+        err = InitializeEffect(context, slot, effect);
+        UnlockEffectList(device);
+
+        if(err != AL_NO_ERROR)
+            SETERR_GOTO(context, err, done, "Effect initialization failed");
+        break;
+
+    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
+        if(!(value == AL_TRUE || value == AL_FALSE))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done,
+                        "Effect slot auxiliary send auto out of range");
+        slot->AuxSendAuto = value;
+        break;
+
+    default:
+        SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid effect slot integer property 0x%04x",
+                    param);
+    }
+    DO_UPDATEPROPS();
+
+done:
+    UnlockEffectSlotList(context);
+    almtx_unlock(&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;
+
+    LockEffectSlotList(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer-vector property 0x%04x",
+                   param);
+    }
+
+done:
+    UnlockEffectSlotList(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;
+
+    almtx_lock(&context->PropLock);
+    LockEffectSlotList(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_GAIN:
+        if(!(value >= 0.0f && value <= 1.0f))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Effect slot gain out of range");
+        slot->Gain = value;
+        break;
+
+    default:
+        SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid effect slot float property 0x%04x",
+                    param);
+    }
+    DO_UPDATEPROPS();
+
+done:
+    UnlockEffectSlotList(context);
+    almtx_unlock(&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;
+
+    LockEffectSlotList(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float-vector property 0x%04x",
+                   param);
+    }
+
+done:
+    UnlockEffectSlotList(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;
+
+    LockEffectSlotList(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
+        *value = slot->AuxSendAuto;
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer property 0x%04x", param);
+    }
+
+done:
+    UnlockEffectSlotList(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;
+
+    LockEffectSlotList(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer-vector property 0x%04x",
+                   param);
+    }
+
+done:
+    UnlockEffectSlotList(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;
+
+    LockEffectSlotList(context);
+    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    case AL_EFFECTSLOT_GAIN:
+        *value = slot->Gain;
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float property 0x%04x", param);
+    }
+
+done:
+    UnlockEffectSlotList(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;
+
+    LockEffectSlotList(context);
+    if(LookupEffectSlot(context, effectslot) == NULL)
+        SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float-vector property 0x%04x",
+                   param);
+    }
+
+done:
+    UnlockEffectSlotList(context);
+    ALCcontext_DecRef(context);
+}
+
+
+ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect)
+{
+    ALCdevice *Device = Context->Device;
+    ALenum newtype = (effect ? effect->type : AL_EFFECT_NULL);
+    struct ALeffectslotProps *props;
+    ALeffectState *State;
+
+    if(newtype != EffectSlot->Effect.Type)
+    {
+        EffectStateFactory *factory;
+
+        factory = getFactoryByType(newtype);
+        if(!factory)
+        {
+            ERR("Failed to find factory for effect type 0x%04x\n", newtype);
+            return AL_INVALID_ENUM;
+        }
+        State = EffectStateFactory_create(factory);
+        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(&Context->FreeEffectslotProps);
+    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);
+}
+
+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))
+{
+}
+
+
+static void AddActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context)
+{
+    struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots,
+                                                     almemory_order_acquire);
+    struct ALeffectslotArray *newarray = NULL;
+    ALsizei newcount = curarray->count + count;
+    ALCdevice *device = context->Device;
+    ALsizei i, j;
+
+    /* Insert the new effect slots into the head of the array, followed by the
+     * existing ones.
+     */
+    newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
+    newarray->count = newcount;
+    for(i = 0;i < count;i++)
+        newarray->slot[i] = LookupEffectSlot(context, slotids[i]);
+    for(j = 0;i < newcount;)
+        newarray->slot[i++] = curarray->slot[j++];
+    /* Remove any duplicates (first instance of each will be kept). */
+    for(i = 1;i < newcount;i++)
+    {
+        for(j = i;j != 0;)
+        {
+            if(UNLIKELY(newarray->slot[i] == newarray->slot[--j]))
+            {
+                newcount--;
+                for(j = i;j < newcount;j++)
+                    newarray->slot[j] = newarray->slot[j+1];
+                i--;
+                break;
+            }
+        }
+    }
+
+    /* Reallocate newarray if the new size ended up smaller from duplicate
+     * removal.
+     */
+    if(UNLIKELY(newcount < newarray->count))
+    {
+        struct ALeffectslotArray *tmpnewarray = al_calloc(DEF_ALIGN,
+            FAM_SIZE(struct ALeffectslotArray, slot, newcount));
+        memcpy(tmpnewarray, newarray, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
+        al_free(newarray);
+        newarray = tmpnewarray;
+        newarray->count = newcount;
+    }
+
+    curarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray, almemory_order_acq_rel);
+    while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
+        althrd_yield();
+    al_free(curarray);
+}
+
+static void RemoveActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context)
+{
+    struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots,
+                                                     almemory_order_acquire);
+    struct ALeffectslotArray *newarray = NULL;
+    ALCdevice *device = context->Device;
+    ALsizei i, j;
+
+    /* Don't shrink the allocated array size since we don't know how many (if
+     * any) of the effect slots to remove are in the array.
+     */
+    newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, curarray->count));
+    newarray->count = 0;
+    for(i = 0;i < curarray->count;i++)
+    {
+        /* Insert this slot into the new array only if it's not one to remove. */
+        ALeffectslot *slot = curarray->slot[i];
+        for(j = count;j != 0;)
+        {
+            if(slot->id == slotids[--j])
+                goto skip_ins;
+        }
+        newarray->slot[newarray->count++] = slot;
+    skip_ins: ;
+    }
+
+    /* TODO: Could reallocate newarray now that we know it's needed size. */
+
+    curarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray, almemory_order_acq_rel);
+    while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
+        althrd_yield();
+    al_free(curarray);
+}
+
+
+ALenum InitEffectSlot(ALeffectslot *slot)
+{
+    EffectStateFactory *factory;
+
+    slot->Effect.Type = AL_EFFECT_NULL;
+
+    factory = getFactoryByType(AL_EFFECT_NULL);
+    slot->Effect.State = EffectStateFactory_create(factory);
+    if(!slot->Effect.State) 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);
+
+    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.DecayLFRatio = 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;
+
+    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);
+    }
+
+    ALeffectState_DecRef(slot->Effect.State);
+    if(slot->Params.EffectState)
+        ALeffectState_DecRef(slot->Params.EffectState);
+}
+
+void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context)
+{
+    struct ALeffectslotProps *props;
+    ALeffectState *oldstate;
+
+    /* Get an unused property container, or allocate a new one as needed. */
+    props = ATOMIC_LOAD(&context->FreeEffectslotProps, 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(&context->FreeEffectslotProps, &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.
+         */
+        if(props->State)
+            ALeffectState_DecRef(props->State);
+        props->State = NULL;
+        ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &context->FreeEffectslotProps, props);
+    }
+
+    if(oldstate)
+        ALeffectState_DecRef(oldstate);
+}
+
+void UpdateAllEffectSlotProps(ALCcontext *context)
+{
+    struct ALeffectslotArray *auxslots;
+    ALsizei i;
+
+    LockEffectSlotList(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, context);
+    }
+    UnlockEffectSlotList(context);
+}
+
+ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *context)
+{
+    ALeffectslotPtr *iter = VECTOR_BEGIN(context->EffectSlotList);
+    ALeffectslotPtr *end = VECTOR_END(context->EffectSlotList);
+    size_t leftover = 0;
+
+    for(;iter != end;iter++)
+    {
+        ALeffectslot *slot = *iter;
+        if(!slot) continue;
+        *iter = NULL;
+
+        DeinitEffectSlot(slot);
+
+        memset(slot, 0, sizeof(*slot));
+        al_free(slot);
+        ++leftover;
+    }
+    if(leftover > 0)
+        WARN("(%p) Deleted "SZFMT" AuxiliaryEffectSlot%s\n", context, leftover, (leftover==1)?"":"s");
+}

