Fix compile on latest vitaSDK

It's currently not used. More stuff is needed anyway which may need a different
approach.

.gitignore

@@ -1,5 +1,5 @@
-build
-winbuild
-include/sndio.h
-include/sys
+build*/
+winbuild/
+win64build/
 openal-soft.kdev4
+.kdev4/

.travis.yml

@@ -0,0 +1,66 @@
+language: c
+matrix:
+  include:
+    - os: linux
+      dist: trusty
+    - os: linux
+      dist: trusty
+      env:
+        - BUILD_ANDROID=true
+    - os: osx
+sudo: required
+install:
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then
+      # Install pulseaudio, portaudio, ALSA, JACK dependencies for
+      # corresponding backends.
+      # Install Qt5 dependency for alsoft-config.
+      sudo apt-get install -qq \
+        libpulse-dev \
+        portaudio19-dev \
+        libasound2-dev \
+        libjack-dev \
+        qtbase5-dev
+    fi
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
+      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-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:
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then
+      cmake \
+        -DALSOFT_REQUIRE_ALSA=ON \
+        -DALSOFT_REQUIRE_OSS=ON \
+        -DALSOFT_REQUIRE_PORTAUDIO=ON \
+        -DALSOFT_REQUIRE_PULSEAUDIO=ON \
+        -DALSOFT_REQUIRE_JACK=ON \
+        -DALSOFT_EMBED_HRTF_DATA=YES \
+        .
+    fi
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
+      cmake \
+        -DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r15/build/cmake/android.toolchain.cmake \
+        -DALSOFT_REQUIRE_OPENSL=ON \
+        -DALSOFT_EMBED_HRTF_DATA=YES \
+        .
+    fi
+  - >
+    if [[ "${TRAVIS_OS_NAME}" == "osx" ]]; then
+      cmake \
+        -DALSOFT_REQUIRE_COREAUDIO=ON \
+        -DALSOFT_EMBED_HRTF_DATA=YES \
+        .
+    fi
+  - make -j2

Alc/alcConfig.c

@@ -1,363 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#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>
-
-#include "alMain.h"
-
-#ifdef _WIN32_IE
-#include <shlobj.h>
-#endif
-
-typedef struct ConfigEntry {
-    char *key;
-    char *value;
-} ConfigEntry;
-
-typedef struct ConfigBlock {
-    char *name;
-    ConfigEntry *entries;
-    unsigned int entryCount;
-} ConfigBlock;
-
-static ConfigBlock *cfgBlocks;
-static unsigned int cfgCount;
-
-static char buffer[1024];
-
-static void LoadConfigFromFile(FILE *f)
-{
-    ConfigBlock *curBlock = cfgBlocks;
-    ConfigEntry *ent;
-
-    while(fgets(buffer, sizeof(buffer), f))
-    {
-        int i = 0;
-
-        while(isspace(buffer[i]))
-            i++;
-        if(!buffer[i] || buffer[i] == '#')
-            continue;
-
-        memmove(buffer, buffer+i, strlen(buffer+i)+1);
-
-        if(buffer[0] == '[')
-        {
-            ConfigBlock *nextBlock;
-            unsigned int i;
-
-            i = 1;
-            while(buffer[i] && buffer[i] != ']')
-                i++;
-
-            if(!buffer[i])
-            {
-                 ERR("config parse error: bad line \"%s\"\n", buffer);
-                 continue;
-            }
-            buffer[i] = 0;
-
-            do {
-                i++;
-                if(buffer[i] && !isspace(buffer[i]))
-                {
-                    if(buffer[i] != '#')
-                        WARN("config warning: extra data after block: \"%s\"\n", buffer+i);
-                    break;
-                }
-            } while(buffer[i]);
-
-            nextBlock = NULL;
-            for(i = 0;i < cfgCount;i++)
-            {
-                if(strcasecmp(cfgBlocks[i].name, buffer+1) == 0)
-                {
-                    nextBlock = cfgBlocks+i;
-                    TRACE("found block '%s'\n", nextBlock->name);
-                    break;
-                }
-            }
-
-            if(!nextBlock)
-            {
-                nextBlock = realloc(cfgBlocks, (cfgCount+1)*sizeof(ConfigBlock));
-                if(!nextBlock)
-                {
-                     ERR("config parse error: error reallocating config blocks\n");
-                     continue;
-                }
-                cfgBlocks = nextBlock;
-                nextBlock = cfgBlocks+cfgCount;
-                cfgCount++;
-
-                nextBlock->name = strdup(buffer+1);
-                nextBlock->entries = NULL;
-                nextBlock->entryCount = 0;
-
-                TRACE("found new block '%s'\n", nextBlock->name);
-            }
-            curBlock = nextBlock;
-            continue;
-        }
-
-        /* Look for the option name */
-        i = 0;
-        while(buffer[i] && buffer[i] != '#' && buffer[i] != '=' &&
-              !isspace(buffer[i]))
-            i++;
-
-        if(!buffer[i] || buffer[i] == '#' || i == 0)
-        {
-            ERR("config parse error: malformed option line: \"%s\"\n", buffer);
-            continue;
-        }
-
-        /* Seperate the option */
-        if(buffer[i] != '=')
-        {
-            buffer[i++] = 0;
-
-            while(isspace(buffer[i]))
-                i++;
-            if(buffer[i] != '=')
-            {
-                ERR("config parse error: option without a value: \"%s\"\n", buffer);
-                continue;
-            }
-        }
-        /* Find the start of the value */
-        buffer[i++] = 0;
-        while(isspace(buffer[i]))
-            i++;
-
-        /* Check if we already have this option set */
-        ent = curBlock->entries;
-        while((unsigned int)(ent-curBlock->entries) < curBlock->entryCount)
-        {
-            if(strcasecmp(ent->key, buffer) == 0)
-                break;
-            ent++;
-        }
-
-        if((unsigned int)(ent-curBlock->entries) >= curBlock->entryCount)
-        {
-            /* Allocate a new option entry */
-            ent = realloc(curBlock->entries, (curBlock->entryCount+1)*sizeof(ConfigEntry));
-            if(!ent)
-            {
-                 ERR("config parse error: error reallocating config entries\n");
-                 continue;
-            }
-            curBlock->entries = ent;
-            ent = curBlock->entries + curBlock->entryCount;
-            curBlock->entryCount++;
-
-            ent->key = strdup(buffer);
-            ent->value = NULL;
-        }
-
-        /* Look for the end of the line (Null term, new-line, or #-symbol) and
-           eat up the trailing whitespace */
-        memmove(buffer, buffer+i, strlen(buffer+i)+1);
-
-        i = 0;
-        while(buffer[i] && buffer[i] != '#' && buffer[i] != '\n')
-            i++;
-        do {
-            i--;
-        } while(i >= 0 && isspace(buffer[i]));
-        buffer[++i] = 0;
-
-        free(ent->value);
-        ent->value = strdup(buffer);
-
-        TRACE("found '%s' = '%s'\n", ent->key, ent->value);
-    }
-}
-
-void ReadALConfig(void)
-{
-    const char *str;
-    FILE *f;
-
-    cfgBlocks = calloc(1, sizeof(ConfigBlock));
-    cfgBlocks->name = strdup("general");
-    cfgCount = 1;
-
-#ifdef _WIN32
-    if(SHGetSpecialFolderPathA(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE)
-    {
-        size_t p = strlen(buffer);
-        snprintf(buffer+p, sizeof(buffer)-p, "\\alsoft.ini");
-        f = fopen(buffer, "rt");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-#else
-    f = fopen("/etc/openal/alsoft.conf", "r");
-    if(f)
-    {
-        LoadConfigFromFile(f);
-        fclose(f);
-    }
-    if((str=getenv("HOME")) != NULL && *str)
-    {
-        snprintf(buffer, sizeof(buffer), "%s/.alsoftrc", str);
-        f = fopen(buffer, "r");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-#endif
-    if((str=getenv("ALSOFT_CONF")) != NULL && *str)
-    {
-        f = fopen(str, "r");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-}
-
-void FreeALConfig(void)
-{
-    unsigned int i;
-
-    for(i = 0;i < cfgCount;i++)
-    {
-        unsigned int j;
-        for(j = 0;j < cfgBlocks[i].entryCount;j++)
-        {
-           free(cfgBlocks[i].entries[j].key);
-           free(cfgBlocks[i].entries[j].value);
-        }
-        free(cfgBlocks[i].entries);
-        free(cfgBlocks[i].name);
-    }
-    free(cfgBlocks);
-    cfgBlocks = NULL;
-    cfgCount = 0;
-}
-
-const char *GetConfigValue(const char *blockName, const char *keyName, const char *def)
-{
-    unsigned int i, j;
-
-    if(!keyName)
-        return def;
-
-    if(!blockName)
-        blockName = "general";
-
-    for(i = 0;i < cfgCount;i++)
-    {
-        if(strcasecmp(cfgBlocks[i].name, blockName) != 0)
-            continue;
-
-        for(j = 0;j < cfgBlocks[i].entryCount;j++)
-        {
-            if(strcasecmp(cfgBlocks[i].entries[j].key, keyName) == 0)
-            {
-                TRACE("Found %s:%s = \"%s\"\n", blockName, keyName,
-                      cfgBlocks[i].entries[j].value);
-                if(cfgBlocks[i].entries[j].value[0])
-                    return cfgBlocks[i].entries[j].value;
-                return def;
-            }
-        }
-    }
-
-    TRACE("Key %s:%s not found\n", blockName, keyName);
-    return def;
-}
-
-int ConfigValueExists(const char *blockName, const char *keyName)
-{
-    const char *val = GetConfigValue(blockName, keyName, "");
-    return !!val[0];
-}
-
-int ConfigValueStr(const char *blockName, const char *keyName, const char **ret)
-{
-    const char *val = GetConfigValue(blockName, keyName, "");
-    if(!val[0]) return 0;
-
-    *ret = val;
-    return 1;
-}
-
-int ConfigValueInt(const char *blockName, const char *keyName, int *ret)
-{
-    const char *val = GetConfigValue(blockName, keyName, "");
-    if(!val[0]) return 0;
-
-    *ret = strtol(val, NULL, 0);
-    return 1;
-}
-
-int ConfigValueUInt(const char *blockName, const char *keyName, unsigned int *ret)
-{
-    const char *val = GetConfigValue(blockName, keyName, "");
-    if(!val[0]) return 0;
-
-    *ret = strtoul(val, NULL, 0);
-    return 1;
-}
-
-int ConfigValueFloat(const char *blockName, const char *keyName, float *ret)
-{
-    const char *val = GetConfigValue(blockName, keyName, "");
-    if(!val[0]) return 0;
-
-#ifdef HAVE_STRTOF
-    *ret = strtof(val, NULL);
-#else
-    *ret = (float)strtod(val, NULL);
-#endif
-    return 1;
-}
-
-int GetConfigValueBool(const char *blockName, const char *keyName, int def)
-{
-    const char *val = GetConfigValue(blockName, keyName, "");
-
-    if(!val[0]) return !!def;
-    return (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||
-            strcasecmp(val, "on") == 0 || atoi(val) != 0);
-}

Alc/alcDedicated.c

@@ -1,100 +0,0 @@
-/**
- * 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., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALdedicatedState {
-    // Must be first in all effects!
-    ALeffectState state;
-
-    ALfloat gains[MaxChannels];
-} ALdedicatedState;
-
-
-static ALvoid DedicatedDestroy(ALeffectState *effect)
-{
-    ALdedicatedState *state = (ALdedicatedState*)effect;
-    free(state);
-}
-
-static ALboolean DedicatedDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
-{
-    (void)effect;
-    (void)Device;
-    return AL_TRUE;
-}
-
-static ALvoid DedicatedUpdate(ALeffectState *effect, ALCdevice *device, const ALeffectslot *Slot)
-{
-    ALdedicatedState *state = (ALdedicatedState*)effect;
-    ALfloat Gain;
-    ALsizei s;
-
-    Gain = Slot->Gain * Slot->effect.Dedicated.Gain;
-    for(s = 0;s < MaxChannels;s++)
-        state->gains[s] = 0.0f;
-
-    if(Slot->effect.type == AL_EFFECT_DEDICATED_DIALOGUE)
-        ComputeAngleGains(device, atan2f(0.0f, 1.0f), 0.0f, Gain, state->gains);
-    else if(Slot->effect.type == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
-        state->gains[LFE] = Gain;
-}
-
-static ALvoid DedicatedProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
-{
-    ALdedicatedState *state = (ALdedicatedState*)effect;
-    const ALfloat *gains = state->gains;
-    ALuint i, c;
-
-    for(c = 0;c < MaxChannels;c++)
-    {
-        for(i = 0;i < SamplesToDo;i++)
-            SamplesOut[c][i] = SamplesIn[i] * gains[c];
-    }
-}
-
-ALeffectState *DedicatedCreate(void)
-{
-    ALdedicatedState *state;
-    ALsizei s;
-
-    state = malloc(sizeof(*state));
-    if(!state)
-        return NULL;
-
-    state->state.Destroy = DedicatedDestroy;
-    state->state.DeviceUpdate = DedicatedDeviceUpdate;
-    state->state.Update = DedicatedUpdate;
-    state->state.Process = DedicatedProcess;
-
-    for(s = 0;s < MaxChannels;s++)
-        state->gains[s] = 0.0f;
-
-    return &state->state;
-}

Alc/alcEcho.c

@@ -1,184 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2009 by Chris Robinson.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALechoState {
-    // Must be first in all effects!
-    ALeffectState state;
-
-    ALfloat *SampleBuffer;
-    ALuint BufferLength;
-
-    // The echo is two tap. The delay is the number of samples from before the
-    // current offset
-    struct {
-        ALuint delay;
-    } Tap[2];
-    ALuint Offset;
-    /* The panning gains for the two taps */
-    ALfloat Gain[2][MaxChannels];
-
-    ALfloat FeedGain;
-
-    FILTER iirFilter;
-    ALfloat history[2];
-} ALechoState;
-
-static ALvoid EchoDestroy(ALeffectState *effect)
-{
-    ALechoState *state = (ALechoState*)effect;
-    if(state)
-    {
-        free(state->SampleBuffer);
-        state->SampleBuffer = NULL;
-        free(state);
-    }
-}
-
-static ALboolean EchoDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
-{
-    ALechoState *state = (ALechoState*)effect;
-    ALuint maxlen, i;
-
-    // Use the next power of 2 for the buffer length, so the tap offsets can be
-    // wrapped using a mask instead of a modulo
-    maxlen  = fastf2u(AL_ECHO_MAX_DELAY * Device->Frequency) + 1;
-    maxlen += fastf2u(AL_ECHO_MAX_LRDELAY * Device->Frequency) + 1;
-    maxlen  = NextPowerOf2(maxlen);
-
-    if(maxlen != state->BufferLength)
-    {
-        void *temp;
-
-        temp = realloc(state->SampleBuffer, maxlen * sizeof(ALfloat));
-        if(!temp)
-            return AL_FALSE;
-        state->SampleBuffer = temp;
-        state->BufferLength = maxlen;
-    }
-    for(i = 0;i < state->BufferLength;i++)
-        state->SampleBuffer[i] = 0.0f;
-
-    return AL_TRUE;
-}
-
-static ALvoid EchoUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
-{
-    ALechoState *state = (ALechoState*)effect;
-    ALuint frequency = Device->Frequency;
-    ALfloat lrpan, cw, g, gain;
-    ALfloat dirGain;
-    ALuint i;
-
-    state->Tap[0].delay = fastf2u(Slot->effect.Echo.Delay * frequency) + 1;
-    state->Tap[1].delay = fastf2u(Slot->effect.Echo.LRDelay * frequency);
-    state->Tap[1].delay += state->Tap[0].delay;
-
-    lrpan = Slot->effect.Echo.Spread;
-
-    state->FeedGain = Slot->effect.Echo.Feedback;
-
-    cw = cosf(F_PI*2.0f * LOWPASSFREQREF / frequency);
-    g = 1.0f - Slot->effect.Echo.Damping;
-    state->iirFilter.coeff = lpCoeffCalc(g, cw);
-
-    gain = Slot->Gain;
-    for(i = 0;i < MaxChannels;i++)
-    {
-        state->Gain[0][i] = 0.0f;
-        state->Gain[1][i] = 0.0f;
-    }
-
-    dirGain = fabsf(lrpan);
-
-    /* First tap panning */
-    ComputeAngleGains(Device, atan2f(-lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[0]);
-
-    /* Second tap panning */
-    ComputeAngleGains(Device, atan2f(+lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[1]);
-}
-
-static ALvoid EchoProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
-{
-    ALechoState *state = (ALechoState*)effect;
-    const ALuint mask = state->BufferLength-1;
-    const ALuint tap1 = state->Tap[0].delay;
-    const ALuint tap2 = state->Tap[1].delay;
-    ALuint offset = state->Offset;
-    ALfloat smp;
-    ALuint i, k;
-
-    for(i = 0;i < SamplesToDo;i++,offset++)
-    {
-        /* First tap */
-        smp = state->SampleBuffer[(offset-tap1) & mask];
-        for(k = 0;k < MaxChannels;k++)
-            SamplesOut[k][i] += smp * state->Gain[0][k];
-
-        /* Second tap */
-        smp = state->SampleBuffer[(offset-tap2) & mask];
-        for(k = 0;k < MaxChannels;k++)
-            SamplesOut[k][i] += smp * state->Gain[1][k];
-
-        // Apply damping and feedback gain to the second tap, and mix in the
-        // new sample
-        smp = lpFilter2P(&state->iirFilter, 0, smp+SamplesIn[i]);
-        state->SampleBuffer[offset&mask] = smp * state->FeedGain;
-    }
-    state->Offset = offset;
-}
-
-ALeffectState *EchoCreate(void)
-{
-    ALechoState *state;
-
-    state = malloc(sizeof(*state));
-    if(!state)
-        return NULL;
-
-    state->state.Destroy = EchoDestroy;
-    state->state.DeviceUpdate = EchoDeviceUpdate;
-    state->state.Update = EchoUpdate;
-    state->state.Process = EchoProcess;
-
-    state->BufferLength = 0;
-    state->SampleBuffer = NULL;
-
-    state->Tap[0].delay = 0;
-    state->Tap[1].delay = 0;
-    state->Offset = 0;
-
-    state->iirFilter.coeff = 0.0f;
-    state->iirFilter.history[0] = 0.0f;
-    state->iirFilter.history[1] = 0.0f;
-
-    return &state->state;
-}

Alc/alcModulator.c

@@ -1,204 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2009 by Chris Robinson.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALmodulatorState {
-    // Must be first in all effects!
-    ALeffectState state;
-
-    enum {
-        SINUSOID,
-        SAWTOOTH,
-        SQUARE
-    } Waveform;
-
-    ALuint index;
-    ALuint step;
-
-    ALfloat Gain[MaxChannels];
-
-    FILTER iirFilter;
-    ALfloat history[1];
-} ALmodulatorState;
-
-#define WAVEFORM_FRACBITS  16
-#define WAVEFORM_FRACONE   (1<<WAVEFORM_FRACBITS)
-#define WAVEFORM_FRACMASK  (WAVEFORM_FRACONE-1)
-
-static __inline ALfloat Sin(ALuint index)
-{
-    return sinf(index * (F_PI*2.0f / WAVEFORM_FRACONE));
-}
-
-static __inline ALfloat Saw(ALuint index)
-{
-    return index*(2.0f/WAVEFORM_FRACONE) - 1.0f;
-}
-
-static __inline ALfloat Square(ALuint index)
-{
-    return ((index>>(WAVEFORM_FRACBITS-1))&1)*2.0f - 1.0f;
-}
-
-
-static __inline ALfloat hpFilter1P(FILTER *iir, ALuint offset, ALfloat input)
-{
-    ALfloat *history = &iir->history[offset];
-    ALfloat a = iir->coeff;
-    ALfloat output = input;
-
-    output = output + (history[0]-output)*a;
-    history[0] = output;
-
-    return input - output;
-}
-
-
-#define DECL_TEMPLATE(func)                                                   \
-static void Process##func(ALmodulatorState *state, ALuint SamplesToDo,        \
-  const ALfloat *RESTRICT SamplesIn,                                          \
-  ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])                                 \
-{                                                                             \
-    const ALuint step = state->step;                                          \
-    ALuint index = state->index;                                              \
-    ALfloat samp;                                                             \
-    ALuint i, k;                                                              \
-                                                                              \
-    for(i = 0;i < SamplesToDo;i++)                                            \
-    {                                                                         \
-        samp = SamplesIn[i];                                                  \
-                                                                              \
-        index += step;                                                        \
-        index &= WAVEFORM_FRACMASK;                                           \
-        samp *= func(index);                                                  \
-                                                                              \
-        samp = hpFilter1P(&state->iirFilter, 0, samp);                        \
-                                                                              \
-        for(k = 0;k < MaxChannels;k++)                                        \
-            SamplesOut[k][i] += state->Gain[k] * samp;                        \
-    }                                                                         \
-    state->index = index;                                                     \
-}
-
-DECL_TEMPLATE(Sin)
-DECL_TEMPLATE(Saw)
-DECL_TEMPLATE(Square)
-
-#undef DECL_TEMPLATE
-
-
-static ALvoid ModulatorDestroy(ALeffectState *effect)
-{
-    ALmodulatorState *state = (ALmodulatorState*)effect;
-    free(state);
-}
-
-static ALboolean ModulatorDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
-{
-    return AL_TRUE;
-    (void)effect;
-    (void)Device;
-}
-
-static ALvoid ModulatorUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
-{
-    ALmodulatorState *state = (ALmodulatorState*)effect;
-    ALfloat gain, cw, a = 0.0f;
-    ALuint index;
-
-    if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
-        state->Waveform = SINUSOID;
-    else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
-        state->Waveform = SAWTOOTH;
-    else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)
-        state->Waveform = SQUARE;
-
-    state->step = fastf2u(Slot->effect.Modulator.Frequency*WAVEFORM_FRACONE /
-                          Device->Frequency);
-    if(state->step == 0) state->step = 1;
-
-    cw = cosf(F_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
-                          Device->Frequency);
-    a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
-    state->iirFilter.coeff = a;
-
-    gain = sqrtf(1.0f/Device->NumChan);
-    gain *= Slot->Gain;
-    for(index = 0;index < MaxChannels;index++)
-        state->Gain[index] = 0.0f;
-    for(index = 0;index < Device->NumChan;index++)
-    {
-        enum Channel chan = Device->Speaker2Chan[index];
-        state->Gain[chan] = gain;
-    }
-}
-
-static ALvoid ModulatorProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
-{
-    ALmodulatorState *state = (ALmodulatorState*)effect;
-
-    switch(state->Waveform)
-    {
-        case SINUSOID:
-            ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut);
-            break;
-
-        case SAWTOOTH:
-            ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut);
-            break;
-
-        case SQUARE:
-            ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut);
-            break;
-    }
-}
-
-ALeffectState *ModulatorCreate(void)
-{
-    ALmodulatorState *state;
-
-    state = malloc(sizeof(*state));
-    if(!state)
-        return NULL;
-
-    state->state.Destroy = ModulatorDestroy;
-    state->state.DeviceUpdate = ModulatorDeviceUpdate;
-    state->state.Update = ModulatorUpdate;
-    state->state.Process = ModulatorProcess;
-
-    state->index = 0;
-    state->step = 1;
-
-    state->iirFilter.coeff = 0.0f;
-    state->iirFilter.history[0] = 0.0f;
-
-    return &state->state;
-}