OpenAL32/alEffect.c

@@ -0,0 +1,819 @@
+/**
+ * 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 "alError.h"
+
+
+extern inline void LockEffectList(ALCdevice *device);
+extern inline void UnlockEffectList(ALCdevice *device);
+extern inline ALboolean IsReverbEffect(ALenum type);
+
+const struct EffectList EffectList[EFFECTLIST_SIZE] = {
+    { "eaxreverb",  EAXREVERB_EFFECT,  AL_EFFECT_EAXREVERB },
+    { "reverb",     REVERB_EFFECT,     AL_EFFECT_REVERB },
+    { "autowah",    AUTOWAH_EFFECT,    AL_EFFECT_AUTOWAH },
+    { "chorus",     CHORUS_EFFECT,     AL_EFFECT_CHORUS },
+    { "compressor", COMPRESSOR_EFFECT, AL_EFFECT_COMPRESSOR },
+    { "distortion", DISTORTION_EFFECT, AL_EFFECT_DISTORTION },
+    { "echo",       ECHO_EFFECT,       AL_EFFECT_ECHO },
+    { "equalizer",  EQUALIZER_EFFECT,  AL_EFFECT_EQUALIZER },
+    { "flanger",    FLANGER_EFFECT,    AL_EFFECT_FLANGER },
+    { "fshifter",   FSHIFTER_EFFECT,   AL_EFFECT_FREQUENCY_SHIFTER },
+    { "modulator",  MODULATOR_EFFECT,  AL_EFFECT_RING_MODULATOR },
+    { "pshifter",   PSHIFTER_EFFECT,   AL_EFFECT_PITCH_SHIFTER },
+    { "dedicated",  DEDICATED_EFFECT,  AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
+    { "dedicated",  DEDICATED_EFFECT,  AL_EFFECT_DEDICATED_DIALOGUE },
+};
+
+ALboolean DisabledEffects[MAX_EFFECTS];
+
+static ALeffect *AllocEffect(ALCcontext *context);
+static void FreeEffect(ALCdevice *device, ALeffect *effect);
+static void InitEffectParams(ALeffect *effect, ALenum type);
+
+static inline ALeffect *LookupEffect(ALCdevice *device, ALuint id)
+{
+    EffectSubList *sublist;
+    ALuint lidx = (id-1) >> 6;
+    ALsizei slidx = (id-1) & 0x3f;
+
+    if(UNLIKELY(lidx >= VECTOR_SIZE(device->EffectList)))
+        return NULL;
+    sublist = &VECTOR_ELEM(device->EffectList, lidx);
+    if(UNLIKELY(sublist->FreeMask & (U64(1)<<slidx)))
+        return NULL;
+    return sublist->Effects + slidx;
+}
+
+
+AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects)
+{
+    ALCcontext *context;
+    ALsizei cur;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(n >= 0))
+        alSetError(context, AL_INVALID_VALUE, "Generating %d effects", n);
+    else for(cur = 0;cur < n;cur++)
+    {
+        ALeffect *effect = AllocEffect(context);
+        if(!effect)
+        {
+            alDeleteEffects(cur, effects);
+            break;
+        }
+        effects[cur] = effect->id;
+    }
+
+    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;
+    LockEffectList(device);
+    if(!(n >= 0))
+        SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d effects", n);
+    for(i = 0;i < n;i++)
+    {
+        if(effects[i] && LookupEffect(device, effects[i]) == NULL)
+            SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect ID %u", effects[i]);
+    }
+    for(i = 0;i < n;i++)
+    {
+        if((effect=LookupEffect(device, effects[i])) != NULL)
+            FreeEffect(device, effect);
+    }
+
+done:
+    UnlockEffectList(device);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect)
+{
+    ALCcontext *Context;
+    ALboolean  result;
+
+    Context = GetContextRef();
+    if(!Context) return AL_FALSE;
+
+    LockEffectList(Context->Device);
+    result = ((!effect || LookupEffect(Context->Device, effect)) ?
+              AL_TRUE : AL_FALSE);
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        if(param == AL_EFFECT_TYPE)
+        {
+            ALboolean isOk = (value == AL_EFFECT_NULL);
+            ALint i;
+            for(i = 0;!isOk && i < EFFECTLIST_SIZE;i++)
+            {
+                if(value == EffectList[i].val &&
+                   !DisabledEffects[EffectList[i].type])
+                    isOk = AL_TRUE;
+            }
+
+            if(isOk)
+                InitEffectParams(ALEffect, value);
+            else
+                alSetError(Context, AL_INVALID_VALUE, "Effect type 0x%04x not supported", value);
+        }
+        else
+        {
+            /* Call the appropriate handler */
+            ALeffect_setParami(ALEffect, Context, param, value);
+        }
+    }
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        /* Call the appropriate handler */
+        ALeffect_setParamiv(ALEffect, Context, param, values);
+    }
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        /* Call the appropriate handler */
+        ALeffect_setParamf(ALEffect, Context, param, value);
+    }
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        /* Call the appropriate handler */
+        ALeffect_setParamfv(ALEffect, Context, param, values);
+    }
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        if(param == AL_EFFECT_TYPE)
+            *value = ALEffect->type;
+        else
+        {
+            /* Call the appropriate handler */
+            ALeffect_getParami(ALEffect, Context, param, value);
+        }
+    }
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        /* Call the appropriate handler */
+        ALeffect_getParamiv(ALEffect, Context, param, values);
+    }
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        /* Call the appropriate handler */
+        ALeffect_getParamf(ALEffect, Context, param, value);
+    }
+    UnlockEffectList(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;
+    LockEffectList(Device);
+    if((ALEffect=LookupEffect(Device, effect)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
+    else
+    {
+        /* Call the appropriate handler */
+        ALeffect_getParamfv(ALEffect, Context, param, values);
+    }
+    UnlockEffectList(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+
+void InitEffect(ALeffect *effect)
+{
+    InitEffectParams(effect, AL_EFFECT_NULL);
+}
+
+static ALeffect *AllocEffect(ALCcontext *context)
+{
+    ALCdevice *device = context->Device;
+    EffectSubList *sublist, *subend;
+    ALeffect *effect = NULL;
+    ALsizei lidx = 0;
+    ALsizei slidx;
+
+    almtx_lock(&device->EffectLock);
+    sublist = VECTOR_BEGIN(device->EffectList);
+    subend = VECTOR_END(device->EffectList);
+    for(;sublist != subend;++sublist)
+    {
+        if(sublist->FreeMask)
+        {
+            slidx = CTZ64(sublist->FreeMask);
+            effect = sublist->Effects + slidx;
+            break;
+        }
+        ++lidx;
+    }
+    if(UNLIKELY(!effect))
+    {
+        const EffectSubList empty_sublist = { 0, NULL };
+        /* Don't allocate so many list entries that the 32-bit ID could
+         * overflow...
+         */
+        if(UNLIKELY(VECTOR_SIZE(device->EffectList) >= 1<<25))
+        {
+            almtx_unlock(&device->EffectLock);
+            alSetError(context, AL_OUT_OF_MEMORY, "Too many effects allocated");
+            return NULL;
+        }
+        lidx = (ALsizei)VECTOR_SIZE(device->EffectList);
+        VECTOR_PUSH_BACK(device->EffectList, empty_sublist);
+        sublist = &VECTOR_BACK(device->EffectList);
+        sublist->FreeMask = ~U64(0);
+        sublist->Effects = al_calloc(16, sizeof(ALeffect)*64);
+        if(UNLIKELY(!sublist->Effects))
+        {
+            VECTOR_POP_BACK(device->EffectList);
+            almtx_unlock(&device->EffectLock);
+            alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate effect batch");
+            return NULL;
+        }
+
+        slidx = 0;
+        effect = sublist->Effects + slidx;
+    }
+
+    memset(effect, 0, sizeof(*effect));
+    InitEffectParams(effect, AL_EFFECT_NULL);
+
+    /* Add 1 to avoid effect ID 0. */
+    effect->id = ((lidx<<6) | slidx) + 1;
+
+    sublist->FreeMask &= ~(U64(1)<<slidx);
+    almtx_unlock(&device->EffectLock);
+
+    return effect;
+}
+
+static void FreeEffect(ALCdevice *device, ALeffect *effect)
+{
+    ALuint id = effect->id - 1;
+    ALsizei lidx = id >> 6;
+    ALsizei slidx = id & 0x3f;
+
+    memset(effect, 0, sizeof(*effect));
+
+    VECTOR_ELEM(device->EffectList, lidx).FreeMask |= U64(1) << slidx;
+}
+
+void ReleaseALEffects(ALCdevice *device)
+{
+    EffectSubList *sublist = VECTOR_BEGIN(device->EffectList);
+    EffectSubList *subend = VECTOR_END(device->EffectList);
+    size_t leftover = 0;
+    for(;sublist != subend;++sublist)
+    {
+        ALuint64 usemask = ~sublist->FreeMask;
+        while(usemask)
+        {
+            ALsizei idx = CTZ64(usemask);
+            ALeffect *effect = sublist->Effects + idx;
+
+            memset(effect, 0, sizeof(*effect));
+            ++leftover;
+
+            usemask &= ~(U64(1) << idx);
+        }
+        sublist->FreeMask = ~usemask;
+    }
+    if(leftover > 0)
+        WARN("(%p) Deleted "SZFMT" Effect%s\n", device, leftover, (leftover==1)?"":"s");
+}
+
+
+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;
+        effect->vtab = &ALeaxreverb_vtable;
+        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;
+        effect->vtab = &ALreverb_vtable;
+        break;
+    case AL_EFFECT_AUTOWAH:
+        effect->Props.Autowah.AttackTime = AL_AUTOWAH_DEFAULT_ATTACK_TIME;
+        effect->Props.Autowah.ReleaseTime = AL_AUTOWAH_DEFAULT_RELEASE_TIME;
+        effect->Props.Autowah.Resonance = AL_AUTOWAH_DEFAULT_RESONANCE;
+        effect->Props.Autowah.PeakGain = AL_AUTOWAH_DEFAULT_PEAK_GAIN;
+        effect->vtab = &ALautowah_vtable;
+        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;
+        effect->vtab = &ALchorus_vtable;
+        break;
+    case AL_EFFECT_COMPRESSOR:
+        effect->Props.Compressor.OnOff = AL_COMPRESSOR_DEFAULT_ONOFF;
+        effect->vtab = &ALcompressor_vtable;
+        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;
+        effect->vtab = &ALdistortion_vtable;
+        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;
+        effect->vtab = &ALecho_vtable;
+        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;
+        effect->vtab = &ALequalizer_vtable;
+        break;
+    case AL_EFFECT_FLANGER:
+        effect->Props.Chorus.Waveform = AL_FLANGER_DEFAULT_WAVEFORM;
+        effect->Props.Chorus.Phase = AL_FLANGER_DEFAULT_PHASE;
+        effect->Props.Chorus.Rate = AL_FLANGER_DEFAULT_RATE;
+        effect->Props.Chorus.Depth = AL_FLANGER_DEFAULT_DEPTH;
+        effect->Props.Chorus.Feedback = AL_FLANGER_DEFAULT_FEEDBACK;
+        effect->Props.Chorus.Delay = AL_FLANGER_DEFAULT_DELAY;
+        effect->vtab = &ALflanger_vtable;
+        break;
+        case AL_EFFECT_FREQUENCY_SHIFTER:
+        effect->Props.Fshifter.Frequency      = AL_FREQUENCY_SHIFTER_DEFAULT_FREQUENCY;
+        effect->Props.Fshifter.LeftDirection  = AL_FREQUENCY_SHIFTER_DEFAULT_LEFT_DIRECTION;
+        effect->Props.Fshifter.RightDirection = AL_FREQUENCY_SHIFTER_DEFAULT_RIGHT_DIRECTION;
+        effect->vtab = &ALfshifter_vtable;
+        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;
+        effect->vtab = &ALmodulator_vtable;
+        break;
+    case AL_EFFECT_PITCH_SHIFTER:
+        effect->Props.Pshifter.CoarseTune      = AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE;
+        effect->Props.Pshifter.FineTune        = AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE;
+        effect->vtab = &ALpshifter_vtable;
+        break;
+    case AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT:
+    case AL_EFFECT_DEDICATED_DIALOGUE:
+        effect->Props.Dedicated.Gain = 1.0f;
+        effect->vtab = &ALdedicated_vtable;
+        break;
+    default:
+        effect->vtab = &ALnull_vtable;
+        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
+
+void 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[EAXREVERB_EFFECT])
+        InitEffectParams(effect, AL_EFFECT_EAXREVERB);
+    else if(!DisabledEffects[REVERB_EFFECT])
+        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