Alc/alcRing.c

@@ -1,127 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <string.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-
-
-struct RingBuffer {
-    ALubyte *mem;
-
-    ALsizei frame_size;
-    ALsizei length;
-    ALint read_pos;
-    ALint write_pos;
-
-    CRITICAL_SECTION cs;
-};
-
-
-RingBuffer *CreateRingBuffer(ALsizei frame_size, ALsizei length)
-{
-    RingBuffer *ring = calloc(1, sizeof(*ring) + ((length+1) * frame_size));
-    if(ring)
-    {
-        ring->mem = (ALubyte*)(ring+1);
-
-        ring->frame_size = frame_size;
-        ring->length = length+1;
-        ring->read_pos = 0;
-        ring->write_pos = 0;
-
-        InitializeCriticalSection(&ring->cs);
-    }
-    return ring;
-}
-
-void DestroyRingBuffer(RingBuffer *ring)
-{
-    if(ring)
-    {
-        DeleteCriticalSection(&ring->cs);
-        free(ring);
-    }
-}
-
-ALsizei RingBufferSize(RingBuffer *ring)
-{
-    ALsizei s;
-
-    EnterCriticalSection(&ring->cs);
-    s = (ring->write_pos-ring->read_pos+ring->length) % ring->length;
-    LeaveCriticalSection(&ring->cs);
-
-    return s;
-}
-
-void WriteRingBuffer(RingBuffer *ring, const ALubyte *data, ALsizei len)
-{
-    int remain;
-
-    EnterCriticalSection(&ring->cs);
-
-    remain = (ring->read_pos-ring->write_pos-1+ring->length) % ring->length;
-    if(remain < len) len = remain;
-
-    if(len > 0)
-    {
-        remain = ring->length - ring->write_pos;
-        if(remain < len)
-        {
-            memcpy(ring->mem+(ring->write_pos*ring->frame_size), data,
-                   remain*ring->frame_size);
-            memcpy(ring->mem, data+(remain*ring->frame_size),
-                   (len-remain)*ring->frame_size);
-        }
-        else
-            memcpy(ring->mem+(ring->write_pos*ring->frame_size), data,
-                   len*ring->frame_size);
-
-        ring->write_pos += len;
-        ring->write_pos %= ring->length;
-    }
-
-    LeaveCriticalSection(&ring->cs);
-}
-
-void ReadRingBuffer(RingBuffer *ring, ALubyte *data, ALsizei len)
-{
-    int remain;
-
-    EnterCriticalSection(&ring->cs);
-
-    remain = ring->length - ring->read_pos;
-    if(remain < len)
-    {
-        memcpy(data, ring->mem+(ring->read_pos*ring->frame_size), remain*ring->frame_size);
-        memcpy(data+(remain*ring->frame_size), ring->mem, (len-remain)*ring->frame_size);
-    }
-    else
-        memcpy(data, ring->mem+(ring->read_pos*ring->frame_size), len*ring->frame_size);
-
-    ring->read_pos += len;
-    ring->read_pos %= ring->length;
-
-    LeaveCriticalSection(&ring->cs);
-}

Alc/alcThread.c

@@ -1,144 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alThunk.h"
-
-#define THREAD_STACK_SIZE (1*1024*1024) /* 1MB */
-
-#ifdef _WIN32
-
-typedef struct {
-    ALuint (*func)(ALvoid*);
-    ALvoid *ptr;
-    HANDLE thread;
-} ThreadInfo;
-
-static DWORD CALLBACK StarterFunc(void *ptr)
-{
-    ThreadInfo *inf = (ThreadInfo*)ptr;
-    ALint ret;
-
-    ret = inf->func(inf->ptr);
-    ExitThread((DWORD)ret);
-
-    return (DWORD)ret;
-}
-
-ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
-{
-    DWORD dummy;
-    ThreadInfo *inf = malloc(sizeof(ThreadInfo));
-    if(!inf) return 0;
-
-    inf->func = func;
-    inf->ptr = ptr;
-
-    inf->thread = CreateThread(NULL, THREAD_STACK_SIZE, StarterFunc, inf, 0, &dummy);
-    if(!inf->thread)
-    {
-        free(inf);
-        return NULL;
-    }
-
-    return inf;
-}
-
-ALuint StopThread(ALvoid *thread)
-{
-    ThreadInfo *inf = thread;
-    DWORD ret = 0;
-
-    WaitForSingleObject(inf->thread, INFINITE);
-    GetExitCodeThread(inf->thread, &ret);
-    CloseHandle(inf->thread);
-
-    free(inf);
-
-    return (ALuint)ret;
-}
-
-#else
-
-#include <pthread.h>
-
-typedef struct {
-    ALuint (*func)(ALvoid*);
-    ALvoid *ptr;
-    ALuint ret;
-    pthread_t thread;
-} ThreadInfo;
-
-static void *StarterFunc(void *ptr)
-{
-    ThreadInfo *inf = (ThreadInfo*)ptr;
-    inf->ret = inf->func(inf->ptr);
-    return NULL;
-}
-
-ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
-{
-    pthread_attr_t attr;
-    ThreadInfo *inf = malloc(sizeof(ThreadInfo));
-    if(!inf) return NULL;
-
-    if(pthread_attr_init(&attr) != 0)
-    {
-        free(inf);
-        return NULL;
-    }
-    if(pthread_attr_setstacksize(&attr, THREAD_STACK_SIZE) != 0)
-    {
-        pthread_attr_destroy(&attr);
-        free(inf);
-        return NULL;
-    }
-
-    inf->func = func;
-    inf->ptr = ptr;
-    if(pthread_create(&inf->thread, &attr, StarterFunc, inf) != 0)
-    {
-        pthread_attr_destroy(&attr);
-        free(inf);
-        return NULL;
-    }
-    pthread_attr_destroy(&attr);
-
-    return inf;
-}
-
-ALuint StopThread(ALvoid *thread)
-{
-    ThreadInfo *inf = thread;
-    ALuint ret;
-
-    pthread_join(inf->thread, NULL);
-    ret = inf->ret;
-
-    free(inf);
-
-    return ret;
-}
-
-#endif

Alc/alconfig.c

@@ -0,0 +1,721 @@
+/**
+ * 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);
+}
+#elif defined __vita__
+void ReadALConfig(void)
+{
+    const char* config_paths[] =
+    {
+        "app0:/alsoft.conf",
+        "ux0:/data/openal/alsoft.conf"
+    };
+
+    FILE* f;
+    unsigned int i;
+    for(i = 0; i < sizeof(config_paths) / sizeof(*config_paths); ++i)
+    {
+        TRACE("Loading config %s...\n", config_paths[i]);
+        f = al_fopen(config_paths[i], "r");
+        if(f)
+        {
+            LoadConfigFromFile(f);
+            fclose(f);
+            break;
+        }
+    }
+}
+#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 +1,179 @@
+#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 *ALCvitaBackendFactory_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

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -26,191 +26,142 @@
 
 #include "alMain.h"
 #include "alu.h"
+#include "ringbuffer.h"
 
-#include <CoreServices/CoreServices.h>
 #include <unistd.h>
 #include <AudioUnit/AudioUnit.h>
 #include <AudioToolbox/AudioToolbox.h>
 
+#include "backends/base.h"
 
-typedef struct {
-    AudioUnit audioUnit;
-
-    ALuint frameSize;
-    ALdouble sampleRateRatio;              // Ratio of hardware sample rate / requested sample rate
-    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
-
-    AudioConverterRef audioConverter;      // Sample rate converter if needed
-    AudioBufferList *bufferList;           // Buffer for data coming from the input device
-    ALCvoid *resampleBuffer;               // Buffer for returned RingBuffer data when resampling
-
-    RingBuffer *ring;
-} ca_data;
 
 static const ALCchar ca_device[] = "CoreAudio Default";
 
 
-static void destroy_buffer_list(AudioBufferList* list)
+typedef struct ALCcoreAudioPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    AudioUnit audioUnit;
+
+    ALuint frameSize;
+    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
+} ALCcoreAudioPlayback;
+
+static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device);
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self);
+static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name);
+static 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)
 {
-    if(list)
-    {
-        UInt32 i;
-        for(i = 0;i < list->mNumberBuffers;i++)
-            free(list->mBuffers[i].mData);
-        free(list);
-    }
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self);
+
+    self->frameSize = 0;
+    memset(&self->format, 0, sizeof(self->format));
 }
 
-static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
 {
-    AudioBufferList *list;
+    AudioUnitUninitialize(self->audioUnit);
+    AudioComponentInstanceDispose(self->audioUnit);
 
-    list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
-    if(list)
-    {
-        list->mNumberBuffers = 1;
-
-        list->mBuffers[0].mNumberChannels = channelCount;
-        list->mBuffers[0].mDataByteSize = byteSize;
-        list->mBuffers[0].mData = malloc(byteSize);
-        if(list->mBuffers[0].mData == NULL)
-        {
-            free(list);
-            list = NULL;
-        }
-    }
-    return list;
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
 }
 
-static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
-                            UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
-{
-    ALCdevice *device = (ALCdevice*)inRefCon;
-    ca_data *data = (ca_data*)device->ExtraData;
 
+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 / data->frameSize);
+               ioData->mBuffers[0].mDataByteSize / self->frameSize);
+    ALCcoreAudioPlayback_unlock(self);
 
     return noErr;
 }
 
-static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
-        AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData)
+
+static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name)
 {
-    ALCdevice *device = (ALCdevice*)inUserData;
-    ca_data *data = (ca_data*)device->ExtraData;
-
-    // Read from the ring buffer and store temporarily in a large buffer
-    ReadRingBuffer(data->ring, data->resampleBuffer, (ALsizei)(*ioNumberDataPackets));
-
-    // Set the input data
-    ioData->mNumberBuffers = 1;
-    ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
-    ioData->mBuffers[0].mData = data->resampleBuffer;
-    ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame;
-
-    return noErr;
-}
-
-static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
-                                    const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
-                                    UInt32 inNumberFrames, AudioBufferList *ioData)
-{
-    ALCdevice *device = (ALCdevice*)inRefCon;
-    ca_data *data = (ca_data*)device->ExtraData;
-    AudioUnitRenderActionFlags flags = 0;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    AudioComponentDescription desc;
+    AudioComponent comp;
     OSStatus err;
 
-    // fill the bufferList with data from the input device
-    err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList);
-    if(err != noErr)
-    {
-        ERR("AudioUnitRender error: %d\n", err);
-        return err;
-    }
-
-    WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames);
-
-    return noErr;
-}
-
-static ALCenum ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
-{
-    ComponentDescription desc;
-    Component comp;
-    ca_data *data;
-    OSStatus err;
-
-    if(!deviceName)
-        deviceName = ca_device;
-    else if(strcmp(deviceName, ca_device) != 0)
+    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 = FindNextComponent(NULL, &desc);
+    comp = AudioComponentFindNext(NULL, &desc);
     if(comp == NULL)
     {
-        ERR("FindNextComponent failed\n");
+        ERR("AudioComponentFindNext failed\n");
         return ALC_INVALID_VALUE;
     }
 
-    data = calloc(1, sizeof(*data));
-
-    err = OpenAComponent(comp, &data->audioUnit);
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
     if(err != noErr)
     {
-        ERR("OpenAComponent failed\n");
-        free(data);
+        ERR("AudioComponentInstanceNew failed\n");
         return ALC_INVALID_VALUE;
     }
 
     /* init and start the default audio unit... */
-    err = AudioUnitInitialize(data->audioUnit);
+    err = AudioUnitInitialize(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioUnitInitialize failed\n");
-        CloseComponent(data->audioUnit);
-        free(data);
+        AudioComponentInstanceDispose(self->audioUnit);
         return ALC_INVALID_VALUE;
     }
 
-    device->DeviceName = strdup(deviceName);
-    device->ExtraData = data;
+    alstr_copy_cstr(&device->DeviceName, name);
     return ALC_NO_ERROR;
 }
 
-static void ca_close_playback(ALCdevice *device)
+static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-
-    AudioUnitUninitialize(data->audioUnit);
-    CloseComponent(data->audioUnit);
-
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean ca_reset_playback(ALCdevice *device)
-{
-    ca_data *data = (ca_data*)device->ExtraData;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
     AudioStreamBasicDescription streamFormat;
     AURenderCallbackStruct input;
     OSStatus err;
     UInt32 size;
 
-    err = AudioUnitUninitialize(data->audioUnit);
+    err = AudioUnitUninitialize(self->audioUnit);
     if(err != noErr)
         ERR("-- AudioUnitUninitialize failed.\n");
 
     /* retrieve default output unit's properties (output side) */
     size = sizeof(AudioStreamBasicDescription);
-    err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
     if(err != noErr || size != sizeof(AudioStreamBasicDescription))
     {
         ERR("AudioUnitGetProperty failed\n");
@@ -228,7 +179,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
 #endif
 
     /* set default output unit's input side to match output side */
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -237,7 +188,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
 
     if(device->Frequency != streamFormat.mSampleRate)
     {
-        device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
+        device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates *
                                       streamFormat.mSampleRate /
                                       device->Frequency);
         device->Frequency = streamFormat.mSampleRate;
@@ -312,7 +263,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
                                  kLinearPCMFormatFlagIsPacked;
 
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -320,11 +271,11 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     }
 
     /* setup callback */
-    data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
-    input.inputProc = ca_callback;
-    input.inputProcRefCon = device;
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    input.inputProc = ALCcoreAudioPlayback_MixerProc;
+    input.inputProcRefCon = self;
 
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -332,7 +283,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     }
 
     /* init the default audio unit... */
-    err = AudioUnitInitialize(data->audioUnit);
+    err = AudioUnitInitialize(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioUnitInitialize failed\n");
@@ -342,12 +293,9 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     return ALC_TRUE;
 }
 
-static ALCboolean ca_start_playback(ALCdevice *device)
+static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    OSStatus err;
-
-    err = AudioOutputUnitStart(data->audioUnit);
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioOutputUnitStart failed\n");
@@ -357,59 +305,196 @@ static ALCboolean ca_start_playback(ALCdevice *device)
     return ALC_TRUE;
 }
 
-static void ca_stop_playback(ALCdevice *device)
+static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    OSStatus err;
-
-    err = AudioOutputUnitStop(data->audioUnit);
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
     if(err != noErr)
         ERR("AudioOutputUnitStop failed\n");
 }
 
-static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
+
+
+
+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;
-    ComponentDescription desc;
-    AudioDeviceID inputDevice;
+    AudioComponentDescription desc;
     UInt32 outputFrameCount;
     UInt32 propertySize;
+    AudioObjectPropertyAddress propertyAddress;
     UInt32 enableIO;
-    Component comp;
-    ca_data *data;
+    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 = FindNextComponent(NULL, &desc);
+    comp = AudioComponentFindNext(NULL, &desc);
     if(comp == NULL)
     {
-        ERR("FindNextComponent failed\n");
+        ERR("AudioComponentFindNext failed\n");
         return ALC_INVALID_VALUE;
     }
 
-    data = calloc(1, sizeof(*data));
-    device->ExtraData = data;
-
     // Open the component
-    err = OpenAComponent(comp, &data->audioUnit);
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
     if(err != noErr)
     {
-        ERR("OpenAComponent failed\n");
+        ERR("AudioComponentInstanceNew failed\n");
         goto error;
     }
 
     // Turn off AudioUnit output
     enableIO = 0;
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -418,22 +503,28 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
 
     // Turn on AudioUnit input
     enableIO = 1;
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
+    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);
-    err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice);
+    propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+    propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
+    propertyAddress.mElement = kAudioObjectPropertyElementMaster;
+
+    err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice);
     if(err != noErr)
     {
-        ERR("AudioHardwareGetProperty failed\n");
+        ERR("AudioObjectGetPropertyData failed\n");
         goto error;
     }
-
     if(inputDevice == kAudioDeviceUnknown)
     {
         ERR("No input device found\n");
@@ -441,18 +532,19 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Track the input device
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
+    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 = ca_capture_callback;
-    input.inputProcRefCon = device;
+    input.inputProc = ALCcoreAudioCapture_RecordProc;
+    input.inputProcRefCon = self;
 
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -460,7 +552,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Initialize the device
-    err = AudioUnitInitialize(data->audioUnit);
+    err = AudioUnitInitialize(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioUnitInitialize failed\n");
@@ -469,7 +561,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
 
     // Get the hardware format
     propertySize = sizeof(AudioStreamBasicDescription);
-    err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
     if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
     {
         ERR("AudioUnitGetProperty failed\n");
@@ -513,9 +605,10 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
 
         case DevFmtQuad:
         case DevFmtX51:
-        case DevFmtX51Side:
+        case DevFmtX51Rear:
         case DevFmtX61:
         case DevFmtX71:
+        case DevFmtAmbi3D:
             ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
             goto error;
     }
@@ -528,8 +621,8 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     requestedFormat.mFramesPerPacket = 1;
 
     // save requested format description for later use
-    data->format = requestedFormat;
-    data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
+    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
@@ -537,11 +630,11 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     outputFormat.mSampleRate = hardwareFormat.mSampleRate;
 
     // Determine sample rate ratio for resampling
-    data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
+    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(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -549,8 +642,8 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Set the AudioUnit output format frame count
-    outputFrameCount = device->UpdateSize * data->sampleRateRatio;
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
+    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);
@@ -558,7 +651,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Set up sample converter
-    err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
+    err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter);
     if(err != noErr)
     {
         ERR("AudioConverterNew failed: %d\n", err);
@@ -566,90 +659,83 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Create a buffer for use in the resample callback
-    data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
+    self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio);
 
     // Allocate buffer for the AudioUnit output
-    data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
-    if(data->bufferList == NULL)
+    self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio);
+    if(self->bufferList == NULL)
         goto error;
 
-    data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
-    if(data->ring == 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:
-    DestroyRingBuffer(data->ring);
-    free(data->resampleBuffer);
-    destroy_buffer_list(data->bufferList);
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+    free(self->resampleBuffer);
+    self->resampleBuffer = NULL;
+    destroy_buffer_list(self->bufferList);
+    self->bufferList = NULL;
 
-    if(data->audioConverter)
-        AudioConverterDispose(data->audioConverter);
-    if(data->audioUnit)
-        CloseComponent(data->audioUnit);
-
-    free(data);
-    device->ExtraData = NULL;
+    if(self->audioConverter)
+        AudioConverterDispose(self->audioConverter);
+    self->audioConverter = NULL;
+    if(self->audioUnit)
+        AudioComponentInstanceDispose(self->audioUnit);
+    self->audioUnit = 0;
 
     return ALC_INVALID_VALUE;
 }
 
-static void ca_close_capture(ALCdevice *device)
+
+static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-
-    DestroyRingBuffer(data->ring);
-    free(data->resampleBuffer);
-    destroy_buffer_list(data->bufferList);
-
-    AudioConverterDispose(data->audioConverter);
-    CloseComponent(data->audioUnit);
-
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static void ca_start_capture(ALCdevice *device)
-{
-    ca_data *data = (ca_data*)device->ExtraData;
-    OSStatus err = AudioOutputUnitStart(data->audioUnit);
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
     if(err != noErr)
+    {
         ERR("AudioOutputUnitStart failed\n");
+        return ALC_FALSE;
+    }
+    return ALC_TRUE;
 }
 