@@ -20,52 +20,46 @@
 
 #include "config.h"
 
-#ifdef _WIN32
+#include <signal.h>
+#include <stdarg.h>
+
+#ifdef HAVE_WINDOWS_H
 #define WIN32_LEAN_AND_MEAN
 #include <windows.h>
 #endif
 
-#include <atomic>
-#include <csignal>
-#include <cstdarg>
-#include <cstdio>
-#include <cstring>
-#include <mutex>
-
-#include "AL/al.h"
+#include "alMain.h"
 #include "AL/alc.h"
+#include "alError.h"
 
-#include "alcontext.h"
-#include "alexcpt.h"
-#include "almalloc.h"
-#include "logging.h"
-#include "opthelpers.h"
-#include "vector.h"
+ALboolean TrapALError = AL_FALSE;
 
-
-bool TrapALError{false};
-
-void ALCcontext::setError(ALenum errorCode, const char *msg, ...)
+void alSetError(ALCcontext *context, ALenum errorCode, const char *msg, ...)
 {
-    auto message = al::vector<char>(256);
+    ALenum curerr = AL_NO_ERROR;
+    char message[1024] = { 0 };
+    va_list args;
+    int msglen;
 
-    va_list args, args2;
     va_start(args, msg);
-    va_copy(args2, args);
-    int msglen{std::vsnprintf(message.data(), message.size(), msg, args)};
-    if(msglen >= 0 && static_cast<size_t>(msglen) >= message.size())
-    {
-        message.resize(static_cast<size_t>(msglen) + 1u);
-        msglen = std::vsnprintf(message.data(), message.size(), msg, args2);
-    }
-    va_end(args2);
+    msglen = vsnprintf(message, sizeof(message), msg, args);
     va_end(args);
 
-    if(msglen >= 0) msg = message.data();
-    else msg = "<internal error constructing message>";
+    if(msglen < 0 || (size_t)msglen >= sizeof(message))
+    {
+        message[sizeof(message)-1] = 0;
+        msglen = (int)strlen(message);
+    }
+    if(msglen > 0)
+        msg = message;
+    else
+    {
+        msg = "<internal error constructing message>";
+        msglen = (int)strlen(msg);
+    }
 
     WARN("Error generated on context %p, code 0x%04x, \"%s\"\n",
-        decltype(std::declval<void*>()){this}, errorCode, msg);
+         context, errorCode, message);
     if(TrapALError)
     {
 #ifdef _WIN32
@@ -77,17 +71,28 @@ void ALCcontext::setError(ALenum errorCode, const char *msg, ...)
 #endif
     }
 
-    ALenum curerr{AL_NO_ERROR};
-    mLastError.compare_exchange_strong(curerr, errorCode);
+    ATOMIC_COMPARE_EXCHANGE_STRONG_SEQ(&context->LastError, &curerr, errorCode);
+    if((ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed)&EventType_Error))
+    {
+        ALbitfieldSOFT enabledevts;
+        almtx_lock(&context->EventCbLock);
+        enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed);
+        if((enabledevts&EventType_Error) && context->EventCb)
+            (*context->EventCb)(AL_EVENT_TYPE_ERROR_SOFT, 0, errorCode, msglen, msg,
+                                context->EventParam);
+        almtx_unlock(&context->EventCbLock);
+    }
 }
 
 AL_API ALenum AL_APIENTRY alGetError(void)
-START_API_FUNC
 {
-    ContextRef context{GetContextRef()};
-    if UNLIKELY(!context)
+    ALCcontext *context;
+    ALenum errorCode;
+
+    context = GetContextRef();
+    if(!context)
     {
-        constexpr ALenum deferror{AL_INVALID_OPERATION};
+        const ALenum deferror = AL_INVALID_OPERATION;
         WARN("Querying error state on null context (implicitly 0x%04x)\n", deferror);
         if(TrapALError)
         {
@@ -101,6 +106,8 @@ START_API_FUNC
         return deferror;
     }
 
-    return context->mLastError.exchange(AL_NO_ERROR);
+    errorCode = ATOMIC_EXCHANGE_SEQ(&context->LastError, AL_NO_ERROR);
+
+    ALCcontext_DecRef(context);
+    return errorCode;
 }
-END_API_FUNC

OpenAL32/alExtension.c

@@ -20,36 +20,45 @@
 
 #include "config.h"
 
-#include <cctype>
-#include <cstdlib>
-#include <cstring>
+#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"
 
-#include "alcontext.h"
-#include "alexcpt.h"
-#include "alstring.h"
-#include "opthelpers.h"
-
 
 AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName)
-START_API_FUNC
 {
-    ContextRef context{GetContextRef()};
-    if UNLIKELY(!context) return AL_FALSE;
+    ALboolean ret = AL_FALSE;
+    ALCcontext *context;
+    const char *ptr;
+    size_t len;
+
+    context = GetContextRef();
+    if(!context) return AL_FALSE;
 
     if(!extName)
-        SETERR_RETURN(context, AL_INVALID_VALUE, AL_FALSE, "NULL pointer");
+        SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer");
 
-    size_t len{strlen(extName)};
-    const char *ptr{context->mExtensionList};
+    len = strlen(extName);
+    ptr = context->ExtensionList;
     while(ptr && *ptr)
     {
-        if(al::strncasecmp(ptr, extName, len) == 0 && (ptr[len] == '\0' || isspace(ptr[len])))
-            return AL_TRUE;
-
-        if((ptr=strchr(ptr, ' ')) != nullptr)
+        if(strncasecmp(ptr, extName, len) == 0 &&
+           (ptr[len] == '\0' || isspace(ptr[len])))
+        {
+            ret = AL_TRUE;
+            break;
+        }
+        if((ptr=strchr(ptr, ' ')) != NULL)
         {
             do {
                 ++ptr;
@@ -57,23 +66,22 @@ START_API_FUNC
         }
     }
 