-static void ca_stop_capture(ALCdevice *device)
+static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    OSStatus err = AudioOutputUnitStop(data->audioUnit);
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
     if(err != noErr)
         ERR("AudioOutputUnitStop failed\n");
 }
 
-static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
+static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    AudioBufferList *list;
+    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;
-
-    // Allocate a temporary AudioBufferList to use as the return resamples data
-    list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
+    if(samples == 0) return ALC_NO_ERROR;
 
     // Point the resampling buffer to the capture buffer
-    list->mNumberBuffers = 1;
-    list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
-    list->mBuffers[0].mDataByteSize = samples * data->frameSize;
-    list->mBuffers[0].mData = 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(data->audioConverter, ca_capture_conversion_callback,
-                                          device, &frameCount, list, NULL);
+    err = AudioConverterFillComplexBuffer(self->audioConverter,
+        ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL
+    );
     if(err != noErr)
     {
         ERR("AudioConverterFillComplexBuffer error: %d\n", err);
@@ -658,49 +744,73 @@ static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint sa
     return ALC_NO_ERROR;
 }
 
-static ALCuint ca_available_samples(ALCdevice *device)
+static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self)
 {
-    ca_data *data = device->ExtraData;
-    return RingBufferSize(data->ring) / data->sampleRateRatio;
+    return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio;
 }
 
 
-static const BackendFuncs ca_funcs = {
-    ca_open_playback,
-    ca_close_playback,
-    ca_reset_playback,
-    ca_start_playback,
-    ca_stop_playback,
-    ca_open_capture,
-    ca_close_capture,
-    ca_start_capture,
-    ca_stop_capture,
-    ca_capture_samples,
-    ca_available_samples,
-    ALCdevice_LockDefault,
-    ALCdevice_UnlockDefault,
-    ALCdevice_GetLatencyDefault
-};
+typedef struct ALCcoreAudioBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCcoreAudioBackendFactory;
+#define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } }
 
-ALCboolean alc_ca_init(BackendFuncs *func_list)
+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))
 {
-    *func_list = ca_funcs;
     return ALC_TRUE;
 }
 
-void alc_ca_deinit(void)
+static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type)
 {
+    if(type == ALCbackend_Playback || ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
 }
 
-void alc_ca_probe(enum DevProbe type)
+static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
 {
     switch(type)
     {
         case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(ca_device);
-            break;
         case CAPTURE_DEVICE_PROBE:
-            AppendCaptureDeviceList(ca_device);
+            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

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -25,64 +25,104 @@
 #include "alMain.h"
 #include "alu.h"
 
+#include "backends/base.h"
 
-static ALCenum loopback_open_playback(ALCdevice *device, const ALCchar *deviceName)
+
+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)
 {
-    device->DeviceName = strdup(deviceName);
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCloopback, ALCbackend, self);
+}
+
+
+static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    alstr_copy_cstr(&device->DeviceName, name);
     return ALC_NO_ERROR;
 }
 
-static void loopback_close_playback(ALCdevice *device)
+static ALCboolean ALCloopback_reset(ALCloopback *self)
 {
-    (void)device;
-}
-
-static ALCboolean loopback_reset_playback(ALCdevice *device)
-{
-    SetDefaultWFXChannelOrder(device);
+    SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
     return ALC_TRUE;
 }
 
-static ALCboolean loopback_start_playback(ALCdevice *device)
+static ALCboolean ALCloopback_start(ALCloopback* UNUSED(self))
 {
     return ALC_TRUE;
-    (void)device;
 }
 
-static void loopback_stop_playback(ALCdevice *device)
+static void ALCloopback_stop(ALCloopback* UNUSED(self))
 {
-    (void)device;
 }
 
 
-static const BackendFuncs loopback_funcs = {
-    loopback_open_playback,
-    loopback_close_playback,
-    loopback_reset_playback,
-    loopback_start_playback,
-    loopback_stop_playback,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    ALCdevice_LockDefault,
-    ALCdevice_UnlockDefault,
-    ALCdevice_GetLatencyDefault
-};
+typedef struct ALCloopbackFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCloopbackFactory;
+#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCloopbackFactory, ALCbackendFactory) } }
 
-ALCboolean alc_loopback_init(BackendFuncs *func_list)
+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))
 {
-    *func_list = loopback_funcs;
     return ALC_TRUE;
 }
 
-void alc_loopback_deinit(void)
+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))
 {
 }
 
-void alc_loopback_probe(enum DevProbe type)
+static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
 {
-    (void)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

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -27,146 +27,194 @@
 
 #include "alMain.h"
 #include "alu.h"
+#include "threads.h"
+#include "compat.h"
+
+#include "backends/base.h"
 
 
-typedef struct {
-    volatile int killNow;
-    ALvoid *thread;
-} null_data;
+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 ALuint NullProc(ALvoid *ptr)
+
+static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device)
 {
-    ALCdevice *Device = (ALCdevice*)ptr;
-    null_data *data = (null_data*)Device->ExtraData;
-    ALuint now, start;
+    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 ALuint restTime = (ALuint64)Device->UpdateSize * 1000 /
-                            Device->Frequency / 2;
+    const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 /
+                                 device->Frequency / 2);
+
+    SetRTPriority();
+    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
 
     done = 0;
-    start = timeGetTime();
-    while(!data->killNow && Device->Connected)
+    if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
     {
-        now = timeGetTime();
+        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 = (ALuint64)(now-start) * Device->Frequency / 1000;
+        avail  = (now.tv_sec - start.tv_sec) * device->Frequency;
+        avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
         if(avail < done)
         {
-            /* Timer wrapped (50 days???). Add the remainder of the cycle to
-             * the available count and reset the number of samples done */
-            avail += ((ALuint64)1<<32)*Device->Frequency/1000 - done;
-            done = 0;
-        }
-        if(avail-done < Device->UpdateSize)
-        {
-            Sleep(restTime);
-            continue;
+            /* 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;
         }
 
-        while(avail-done >= Device->UpdateSize)
+        if(avail-done < device->UpdateSize)
+            al_nssleep(restTime);
+        else while(avail-done >= device->UpdateSize)
         {
-            aluMixData(Device, NULL, Device->UpdateSize);
-            done += Device->UpdateSize;
+            ALCnullBackend_lock(self);
+            aluMixData(device, NULL, device->UpdateSize);
+            ALCnullBackend_unlock(self);
+            done += device->UpdateSize;
         }
     }
 
     return 0;
 }
 
-static ALCenum null_open_playback(ALCdevice *device, const ALCchar *deviceName)
-{
-    null_data *data;
 
-    if(!deviceName)
-        deviceName = nullDevice;
-    else if(strcmp(deviceName, nullDevice) != 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;
 
-    data = (null_data*)calloc(1, sizeof(*data));
+    device = STATIC_CAST(ALCbackend, self)->mDevice;
+    alstr_copy_cstr(&device->DeviceName, name);
 
-    device->DeviceName = strdup(deviceName);
-    device->ExtraData = data;
     return ALC_NO_ERROR;
 }
 
-static void null_close_playback(ALCdevice *device)
+static ALCboolean ALCnullBackend_reset(ALCnullBackend *self)
 {
-    null_data *data = (null_data*)device->ExtraData;
-
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean null_reset_playback(ALCdevice *device)
-{
-    SetDefaultWFXChannelOrder(device);
+    SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
     return ALC_TRUE;
 }
 
-static ALCboolean null_start_playback(ALCdevice *device)
+static ALCboolean ALCnullBackend_start(ALCnullBackend *self)
 {
-    null_data *data = (null_data*)device->ExtraData;
-
-    data->thread = StartThread(NullProc, device);
-    if(data->thread == NULL)
+    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 null_stop_playback(ALCdevice *device)
+static void ALCnullBackend_stop(ALCnullBackend *self)
 {
-    null_data *data = (null_data*)device->ExtraData;
+    int res;
 
-    if(!data->thread)
+    if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
         return;
-
-    data->killNow = 1;
-    StopThread(data->thread);
-    data->thread = NULL;
-
-    data->killNow = 0;
+    althrd_join(self->thread, &res);
 }
 
 
-static const BackendFuncs null_funcs = {
-    null_open_playback,
-    null_close_playback,
-    null_reset_playback,
-    null_start_playback,
-    null_stop_playback,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    ALCdevice_LockDefault,
-    ALCdevice_UnlockDefault,
-    ALCdevice_GetLatencyDefault
-};
+typedef struct ALCnullBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCnullBackendFactory;
+#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCnullBackendFactory, ALCbackendFactory) } }
 
-ALCboolean alc_null_init(BackendFuncs *func_list)
+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))
 {
-    *func_list = null_funcs;
     return ALC_TRUE;
 }
 
-void alc_null_deinit(void)
+static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory* UNUSED(self), ALCbackend_Type type)
 {
+    if(type == ALCbackend_Playback)
+        return ALC_TRUE;
+    return ALC_FALSE;
 }
 
-void alc_null_probe(enum DevProbe type)
+static void ALCnullBackendFactory_probe(ALCnullBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
 {
     switch(type)
     {
         case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(nullDevice);
-            break;
         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/portaudio.c

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -26,6 +26,11 @@
 
 #include "alMain.h"
 #include "alu.h"
+#include "alconfig.h"
+#include "ringbuffer.h"
+#include "compat.h"
+
+#include "backends/base.h"
 
 #include <portaudio.h>
 
@@ -35,7 +40,7 @@ static const ALCchar pa_device[] = "PortAudio Default";
 
 #ifdef HAVE_DYNLOAD
 static void *pa_handle;
-#define MAKE_FUNC(x) static typeof(x) * p##x
+#define MAKE_FUNC(x) static __typeof(x) * p##x
 MAKE_FUNC(Pa_Initialize);
 MAKE_FUNC(Pa_Terminate);
 MAKE_FUNC(Pa_GetErrorText);
@@ -44,6 +49,7 @@ 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
 
@@ -55,6 +61,7 @@ MAKE_FUNC(Pa_GetStreamInfo);
 #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
 
@@ -96,6 +103,7 @@ static ALCboolean pa_load(void)
         LOAD_FUNC(Pa_OpenStream);
         LOAD_FUNC(Pa_CloseStream);
         LOAD_FUNC(Pa_GetDefaultOutputDevice);
+        LOAD_FUNC(Pa_GetDefaultInputDevice);
         LOAD_FUNC(Pa_GetStreamInfo);
 #undef LOAD_FUNC
 
@@ -118,157 +126,165 @@ static ALCboolean pa_load(void)
 }
 
 
-typedef struct {
+typedef struct ALCportPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
+
     PaStream *stream;
     PaStreamParameters params;
     ALuint update_size;
+} ALCportPlayback;
 
-    RingBuffer *ring;
-} pa_data;
+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 int pa_callback(const void *inputBuffer, void *outputBuffer,
-                       unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
-                       const PaStreamCallbackFlags statusFlags, void *userData)
+static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device)
 {
-    ALCdevice *device = (ALCdevice*)userData;
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCportPlayback, ALCbackend, self);
 
-    (void)inputBuffer;
-    (void)timeInfo;
-    (void)statusFlags;
-
-    aluMixData(device, outputBuffer, framesPerBuffer);
-    return 0;
+    self->stream = NULL;
 }
 
-static int pa_capture_cb(const void *inputBuffer, void *outputBuffer,
-                         unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
-                         const PaStreamCallbackFlags statusFlags, void *userData)
+static void ALCportPlayback_Destruct(ALCportPlayback *self)
 {
-    ALCdevice *device = (ALCdevice*)userData;
-    pa_data *data = (pa_data*)device->ExtraData;
+    PaError err = self->stream ? Pa_CloseStream(self->stream) : paNoError;
+    if(err != paNoError)
+        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
+    self->stream = NULL;
 
-    (void)outputBuffer;
-    (void)timeInfo;
-    (void)statusFlags;
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
 
-    WriteRingBuffer(data->ring, inputBuffer, framesPerBuffer);
+
+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 pa_open_playback(ALCdevice *device, const ALCchar *deviceName)
+static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name)
 {
-    pa_data *data;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
     PaError err;
 
-    if(!deviceName)
-        deviceName = pa_device;
-    else if(strcmp(deviceName, pa_device) != 0)
+    if(!name)
+        name = pa_device;
+    else if(strcmp(name, pa_device) != 0)
         return ALC_INVALID_VALUE;
 
-    data = (pa_data*)calloc(1, sizeof(pa_data));
-    data->update_size = device->UpdateSize;
+    self->update_size = device->UpdateSize;
 
-    data->params.device = -1;
-    if(!ConfigValueInt("port", "device", &data->params.device) ||
-       data->params.device < 0)
-        data->params.device = Pa_GetDefaultOutputDevice();
-    data->params.suggestedLatency = (device->UpdateSize*device->NumUpdates) /
+    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;
-    data->params.hostApiSpecificStreamInfo = NULL;
+    self->params.hostApiSpecificStreamInfo = NULL;
 
-    data->params.channelCount = ((device->FmtChans == DevFmtMono) ? 1 : 2);
+    self->params.channelCount = ((device->FmtChans == DevFmtMono) ? 1 : 2);
 
     switch(device->FmtType)
     {
         case DevFmtByte:
-            data->params.sampleFormat = paInt8;
+            self->params.sampleFormat = paInt8;
             break;
         case DevFmtUByte:
-            data->params.sampleFormat = paUInt8;
+            self->params.sampleFormat = paUInt8;
             break;
         case DevFmtUShort:
             /* fall-through */
         case DevFmtShort:
-            data->params.sampleFormat = paInt16;
+            self->params.sampleFormat = paInt16;
             break;
         case DevFmtUInt:
             /* fall-through */
         case DevFmtInt:
-            data->params.sampleFormat = paInt32;
+            self->params.sampleFormat = paInt32;
             break;
         case DevFmtFloat:
-            data->params.sampleFormat = paFloat32;
+            self->params.sampleFormat = paFloat32;
             break;
     }
 
 retry_open:
-    err = Pa_OpenStream(&data->stream, NULL, &data->params, device->Frequency,
-                        device->UpdateSize, paNoFlag, pa_callback, device);
+    err = Pa_OpenStream(&self->stream, NULL, &self->params,
+        device->Frequency, device->UpdateSize, paNoFlag,
+        ALCportPlayback_WriteCallback, self
+    );
     if(err != paNoError)
     {
-        if(data->params.sampleFormat == paFloat32)
+        if(self->params.sampleFormat == paFloat32)
         {
-            data->params.sampleFormat = paInt16;
+            self->params.sampleFormat = paInt16;
             goto retry_open;
         }
         ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
-        free(data);
         return ALC_INVALID_VALUE;
     }
 
-    device->ExtraData = data;
-    device->DeviceName = strdup(deviceName);
+    alstr_copy_cstr(&device->DeviceName, name);
 
     return ALC_NO_ERROR;
+
 }
 
-static void pa_close_playback(ALCdevice *device)
+static ALCboolean ALCportPlayback_reset(ALCportPlayback *self)
 {
-    pa_data *data = (pa_data*)device->ExtraData;
-    PaError err;
-
-    err = Pa_CloseStream(data->stream);
-    if(err != paNoError)
-        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
-
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean pa_reset_playback(ALCdevice *device)
-{
-    pa_data *data = (pa_data*)device->ExtraData;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
     const PaStreamInfo *streamInfo;
 
-    streamInfo = Pa_GetStreamInfo(data->stream);
+    streamInfo = Pa_GetStreamInfo(self->stream);
     device->Frequency = streamInfo->sampleRate;
-    device->UpdateSize = data->update_size;
+    device->UpdateSize = self->update_size;
 
-    if(data->params.sampleFormat == paInt8)
+    if(self->params.sampleFormat == paInt8)
         device->FmtType = DevFmtByte;
-    else if(data->params.sampleFormat == paUInt8)
+    else if(self->params.sampleFormat == paUInt8)
         device->FmtType = DevFmtUByte;
-    else if(data->params.sampleFormat == paInt16)
+    else if(self->params.sampleFormat == paInt16)
         device->FmtType = DevFmtShort;
-    else if(data->params.sampleFormat == paInt32)
+    else if(self->params.sampleFormat == paInt32)
         device->FmtType = DevFmtInt;
-    else if(data->params.sampleFormat == paFloat32)
+    else if(self->params.sampleFormat == paFloat32)
         device->FmtType = DevFmtFloat;
     else
     {
-        ERR("Unexpected sample format: 0x%lx\n", data->params.sampleFormat);
+        ERR("Unexpected sample format: 0x%lx\n", self->params.sampleFormat);
         return ALC_FALSE;
     }
 
-    if(data->params.channelCount == 2)
+    if(self->params.channelCount == 2)
         device->FmtChans = DevFmtStereo;
-    else if(data->params.channelCount == 1)
+    else if(self->params.channelCount == 1)
         device->FmtChans = DevFmtMono;
     else
     {
-        ERR("Unexpected channel count: %u\n", data->params.channelCount);
+        ERR("Unexpected channel count: %u\n", self->params.channelCount);
         return ALC_FALSE;
     }
     SetDefaultChannelOrder(device);
@@ -276,12 +292,11 @@ static ALCboolean pa_reset_playback(ALCdevice *device)
     return ALC_TRUE;
 }
 
-static ALCboolean pa_start_playback(ALCdevice *device)
+static ALCboolean ALCportPlayback_start(ALCportPlayback *self)
 {
-    pa_data *data = (pa_data*)device->ExtraData;
     PaError err;
 
-    err = Pa_StartStream(data->stream);
+    err = Pa_StartStream(self->stream);
     if(err != paNoError)
     {
         ERR("Pa_StartStream() returned an error: %s\n", Pa_GetErrorText(err));
@@ -291,160 +306,197 @@ static ALCboolean pa_start_playback(ALCdevice *device)
     return ALC_TRUE;
 }
 
-static void pa_stop_playback(ALCdevice *device)
+static void ALCportPlayback_stop(ALCportPlayback *self)
 {
-    pa_data *data = (pa_data*)device->ExtraData;
-    PaError err;
-
-    err = Pa_StopStream(data->stream);
+    PaError err = Pa_StopStream(self->stream);
     if(err != paNoError)
         ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
 }
 
 
-static ALCenum pa_open_capture(ALCdevice *device, const ALCchar *deviceName)
+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)
 {
-    ALuint frame_size;
-    pa_data *data;
+    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(!deviceName)
-        deviceName = pa_device;
-    else if(strcmp(deviceName, pa_device) != 0)
+    if(!name)
+        name = pa_device;
+    else if(strcmp(name, pa_device) != 0)
         return ALC_INVALID_VALUE;
 
-    data = (pa_data*)calloc(1, sizeof(pa_data));
-    if(data == NULL)
-        return ALC_OUT_OF_MEMORY;
+    samples = device->UpdateSize * device->NumUpdates;
+    samples = maxu(samples, 100 * device->Frequency / 1000);
+    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
 
-    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
-    data->ring = CreateRingBuffer(frame_size, device->UpdateSize*device->NumUpdates);
-    if(data->ring == NULL)
-        goto error;
+    self->ring = ll_ringbuffer_create(samples, frame_size, false);
+    if(self->ring == NULL) return ALC_INVALID_VALUE;
 
-    data->params.device = -1;
-    if(!ConfigValueInt("port", "capture", &data->params.device) ||
-       data->params.device < 0)
-        data->params.device = Pa_GetDefaultOutputDevice();
-    data->params.suggestedLatency = 0.0f;
-    data->params.hostApiSpecificStreamInfo = NULL;
+    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:
-            data->params.sampleFormat = paInt8;
+            self->params.sampleFormat = paInt8;
             break;
         case DevFmtUByte:
-            data->params.sampleFormat = paUInt8;
+            self->params.sampleFormat = paUInt8;
             break;
         case DevFmtShort:
-            data->params.sampleFormat = paInt16;
+            self->params.sampleFormat = paInt16;
             break;
         case DevFmtInt:
-            data->params.sampleFormat = paInt32;
+            self->params.sampleFormat = paInt32;
             break;
         case DevFmtFloat:
-            data->params.sampleFormat = paFloat32;
+            self->params.sampleFormat = paFloat32;
             break;
         case DevFmtUInt:
         case DevFmtUShort:
             ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
-            goto error;
+            return ALC_INVALID_VALUE;
     }
-    data->params.channelCount = ChannelsFromDevFmt(device->FmtChans);
+    self->params.channelCount = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
 
-    err = Pa_OpenStream(&data->stream, &data->params, NULL, device->Frequency,
-                        paFramesPerBufferUnspecified, paNoFlag, pa_capture_cb, device);
+    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));
-        goto error;
+        return ALC_INVALID_VALUE;
     }
 
-    device->DeviceName = strdup(deviceName);
+    alstr_copy_cstr(&device->DeviceName, name);
 
-    device->ExtraData = data;
     return ALC_NO_ERROR;
-
-error:
-    DestroyRingBuffer(data->ring);
-    free(data);
-    return ALC_INVALID_VALUE;
 }
 
-static void pa_close_capture(ALCdevice *device)
+
+static ALCboolean ALCportCapture_start(ALCportCapture *self)
 {
-    pa_data *data = (pa_data*)device->ExtraData;
-    PaError err;
-
-    err = Pa_CloseStream(data->stream);
-    if(err != paNoError)
-        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
-
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static void pa_start_capture(ALCdevice *device)
-{
-    pa_data *data = device->ExtraData;
-    PaError err;
-
-    err = Pa_StartStream(data->stream);
+    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 pa_stop_capture(ALCdevice *device)
+static void ALCportCapture_stop(ALCportCapture *self)
 {
-    pa_data *data = (pa_data*)device->ExtraData;
-    PaError err;
-
-    err = Pa_StopStream(data->stream);
+    PaError err = Pa_StopStream(self->stream);
     if(err != paNoError)
         ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
 }
 
-static ALCenum pa_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
+
+static ALCuint ALCportCapture_availableSamples(ALCportCapture *self)
 {
-    pa_data *data = device->ExtraData;
-    ReadRingBuffer(data->ring, buffer, samples);
+    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;
 }
 
-static ALCuint pa_available_samples(ALCdevice *device)
-{
-    pa_data *data = device->ExtraData;
-    return RingBufferSize(data->ring);
-}
+
+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 const BackendFuncs pa_funcs = {
-    pa_open_playback,
-    pa_close_playback,
-    pa_reset_playback,
-    pa_start_playback,
-    pa_stop_playback,
-    pa_open_capture,
-    pa_close_capture,
-    pa_start_capture,
-    pa_stop_capture,
-    pa_capture_samples,
-    pa_available_samples,
-    ALCdevice_LockDefault,
-    ALCdevice_UnlockDefault,
-    ALCdevice_GetLatencyDefault
-};
-
-ALCboolean alc_pa_init(BackendFuncs *func_list)
+static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory* UNUSED(self))
 {
     if(!pa_load())
         return ALC_FALSE;
-    *func_list = pa_funcs;
     return ALC_TRUE;
 }
 
-void alc_pa_deinit(void)
+static void ALCportBackendFactory_deinit(ALCportBackendFactory* UNUSED(self))
 {
 #ifdef HAVE_DYNLOAD
     if(pa_handle)
@@ -458,15 +510,46 @@ void alc_pa_deinit(void)
 #endif
 }
 
-void alc_pa_probe(enum DevProbe type)
+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:
-            AppendAllDevicesList(pa_device);
-            break;
         case CAPTURE_DEVICE_PROBE:
-            AppendCaptureDeviceList(pa_device);
+            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

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -26,6 +26,10 @@
 
 #include "alMain.h"
 #include "alu.h"
+#include "threads.h"
+#include "ringbuffer.h"
+
+#include "backends/base.h"
 
 #include <sndio.h>
 
@@ -33,48 +37,88 @@
 static const ALCchar sndio_device[] = "SndIO Default";
 
 
-static ALCboolean sndio_load(void)
-{
-    return ALC_TRUE;
-}
+typedef struct SndioPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
 
-
-typedef struct {
     struct sio_hdl *sndHandle;
 
     ALvoid *mix_data;
     ALsizei data_size;
 
-    volatile int killNow;
-    ALvoid *thread;
-} sndio_data;
+    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 ALuint sndio_proc(ALvoid *ptr)
+static void SndioPlayback_Construct(SndioPlayback *self, ALCdevice *device)
 {
-    ALCdevice *device = ptr;
-    sndio_data *data = device->ExtraData;
+    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);
+    frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
 
-    while(!data->killNow && device->Connected)
+    while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
     {
-        ALsizei len = data->data_size;
-        ALubyte *WritePtr = data->mix_data;
+        ALsizei len = self->data_size;
+        ALubyte *WritePtr = self->mix_data;
 
+        SndioPlayback_lock(self);
         aluMixData(device, WritePtr, len/frameSize);
-        while(len > 0 && !data->killNow)
+        SndioPlayback_unlock(self);
+        while(len > 0 && !ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
         {
-            wrote = sio_write(data->sndHandle, WritePtr, len);
+            wrote = sio_write(self->sndHandle, WritePtr, len);
             if(wrote == 0)
             {
                 ERR("sio_write failed\n");
                 ALCdevice_Lock(device);
-                aluHandleDisconnect(device);
+                aluHandleDisconnect(device, "Failed to write playback samples");
                 ALCdevice_Unlock(device);
                 break;
             }
@@ -88,45 +132,30 @@ static ALuint sndio_proc(ALvoid *ptr)
 }
 
 
-
-static ALCenum sndio_open_playback(ALCdevice *device, const ALCchar *deviceName)
+static ALCenum SndioPlayback_open(SndioPlayback *self, const ALCchar *name)
 {
-    sndio_data *data;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
 
-    if(!deviceName)
-        deviceName = sndio_device;
-    else if(strcmp(deviceName, sndio_device) != 0)
+    if(!name)
+        name = sndio_device;
+    else if(strcmp(name, sndio_device) != 0)
         return ALC_INVALID_VALUE;
 
-    data = calloc(1, sizeof(*data));
-    data->killNow = 0;
-
-    data->sndHandle = sio_open(NULL, SIO_PLAY, 0);
-    if(data->sndHandle == NULL)
+    self->sndHandle = sio_open(NULL, SIO_PLAY, 0);
+    if(self->sndHandle == NULL)
     {
-        free(data);
         ERR("Could not open device\n");
         return ALC_INVALID_VALUE;
     }
 
-    device->DeviceName = strdup(deviceName);
-    device->ExtraData = data;
+    alstr_copy_cstr(&device->DeviceName, name);
 
     return ALC_NO_ERROR;
 }
 
-static void sndio_close_playback(ALCdevice *device)
+static ALCboolean SndioPlayback_reset(SndioPlayback *self)
 {
-    sndio_data *data = device->ExtraData;
-
-    sio_close(data->sndHandle);
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean sndio_reset_playback(ALCdevice *device)
-{
-    sndio_data *data = device->ExtraData;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
     struct sio_par par;
 
     sio_initpar(&par);
@@ -168,7 +197,7 @@ static ALCboolean sndio_reset_playback(ALCdevice *device)
     par.appbufsz = device->UpdateSize * (device->NumUpdates-1);
     if(!par.appbufsz) par.appbufsz = device->UpdateSize;
 
-    if(!sio_setpar(data->sndHandle, &par) || !sio_getpar(data->sndHandle, &par))
+    if(!sio_setpar(self->sndHandle, &par) || !sio_getpar(self->sndHandle, &par))
     {
         ERR("Failed to set device parameters\n");
         return ALC_FALSE;
@@ -209,88 +238,363 @@ static ALCboolean sndio_reset_playback(ALCdevice *device)
     return ALC_TRUE;
 }
 
-static ALCboolean sndio_start_playback(ALCdevice *device)
+static ALCboolean SndioPlayback_start(SndioPlayback *self)
 {
-    sndio_data *data = device->ExtraData;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
 
-    if(!sio_start(data->sndHandle))
+    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;
     }
 
-    data->data_size = device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
-    data->mix_data = calloc(1, data->data_size);
-
-    data->thread = StartThread(sndio_proc, device);
-    if(data->thread == NULL)
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    if(althrd_create(&self->thread, SndioPlayback_mixerProc, self) != althrd_success)
     {
-        sio_stop(data->sndHandle);
-        free(data->mix_data);
-        data->mix_data = NULL;
+        sio_stop(self->sndHandle);
         return ALC_FALSE;
     }
 
     return ALC_TRUE;
 }
 
-static void sndio_stop_playback(ALCdevice *device)
+static void SndioPlayback_stop(SndioPlayback *self)
 {
-    sndio_data *data = device->ExtraData;
+    int res;
 
-    if(!data->thread)
+    if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
         return;
+    althrd_join(self->thread, &res);
 
-    data->killNow = 1;
-    StopThread(data->thread);
-    data->thread = NULL;
-
-    data->killNow = 0;
-    if(!sio_stop(data->sndHandle))
+    if(!sio_stop(self->sndHandle))
         ERR("Error stopping device\n");
 
-    free(data->mix_data);
-    data->mix_data = NULL;
+    al_free(self->mix_data);
+    self->mix_data = NULL;
 }
 
 
-static const BackendFuncs sndio_funcs = {
-    sndio_open_playback,
-    sndio_close_playback,
-    sndio_reset_playback,
-    sndio_start_playback,
-    sndio_stop_playback,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    ALCdevice_LockDefault,
-    ALCdevice_UnlockDefault,
-    ALCdevice_GetLatencyDefault
-};
+typedef struct SndioCapture {
+    DERIVE_FROM_TYPE(ALCbackend);
 
-ALCboolean alc_sndio_init(BackendFuncs *func_list)
+    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)
 {
-    if(!sndio_load())
+    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;
-    *func_list = sndio_funcs;
+    }
+
+    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;
 }
 
-void alc_sndio_deinit(void)
+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");
 }
 
-void alc_sndio_probe(enum DevProbe type)
+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:
-            AppendAllDevicesList(sndio_device);
-            break;
         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

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -22,6 +22,7 @@
 
 #include <sys/ioctl.h>
 #include <sys/types.h>
+#include <sys/time.h>
 #include <sys/stat.h>
 #include <fcntl.h>
 #include <stdlib.h>
@@ -33,108 +34,171 @@
 
 #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";
 
-typedef struct {
-    int fd;
-    volatile int killNow;
-    ALvoid *thread;
 
-    ALubyte *mix_data;
-    int data_size;
-} solaris_data;
-
-
-static ALuint SolarisProc(ALvoid *ptr)
+static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device)
 {
-    ALCdevice *Device = (ALCdevice*)ptr;
-    solaris_data *data = (solaris_data*)Device->ExtraData;
-    ALint frameSize;
-    int wrote;
+    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);
 
-    frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
+    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
 
-    while(!data->killNow && Device->Connected)
+    ALCsolarisBackend_lock(self);
+    while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
+          ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
     {
-        ALint len = data->data_size;
-        ALubyte *WritePtr = data->mix_data;
+        FD_ZERO(&wfds);
+        FD_SET(self->fd, &wfds);
+        timeout.tv_sec = 1;
+        timeout.tv_usec = 0;
 
-        aluMixData(Device, WritePtr, len/frameSize);
-        while(len > 0 && !data->killNow)
+        ALCsolarisBackend_unlock(self);
+        sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
+        ALCsolarisBackend_lock(self);
+        if(sret < 0)
         {
-            wrote = write(data->fd, WritePtr, len);
+            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)
-                {
-                    ERR("write failed: %s\n", strerror(errno));
-                    ALCdevice_Lock(Device);
-                    aluHandleDisconnect(Device);
-                    ALCdevice_Unlock(Device);
-                    break;
-                }
-
-                Sleep(1);
-                continue;
+                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;
             }
 
-            len -= wrote;
-            WritePtr += wrote;
+            to_write -= wrote;
+            write_ptr += wrote;
         }
     }
+    ALCsolarisBackend_unlock(self);
 
     return 0;
 }
 
 
-static ALCenum solaris_open_playback(ALCdevice *device, const ALCchar *deviceName)
+static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name)
 {
-    solaris_data *data;
+    ALCdevice *device;
 
-    if(!deviceName)
-        deviceName = solaris_device;
-    else if(strcmp(deviceName, solaris_device) != 0)
+    if(!name)
+        name = solaris_device;
+    else if(strcmp(name, solaris_device) != 0)
         return ALC_INVALID_VALUE;
 
-    data = (solaris_data*)calloc(1, sizeof(solaris_data));
-    data->killNow = 0;
-
-    data->fd = open(solaris_driver, O_WRONLY);
-    if(data->fd == -1)
+    self->fd = open(solaris_driver, O_WRONLY);
+    if(self->fd == -1)
     {
-        free(data);
         ERR("Could not open %s: %s\n", solaris_driver, strerror(errno));
         return ALC_INVALID_VALUE;
     }
 
-    device->DeviceName = strdup(deviceName);
-    device->ExtraData = data;
+    device = STATIC_CAST(ALCbackend,self)->mDevice;
+    alstr_copy_cstr(&device->DeviceName, name);
+
     return ALC_NO_ERROR;
 }
 
-static void solaris_close_playback(ALCdevice *device)
+static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self)
 {
-    solaris_data *data = (solaris_data*)device->ExtraData;
-
-    close(data->fd);
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean solaris_reset_playback(ALCdevice *device)
-{
-    solaris_data *data = (solaris_data*)device->ExtraData;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
     audio_info_t info;
-    ALuint frameSize;
-    int numChannels;
+    ALsizei frameSize;
+    ALsizei numChannels;
 
     AUDIO_INITINFO(&info);
 
@@ -142,7 +206,7 @@ static ALCboolean solaris_reset_playback(ALCdevice *device)
 
     if(device->FmtChans != DevFmtMono)
         device->FmtChans = DevFmtStereo;
-    numChannels = ChannelsFromDevFmt(device->FmtChans);
+    numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
     info.play.channels = numChannels;
 
     switch(device->FmtType)
@@ -170,15 +234,15 @@ static ALCboolean solaris_reset_playback(ALCdevice *device)
     frameSize = numChannels * BytesFromDevFmt(device->FmtType);
     info.play.buffer_size = device->UpdateSize*device->NumUpdates * frameSize;
 
-    if(ioctl(data->fd, AUDIO_SETINFO, &info) < 0)
+    if(ioctl(self->fd, AUDIO_SETINFO, &info) < 0)
     {
         ERR("ioctl failed: %s\n", strerror(errno));
         return ALC_FALSE;
     }
 
-    if(ChannelsFromDevFmt(device->FmtChans) != info.play.channels)
+    if(ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)info.play.channels)
     {
-        ERR("Could not set %d channels, got %d instead\n", ChannelsFromDevFmt(device->FmtChans), info.play.channels);
+        ERR("Failed to set %s, got %u channels instead\n", DevFmtChannelsString(device->FmtChans), info.play.channels);
         return ALC_FALSE;
     }
 
@@ -197,77 +261,73 @@ static ALCboolean solaris_reset_playback(ALCdevice *device)
 
     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 solaris_start_playback(ALCdevice *device)
+static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self)
 {
-    solaris_data *data = (solaris_data*)device->ExtraData;
-
-    data->data_size = device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
-    data->mix_data = calloc(1, data->data_size);
-
-    data->thread = StartThread(SolarisProc, device);
-    if(data->thread == NULL)
-    {
-        free(data->mix_data);
-        data->mix_data = NULL;
+    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 solaris_stop_playback(ALCdevice *device)
+static void ALCsolarisBackend_stop(ALCsolarisBackend *self)
 {
-    solaris_data *data = (solaris_data*)device->ExtraData;
+    int res;
 
-    if(!data->thread)
+    if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
         return;
 
-    data->killNow = 1;
-    StopThread(data->thread);
-    data->thread = NULL;
+    althrd_join(self->thread, &res);
 
-    data->killNow = 0;
-    if(ioctl(data->fd, AUDIO_DRAIN) < 0)
+    if(ioctl(self->fd, AUDIO_DRAIN) < 0)
         ERR("Error draining device: %s\n", strerror(errno));
-
-    free(data->mix_data);
-    data->mix_data = NULL;
 }
 
 
-static const BackendFuncs solaris_funcs = {
-    solaris_open_playback,
-    solaris_close_playback,
-    solaris_reset_playback,
-    solaris_start_playback,
-    solaris_stop_playback,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    ALCdevice_LockDefault,
-    ALCdevice_UnlockDefault,
-    ALCdevice_GetLatencyDefault
-};
+typedef struct ALCsolarisBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCsolarisBackendFactory;
+#define ALCSOLARISBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsolarisBackendFactory, ALCbackendFactory) } }
 
-ALCboolean alc_solaris_init(BackendFuncs *func_list)
+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)
 {
-    ConfigValueStr("solaris", "device", &solaris_driver);
+    static ALCsolarisBackendFactory factory = ALCSOLARISBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
 
-    *func_list = solaris_funcs;
+
+static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory* UNUSED(self))
+{
+    ConfigValueStr(NULL, "solaris", "device", &solaris_driver);
     return ALC_TRUE;
 }
 
-void alc_solaris_deinit(void)
+static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory* UNUSED(self), ALCbackend_Type type)
 {
+    if(type == ALCbackend_Playback)
+        return ALC_TRUE;
+    return ALC_FALSE;
 }
 
-void alc_solaris_probe(enum DevProbe type)
+static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
 {
     switch(type)
     {
@@ -277,7 +337,7 @@ void alc_solaris_probe(enum DevProbe type)
             struct stat buf;
             if(stat(solaris_driver, &buf) == 0)
 #endif
-                AppendAllDevicesList(solaris_device);
+                alstr_append_range(outnames, solaris_device, solaris_device+sizeof(solaris_device));
         }
         break;
 
@@ -285,3 +345,16 @@ void alc_solaris_probe(enum DevProbe type)
             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/vita.c