-    return AL_FALSE;
+done:
+    ALCcontext_DecRef(context);
+    return ret;
 }
-END_API_FUNC
 
 
 AL_API ALvoid* AL_APIENTRY alGetProcAddress(const ALchar *funcName)
-START_API_FUNC
 {
-    if(!funcName) return nullptr;
-    return alcGetProcAddress(nullptr, funcName);
+    if(!funcName)
+        return NULL;
+    return alcGetProcAddress(NULL, funcName);
 }
-END_API_FUNC
 
 AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *enumName)
-START_API_FUNC
 {
-    if(!enumName) return static_cast<ALenum>(0);
-    return alcGetEnumValue(nullptr, enumName);
+    if(!enumName)
+        return (ALenum)0;
+    return alcGetEnumValue(NULL, enumName);
 }
-END_API_FUNC

OpenAL32/alFilter.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 <stdlib.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "alFilter.h"
+#include "alError.h"
+
+
+#define FILTER_MIN_GAIN 0.0f
+#define FILTER_MAX_GAIN 4.0f /* +12dB */
+
+extern inline void LockFilterList(ALCdevice *device);
+extern inline void UnlockFilterList(ALCdevice *device);
+
+static ALfilter *AllocFilter(ALCcontext *context);
+static void FreeFilter(ALCdevice *device, ALfilter *filter);
+static void InitFilterParams(ALfilter *filter, ALenum type);
+
+static inline ALfilter *LookupFilter(ALCdevice *device, ALuint id)
+{
+    FilterSubList *sublist;
+    ALuint lidx = (id-1) >> 6;
+    ALsizei slidx = (id-1) & 0x3f;
+
+    if(UNLIKELY(lidx >= VECTOR_SIZE(device->FilterList)))
+        return NULL;
+    sublist = &VECTOR_ELEM(device->FilterList, lidx);
+    if(UNLIKELY(sublist->FreeMask & (U64(1)<<slidx)))
+        return NULL;
+    return sublist->Filters + slidx;
+}
+
+
+AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
+{
+    ALCcontext *context;
+    ALsizei cur = 0;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(n >= 0))
+        alSetError(context, AL_INVALID_VALUE, "Generating %d filters", n);
+    else for(cur = 0;cur < n;cur++)
+    {
+        ALfilter *filter = AllocFilter(context);
+        if(!filter)
+        {
+            alDeleteFilters(cur, filters);
+            break;
+        }
+
+        filters[cur] = filter->id;
+    }
+
+    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;
+    LockFilterList(device);
+    if(!(n >= 0))
+        SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d filters", n);
+    for(i = 0;i < n;i++)
+    {
+        if(filters[i] && LookupFilter(device, filters[i]) == NULL)
+            SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid filter ID %u", filters[i]);
+    }
+    for(i = 0;i < n;i++)
+    {
+        if((filter=LookupFilter(device, filters[i])) != NULL)
+            FreeFilter(device, filter);
+    }
+
+done:
+    UnlockFilterList(device);
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter)
+{
+    ALCcontext *Context;
+    ALboolean  result;
+
+    Context = GetContextRef();
+    if(!Context) return AL_FALSE;
+
+    LockFilterList(Context->Device);
+    result = ((!filter || LookupFilter(Context->Device, filter)) ?
+              AL_TRUE : AL_FALSE);
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    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, "Invalid filter type 0x%04x", value);
+        }
+        else
+        {
+            /* Call the appropriate handler */
+            ALfilter_setParami(ALFilter, Context, param, value);
+        }
+    }
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_setParamiv(ALFilter, Context, param, values);
+    }
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_setParamf(ALFilter, Context, param, value);
+    }
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_setParamfv(ALFilter, Context, param, values);
+    }
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    else
+    {
+        if(param == AL_FILTER_TYPE)
+            *value = ALFilter->type;
+        else
+        {
+            /* Call the appropriate handler */
+            ALfilter_getParami(ALFilter, Context, param, value);
+        }
+    }
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_getParamiv(ALFilter, Context, param, values);
+    }
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_getParamf(ALFilter, Context, param, value);
+    }
+    UnlockFilterList(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;
+    LockFilterList(Device);
+    if((ALFilter=LookupFilter(Device, filter)) == NULL)
+        alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
+    else
+    {
+        /* Call the appropriate handler */
+        ALfilter_getParamfv(ALFilter, Context, param, values);
+    }
+    UnlockFilterList(Device);
+
+    ALCcontext_DecRef(Context);
+}
+
+
+static void ALlowpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); }
+static void ALlowpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); }
+static void ALlowpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
+{
+    switch(param)
+    {
+        case AL_LOWPASS_GAIN:
+            if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gain %f out of range", val);
+            filter->Gain = val;
+            break;
+
+        case AL_LOWPASS_GAINHF:
+            if(!(val >= AL_LOWPASS_MIN_GAINHF && val <= AL_LOWPASS_MAX_GAINHF))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gainhf %f out of range", val);
+            filter->GainHF = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid low-pass float property 0x%04x", param);
+    }
+}
+static void ALlowpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
+{ ALlowpass_setParamf(filter, context, param, vals[0]); }
+
+static void ALlowpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); }
+static void ALlowpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); }
+static void ALlowpass_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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid low-pass float property 0x%04x", param);
+    }
+}
+static void ALlowpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
+{ ALlowpass_getParamf(filter, context, param, vals); }
+
+DEFINE_ALFILTER_VTABLE(ALlowpass);
+
+
+static void ALhighpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); }
+static void ALhighpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); }
+static void ALhighpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
+{
+    switch(param)
+    {
+        case AL_HIGHPASS_GAIN:
+            if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gain out of range");
+            filter->Gain = val;
+            break;
+
+        case AL_HIGHPASS_GAINLF:
+            if(!(val >= AL_HIGHPASS_MIN_GAINLF && val <= AL_HIGHPASS_MAX_GAINLF))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gainlf out of range");