@@ -0,0 +1,513 @@
+/**
+ * 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 <psp2/audioin.h>
+#include <psp2/audioout.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "ringbuffer.h"
+#include "threads.h"
+
+#include "backends/base.h"
+
+#define AUDIO_SAMPLE_ALIGN(s)   (((s) + 63) & ~63)
+
+static const ALCchar playbackDeviceName[] = "PS Vita Speakers/Headphones";
+static const ALCchar captureDeviceName[] = "PS Vita Microphone";
+
+extern unsigned int _oal_thread_priority __attribute__((weak));
+extern unsigned int _oal_thread_affinity __attribute__((weak));
+
+// -----------------------------------------------------------------------------
+// Playback
+// -----------------------------------------------------------------------------
+typedef struct ALCvitaPlayback
+{
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    ATOMIC(int) killNow;
+    SceUID thread;
+    SceKernelLwMutexWork lock;
+
+    int portNumber;
+    ALsizei frameSize;
+    void* waveBuffer;
+
+    ALuint Frequency;
+    enum DevFmtChannels FmtChans;
+    enum DevFmtType     FmtType;
+    ALuint UpdateSize;
+} ALCvitaPlayback;
+
+static void ALCvitaPlayback_Construct(ALCvitaPlayback *self, ALCdevice *device);
+static void ALCvitaPlayback_Destruct(ALCvitaPlayback *self);
+static ALCenum ALCvitaPlayback_open(ALCvitaPlayback *self, const ALCchar *name);
+static ALCboolean ALCvitaPlayback_reset(ALCvitaPlayback *self);
+static ALCboolean ALCvitaPlayback_start(ALCvitaPlayback *self);
+static void ALCvitaPlayback_stop(ALCvitaPlayback *self);
+static void ALCvitaPlayback_lock(ALCvitaPlayback *self);
+static void ALCvitaPlayback_unlock(ALCvitaPlayback *self);
+
+static DECLARE_FORWARD2(ALCvitaPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCvitaPlayback, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCvitaPlayback, ALCbackend, ClockLatency, getClockLatency)
+DECLARE_DEFAULT_ALLOCATORS(ALCvitaPlayback)
+
+DEFINE_ALCBACKEND_VTABLE(ALCvitaPlayback);
+
+static void ALCvitaPlayback_Construct(ALCvitaPlayback *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCvitaPlayback, ALCbackend, self);
+
+    device->UpdateSize = AUDIO_SAMPLE_ALIGN(device->UpdateSize);
+
+    self->portNumber = 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;
+
+    sceKernelCreateLwMutex(
+        &self->lock,
+        "OpenAL Vita playback mutex",
+        SCE_KERNEL_MUTEX_ATTR_RECURSIVE, // No SCE_KERNEL_LW_MUTEX_ATTR_RECURSIVE in VitaSDK, but it's the same
+        0,
+        NULL
+    );
+}
+
+static void ALCvitaPlayback_Destruct(ALCvitaPlayback *self)
+{
+    if (self->portNumber)
+    {
+        sceAudioOutReleasePort(self->portNumber);
+        self->portNumber = 0;
+    }
+
+    if (self->waveBuffer)
+    {
+        free(self->waveBuffer);
+        self->waveBuffer = NULL;
+    }
+
+    sceKernelDeleteLwMutex(&self->lock);
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+static int ALCvitaPlayback_MixerProc(SceSize args, void *argp)
+{
+    (void)args;
+    ALCvitaPlayback *self = *(ALCvitaPlayback **) argp;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    while (!ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
+    {
+        ALCvitaPlayback_lock(self);
+        aluMixData(device, self->waveBuffer, device->UpdateSize);
+        ALCvitaPlayback_unlock(self);
+        sceAudioOutOutput(self->portNumber, self->waveBuffer);
+    }
+
+    return 0;
+}
+
+static ALCenum ALCvitaPlayback_open(ALCvitaPlayback *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    /* Only signed short output sample format is supported */
+    device->FmtType = DevFmtShort;
+
+    /* Only mono/stereo channel configurations are supported */
+    if (device->FmtChans != DevFmtMono && device->FmtChans != DevFmtStereo)
+        device->FmtChans = DevFmtStereo;
+
+    device->UpdateSize = AUDIO_SAMPLE_ALIGN(device->UpdateSize);
+
+    self->portNumber = sceAudioOutOpenPort(
+        SCE_AUDIO_OUT_PORT_TYPE_BGM,
+        device->UpdateSize,
+        device->Frequency,
+        device->FmtChans == DevFmtStereo ? SCE_AUDIO_OUT_MODE_STEREO : SCE_AUDIO_OUT_MODE_MONO
+    );
+
+    if (self->portNumber < 0)
+        return ALC_INVALID_VALUE;
+
+
+    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;
+
+    self->waveBuffer = calloc(device->UpdateSize * self->frameSize, 1);
+
+    alstr_copy_cstr(&device->DeviceName, name ? name : playbackDeviceName);
+
+    return ALC_NO_ERROR;
+}
+
+static ALCboolean ALCvitaPlayback_reset(ALCvitaPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    if (device->FmtChans != DevFmtMono && device->FmtChans != DevFmtStereo)
+        device->FmtChans = DevFmtStereo;
+
+    device->UpdateSize = AUDIO_SAMPLE_ALIGN(device->UpdateSize);
+
+    sceAudioOutSetConfig(
+        self->portNumber,
+        device->UpdateSize,
+        device->Frequency,
+        device->FmtChans == DevFmtStereo ? SCE_AUDIO_OUT_MODE_STEREO : SCE_AUDIO_OUT_MODE_MONO
+    );
+
+    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;
+
+    if (self->waveBuffer)
+    {
+        free(self->waveBuffer);
+    }
+
+    self->waveBuffer = calloc(device->UpdateSize * self->frameSize, 1);
+
+    SetDefaultWFXChannelOrder(device);
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCvitaPlayback_start(ALCvitaPlayback *self)
+{
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+
+    int priority = 128; // middle
+    int affinity = 0; // DEFAULT
+    int stack_size = 0x10000; // 64Kib
+
+    if (&_oal_thread_priority != NULL) {
+        priority = _oal_thread_priority;
+    } else {
+        SceKernelThreadInfo info;
+        info.size = sizeof(SceKernelThreadInfo);
+        if (sceKernelGetThreadInfo(sceKernelGetThreadId(), &info) == 0) {
+            priority = info.currentPriority - 1;
+        }
+    }
+
+    if (priority < 64 && priority != 0) priority = 64;
+    if (priority > 191) priority = 191;
+
+    if (&_oal_thread_affinity != NULL) {
+        affinity = _oal_thread_affinity;
+    }
+
+    self->thread = sceKernelCreateThread("OpenAL Vita playback thread", ALCvitaPlayback_MixerProc,
+                           priority, stack_size, 0, affinity, NULL);
+
+    if (self->thread < 0)
+        return ALC_FALSE;
+
+    int ret = sceKernelStartThread(self->thread, 4, &self);
+    if (ret < 0)
+        return ALC_FALSE;
+
+    return ALC_TRUE;
+}
+
+static void ALCvitaPlayback_stop(ALCvitaPlayback *self)
+{
+    if (ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+
+    sceKernelWaitThreadEnd(self->thread, NULL, NULL);
+    sceKernelDeleteThread(self->thread);
+}
+
+static void ALCvitaPlayback_lock(ALCvitaPlayback *self)
+{
+    sceKernelLockLwMutex(&self->lock, 1, NULL);
+}
+
+static void ALCvitaPlayback_unlock(ALCvitaPlayback *self)
+{
+    sceKernelUnlockLwMutex(&self->lock, 1);
+}
+
+// -----------------------------------------------------------------------------
+// Capture
+// -----------------------------------------------------------------------------
+typedef struct ALCvitaCapture
+{
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    ATOMIC(int) killNow;
+    althrd_t thread;
+    SceKernelLwMutexWork lock;
+
+    int portNumber;
+    ALsizei frameSize;
+    ll_ringbuffer_t *ring;
+
+    ALuint Frequency;
+    enum DevFmtChannels FmtChans;
+    enum DevFmtType     FmtType;
+    ALuint UpdateSize;
+} ALCvitaCapture;
+
+static void ALCvitaCapture_Construct(ALCvitaCapture *self, ALCdevice *device);
+static void ALCvitaCapture_Destruct(ALCvitaCapture *self);
+static ALCenum ALCvitaCapture_open(ALCvitaCapture *self, const ALCchar *name);
+static ALCboolean ALCvitaCapture_reset(ALCvitaCapture *self);
+static ALCboolean ALCvitaCapture_start(ALCvitaCapture *self);
+static void ALCvitaCapture_stop(ALCvitaCapture *self);
+static void ALCvitaCapture_lock(ALCvitaCapture *self);
+static void ALCvitaCapture_unlock(ALCvitaCapture *self);
+static ALCenum ALCvitaCapture_captureSamples(ALCvitaCapture *self, ALCvoid *buffer, ALCuint samples);
+static ALCuint ALCvitaCapture_availableSamples(ALCvitaCapture *self);
+static DECLARE_FORWARD(ALCvitaCapture, ALCbackend, ClockLatency, getClockLatency)
+DECLARE_DEFAULT_ALLOCATORS(ALCvitaCapture)
+
+DEFINE_ALCBACKEND_VTABLE(ALCvitaCapture);
+
+static void ALCvitaCapture_Construct(ALCvitaCapture *self, ALCdevice *device)
+{
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCvitaCapture, ALCbackend, self);
+
+    self->portNumber = 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 ALCvitaCapture_Destruct(ALCvitaCapture *self)
+{
+    if (self->portNumber)
+    {
+        sceAudioOutReleasePort(self->portNumber);
+        self->portNumber = 0;
+    }
+
+    if (self->ring)
+    {
+        ll_ringbuffer_free(self->ring);
+        self->ring = NULL;
+    }
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+static int ALCvitaCapture_MixerProc(void *ptr)
+{
+    ALCvitaCapture *self = (ALCvitaCapture*)ptr;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    void* buf = malloc(self->frameSize * device->UpdateSize);
+
+    while (!ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
+    {
+        sceAudioInInput(self->portNumber, buf);
+        ll_ringbuffer_write(self->ring, buf, self->frameSize * device->UpdateSize);
+    }
+
+    free(buf);
+
+    return 0;
+}
+
+static ALCenum ALCvitaCapture_open(ALCvitaCapture *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+
+    /* Only signed short output sample format is supported */
+    device->FmtType = DevFmtShort;
+
+    /* Only mono channel configuration is supported */
+    if (device->FmtChans != DevFmtMono)
+        device->FmtChans = DevFmtMono;
+
+    /* TODO: Validate samplerate and update size */
+
+    self->portNumber = sceAudioInOpenPort(
+        SCE_AUDIO_IN_PORT_TYPE_RAW,
+        device->UpdateSize,
+        device->Frequency,
+        SCE_AUDIO_IN_PARAM_FORMAT_S16_MONO
+    );
+
+    if (self->portNumber < 0)
+    {
+        return ALC_INVALID_VALUE;
+    }
+
+    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;
+
+    self->ring = ll_ringbuffer_create(device->UpdateSize * device->NumUpdates, self->frameSize, false);
+    if (self->ring == NULL)
+        return ALC_INVALID_VALUE;
+
+    alstr_copy_cstr(&device->DeviceName, name ? name : captureDeviceName);
+
+    return ALC_NO_ERROR;
+}
+
+static ALCboolean ALCvitaCapture_reset(ALCvitaCapture *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    SetDefaultWFXChannelOrder(device);
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCvitaCapture_start(ALCvitaCapture *self)
+{
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    if (althrd_create(&self->thread, ALCvitaCapture_MixerProc, self) != althrd_success)
+        return ALC_FALSE;
+
+    return ALC_TRUE;
+}
+
+static void ALCvitaCapture_stop(ALCvitaCapture *self)
+{
+    int res;
+
+    if (ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
+        return;
+
+    althrd_join(self->thread, &res);
+}
+
+static void ALCvitaCapture_lock(ALCvitaCapture *self)
+{
+    sceKernelLockLwMutex(&self->lock, 1, NULL);
+}
+
+static void ALCvitaCapture_unlock(ALCvitaCapture *self)
+{
+    sceKernelUnlockLwMutex(&self->lock, 1);
+}
+
+
+static ALCuint ALCvitaCapture_availableSamples(ALCvitaCapture *self)
+{
+    return ll_ringbuffer_read_space(self->ring);
+}
+
+static ALCenum ALCvitaCapture_captureSamples(ALCvitaCapture *self, ALCvoid *buffer, ALCuint samples)
+{
+    ll_ringbuffer_read(self->ring, buffer, samples);
+    return ALC_NO_ERROR;
+}
+
+// -----------------------------------------------------------------------------
+// Backends
+// -----------------------------------------------------------------------------
+
+typedef struct ALCvitaBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCvitaBackendFactory;
+#define ALCvitaBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCvitaBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCvitaBackendFactory_getFactory(void);
+
+static ALCboolean ALCvitaBackendFactory_init(ALCvitaBackendFactory *self);
+static void ALCvitaBackendFactory_deinit(ALCvitaBackendFactory *self);
+static ALCboolean ALCvitaBackendFactory_querySupport(ALCvitaBackendFactory *self, ALCbackend_Type type);
+static void ALCvitaBackendFactory_probe(ALCvitaBackendFactory *self, enum DevProbe type, al_string *outnames);
+static ALCbackend* ALCvitaBackendFactory_createBackend(ALCvitaBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCvitaBackendFactory);
+
+ALCbackendFactory *ALCvitaBackendFactory_getFactory(void)
+{
+    static ALCvitaBackendFactory factory = ALCvitaBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCvitaBackendFactory_init(ALCvitaBackendFactory* UNUSED(self))
+{
+    return AL_TRUE;
+}
+
+static void ALCvitaBackendFactory_deinit(ALCvitaBackendFactory* UNUSED(self))
+{
+}
+
+static ALCboolean ALCvitaBackendFactory_querySupport(ALCvitaBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if (type == ALCbackend_Playback || type == ALCbackend_Capture)
+        return ALC_TRUE;
+
+    return ALC_FALSE;
+}
+
+static void ALCvitaBackendFactory_probe(ALCvitaBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
+{
+    if (type == ALL_DEVICE_PROBE)
+        alstr_append_range(outnames, playbackDeviceName, playbackDeviceName+sizeof(playbackDeviceName));
+    else if (type == CAPTURE_DEVICE_PROBE)
+        alstr_append_range(outnames, captureDeviceName, playbackDeviceName+sizeof(captureDeviceName));
+}
+
+static ALCbackend* ALCvitaBackendFactory_createBackend(ALCvitaBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if (type == ALCbackend_Playback)
+    {
+        ALCvitaPlayback *backend;
+        NEW_OBJ(backend, ALCvitaPlayback)(device);
+
+        if (!backend)
+            return NULL;
+
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    if (type == ALCbackend_Capture)
+    {
+        ALCvitaCapture *backend;
+        NEW_OBJ(backend, ALCvitaCapture)(device);
+
+        if (!backend)
+            return NULL;
+
+        return STATIC_CAST(ALCbackend, backend);
+    }
+    return NULL;
+}

Alc/backends/wave.c

@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  * Or go to http://www.gnu.org/copyleft/lgpl.html
  */
 
@@ -23,24 +23,15 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include <memory.h>
-#ifdef HAVE_WINDOWS_H
-#include <windows.h>
-#endif
+#include <errno.h>
 
 #include "alMain.h"
 #include "alu.h"
+#include "alconfig.h"
+#include "threads.h"
+#include "compat.h"
 
-
-typedef struct {
-    FILE *f;
-    long DataStart;
-
-    ALvoid *buffer;
-    ALuint size;
-
-    volatile int killNow;
-    ALvoid *thread;
-} wave_data;
+#include "backends/base.h"
 
 
 static const ALCchar waveDevice[] = "Wave File Writer";
@@ -54,106 +45,168 @@ static const ALubyte SUBTYPE_FLOAT[] = {
     0x00, 0x38, 0x9b, 0x71
 };
 
-static const ALuint channel_masks[] = {
-    0, /* invalid */
-    0x4, /* Mono */
-    0x1 | 0x2, /* Stereo */
-    0, /* 3 channel */
-    0x1 | 0x2 | 0x10 | 0x20, /* Quad */
-    0, /* 5 channel */
-    0x1 | 0x2 | 0x4 | 0x8 | 0x10 | 0x20, /* 5.1 */
-    0x1 | 0x2 | 0x4 | 0x8 | 0x100 | 0x200 | 0x400, /* 6.1 */
-    0x1 | 0x2 | 0x4 | 0x8 | 0x10 | 0x20 | 0x200 | 0x400, /* 7.1 */
+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)
 {
-    fputc(val&0xff, f);
-    fputc((val>>8)&0xff, f);
+    ALubyte data[2] = { val&0xff, (val>>8)&0xff };
+    fwrite(data, 1, 2, f);
 }
 
 static void fwrite32le(ALuint val, FILE *f)
 {
-    fputc(val&0xff, f);
-    fputc((val>>8)&0xff, f);
-    fputc((val>>16)&0xff, f);
-    fputc((val>>24)&0xff, f);
+    ALubyte data[4] = { val&0xff, (val>>8)&0xff, (val>>16)&0xff, (val>>24)&0xff };
+    fwrite(data, 1, 4, f);
 }
 
 
-static ALuint WaveProc(ALvoid *ptr)
-{
-    ALCdevice *Device = (ALCdevice*)ptr;
-    wave_data *data = (wave_data*)Device->ExtraData;
-    ALuint frameSize;
-    ALuint now, start;
-    ALuint64 avail, done;
-    size_t fs;
-    const ALuint restTime = (ALuint64)Device->UpdateSize * 1000 /
-                            Device->Frequency / 2;
+typedef struct ALCwaveBackend {
+    DERIVE_FROM_TYPE(ALCbackend);
 
-    frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
+    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;
-    start = timeGetTime();
-    while(!data->killNow && Device->Connected)
+    if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
     {
-        now = timeGetTime();
+        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 = (ALuint64)(now-start) * Device->Frequency / 1000;
+        avail  = (now.tv_sec - start.tv_sec) * device->Frequency;
+        avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
         if(avail < done)
         {
-            /* Timer wrapped (50 days???). Add the remainder of the cycle to
-             * the available count and reset the number of samples done */
-            avail += ((ALuint64)1<<32)*Device->Frequency/1000 - done;
-            done = 0;
-        }
-        if(avail-done < Device->UpdateSize)
-        {
-            Sleep(restTime);
-            continue;
+            /* 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;
         }
 
-        while(avail-done >= Device->UpdateSize)
+        if(avail-done < device->UpdateSize)
+            al_nssleep(restTime);
+        else while(avail-done >= device->UpdateSize)
         {
-            aluMixData(Device, data->buffer, Device->UpdateSize);
-            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);
-                ALubyte *bytes = data->buffer;
+                ALuint bytesize = BytesFromDevFmt(device->FmtType);
                 ALuint i;
 
-                if(bytesize == 1)
+                if(bytesize == 2)
                 {
-                    for(i = 0;i < data->size;i++)
-                        fputc(bytes[i], data->f);
-                }
-                else if(bytesize == 2)
-                {
-                    for(i = 0;i < data->size;i++)
-                        fputc(bytes[i^1], data->f);
+                    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)
                 {
-                    for(i = 0;i < data->size;i++)
-                        fputc(bytes[i^3], data->f);
+                    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);
+                    }
                 }
             }
-            else
-            {
-                fs = fwrite(data->buffer, frameSize, Device->UpdateSize,
-                            data->f);
-                fs = fs;
-            }
-            if(ferror(data->f))
+
+            fs = fwrite(self->mBuffer, frameSize, device->UpdateSize, self->mFile);
+            (void)fs;
+            if(ferror(self->mFile))
             {
                 ERR("Error writing to file\n");
-                ALCdevice_Lock(Device);
-                aluHandleDisconnect(Device);
-                ALCdevice_Unlock(Device);
+                ALCdevice_Lock(device);
+                aluHandleDisconnect(device, "Failed to write playback samples");
+                ALCdevice_Unlock(device);
                 break;
             }
         }
@@ -162,52 +215,48 @@ static ALuint WaveProc(ALvoid *ptr)
     return 0;
 }
 
-static ALCenum wave_open_playback(ALCdevice *device, const ALCchar *deviceName)
+
+static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name)
 {
-    wave_data *data;
+    ALCdevice *device;
     const char *fname;
 
-    fname = GetConfigValue("wave", "file", "");
-    if(!fname[0])
+    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;
 
-    if(!deviceName)
-        deviceName = waveDevice;
-    else if(strcmp(deviceName, waveDevice) != 0)
-        return ALC_INVALID_VALUE;
-
-    data = (wave_data*)calloc(1, sizeof(wave_data));
-
-    data->f = fopen(fname, "wb");
-    if(!data->f)
+    self->mFile = al_fopen(fname, "wb");
+    if(!self->mFile)
     {
-        free(data);
         ERR("Could not open file '%s': %s\n", fname, strerror(errno));
         return ALC_INVALID_VALUE;
     }
 
-    device->DeviceName = strdup(deviceName);
-    device->ExtraData = data;
+    device = STATIC_CAST(ALCbackend, self)->mDevice;
+    alstr_copy_cstr(&device->DeviceName, name);
+
     return ALC_NO_ERROR;
 }
 
-static void wave_close_playback(ALCdevice *device)
+static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self)
 {
-    wave_data *data = (wave_data*)device->ExtraData;
-
-    fclose(data->f);
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean wave_reset_playback(ALCdevice *device)
-{
-    wave_data *data = (wave_data*)device->ExtraData;
-    ALuint channels=0, bits=0;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
+    ALuint channels=0, bits=0, chanmask=0;
+    int isbformat = 0;
     size_t val;
 
-    fseek(data->f, 0, SEEK_SET);
-    clearerr(data->f);
+    fseek(self->mFile, 0, SEEK_SET);
+    clearerr(self->mFile);
+
+    if(GetConfigValueBool(NULL, "wave", "bformat", 0))
+    {
+        device->FmtChans = DevFmtAmbi3D;
+        device->AmbiOrder = 1;
+    }
 
     switch(device->FmtType)
     {
@@ -226,145 +275,179 @@ static ALCboolean wave_reset_playback(ALCdevice *device)
         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);
+    channels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
 
-    fprintf(data->f, "RIFF");
-    fwrite32le(0xFFFFFFFF, data->f); // 'RIFF' header len; filled in at close
+    fputs("RIFF", self->mFile);
+    fwrite32le(0xFFFFFFFF, self->mFile); // 'RIFF' header len; filled in at close
 
-    fprintf(data->f, "WAVE");
+    fputs("WAVE", self->mFile);
 
-    fprintf(data->f, "fmt ");
-    fwrite32le(40, data->f); // 'fmt ' header len; 40 bytes for EXTENSIBLE
+    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, data->f);
+    fwrite16le(0xFFFE, self->mFile);
     // 16-bit val, channel count
-    fwrite16le(channels, data->f);
+    fwrite16le(channels, self->mFile);
     // 32-bit val, frequency
-    fwrite32le(device->Frequency, data->f);
+    fwrite32le(device->Frequency, self->mFile);
     // 32-bit val, bytes per second
-    fwrite32le(device->Frequency * channels * bits / 8, data->f);
+    fwrite32le(device->Frequency * channels * bits / 8, self->mFile);
     // 16-bit val, frame size
-    fwrite16le(channels * bits / 8, data->f);
+    fwrite16le(channels * bits / 8, self->mFile);
     // 16-bit val, bits per sample
-    fwrite16le(bits, data->f);
+    fwrite16le(bits, self->mFile);
     // 16-bit val, extra byte count
-    fwrite16le(22, data->f);
+    fwrite16le(22, self->mFile);
     // 16-bit val, valid bits per sample
-    fwrite16le(bits, data->f);
+    fwrite16le(bits, self->mFile);
     // 32-bit val, channel mask
-    fwrite32le(channel_masks[channels], data->f);
+    fwrite32le(chanmask, self->mFile);
     // 16 byte GUID, sub-type format
-    val = fwrite(((bits==32) ? SUBTYPE_FLOAT : SUBTYPE_PCM), 1, 16, data->f);
-    val = val;
+    val = fwrite((device->FmtType == DevFmtFloat) ?
+                 (isbformat ? SUBTYPE_BFORMAT_FLOAT : SUBTYPE_FLOAT) :
+                 (isbformat ? SUBTYPE_BFORMAT_PCM : SUBTYPE_PCM), 1, 16, self->mFile);
+    (void)val;
 
-    fprintf(data->f, "data");
-    fwrite32le(0xFFFFFFFF, data->f); // 'data' header len; filled in at close
+    fputs("data", self->mFile);
+    fwrite32le(0xFFFFFFFF, self->mFile); // 'data' header len; filled in at close
 
-    if(ferror(data->f))
+    if(ferror(self->mFile))
     {
         ERR("Error writing header: %s\n", strerror(errno));
         return ALC_FALSE;
     }
-    data->DataStart = ftell(data->f);
+    self->mDataStart = ftell(self->mFile);
 
     SetDefaultWFXChannelOrder(device);
 
     return ALC_TRUE;
 }
 
-static ALCboolean wave_start_playback(ALCdevice *device)
+static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self)
 {
-    wave_data *data = (wave_data*)device->ExtraData;
+    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
 
-    data->size = device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
-    data->buffer = malloc(data->size);
-    if(!data->buffer)
+    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;
     }
 