+            filter->GainLF = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param);
+    }
+}
+static void ALhighpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
+{ ALhighpass_setParamf(filter, context, param, vals[0]); }
+
+static void ALhighpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); }
+static void ALhighpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); }
+static void ALhighpass_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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param);
+    }
+}
+static void ALhighpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
+{ ALhighpass_getParamf(filter, context, param, vals); }
+
+DEFINE_ALFILTER_VTABLE(ALhighpass);
+
+
+static void ALbandpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); }
+static void ALbandpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); }
+static void ALbandpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
+{
+    switch(param)
+    {
+        case AL_BANDPASS_GAIN:
+            if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gain out of range");
+            filter->Gain = val;
+            break;
+
+        case AL_BANDPASS_GAINHF:
+            if(!(val >= AL_BANDPASS_MIN_GAINHF && val <= AL_BANDPASS_MAX_GAINHF))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainhf out of range");
+            filter->GainHF = val;
+            break;
+
+        case AL_BANDPASS_GAINLF:
+            if(!(val >= AL_BANDPASS_MIN_GAINLF && val <= AL_BANDPASS_MAX_GAINLF))
+                SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainlf out of range");
+            filter->GainLF = val;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param);
+    }
+}
+static void ALbandpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
+{ ALbandpass_setParamf(filter, context, param, vals[0]); }
+
+static void ALbandpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); }
+static void ALbandpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); }
+static void ALbandpass_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:
+            alSetError(context, AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param);
+    }
+}
+static void ALbandpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
+{ ALbandpass_getParamf(filter, context, param, vals); }
+
+DEFINE_ALFILTER_VTABLE(ALbandpass);
+
+
+static void ALnullfilter_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+static void ALnullfilter_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+static void ALnullfilter_setParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+static void ALnullfilter_setParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+
+static void ALnullfilter_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+static void ALnullfilter_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+static void ALnullfilter_getParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat *UNUSED(val))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+static void ALnullfilter_getParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals))
+{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
+
+DEFINE_ALFILTER_VTABLE(ALnullfilter);
+
+
+static ALfilter *AllocFilter(ALCcontext *context)
+{
+    ALCdevice *device = context->Device;
+    FilterSubList *sublist, *subend;
+    ALfilter *filter = NULL;
+    ALsizei lidx = 0;
+    ALsizei slidx;
+
+    almtx_lock(&device->FilterLock);
+    sublist = VECTOR_BEGIN(device->FilterList);
+    subend = VECTOR_END(device->FilterList);
+    for(;sublist != subend;++sublist)
+    {
+        if(sublist->FreeMask)
+        {
+            slidx = CTZ64(sublist->FreeMask);
+            filter = sublist->Filters + slidx;
+            break;
+        }
+        ++lidx;
+    }
+    if(UNLIKELY(!filter))
+    {
+        const FilterSubList empty_sublist = { 0, NULL };
+        /* Don't allocate so many list entries that the 32-bit ID could
+         * overflow...
+         */
+        if(UNLIKELY(VECTOR_SIZE(device->FilterList) >= 1<<25))
+        {
+            almtx_unlock(&device->FilterLock);
+            alSetError(context, AL_OUT_OF_MEMORY, "Too many filters allocated");
+            return NULL;
+        }
+        lidx = (ALsizei)VECTOR_SIZE(device->FilterList);
+        VECTOR_PUSH_BACK(device->FilterList, empty_sublist);
+        sublist = &VECTOR_BACK(device->FilterList);
+        sublist->FreeMask = ~U64(0);
+        sublist->Filters = al_calloc(16, sizeof(ALfilter)*64);
+        if(UNLIKELY(!sublist->Filters))
+        {
+            VECTOR_POP_BACK(device->FilterList);
+            almtx_unlock(&device->FilterLock);
+            alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate filter batch");
+            return NULL;
+        }
+
+        slidx = 0;
+        filter = sublist->Filters + slidx;
+    }
+
+    memset(filter, 0, sizeof(*filter));
+    InitFilterParams(filter, AL_FILTER_NULL);
+
+    /* Add 1 to avoid filter ID 0. */
+    filter->id = ((lidx<<6) | slidx) + 1;
+
+    sublist->FreeMask &= ~(U64(1)<<slidx);
+    almtx_unlock(&device->FilterLock);
+
+    return filter;
+}
+
+static void FreeFilter(ALCdevice *device, ALfilter *filter)
+{
+    ALuint id = filter->id - 1;
+    ALsizei lidx = id >> 6;
+    ALsizei slidx = id & 0x3f;
+
+    memset(filter, 0, sizeof(*filter));
+
+    VECTOR_ELEM(device->FilterList, lidx).FreeMask |= U64(1) << slidx;
+}
+
+void ReleaseALFilters(ALCdevice *device)
+{
+    FilterSubList *sublist = VECTOR_BEGIN(device->FilterList);
+    FilterSubList *subend = VECTOR_END(device->FilterList);
+    size_t leftover = 0;
+    for(;sublist != subend;++sublist)
+    {
+        ALuint64 usemask = ~sublist->FreeMask;
+        while(usemask)
+        {
+            ALsizei idx = CTZ64(usemask);
+            ALfilter *filter = sublist->Filters + idx;
+
+            memset(filter, 0, sizeof(*filter));
+            ++leftover;
+
+            usemask &= ~(U64(1) << idx);
+        }
+        sublist->FreeMask = ~usemask;
+    }
+    if(leftover > 0)
+        WARN("(%p) Deleted "SZFMT" Filter%s\n", device, leftover, (leftover==1)?"":"s");
+}
+
+
+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->vtab = &ALlowpass_vtable;
+    }
+    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->vtab = &ALhighpass_vtable;
+    }
+    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->vtab = &ALbandpass_vtable;
+    }
+    else
+    {
+        filter->Gain = 1.0f;
+        filter->GainHF = 1.0f;
+        filter->HFReference = LOWPASSFREQREF;
+        filter->GainLF = 1.0f;
+        filter->LFReference = HIGHPASSFREQREF;
+        filter->vtab = &ALnullfilter_vtable;
+    }
+    filter->type = type;
+}

OpenAL32/alListener.c

@@ -0,0 +1,502 @@
+/**
+ * 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 "alu.h"
+#include "alError.h"
+#include "alListener.h"
+#include "alSource.h"
+
+#define DO_UPDATEPROPS() do {                                                 \
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))          \
+        UpdateListenerProps(context);                                         \
+    else                                                                      \
+        ATOMIC_FLAG_CLEAR(&listener->PropsClean, almemory_order_release);     \
+} while(0)
+
+
+AL_API ALvoid AL_APIENTRY alListenerf(ALenum param, ALfloat value)
+{
+    ALlistener *listener;
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    listener = context->Listener;
+    almtx_lock(&context->PropLock);
+    switch(param)
+    {
+    case AL_GAIN:
+        if(!(value >= 0.0f && isfinite(value)))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener gain out of range");
+        listener->Gain = value;
+        DO_UPDATEPROPS();
+        break;
+
+    case AL_METERS_PER_UNIT:
+        if(!(value >= AL_MIN_METERS_PER_UNIT && value <= AL_MAX_METERS_PER_UNIT))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener meters per unit out of range");
+        context->MetersPerUnit = value;
+        if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
+            UpdateContextProps(context);
+        else
+            ATOMIC_FLAG_CLEAR(&context->PropsClean, almemory_order_release);
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener float property");
+    }
+
+done:
+    almtx_unlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alListener3f(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3)
+{
+    ALlistener *listener;
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    listener = context->Listener;
+    almtx_lock(&context->PropLock);
+    switch(param)
+    {
+    case AL_POSITION:
+        if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener position out of range");
+        listener->Position[0] = value1;
+        listener->Position[1] = value2;
+        listener->Position[2] = value3;
+        DO_UPDATEPROPS();
+        break;
+
+    case AL_VELOCITY:
+        if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener velocity out of range");
+        listener->Velocity[0] = value1;
+        listener->Velocity[1] = value2;
+        listener->Velocity[2] = value3;
+        DO_UPDATEPROPS();
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-float property");
+    }
+
+done:
+    almtx_unlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alListenerfv(ALenum param, const ALfloat *values)
+{
+    ALlistener *listener;
+    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;
+
+    listener = context->Listener;
+    almtx_lock(&context->PropLock);
+    if(!values) SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer");
+    switch(param)
+    {
+    case AL_ORIENTATION:
+        if(!(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2]) &&
+             isfinite(values[3]) && isfinite(values[4]) && isfinite(values[5])))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener orientation out of range");
+        /* AT then UP */
+        listener->Forward[0] = values[0];
+        listener->Forward[1] = values[1];
+        listener->Forward[2] = values[2];
+        listener->Up[0] = values[3];
+        listener->Up[1] = values[4];
+        listener->Up[2] = values[5];
+        DO_UPDATEPROPS();
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener float-vector property");
+    }
+
+done:
+    almtx_unlock(&context->PropLock);
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alListeneri(ALenum param, ALint UNUSED(value))
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    almtx_lock(&context->PropLock);
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener integer property");
+    }
+    almtx_unlock(&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;
+
+    almtx_lock(&context->PropLock);
+    switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-integer property");
+    }
+    almtx_unlock(&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;
+
+    almtx_lock(&context->PropLock);
+    if(!values)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    else switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener integer-vector property");
+    }
+    almtx_unlock(&context->PropLock);
+
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    almtx_lock(&context->PropLock);
+    if(!value)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    else switch(param)
+    {
+    case AL_GAIN:
+        *value = context->Listener->Gain;
+        break;
+
+    case AL_METERS_PER_UNIT:
+        *value = context->MetersPerUnit;
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener float property");
+    }
+    almtx_unlock(&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;
+
+    almtx_lock(&context->PropLock);
+    if(!value1 || !value2 || !value3)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    else 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:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-float property");
+    }
+    almtx_unlock(&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;
+
+    almtx_lock(&context->PropLock);
+    if(!values)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    else 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:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener float-vector property");
+    }
+    almtx_unlock(&context->PropLock);
+
+    ALCcontext_DecRef(context);
+}
+
+
+AL_API ALvoid AL_APIENTRY alGetListeneri(ALenum param, ALint *value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    almtx_lock(&context->PropLock);
+    if(!value)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    else switch(param)
+    {
+    default:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener integer property");
+    }
+    almtx_unlock(&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;
+
+    almtx_lock(&context->PropLock);
+    if(!value1 || !value2 || !value3)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    else 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:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-integer property");
+    }
+    almtx_unlock(&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;
+
+    almtx_lock(&context->PropLock);
+    if(!values)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    else 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:
+        alSetError(context, AL_INVALID_ENUM, "Invalid listener integer-vector property");
+    }
+    almtx_unlock(&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(&context->FreeListenerProps, 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(&context->FreeListenerProps, &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;
+
+    /* 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*, &context->FreeListenerProps, props);
+    }
+}