-    data->thread = StartThread(WaveProc, device);
-    if(data->thread == NULL)
+    ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
+    if(althrd_create(&self->thread, ALCwaveBackend_mixerProc, self) != althrd_success)
     {
-        free(data->buffer);
-        data->buffer = NULL;
+        free(self->mBuffer);
+        self->mBuffer = NULL;
+        self->mSize = 0;
         return ALC_FALSE;
     }
 
     return ALC_TRUE;
 }
 
-static void wave_stop_playback(ALCdevice *device)
+static void ALCwaveBackend_stop(ALCwaveBackend *self)
 {
-    wave_data *data = (wave_data*)device->ExtraData;
     ALuint dataLen;
     long size;
+    int res;
 
-    if(!data->thread)
+    if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
         return;
+    althrd_join(self->thread, &res);
 
-    data->killNow = 1;
-    StopThread(data->thread);
-    data->thread = NULL;
+    free(self->mBuffer);
+    self->mBuffer = NULL;
 
-    data->killNow = 0;
-
-    free(data->buffer);
-    data->buffer = NULL;
-
-    size = ftell(data->f);
+    size = ftell(self->mFile);
     if(size > 0)
     {
-        dataLen = size - data->DataStart;
-        if(fseek(data->f, data->DataStart-4, SEEK_SET) == 0)
-            fwrite32le(dataLen, data->f); // 'data' header len
-        if(fseek(data->f, 4, SEEK_SET) == 0)
-            fwrite32le(size-8, data->f); // 'WAVE' header len
+        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
     }
 }
 
 
-static const BackendFuncs wave_funcs = {
-    wave_open_playback,
-    wave_close_playback,
-    wave_reset_playback,
-    wave_start_playback,
-    wave_stop_playback,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    NULL,
-    ALCdevice_LockDefault,
-    ALCdevice_UnlockDefault,
-    ALCdevice_GetLatencyDefault
-};
+typedef struct ALCwaveBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCwaveBackendFactory;
+#define ALCWAVEBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwaveBackendFactory, ALCbackendFactory) } }
 
-ALCboolean alc_wave_init(BackendFuncs *func_list)
+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))
 {
-    *func_list = wave_funcs;
     return ALC_TRUE;
 }
 
-void alc_wave_deinit(void)
+static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory* UNUSED(self), ALCbackend_Type type)
 {
+    if(type == ALCbackend_Playback)
+        return !!ConfigValueExists(NULL, "wave", "file");
+    return ALC_FALSE;
 }
 
-void alc_wave_probe(enum DevProbe type)
+static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
 {
-    if(!ConfigValueExists("wave", "file"))
-        return;
-
     switch(type)
     {
         case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(waveDevice);
+            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/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.
+ */
+extern const ALfloat N3D2N3DScale[MAX_AMBI_COEFFS];
+extern const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS];
+extern 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

@@ -27,120 +27,98 @@
 #include <string.h>
 
 #include "bs2b.h"
+#include "alu.h"
 
-#ifndef M_PI
-#define M_PI  3.14159265358979323846
-#endif
-
-/* Single pole IIR filter.
- * O[n] = a0*I[n] + a1*I[n-1] + b1*O[n-1]
- */
-
-/* Lowpass filter */
-#define lo_filter(in, out_1) (bs2b->a0_lo*(in) + bs2b->b1_lo*(out_1))
-
-/* Highboost filter */
-#define hi_filter(in, in_1, out_1) (bs2b->a0_hi*(in) + bs2b->a1_hi*(in_1) + bs2b->b1_hi*(out_1))
 
 /* Set up all data. */
 static void init(struct bs2b *bs2b)
 {
-    double Fc_lo, Fc_hi;
-    double G_lo,  G_hi;
-    double x;
-
-    if ((bs2b->srate > 192000) || (bs2b->srate < 2000))
-        bs2b->srate = BS2B_DEFAULT_SRATE;
+    float Fc_lo, Fc_hi;
+    float G_lo,  G_hi;
+    float x, g;
 
     switch(bs2b->level)
     {
     case BS2B_LOW_CLEVEL: /* Low crossfeed level */
-        Fc_lo = 360.0;
-        Fc_hi = 501.0;
-        G_lo  = 0.398107170553497;
-        G_hi  = 0.205671765275719;
+        Fc_lo = 360.0f;
+        Fc_hi = 501.0f;
+        G_lo  = 0.398107170553497f;
+        G_hi  = 0.205671765275719f;
         break;
 
     case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */
-        Fc_lo = 500.0;
-        Fc_hi = 711.0;
-        G_lo  = 0.459726988530872;
-        G_hi  = 0.228208484414988;
+        Fc_lo = 500.0f;
+        Fc_hi = 711.0f;
+        G_lo  = 0.459726988530872f;
+        G_hi  = 0.228208484414988f;
         break;
 
     case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */
-        Fc_lo = 700.0;
-        Fc_hi = 1021.0;
-        G_lo  = 0.530884444230988;
-        G_hi  = 0.250105790667544;
+        Fc_lo = 700.0f;
+        Fc_hi = 1021.0f;
+        G_lo  = 0.530884444230988f;
+        G_hi  = 0.250105790667544f;
         break;
 
     case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */
-        Fc_lo = 360.0;
-        Fc_hi = 494.0;
-        G_lo  = 0.316227766016838;
-        G_hi  = 0.168236228897329;
+        Fc_lo = 360.0f;
+        Fc_hi = 494.0f;
+        G_lo  = 0.316227766016838f;
+        G_hi  = 0.168236228897329f;
         break;
 
     case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */
-        Fc_lo = 500.0;
-        Fc_hi = 689.0;
-        G_lo  = 0.354813389233575;
-        G_hi  = 0.187169483835901;
+        Fc_lo = 500.0f;
+        Fc_hi = 689.0f;
+        G_lo  = 0.354813389233575f;
+        G_hi  = 0.187169483835901f;
         break;
 
     default: /* High easy crossfeed level */
         bs2b->level = BS2B_HIGH_ECLEVEL;
 
-        Fc_lo = 700.0;
-        Fc_hi = 975.0;
-        G_lo  = 0.398107170553497;
-        G_hi  = 0.205671765275719;
+        Fc_lo = 700.0f;
+        Fc_hi = 975.0f;
+        G_lo  = 0.398107170553497f;
+        G_hi  = 0.205671765275719f;
         break;
     } /* switch */
 
+    g = 1.0f / (1.0f - G_hi + G_lo);
+
     /* $fc = $Fc / $s;
      * $d  = 1 / 2 / pi / $fc;
      * $x  = exp(-1 / $d);
      */
-
-    x           = exp(-2.0 * M_PI * Fc_lo / bs2b->srate);
+    x           = expf(-2.0f * F_PI * Fc_lo / bs2b->srate);
     bs2b->b1_lo = x;
-    bs2b->a0_lo = G_lo * (1.0 - x);
+    bs2b->a0_lo = G_lo * (1.0f - x) * g;
 
-    x           = exp(-2.0 * M_PI * Fc_hi / bs2b->srate);
+    x           = expf(-2.0f * F_PI * Fc_hi / bs2b->srate);
     bs2b->b1_hi = x;
-    bs2b->a0_hi = 1.0 - G_hi * (1.0 - x);
-    bs2b->a1_hi = -x;
-
-    bs2b->gain  = 1.0f / (float)(1.0 - G_hi + G_lo);
+    bs2b->a0_hi = (1.0f - G_hi * (1.0f - x)) * g;
+    bs2b->a1_hi = -x * g;
 } /* init */
 
+
 /* Exported functions.
  * See descriptions in "bs2b.h"
  */
 
-void bs2b_set_level(struct bs2b *bs2b, int level)
+void bs2b_set_params(struct bs2b *bs2b, int level, int srate)
 {
-    if(level == bs2b->level)
-        return;
+    if(srate <= 0) srate = 1;
+
     bs2b->level = level;
+    bs2b->srate = srate;
     init(bs2b);
-} /* bs2b_set_level */
+} /* bs2b_set_params */
 
 int bs2b_get_level(struct bs2b *bs2b)
 {
     return bs2b->level;
 } /* bs2b_get_level */
 
-void bs2b_set_srate(struct bs2b *bs2b, int srate)
-{
-    if (srate == bs2b->srate)
-        return;
-    bs2b->srate = srate;
-    init(bs2b);
-} /* bs2b_set_srate */
-
 int bs2b_get_srate(struct bs2b *bs2b)
 {
     return bs2b->srate;
@@ -151,35 +129,59 @@ void bs2b_clear(struct bs2b *bs2b)
     memset(&bs2b->last_sample, 0, sizeof(bs2b->last_sample));
 } /* bs2b_clear */
 
-void bs2b_cross_feed(struct bs2b *bs2b, float *sample)
+void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo)
 {
-    /* Lowpass filter */
-    bs2b->last_sample.lo[0] = lo_filter(sample[0], bs2b->last_sample.lo[0]);
-    bs2b->last_sample.lo[1] = lo_filter(sample[1], bs2b->last_sample.lo[1]);
+    float lsamples[128][2];
+    float rsamples[128][2];
+    int base;
 
-    /* Highboost filter */
-    bs2b->last_sample.hi[0] = hi_filter(sample[0], bs2b->last_sample.asis[0], bs2b->last_sample.hi[0]);
-    bs2b->last_sample.hi[1] = hi_filter(sample[1], bs2b->last_sample.asis[1], bs2b->last_sample.hi[1]);
-    bs2b->last_sample.asis[0] = sample[0];
-    bs2b->last_sample.asis[1] = sample[1];
+    for(base = 0;base < SamplesToDo;)
+    {
+        int todo = mini(128, SamplesToDo-base);
+        int i;
 
-    /* Crossfeed */
-    sample[0] = (float)(bs2b->last_sample.hi[0] + bs2b->last_sample.lo[1]);
-    sample[1] = (float)(bs2b->last_sample.hi[1] + bs2b->last_sample.lo[0]);
+        /* Process left input */
+        lsamples[0][0] = bs2b->a0_lo*Left[0] +
+                         bs2b->b1_lo*bs2b->last_sample[0].lo;
+        lsamples[0][1] = bs2b->a0_hi*Left[0] +
+                         bs2b->a1_hi*bs2b->last_sample[0].asis +
+                         bs2b->b1_hi*bs2b->last_sample[0].hi;
+        for(i = 1;i < todo;i++)
+        {
+            lsamples[i][0] = bs2b->a0_lo*Left[i] +
+                             bs2b->b1_lo*lsamples[i-1][0];
+            lsamples[i][1] = bs2b->a0_hi*Left[i] +
+                             bs2b->a1_hi*Left[i-1] +
+                             bs2b->b1_hi*lsamples[i-1][1];
+        }
+        bs2b->last_sample[0].asis = Left[i-1];
+        bs2b->last_sample[0].lo = lsamples[i-1][0];
+        bs2b->last_sample[0].hi = lsamples[i-1][1];
 
-    /* Bass boost cause allpass attenuation */
-    sample[0] *= bs2b->gain;
-    sample[1] *= bs2b->gain;
+        /* Process right input */
+        rsamples[0][0] = bs2b->a0_lo*Right[0] +
+                         bs2b->b1_lo*bs2b->last_sample[1].lo;
+        rsamples[0][1] = bs2b->a0_hi*Right[0] +
+                         bs2b->a1_hi*bs2b->last_sample[1].asis +
+                         bs2b->b1_hi*bs2b->last_sample[1].hi;
+        for(i = 1;i < todo;i++)
+        {
+            rsamples[i][0] = bs2b->a0_lo*Right[i] +
+                             bs2b->b1_lo*rsamples[i-1][0];
+            rsamples[i][1] = bs2b->a0_hi*Right[i] +
+                             bs2b->a1_hi*Right[i-1] +
+                             bs2b->b1_hi*rsamples[i-1][1];
+        }
+        bs2b->last_sample[1].asis = Right[i-1];
+        bs2b->last_sample[1].lo = rsamples[i-1][0];
+        bs2b->last_sample[1].hi = rsamples[i-1][1];
 
-    /* Clipping of overloaded samples */
-#if 0
-    if (sample[0] > 1.0)
-        sample[0] = 1.0;
-    if (sample[0] < -1.0)
-        sample[0] = -1.0;
-    if (sample[1] > 1.0)
-        sample[1] = 1.0;
-    if (sample[1] < -1.0)
-        sample[1] = -1.0;
-#endif
+        /* Crossfeed */
+        for(i = 0;i < todo;i++)
+            *(Left++) = lsamples[i][1] + rsamples[i][0];
+        for(i = 0;i < todo;i++)
+            *(Right++) = rsamples[i][1] + lsamples[i][0];
+
+        base += todo;
+    }
 } /* bs2b_cross_feed */

Alc/compat.h

@@ -0,0 +1,57 @@
+#ifndef AL_COMPAT_H
+#define AL_COMPAT_H
+
+#include "alstring.h"
+
+#ifdef __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

@@ -0,0 +1,15 @@
+#ifndef CPU_CAPS_H
+#define CPU_CAPS_H
+
+extern int CPUCapFlags;
+enum {
+    CPU_CAP_SSE    = 1<<0,
+    CPU_CAP_SSE2   = 1<<1,
+    CPU_CAP_SSE3   = 1<<2,
+    CPU_CAP_SSE4_1 = 1<<3,
+    CPU_CAP_NEON   = 1<<4,
+};
+
+void FillCPUCaps(int capfilter);
+
+#endif /* CPU_CAPS_H */

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

@@ -0,0 +1,129 @@
+
+#include "config.h"
+
+#include "AL/alc.h"
+#include "AL/al.h"
+
+#include "alMain.h"
+#include "defs.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);
+
+
+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);
+
+    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 * sqrtf(gain) * alpha;
+            b[0] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
+            b[1] = -2.0f*gain*((gain-1.0f) + (gain+1.0f)*cos_w0                   );
+            b[2] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2);
+            a[0] =             (gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
+            a[1] =  2.0f*     ((gain-1.0f) - (gain+1.0f)*cos_w0                   );
+            a[2] =             (gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
+            break;
+        case 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);
+            a[0] =             (gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
+            a[1] = -2.0f*     ((gain-1.0f) + (gain+1.0f)*cos_w0                   );
+            a[2] =             (gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
+            break;
+        case BiquadType_Peaking:
+            gain = sqrtf(gain);
+            b[0] =  1.0f + alpha * gain;
+            b[1] = -2.0f * cos_w0;
+            b[2] =  1.0f - alpha * gain;
+            a[0] =  1.0f + alpha / gain;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha / gain;
+            break;
+
+        case BiquadType_LowPass:
+            b[0] = (1.0f - cos_w0) / 2.0f;
+            b[1] =  1.0f - cos_w0;
+            b[2] = (1.0f - cos_w0) / 2.0f;
+            a[0] =  1.0f + alpha;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha;
+            break;
+        case BiquadType_HighPass:
+            b[0] =  (1.0f + cos_w0) / 2.0f;
+            b[1] = -(1.0f + cos_w0);
+            b[2] =  (1.0f + cos_w0) / 2.0f;
+            a[0] =   1.0f + alpha;
+            a[1] =  -2.0f * cos_w0;
+            a[2] =   1.0f - alpha;
+            break;
+        case BiquadType_BandPass:
+            b[0] =  alpha;
+            b[1] =  0;
+            b[2] = -alpha;
+            a[0] =  1.0f + alpha;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha;
+            break;
+    }
+
+    filter->a1 = a[1] / a[0];
+    filter->a2 = a[2] / a[0];
+    filter->b0 = b[0] / a[0];
+    filter->b1 = b[1] / a[0];
+    filter->b2 = b[2] / a[0];
+}
+
+
+void BiquadFilter_processC(BiquadFilter *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples)
+{
+    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-
+     * sample history; the last two inputs and outputs), and works better for
+     * floating-point which favors summing similarly-sized values while being
+     * less bothered by overflow.
+     *
+     * See: http://www.earlevel.com/main/2003/02/28/biquads/
+     */
+    for(i = 0;i < numsamples;i++)
+    {
+        ALfloat input = src[i];
+        ALfloat output = input*b0 + z1;
+        z1 = input*b1 - output*a1 + z2;
+        z2 = input*b2 - output*a2;
+        dst[i] = output;
+    }
+
+    filter->z1 = z1;
+    filter->z2 = z2;
+}

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/hrtf_tables.inc

@@ -1,848 +0,0 @@
-/* Elevation metrics */
-static const ALubyte defaultAzCount[19] = { 1, 12, 24, 36, 45, 56, 60, 72, 72, 72, 72, 72, 60, 56, 45, 36, 24, 12, 1, };
-static const ALushort defaultEvOffset[19] = { 0, 1, 13, 37, 73, 118, 174, 234, 306, 378, 450, 522, 594, 654, 710, 755, 791, 815, 827, };
-
-/* HRIR Coefficients */
-static const ALshort defaultCoeffs[26496] =
-{
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-    +7749, +9628, +3084, +3049, +870, +16, +727, -1159, -2880, -1819, +969, +83, -717, -1031, -1759, -1055, -39, -9, -26, -31, -379, -324, -646, -479, -344, -651, -608, -341, -196, -48, +22, -160,
-    +7872, +8889, +2424, +3846, +1113, -405, +870, -636, -2562, -2175, +535, +438, -627, -691, -988, -1594, -378, +123, -383, -287, -574, -435, -436, -421, -370, -475, -575, -138, +9, -79, -10, -210,
-    +7007, +8290, +2692, +3507, +1037, -125, +838, -876, -2265, -1597, +225, -49, -510, -580, -573, -987, -677, +32, +42, -236, -535, -281, -374, -349, -433, -522, -408, -257, -182, -133, +15, -132,
-    +6475, +7694, +2419, +2913, +1152, +342, +794, -920, -2186, -1265, +352, -196, -457, -569, -494, -665, -413, +113, +33, -239, -428, -173, -379, -384, -517, -559, -408, -308, -159, -59, +31, -118,
-    +5981, +7105, +2165, +2692, +1134, +347, +767, -876, -2030, -957, +421, -294, -318, -441, -466, -416, -281, -39, -27, -194, -378, -156, -345, -274, -384, -498, -440, -344, -158, -51, +21, -134,
-    +5468, +6465, +2050, +2460, +952, +446, +858, -776, -1849, -646, +711, -106, -171, -377, -412, -225, -208, -139, -49, -122, -280, -59, -288, -217, -330, -482, -372, -276, -153, -54, +11, -123,
-    +5247, +6086, +1760, +2189, +842, +506, +1054, -433, -1470, -298, +965, +30, -189, -498, -477, -163, -126, -106, -24, -94, -283, -72, -247, -143, -266, -434, -302, -196, -110, -68, -52, -213,
-    +5232, +6232, +1996, +2383, +983, +625, +1098, -519, -1614, -432, +848, -66, -286, -585, -461, -110, -157, -165, -46, -92, -261, -20, -217, -109, -190, -394, -333, -281, -239, -151, -66, -182,
-    +5323, +6297, +1900, +2226, +867, +582, +1114, -514, -1700, -507, +886, -53, -362, -695, -545, -42, +42, -2, +24, -89, -290, -90, -302, -188, -238, -341, -202, -173, -185, -163, -115, -218,
-    +5574, +6674, +1997, +2183, +707, +407, +936, -776, -2047, -829, +914, +130, -285, -696, -573, -93, +9, -5, +50, +20, -186, -16, -247, -182, -271, -441, -315, -175, -47, +63, +58, -176,
-    +6045, +7217, +2109, +2333, +722, +394, +912, -1008, -2484, -1201, +940, +97, -533, -929, -717, -112, +69, +18, +107, +96, -204, -31, -262, -175, -230, -397, -254, -106, -72, -21, +17, -112,
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-    +20434, +13223, -8548, -342, +182, +828, -924, -2160, -2725, +468, +2161, -1789, +445, -1242, +435, -205, -999, -575, -649, -4, -1029, -158, -482, +542, -293, +181, -169, -874, -514, -927, -442, +127,
-    +21071, +12403, -8865, +245, -318, +823, -1130, -2243, -2635, +724, +2101, -1684, +725, -1293, +626, -269, -883, -830, -829, +105, -1203, -105, -238, +653, -371, +371, -143, -1016, -532, -886, -493, +33,
-    +20789, +11662, -8363, +332, -911, +1138, -984, -2430, -2493, +1026, +1715, -1379, +1228, -1581, +803, -170, -1256, -633, -926, +9, -905, -125, -208, +782, -256, +313, -208, -993, -501, -825, -526, -5,
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-    +13984, +10256, -4374, -387, +520, +1170, -465, -1651, -1737, +620, +1299, -598, +593, -692, +507, -106, -759, -500, -601, +13, -482, -134, -166, +603, -169, +64, -79, -688, -462, -543, -415, -231,
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-    +11076, +10636, -2726, -2397, +319, +2006, +654, -1075, -1802, -427, +898, -371, +393, -170, +406, +444, -274, -100, +119, -97, -686, -259, -554, +17, -61, -105, -68, -278, -240, -444, -157, +31,
-    +11612, +10554, -3129, -2035, +925, +2135, +343, -1322, -1834, -253, +981, -508, +127, -339, +524, +263, -428, +146, +113, -216, -691, -403, -615, +161, -3, -171, -98, -195, -334, -500, -145, +70,
-    +12375, +10683, -3543, -1496, +1375, +1878, -63, -1524, -1717, +100, +1163, -755, -161, -335, +443, +21, -544, +0, +88, -275, -693, -316, -565, +159, -85, -8, -150, -333, -390, -555, -137, +51,
-    +13451, +11029, -4018, -1046, +1272, +1286, -363, -1499, -1496, +592, +1316, -1108, -273, -352, +296, -275, -635, -344, -118, -175, -740, -127, -424, +105, -164, +113, -179, -568, -399, -627, -271, +78,
-    +14734, +11620, -4801, -879, +695, +632, -389, -1350, -1318, +1007, +1424, -1500, -74, -505, -4, -282, -945, -711, -166, -124, -792, +39, -391, +77, -83, +118, -326, -662, -381, -867, -393, +164,
-    +15985, +11630, -5516, -532, +203, +208, -354, -1214, -1189, +1341, +1442, -1615, +268, -744, -62, -254, -1183, -868, -259, -11, -770, +72, -261, +154, +24, +109, -377, -617, -514, -967, -389, +157,
-    +16167, +11377, -5735, -169, +103, +84, -542, -1299, -1254, +1362, +1544, -1556, +422, -670, -103, -248, -1015, -1157, -377, +88, -926, +197, -106, +201, +13, +190, -354, -777, -517, -923, -458, +58,
-    +15468, +10670, -5248, +0, +346, +442, -693, -1493, -1502, +1200, +1468, -1209, +625, -745, +218, -158, -964, -895, -623, +11, -688, +93, -44, +335, -4, +183, -327, -757, -508, -710, -494, -108,
-    +11864, +10338, -3078, -621, +1333, +599, -268, -1143, -1003, +1077, +1244, -764, -17, -195, +147, -60, -510, -758, -332, -70, -719, -16, -15, +316, +287, +250, -355, -886, -747, -549, -54, +215,
-};
-
-/* HRIR Delays */
-static const ALubyte defaultDelays[828] =
-{
-    12, 12, 13, 14, 14, 14, 13, 12, 11, 10, 10, 10, 11, 12, 13, 14, 15, 15, 16, 16, 16, 15, 15, 14, 13, 12, 11, 10, 9, 8, 8, 8, 8, 8, 9, 10, 11, 12, 13, 14, 15, 16, 16, 17, 18, 18, 18, 18, 18, 17, 16, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 7, 6, 6, 6, 6, 6, 7, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 18, 19, 20, 20, 20, 20, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 8, 7, 6, 5, 5, 5, 4, 4, 4, 5, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 21, 22, 22, 22, 22, 22, 21, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 4, 3, 3, 3, 3, 3, 3, 3, 4, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 22, 23, 24, 24, 24, 24, 24, 23, 22, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 3, 2, 2, 2, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 22, 23, 24, 24, 25, 26, 26, 26, 26, 26, 25, 24, 24, 23, 22, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 3, 3, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 3, 3, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 26, 27, 28, 28, 28, 27, 26, 26, 25, 24, 23, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 2, 2, 1, 1, 1, 0, 0, 0, 0, 0, 1, 1, 1, 2, 2, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 29, 28, 27, 26, 25, 24, 23, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 2, 2, 1, 1, 0, 0, 0, 0, 0, 0, 0, 1, 1, 2, 2, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 26, 27, 28, 28, 28, 27, 26, 26, 25, 24, 23, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 2, 2, 1, 1, 1, 0, 0, 0, 0, 0, 1, 1, 1, 2, 2, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 22, 23, 24, 24, 25, 26, 26, 26, 26, 26, 25, 24, 24, 23, 22, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 3, 3, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 3, 3, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 22, 23, 24, 24, 24, 24, 24, 23, 22, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 3, 2, 2, 2, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 21, 22, 22, 22, 22, 22, 21, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 4, 3, 3, 3, 3, 3, 3, 3, 4, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 18, 19, 20, 20, 20, 20, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 8, 7, 6, 5, 5, 5, 4, 4, 4, 5, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 16, 17, 18, 18, 18, 18, 18, 17, 16, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 7, 6, 6, 6, 6, 6, 7, 7, 8, 9, 10, 11, 12, 13, 14, 15, 15, 16, 16, 16, 15, 15, 14, 13, 12, 11, 10, 9, 8, 8, 8, 8, 8, 9, 10, 11, 12, 13, 14, 14, 14, 13, 12, 11, 10, 10, 10, 11, 12,
-};
-
-/* Default HRTF Definition */
-static const struct Hrtf DefaultHrtf = {
-    44100, 32, 19, defaultAzCount, defaultEvOffset,
-    defaultCoeffs, defaultDelays, NULL
-};