OpenAL32/alState.c

@@ -0,0 +1,900 @@
+/**
+ * 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 alCubicResampler[] = "Cubic";
+static const ALchar alBSinc12Resampler[] = "11th order Sinc";
+static const ALchar alBSinc24Resampler[] = "23rd order Sinc";
+
+/* WARNING: Non-standard export! Not part of any extension, or exposed in the
+ * alcFunctions list.
+ */
+AL_API const ALchar* AL_APIENTRY alsoft_get_version(void)
+{
+    const char *spoof = getenv("ALSOFT_SPOOF_VERSION");
+    if(spoof && spoof[0] != '\0') return spoof;
+    return ALSOFT_VERSION;
+}
+
+#define DO_UPDATEPROPS() do {                                                 \
+    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))          \
+        UpdateContextProps(context);                                          \
+    else                                                                      \
+        ATOMIC_FLAG_CLEAR(&context->PropsClean, almemory_order_release);      \
+} while(0)
+
+
+AL_API ALvoid AL_APIENTRY alEnable(ALenum capability)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    almtx_lock(&context->PropLock);
+    switch(capability)
+    {
+    case AL_SOURCE_DISTANCE_MODEL:
+        context->SourceDistanceModel = AL_TRUE;
+        DO_UPDATEPROPS();
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid enable property 0x%04x", capability);
+    }
+    almtx_unlock(&context->PropLock);
+
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alDisable(ALenum capability)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    almtx_lock(&context->PropLock);
+    switch(capability)
+    {
+    case AL_SOURCE_DISTANCE_MODEL:
+        context->SourceDistanceModel = AL_FALSE;
+        DO_UPDATEPROPS();
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid disable property 0x%04x", capability);
+    }
+    almtx_unlock(&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;
+
+    almtx_lock(&context->PropLock);
+    switch(capability)
+    {
+    case AL_SOURCE_DISTANCE_MODEL:
+        value = context->SourceDistanceModel;
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid is enabled property 0x%04x", capability);
+    }
+    almtx_unlock(&context->PropLock);
+
+    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;
+
+    almtx_lock(&context->PropLock);
+    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:
+        alSetError(context, AL_INVALID_VALUE, "Invalid boolean property 0x%04x", pname);
+    }
+    almtx_unlock(&context->PropLock);
+
+    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;
+
+    almtx_lock(&context->PropLock);
+    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:
+        alSetError(context, AL_INVALID_VALUE, "Invalid double property 0x%04x", pname);
+    }
+    almtx_unlock(&context->PropLock);
+
+    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;
+
+    almtx_lock(&context->PropLock);
+    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:
+        alSetError(context, AL_INVALID_VALUE, "Invalid float property 0x%04x", pname);
+    }
+    almtx_unlock(&context->PropLock);
+
+    ALCcontext_DecRef(context);
+    return value;
+}
+
+AL_API ALint AL_APIENTRY alGetInteger(ALenum pname)
+{
+    ALCcontext *context;
+    ALint value = 0;
+
+    context = GetContextRef();
+    if(!context) return 0;
+
+    almtx_lock(&context->PropLock);
+    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:
+        alSetError(context, AL_INVALID_VALUE, "Invalid integer property 0x%04x", pname);
+    }
+    almtx_unlock(&context->PropLock);
+
+    ALCcontext_DecRef(context);
+    return value;
+}
+
+AL_API ALint64SOFT AL_APIENTRY alGetInteger64SOFT(ALenum pname)
+{
+    ALCcontext *context;
+    ALint64SOFT value = 0;
+
+    context = GetContextRef();
+    if(!context) return 0;
+
+    almtx_lock(&context->PropLock);
+    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:
+        alSetError(context, AL_INVALID_VALUE, "Invalid integer64 property 0x%04x", pname);
+    }
+    almtx_unlock(&context->PropLock);
+
+    ALCcontext_DecRef(context);
+    return value;
+}
+
+AL_API void* AL_APIENTRY alGetPointerSOFT(ALenum pname)
+{
+    ALCcontext *context;
+    void *value = NULL;
+
+    context = GetContextRef();
+    if(!context) return NULL;
+
+    almtx_lock(&context->PropLock);
+    switch(pname)
+    {
+    case AL_EVENT_CALLBACK_FUNCTION_SOFT:
+        value = context->EventCb;
+        break;
+
+    case AL_EVENT_CALLBACK_USER_PARAM_SOFT:
+        value = context->EventParam;
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid pointer property 0x%04x", pname);
+    }
+    almtx_unlock(&context->PropLock);
+
+    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)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    switch(pname)
+    {
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid boolean-vector property 0x%04x", pname);
+    }
+
+    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)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    switch(pname)
+    {
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid double-vector property 0x%04x", pname);
+    }
+
+    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)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    switch(pname)
+    {
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid float-vector property 0x%04x", pname);
+    }
+
+    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;
+
+    if(!values)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    switch(pname)
+    {
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid integer-vector property 0x%04x", pname);
+    }
+
+    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;
+
+    if(!values)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    switch(pname)
+    {
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid integer64-vector property 0x%04x", pname);
+    }
+
+    ALCcontext_DecRef(context);
+}
+
+AL_API void AL_APIENTRY alGetPointervSOFT(ALenum pname, void **values)
+{
+    ALCcontext *context;
+
+    if(values)
+    {
+        switch(pname)
+        {
+            case AL_EVENT_CALLBACK_FUNCTION_SOFT:
+            case AL_EVENT_CALLBACK_USER_PARAM_SOFT:
+                values[0] = alGetPointerSOFT(pname);
+                return;
+        }
+    }
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!values)
+        alSetError(context, AL_INVALID_VALUE, "NULL pointer");
+    switch(pname)
+    {
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid pointer-vector property 0x%04x", pname);
+    }
+
+    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:
+        alSetError(context, AL_INVALID_VALUE, "Invalid string property 0x%04x", pname);
+    }
+
+    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)))
+        alSetError(context, AL_INVALID_VALUE, "Doppler factor %f out of range", value);
+    else
+    {
+        almtx_lock(&context->PropLock);
+        context->DopplerFactor = value;
+        DO_UPDATEPROPS();
+        almtx_unlock(&context->PropLock);
+    }
+
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alDopplerVelocity(ALfloat value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if((ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed)&EventType_Deprecated))
+    {
+        static const ALCchar msg[] =
+            "alDopplerVelocity is deprecated in AL1.1, use alSpeedOfSound";
+        const ALsizei msglen = (ALsizei)strlen(msg);
+        ALbitfieldSOFT enabledevts;
+        almtx_lock(&context->EventCbLock);
+        enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed);
+        if((enabledevts&EventType_Deprecated) && context->EventCb)
+            (*context->EventCb)(AL_EVENT_TYPE_DEPRECATED_SOFT, 0, 0, msglen, msg,
+                                context->EventParam);
+        almtx_unlock(&context->EventCbLock);
+    }
+
+    if(!(value >= 0.0f && isfinite(value)))
+        alSetError(context, AL_INVALID_VALUE, "Doppler velocity %f out of range", value);
+    else
+    {
+        almtx_lock(&context->PropLock);
+        context->DopplerVelocity = value;
+        DO_UPDATEPROPS();
+        almtx_unlock(&context->PropLock);
+    }
+
+    ALCcontext_DecRef(context);
+}
+
+AL_API ALvoid AL_APIENTRY alSpeedOfSound(ALfloat value)
+{
+    ALCcontext *context;
+
+    context = GetContextRef();
+    if(!context) return;
+
+    if(!(value > 0.0f && isfinite(value)))
+        alSetError(context, AL_INVALID_VALUE, "Speed of sound %f out of range", value);
+    else
+    {
+        almtx_lock(&context->PropLock);
+        context->SpeedOfSound = value;
+        DO_UPDATEPROPS();
+        almtx_unlock(&context->PropLock);
+    }
+
+    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))
+        alSetError(context, AL_INVALID_VALUE, "Distance model 0x%04x out of range", value);
+    else
+    {
+        almtx_lock(&context->PropLock);
+        context->DistanceModel = value;
+        if(!context->SourceDistanceModel)
+            DO_UPDATEPROPS();
+        almtx_unlock(&context->PropLock);
+    }
+
+    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,
+        alCubicResampler, alBSinc12Resampler,
+        alBSinc24Resampler,
+    };
+    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))
+            SETERR_GOTO(context, AL_INVALID_VALUE, done, "Resampler name index %d out of range",
+                        index);
+        value = ResamplerNames[index];
+        break;
+
+    default:
+        alSetError(context, AL_INVALID_VALUE, "Invalid string indexed property");
+    }
+
+done:
+    ALCcontext_DecRef(context);
+    return value;
+}
+
+
+void UpdateContextProps(ALCcontext *context)
+{
+    struct ALcontextProps *props;
+
+    /* Get an unused proprty container, or allocate a new one as needed. */
+    props = ATOMIC_LOAD(&context->FreeContextProps, almemory_order_acquire);
+    if(!props)
+        props = al_calloc(16, sizeof(*props));
+    else
+    {
+        struct ALcontextProps *next;
+        do {
+            next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
+        } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeContextProps, &props, next,
+                almemory_order_seq_cst, almemory_order_acquire) == 0);
+    }
+
+    /* Copy in current property values. */
+    props->MetersPerUnit = context->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(&context->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 ALcontextProps*, &context->FreeContextProps, props);
+    }
+}