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 +1,530 @@
+#include "config.h"
+
+#include <math.h>
+
+#include "mastering.h"
+#include "alu.h"
+#include "almalloc.h"
+#include "static_assert.h"
+
+
+/* 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] = -INFINITY;
+            Comp->Hold->Expiries[0] = hold;
+            Comp->Hold->Length = hold;
+            Comp->Delay = (ALfloat(*)[])(Comp->Hold + 1);
+        }
+        else
+        {
+            Comp->Delay = (ALfloat(*)[])(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.c

@@ -1,421 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-#include <ctype.h>
-#include <assert.h>
-
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alSource.h"
-#include "alBuffer.h"
-#include "alListener.h"
-#include "alAuxEffectSlot.h"
-#include "alu.h"
-#include "bs2b.h"
-
-
-static __inline ALfloat Sample_ALbyte(ALbyte val)
-{ return val * (1.0f/127.0f); }
-
-static __inline ALfloat Sample_ALshort(ALshort val)
-{ return val * (1.0f/32767.0f); }
-
-static __inline ALfloat Sample_ALfloat(ALfloat val)
-{ return val; }
-
-#define DECL_TEMPLATE(T)                                                      \
-static void Load_##T(ALfloat *dst, const T *src, ALuint srcstep, ALuint samples)\
-{                                                                             \
-    ALuint i;                                                                 \
-    for(i = 0;i < samples;i++)                                                \
-        dst[i] = Sample_##T(src[i*srcstep]);                                  \
-}
-
-DECL_TEMPLATE(ALbyte)
-DECL_TEMPLATE(ALshort)
-DECL_TEMPLATE(ALfloat)
-
-#undef DECL_TEMPLATE
-
-static void LoadData(ALfloat *dst, const ALvoid *src, ALuint srcstep, enum FmtType srctype, ALuint samples)
-{
-    switch(srctype)
-    {
-        case FmtByte:
-            Load_ALbyte(dst, src, srcstep, samples);
-            break;
-        case FmtShort:
-            Load_ALshort(dst, src, srcstep, samples);
-            break;
-        case FmtFloat:
-            Load_ALfloat(dst, src, srcstep, samples);
-            break;
-    }
-}
-
-static void SilenceData(ALfloat *dst, ALuint samples)
-{
-    ALuint i;
-    for(i = 0;i < samples;i++)
-        dst[i] = 0.0f;
-}
-
-
-static void Filter2P(FILTER *filter, ALuint chan, ALfloat *RESTRICT dst,
-                     const ALfloat *RESTRICT src, ALuint numsamples)
-{
-    ALuint i;
-    for(i = 0;i < numsamples;i++)
-        dst[i] = lpFilter2P(filter, chan, src[i]);
-    dst[i] = lpFilter2PC(filter, chan, src[i]);
-}
-
-
-ALvoid MixSource(ALsource *Source, ALCdevice *Device, ALuint SamplesToDo)
-{
-    ALbufferlistitem *BufferListItem;
-    ALuint DataPosInt, DataPosFrac;
-    ALuint BuffersPlayed;
-    ALboolean Looping;
-    ALuint increment;
-    enum Resampler Resampler;
-    ALenum State;
-    ALuint OutPos;
-    ALuint NumChannels;
-    ALuint SampleSize;
-    ALint64 DataSize64;
-    ALuint chan, j;
-
-    /* Get source info */
-    State         = Source->state;
-    BuffersPlayed = Source->BuffersPlayed;
-    DataPosInt    = Source->position;
-    DataPosFrac   = Source->position_fraction;
-    Looping       = Source->Looping;
-    increment     = Source->Params.Step;
-    Resampler     = (increment==FRACTIONONE) ? PointResampler : Source->Resampler;
-    NumChannels   = Source->NumChannels;
-    SampleSize    = Source->SampleSize;
-
-    /* Get current buffer queue item */
-    BufferListItem = Source->queue;
-    for(j = 0;j < BuffersPlayed;j++)
-        BufferListItem = BufferListItem->next;
-
-    OutPos = 0;
-    do {
-        const ALuint BufferPrePadding = ResamplerPrePadding[Resampler];
-        const ALuint BufferPadding = ResamplerPadding[Resampler];
-        ALuint SrcBufferSize, DstBufferSize;
-
-        /* Figure out how many buffer bytes will be needed */
-        DataSize64  = SamplesToDo-OutPos+1;
-        DataSize64 *= increment;
-        DataSize64 += DataPosFrac+FRACTIONMASK;
-        DataSize64 >>= FRACTIONBITS;
-        DataSize64 += BufferPadding+BufferPrePadding;
-
-        SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE);
-
-        /* Figure out how many samples we can actually mix from this. */
-        DataSize64  = SrcBufferSize;
-        DataSize64 -= BufferPadding+BufferPrePadding;
-        DataSize64 <<= FRACTIONBITS;
-        DataSize64 -= increment;
-        DataSize64 -= DataPosFrac;
-
-        DstBufferSize = (ALuint)((DataSize64+(increment-1)) / increment);
-        DstBufferSize = minu(DstBufferSize, (SamplesToDo-OutPos));
-
-        /* Some mixers like having a multiple of 4, so try to give that unless
-         * this is the last update. */
-        if(OutPos+DstBufferSize < SamplesToDo)
-            DstBufferSize &= ~3;
-
-        for(chan = 0;chan < NumChannels;chan++)
-        {
-            ALfloat *SrcData = Device->SampleData1;
-            ALfloat *ResampledData = Device->SampleData2;
-            ALuint SrcDataSize = 0;
-
-            if(Source->SourceType == AL_STATIC)
-            {
-                const ALbuffer *ALBuffer = Source->queue->buffer;
-                const ALubyte *Data = ALBuffer->data;
-                ALuint DataSize;
-                ALuint pos;
-
-                /* If current pos is beyond the loop range, do not loop */
-                if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
-                {
-                    Looping = AL_FALSE;
-
-                    if(DataPosInt >= BufferPrePadding)
-                        pos = DataPosInt - BufferPrePadding;
-                    else
-                    {
-                        DataSize = BufferPrePadding - DataPosInt;
-                        DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
-
-                        SilenceData(&SrcData[SrcDataSize], DataSize);
-                        SrcDataSize += DataSize;
-
-                        pos = 0;
-                    }
-
-                    /* Copy what's left to play in the source buffer, and clear the
-                     * rest of the temp buffer */
-                    DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos);
-
-                    LoadData(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
-                             NumChannels, ALBuffer->FmtType, DataSize);
-                    SrcDataSize += DataSize;
-
-                    SilenceData(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
-                    SrcDataSize += SrcBufferSize - SrcDataSize;
-                }
-                else
-                {
-                    ALuint LoopStart = ALBuffer->LoopStart;
-                    ALuint LoopEnd   = ALBuffer->LoopEnd;
-
-                    if(DataPosInt >= LoopStart)
-                    {
-                        pos = DataPosInt-LoopStart;
-                        while(pos < BufferPrePadding)
-                            pos += LoopEnd-LoopStart;
-                        pos -= BufferPrePadding;
-                        pos += LoopStart;
-                    }
-                    else if(DataPosInt >= BufferPrePadding)
-                        pos = DataPosInt - BufferPrePadding;
-                    else
-                    {
-                        DataSize = BufferPrePadding - DataPosInt;
-                        DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
-
-                        SilenceData(&SrcData[SrcDataSize], DataSize);
-                        SrcDataSize += DataSize;
-
-                        pos = 0;
-                    }
-
-                    /* Copy what's left of this loop iteration, then copy repeats
-                     * of the loop section */
-                    DataSize = LoopEnd - pos;
-                    DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
-
-                    LoadData(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
-                             NumChannels, ALBuffer->FmtType, DataSize);
-                    SrcDataSize += DataSize;
-
-                    DataSize = LoopEnd-LoopStart;
-                    while(SrcBufferSize > SrcDataSize)
-                    {
-                        DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
-
-                        LoadData(&SrcData[SrcDataSize], &Data[(LoopStart*NumChannels + chan)*SampleSize],
-                                 NumChannels, ALBuffer->FmtType, DataSize);
-                        SrcDataSize += DataSize;
-                    }
-                }
-            }
-            else
-            {
-                /* Crawl the buffer queue to fill in the temp buffer */
-                ALbufferlistitem *tmpiter = BufferListItem;
-                ALuint pos;
-
-                if(DataPosInt >= BufferPrePadding)
-                    pos = DataPosInt - BufferPrePadding;
-                else
-                {
-                    pos = BufferPrePadding - DataPosInt;
-                    while(pos > 0)
-                    {
-                        if(!tmpiter->prev && !Looping)
-                        {
-                            ALuint DataSize = minu(SrcBufferSize - SrcDataSize, pos);
-
-                            SilenceData(&SrcData[SrcDataSize], DataSize);
-                            SrcDataSize += DataSize;
-
-                            pos = 0;
-                            break;
-                        }
-
-                        if(tmpiter->prev)
-                            tmpiter = tmpiter->prev;
-                        else
-                        {
-                            while(tmpiter->next)
-                                tmpiter = tmpiter->next;
-                        }
-
-                        if(tmpiter->buffer)
-                        {
-                            if((ALuint)tmpiter->buffer->SampleLen > pos)
-                            {
-                                pos = tmpiter->buffer->SampleLen - pos;
-                                break;
-                            }
-                            pos -= tmpiter->buffer->SampleLen;
-                        }
-                    }
-                }
-
-                while(tmpiter && SrcBufferSize > SrcDataSize)
-                {
-                    const ALbuffer *ALBuffer;
-                    if((ALBuffer=tmpiter->buffer) != NULL)
-                    {
-                        const ALubyte *Data = ALBuffer->data;
-                        ALuint DataSize = ALBuffer->SampleLen;
-
-                        /* Skip the data already played */
-                        if(DataSize <= pos)
-                            pos -= DataSize;
-                        else
-                        {
-                            Data += (pos*NumChannels + chan)*SampleSize;
-                            DataSize -= pos;
-                            pos -= pos;
-
-                            DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
-                            LoadData(&SrcData[SrcDataSize], Data, NumChannels,
-                                     ALBuffer->FmtType, DataSize);
-                            SrcDataSize += DataSize;
-                        }
-                    }
-                    tmpiter = tmpiter->next;
-                    if(!tmpiter && Looping)
-                        tmpiter = Source->queue;
-                    else if(!tmpiter)
-                    {
-                        SilenceData(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
-                        SrcDataSize += SrcBufferSize - SrcDataSize;
-                    }
-                }
-            }
-
-            /* Now resample, then filter and mix to the appropriate outputs. */
-            Source->Params.Resample(&SrcData[BufferPrePadding], DataPosFrac,
-                                    increment, ResampledData, DstBufferSize);
-
-            {
-                DirectParams *directparms = &Source->Params.Direct;
-
-                Filter2P(&directparms->iirFilter, chan, SrcData, ResampledData,
-                         DstBufferSize);
-                Source->Params.DryMix(directparms, SrcData, chan, OutPos,
-                                      SamplesToDo, DstBufferSize);
-            }
-
-            for(j = 0;j < Device->NumAuxSends;j++)
-            {
-                SendParams *sendparms = &Source->Params.Send[j];
-                if(!sendparms->Slot)
-                    continue;
-
-                Filter2P(&sendparms->iirFilter, chan, SrcData, ResampledData,
-                         DstBufferSize);
-                Source->Params.WetMix(sendparms, SrcData, OutPos,
-                                      SamplesToDo, DstBufferSize);
-            }
-        }
-        /* Update positions */
-        for(j = 0;j < DstBufferSize;j++)
-        {
-            DataPosFrac += increment;
-            DataPosInt  += DataPosFrac>>FRACTIONBITS;
-            DataPosFrac &= FRACTIONMASK;
-        }
-        OutPos += DstBufferSize;
-
-        /* Handle looping sources */
-        while(1)
-        {
-            const ALbuffer *ALBuffer;
-            ALuint DataSize = 0;
-            ALuint LoopStart = 0;
-            ALuint LoopEnd = 0;
-
-            if((ALBuffer=BufferListItem->buffer) != NULL)
-            {
-                DataSize = ALBuffer->SampleLen;
-                LoopStart = ALBuffer->LoopStart;
-                LoopEnd = ALBuffer->LoopEnd;
-                if(LoopEnd > DataPosInt)
-                    break;
-            }
-
-            if(Looping && Source->SourceType == AL_STATIC)
-            {
-                DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
-                break;
-            }
-
-            if(DataSize > DataPosInt)
-                break;
-
-            if(BufferListItem->next)
-            {
-                BufferListItem = BufferListItem->next;
-                BuffersPlayed++;
-            }
-            else if(Looping)
-            {
-                BufferListItem = Source->queue;
-                BuffersPlayed = 0;
-            }
-            else
-            {
-                State = AL_STOPPED;
-                BufferListItem = Source->queue;
-                BuffersPlayed = Source->BuffersInQueue;
-                DataPosInt = 0;
-                DataPosFrac = 0;
-                break;
-            }
-
-            DataPosInt -= DataSize;
-        }
-    } while(State == AL_PLAYING && OutPos < SamplesToDo);
-
-    /* Update source info */
-    Source->state             = State;
-    Source->BuffersPlayed     = BuffersPlayed;
-    Source->position          = DataPosInt;
-    Source->position_fraction = DataPosFrac;
-    Source->Hrtf.Offset      += OutPos;
-    if(State == AL_PLAYING)
-        Source->Hrtf.Counter = maxu(Source->Hrtf.Counter, OutPos) - OutPos;
-    else
-    {
-        Source->Hrtf.Counter = 0;
-        Source->Hrtf.Moving  = AL_FALSE;
-    }
-}

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/mixer_c.c

@@ -1,131 +0,0 @@
-#include "config.h"
-
-#include <assert.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "alSource.h"
-#include "alAuxEffectSlot.h"
-
-
-static __inline ALfloat point32(const ALfloat *vals, ALuint frac)
-{ return vals[0]; (void)frac; }
-static __inline ALfloat lerp32(const ALfloat *vals, ALuint frac)
-{ return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); }
-static __inline ALfloat cubic32(const ALfloat *vals, ALuint frac)
-{ return cubic(vals[-1], vals[0], vals[1], vals[2], frac * (1.0f/FRACTIONONE)); }
-
-void Resample_copy32_C(const ALfloat *data, ALuint frac,
-  ALuint increment, ALfloat *RESTRICT OutBuffer, ALuint BufferSize)
-{
-    (void)frac;
-    assert(increment==FRACTIONONE);
-    memcpy(OutBuffer, data, (BufferSize+1)*sizeof(ALfloat));
-}
-
-#define DECL_TEMPLATE(Sampler)                                                \
-void Resample_##Sampler##_C(const ALfloat *data, ALuint frac,                 \
-  ALuint increment, ALfloat *RESTRICT OutBuffer, ALuint BufferSize)           \
-{                                                                             \
-    ALuint pos = 0;                                                           \
-    ALuint i;                                                                 \
-                                                                              \
-    for(i = 0;i < BufferSize+1;i++)                                           \
-    {                                                                         \
-        OutBuffer[i] = Sampler(data + pos, frac);                             \
-                                                                              \
-        frac += increment;                                                    \
-        pos  += frac>>FRACTIONBITS;                                           \
-        frac &= FRACTIONMASK;                                                 \
-    }                                                                         \
-}
-
-DECL_TEMPLATE(point32)
-DECL_TEMPLATE(lerp32)
-DECL_TEMPLATE(cubic32)
-
-#undef DECL_TEMPLATE
-
-
-static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                     const ALuint IrSize,
-                                     ALfloat (*RESTRICT Coeffs)[2],
-                                     const ALfloat (*RESTRICT CoeffStep)[2],
-                                     ALfloat left, ALfloat right)
-{
-    ALuint c;
-    for(c = 0;c < IrSize;c++)
-    {
-        const ALuint off = (Offset+c)&HRIR_MASK;
-        Values[off][0] += Coeffs[c][0] * left;
-        Values[off][1] += Coeffs[c][1] * right;
-        Coeffs[c][0] += CoeffStep[c][0];
-        Coeffs[c][1] += CoeffStep[c][1];
-    }
-}
-
-static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                 const ALuint IrSize,
-                                 ALfloat (*RESTRICT Coeffs)[2],
-                                 ALfloat left, ALfloat right)
-{
-    ALuint c;
-    for(c = 0;c < IrSize;c++)
-    {
-        const ALuint off = (Offset+c)&HRIR_MASK;
-        Values[off][0] += Coeffs[c][0] * left;
-        Values[off][1] += Coeffs[c][1] * right;
-    }
-}
-
-#define SUFFIX C
-#include "mixer_inc.c"
-#undef SUFFIX
-
-
-void MixDirect_C(const DirectParams *params, const ALfloat *RESTRICT data, ALuint srcchan,
-  ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
-{
-    ALfloat (*RESTRICT DryBuffer)[BUFFERSIZE] = params->OutBuffer;
-    ALfloat *RESTRICT ClickRemoval = params->ClickRemoval;
-    ALfloat *RESTRICT PendingClicks = params->PendingClicks;
-    ALfloat DrySend;
-    ALuint pos;
-    ALuint c;
-
-    for(c = 0;c < MaxChannels;c++)
-    {
-        DrySend = params->Gains[srcchan][c];
-        if(DrySend < 0.00001f)
-            continue;
-
-        if(OutPos == 0)
-            ClickRemoval[c] -= data[0]*DrySend;
-        for(pos = 0;pos < BufferSize;pos++)
-            DryBuffer[c][OutPos+pos] += data[pos]*DrySend;
-        if(OutPos+pos == SamplesToDo)
-            PendingClicks[c] += data[pos]*DrySend;
-    }
-}
-
-
-void MixSend_C(const SendParams *params, const ALfloat *RESTRICT data,
-  ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
-{
-    ALeffectslot *Slot = params->Slot;
-    ALfloat (*RESTRICT WetBuffer)[BUFFERSIZE] = Slot->WetBuffer;
-    ALfloat *RESTRICT WetClickRemoval = Slot->ClickRemoval;
-    ALfloat *RESTRICT WetPendingClicks = Slot->PendingClicks;
-    ALfloat  WetSend = params->Gain;
-    ALuint pos;
-
-    if(WetSend < 0.00001f)
-        return;
-
-    if(OutPos == 0)
-        WetClickRemoval[0] -= data[0] * WetSend;
-    for(pos = 0;pos < BufferSize;pos++)
-        WetBuffer[0][OutPos+pos] += data[pos] * WetSend;
-    if(OutPos+pos == SamplesToDo)
-        WetPendingClicks[0] += data[pos] * WetSend;
-}

Alc/mixer_defs.h

@@ -1,31 +0,0 @@
-#ifndef MIXER_DEFS_H
-#define MIXER_DEFS_H
-
-#include "AL/alc.h"
-#include "AL/al.h"
-#include "alMain.h"
-
-struct DirectParams;
-struct SendParams;
-
-/* C resamplers */
-void Resample_copy32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
-void Resample_point32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
-void Resample_lerp32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
-void Resample_cubic32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
-
-
-/* C mixers */
-void MixDirect_Hrtf_C(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
-void MixDirect_C(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
-void MixSend_C(const struct SendParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint);
-
-/* SSE mixers */
-void MixDirect_Hrtf_SSE(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
-void MixDirect_SSE(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
-void MixSend_SSE(const struct SendParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint);
-
-/* Neon mixers */
-void MixDirect_Hrtf_Neon(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
-
-#endif /* MIXER_DEFS_H */

Alc/mixer_inc.c

@@ -1,141 +0,0 @@
-#include "config.h"
-
-#include "alMain.h"
-#include "alSource.h"
-#include "mixer_defs.h"
-
-#ifdef __GNUC__
-#define LIKELY(x) __builtin_expect(!!(x), 1)
-#define UNLIKELY(x) __builtin_expect(!!(x), 0)
-#else
-#define LIKELY(x) (x)
-#define UNLIKELY(x) (x)
-#endif
-
-#define REAL_MERGE2(a,b) a##b
-#define MERGE2(a,b) REAL_MERGE2(a,b)
-
-#define MixDirect_Hrtf MERGE2(MixDirect_Hrtf_,SUFFIX)
-
-
-static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                     const ALuint irSize,
-                                     ALfloat (*RESTRICT Coeffs)[2],
-                                     const ALfloat (*RESTRICT CoeffStep)[2],
-                                     ALfloat left, ALfloat right);
-static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                 const ALuint irSize,
-                                 ALfloat (*RESTRICT Coeffs)[2],
-                                 ALfloat left, ALfloat right);
-
-
-void MixDirect_Hrtf(const DirectParams *params, const ALfloat *RESTRICT data, ALuint srcchan,
-  ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
-{
-    ALfloat (*RESTRICT DryBuffer)[BUFFERSIZE] = params->OutBuffer;
-    ALfloat *RESTRICT ClickRemoval = params->ClickRemoval;
-    ALfloat *RESTRICT PendingClicks = params->PendingClicks;
-    const ALuint IrSize = params->Hrtf.Params.IrSize;
-    const ALint *RESTRICT DelayStep = params->Hrtf.Params.DelayStep;
-    const ALfloat (*RESTRICT CoeffStep)[2] = params->Hrtf.Params.CoeffStep;
-    const ALfloat (*RESTRICT TargetCoeffs)[2] = params->Hrtf.Params.Coeffs[srcchan];
-    const ALuint *RESTRICT TargetDelay = params->Hrtf.Params.Delay[srcchan];
-    ALfloat *RESTRICT History = params->Hrtf.State->History[srcchan];
-    ALfloat (*RESTRICT Values)[2] = params->Hrtf.State->Values[srcchan];
-    ALint Counter = maxu(params->Hrtf.State->Counter, OutPos) - OutPos;
-    ALuint Offset = params->Hrtf.State->Offset + OutPos;
-    ALIGN(16) ALfloat Coeffs[HRIR_LENGTH][2];
-    ALuint Delay[2];
-    ALfloat left, right;
-    ALuint pos;
-    ALuint c;
-
-    pos = 0;
-    for(c = 0;c < IrSize;c++)
-    {
-        Coeffs[c][0] = TargetCoeffs[c][0] - (CoeffStep[c][0]*Counter);
-        Coeffs[c][1] = TargetCoeffs[c][1] - (CoeffStep[c][1]*Counter);
-    }
-
-    Delay[0] = TargetDelay[0] - (DelayStep[0]*Counter);
-    Delay[1] = TargetDelay[1] - (DelayStep[1]*Counter);
-
-    if(LIKELY(OutPos == 0))
-    {
-        History[Offset&SRC_HISTORY_MASK] = data[pos];
-        left  = lerp(History[(Offset-(Delay[0]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
-                     History[(Offset-(Delay[0]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
-                     (Delay[0]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
-        right = lerp(History[(Offset-(Delay[1]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
-                     History[(Offset-(Delay[1]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
-                     (Delay[1]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
-
-        ClickRemoval[FrontLeft]  -= Values[(Offset+1)&HRIR_MASK][0] +
-                                    Coeffs[0][0] * left;
-        ClickRemoval[FrontRight] -= Values[(Offset+1)&HRIR_MASK][1] +
-                                    Coeffs[0][1] * right;
-    }
-    for(pos = 0;pos < BufferSize && Counter > 0;pos++)
-    {
-        History[Offset&SRC_HISTORY_MASK] = data[pos];
-        left  = lerp(History[(Offset-(Delay[0]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
-                     History[(Offset-(Delay[0]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
-                     (Delay[0]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
-        right = lerp(History[(Offset-(Delay[1]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
-                     History[(Offset-(Delay[1]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
-                     (Delay[1]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
-
-        Delay[0] += DelayStep[0];
-        Delay[1] += DelayStep[1];
-
-        Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f;
-        Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f;
-        Offset++;
-
-        ApplyCoeffsStep(Offset, Values, IrSize, Coeffs, CoeffStep, left, right);
-        DryBuffer[FrontLeft][OutPos]  += Values[Offset&HRIR_MASK][0];
-        DryBuffer[FrontRight][OutPos] += Values[Offset&HRIR_MASK][1];
-
-        OutPos++;
-        Counter--;
-    }
-
-    Delay[0] >>= HRTFDELAY_BITS;
-    Delay[1] >>= HRTFDELAY_BITS;
-    for(;pos < BufferSize;pos++)
-    {
-        History[Offset&SRC_HISTORY_MASK] = data[pos];
-        left = History[(Offset-Delay[0])&SRC_HISTORY_MASK];
-        right = History[(Offset-Delay[1])&SRC_HISTORY_MASK];
-
-        Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f;
-        Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f;
-        Offset++;
-
-        ApplyCoeffs(Offset, Values, IrSize, Coeffs, left, right);
-        DryBuffer[FrontLeft][OutPos]  += Values[Offset&HRIR_MASK][0];
-        DryBuffer[FrontRight][OutPos] += Values[Offset&HRIR_MASK][1];
-
-        OutPos++;
-    }
-    if(LIKELY(OutPos == SamplesToDo))
-    {
-        History[Offset&SRC_HISTORY_MASK] = data[pos];
-        left = History[(Offset-Delay[0])&SRC_HISTORY_MASK];
-        right = History[(Offset-Delay[1])&SRC_HISTORY_MASK];
-
-        PendingClicks[FrontLeft]  += Values[(Offset+1)&HRIR_MASK][0] +
-                                     Coeffs[0][0] * left;
-        PendingClicks[FrontRight] += Values[(Offset+1)&HRIR_MASK][1] +
-                                     Coeffs[0][1] * right;
-    }
-}
-
-
-#undef MixDirect_Hrtf
-
-#undef MERGE2
-#undef REAL_MERGE2
-
-#undef UNLIKELY
-#undef LIKELY

Alc/mixer_neon.c

@@ -1,61 +0,0 @@
-#include "config.h"
-
-#ifdef HAVE_ARM_NEON_H
-#include <arm_neon.h>
-#endif
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alu.h"
-
-
-static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                     const ALuint IrSize,
-                                     ALfloat (*RESTRICT Coeffs)[2],
-                                     const ALfloat (*RESTRICT CoeffStep)[2],
-                                     ALfloat left, ALfloat right)
-{
-    ALuint c;
-    for(c = 0;c < IrSize;c++)
-    {
-        const ALuint off = (Offset+c)&HRIR_MASK;
-        Values[off][0] += Coeffs[c][0] * left;
-        Values[off][1] += Coeffs[c][1] * right;
-        Coeffs[c][0] += CoeffStep[c][0];
-        Coeffs[c][1] += CoeffStep[c][1];
-    }
-}
-
-static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                 const ALuint IrSize,
-                                 ALfloat (*RESTRICT Coeffs)[2],
-                                 ALfloat left, ALfloat right)
-{
-    ALuint 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);
-    }
-    for(c = 0;c < IrSize;c += 2)
-    {
-        const ALuint o0 = (Offset+c)&HRIR_MASK;
-        const ALuint 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 SUFFIX Neon
-#include "mixer_inc.c"
-#undef SUFFIX

Alc/mixer_sse.c

@@ -1,204 +0,0 @@
-#include "config.h"
-
-#ifdef HAVE_XMMINTRIN_H
-#include <xmmintrin.h>
-#endif
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alu.h"
-
-#include "alSource.h"
-#include "alAuxEffectSlot.h"
-#include "mixer_defs.h"
-
-
-static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                     const ALuint IrSize,
-                                     ALfloat (*RESTRICT Coeffs)[2],
-                                     const ALfloat (*RESTRICT CoeffStep)[2],
-                                     ALfloat left, ALfloat right)
-{
-    const __m128 lrlr = { left, right, left, right };
-    __m128 coeffs, deltas, imp0, imp1;
-    __m128 vals = _mm_setzero_ps();
-    ALuint i;
-
-    if((Offset&1))
-    {
-        const ALuint o0 = Offset&HRIR_MASK;
-        const ALuint o1 = (Offset+IrSize-1)&HRIR_MASK;
-
-        coeffs = _mm_load_ps(&Coeffs[0][0]);
-        deltas = _mm_load_ps(&CoeffStep[0][0]);
-        vals = _mm_loadl_pi(vals, (__m64*)&Values[o0][0]);
-        imp0 = _mm_mul_ps(lrlr, coeffs);
-        coeffs = _mm_add_ps(coeffs, deltas);
-        vals = _mm_add_ps(imp0, vals);
-        _mm_store_ps(&Coeffs[0][0], coeffs);
-        _mm_storel_pi((__m64*)&Values[o0][0], vals);
-        for(i = 1;i < IrSize-1;i += 2)
-        {
-            const ALuint o2 = (Offset+i)&HRIR_MASK;
-
-            coeffs = _mm_load_ps(&Coeffs[i+1][0]);
-            deltas = _mm_load_ps(&CoeffStep[i+1][0]);
-            vals = _mm_load_ps(&Values[o2][0]);
-            imp1 = _mm_mul_ps(lrlr, coeffs);
-            coeffs = _mm_add_ps(coeffs, deltas);
-            imp0 = _mm_shuffle_ps(imp0, imp1, _MM_SHUFFLE(1, 0, 3, 2));
-            vals = _mm_add_ps(imp0, vals);
-            _mm_store_ps(&Coeffs[i+1][0], coeffs);
-            _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 ALuint o = (Offset + i)&HRIR_MASK;
-
-            coeffs = _mm_load_ps(&Coeffs[i][0]);
-            deltas = _mm_load_ps(&CoeffStep[i][0]);
-            vals = _mm_load_ps(&Values[o][0]);
-            imp0 = _mm_mul_ps(lrlr, coeffs);
-            coeffs = _mm_add_ps(coeffs, deltas);
-            vals = _mm_add_ps(imp0, vals);
-            _mm_store_ps(&Coeffs[i][0], coeffs);
-            _mm_store_ps(&Values[o][0], vals);
-        }
-    }
-}
-
-static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
-                                 const ALuint IrSize,
-                                 ALfloat (*RESTRICT Coeffs)[2],
-                                 ALfloat left, ALfloat right)
-{
-    const __m128 lrlr = { left, right, left, right };
-    __m128 vals = _mm_setzero_ps();
-    __m128 coeffs;
-    ALuint i;
-
-    if((Offset&1))
-    {
-        const ALuint o0 = Offset&HRIR_MASK;
-        const ALuint 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 ALuint 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 ALuint 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 SUFFIX SSE
-#include "mixer_inc.c"
-#undef SUFFIX
-
-
-void MixDirect_SSE(const DirectParams *params, const ALfloat *RESTRICT data, ALuint srcchan,
-  ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
-{
-    ALfloat (*RESTRICT DryBuffer)[BUFFERSIZE] = params->OutBuffer;
-    ALfloat *RESTRICT ClickRemoval = params->ClickRemoval;
-    ALfloat *RESTRICT PendingClicks = params->PendingClicks;
-    ALfloat DrySend;
-    ALuint pos;
-    ALuint c;
-
-    for(c = 0;c < MaxChannels;c++)
-    {
-        __m128 gain;
-
-        DrySend = params->Gains[srcchan][c];
-        if(DrySend < 0.00001f)
-            continue;
-
-        if(OutPos == 0)
-            ClickRemoval[c] -= data[0]*DrySend;
-
-        gain = _mm_set1_ps(DrySend);
-        for(pos = 0;pos < BufferSize-3;pos += 4)
-        {
-            const __m128 val4 = _mm_load_ps(&data[pos]);
-            __m128 dry4 = _mm_load_ps(&DryBuffer[c][OutPos+pos]);
-            dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain));
-            _mm_store_ps(&DryBuffer[c][OutPos+pos], dry4);
-        }
-        for(;pos < BufferSize;pos++)
-            DryBuffer[c][OutPos+pos] += data[pos]*DrySend;
-
-        if(OutPos+pos == SamplesToDo)
-            PendingClicks[c] += data[pos]*DrySend;
-    }
-}
-
-
-void MixSend_SSE(const SendParams *params, const ALfloat *RESTRICT data,
-  ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
-{
-    ALeffectslot *Slot = params->Slot;
-    ALfloat (*RESTRICT WetBuffer)[BUFFERSIZE] = Slot->WetBuffer;
-    ALfloat *RESTRICT WetClickRemoval = Slot->ClickRemoval;
-    ALfloat *RESTRICT WetPendingClicks = Slot->PendingClicks;
-    const ALfloat WetGain = params->Gain;
-    __m128 gain;
-    ALuint pos;
-
-    if(WetGain < 0.00001f)
-        return;
-
-    if(OutPos == 0)
-        WetClickRemoval[0] -= data[0] * WetGain;
-
-    gain = _mm_set1_ps(WetGain);
-    for(pos = 0;pos < BufferSize-3;pos+=4)
-    {
-        const __m128 val4 = _mm_load_ps(&data[pos]);
-        __m128 wet4 = _mm_load_ps(&WetBuffer[0][OutPos+pos]);
-        wet4 = _mm_add_ps(wet4, _mm_mul_ps(val4, gain));
-        _mm_store_ps(&WetBuffer[0][OutPos+pos], wet4);
-    }
-    for(;pos < BufferSize;pos++)
-        WetBuffer[0][OutPos+pos] += data[pos] * WetGain;
-
-    if(OutPos+pos == SamplesToDo)
-        WetPendingClicks[0] += data[pos] * WetGain;
-}

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 */

COPYING

@@ -1,17 +1,15 @@
- 
-		  GNU LIBRARY GENERAL PUBLIC LICENSE
-		       Version 2, June 1991
-
+                  GNU LIBRARY GENERAL PUBLIC LICENSE
+                       Version 2, June 1991
 
  Copyright (C) 1991 Free Software Foundation, Inc.
-                    675 Mass Ave, Cambridge, MA 02139, USA
+ 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
  Everyone is permitted to copy and distribute verbatim copies
  of this license document, but changing it is not allowed.
 
 [This is the first released version of the library GPL.  It is
  numbered 2 because it goes with version 2 of the ordinary GPL.]
 
-			    Preamble
+                            Preamble
 
   The licenses for most software are designed to take away your
 freedom to share and change it.  By contrast, the GNU General Public
@@ -53,7 +51,7 @@ library.  If the library is modified by someone else and passed on, we
 want its recipients to know that what they have is not the original
 version, so that any problems introduced by others will not reflect on
 the original authors' reputations.
-
+
   Finally, any free program is threatened constantly by software
 patents.  We wish to avoid the danger that companies distributing free
 software will individually obtain patent licenses, thus in effect
@@ -100,8 +98,8 @@ works together with the library.
 
   Note that it is possible for a library to be covered by the ordinary
 General Public License rather than by this special one.
-
-		  GNU LIBRARY GENERAL PUBLIC LICENSE
+
+                  GNU LIBRARY GENERAL PUBLIC LICENSE
    TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
 
   0. This License Agreement applies to any software library which
@@ -147,7 +145,7 @@ Library.
   You may charge a fee for the physical act of transferring a copy,
 and you may at your option offer warranty protection in exchange for a
 fee.
-
+
   2. You may modify your copy or copies of the Library or any portion
 of it, thus forming a work based on the Library, and copy and
 distribute such modifications or work under the terms of Section 1
@@ -205,7 +203,7 @@ instead of to this License.  (If a newer version than version 2 of the
 ordinary GNU General Public License has appeared, then you can specify
 that version instead if you wish.)  Do not make any other change in
 these notices.
-
+
   Once this change is made in a given copy, it is irreversible for
 that copy, so the ordinary GNU General Public License applies to all
 subsequent copies and derivative works made from that copy.
@@ -256,7 +254,7 @@ Library will still fall under Section 6.)
 distribute the object code for the work under the terms of Section 6.
 Any executables containing that work also fall under Section 6,
 whether or not they are linked directly with the Library itself.
-
+
   6. As an exception to the Sections above, you may also compile or
 link a "work that uses the Library" with the Library to produce a
 work containing portions of the Library, and distribute that work
@@ -310,7 +308,7 @@ restrictions of other proprietary libraries that do not normally
 accompany the operating system.  Such a contradiction means you cannot
 use both them and the Library together in an executable that you
 distribute.
-
+
   7. You may place library facilities that are a work based on the
 Library side-by-side in a single library together with other library
 facilities not covered by this License, and distribute such a combined
@@ -351,7 +349,7 @@ subject to these terms and conditions.  You may not impose any further
 restrictions on the recipients' exercise of the rights granted herein.
 You are not responsible for enforcing compliance by third parties to
 this License.
-
+
   11. If, as a consequence of a court judgment or allegation of patent
 infringement or for any other reason (not limited to patent issues),
 conditions are imposed on you (whether by court order, agreement or
@@ -403,7 +401,7 @@ conditions either of that version or of any later version published by
 the Free Software Foundation.  If the Library does not specify a
 license version number, you may choose any version ever published by
 the Free Software Foundation.
-
+
   14. If you wish to incorporate parts of the Library into other free
 programs whose distribution conditions are incompatible with these,
 write to the author to ask for permission.  For software which is
@@ -413,7 +411,7 @@ decision will be guided by the two goals of preserving the free status
 of all derivatives of our free software and of promoting the sharing
 and reuse of software generally.
 
-			    NO WARRANTY
+                            NO WARRANTY
 
   15. BECAUSE THE LIBRARY IS LICENSED FREE OF CHARGE, THERE IS NO
 WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW.
@@ -436,49 +434,4 @@ FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF
 SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
 DAMAGES.
 
-		     END OF TERMS AND CONDITIONS
-
-     Appendix: How to Apply These Terms to Your New Libraries
-
-  If you develop a new library, and you want it to be of the greatest
-possible use to the public, we recommend making it free software that
-everyone can redistribute and change.  You can do so by permitting
-redistribution under these terms (or, alternatively, under the terms of the
-ordinary General Public License).
-
-  To apply these terms, attach the following notices to the library.  It is
-safest to attach them to the start of each source file to most effectively
-convey the exclusion of warranty; and each file should have at least the
-"copyright" line and a pointer to where the full notice is found.
-
-    <one line to give the library's name and a brief idea of what it does.>
-    Copyright (C) <year>  <name of author>
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
-
-Also add information on how to contact you by electronic and paper mail.
-
-You should also get your employer (if you work as a programmer) or your
-school, if any, to sign a "copyright disclaimer" for the library, if
-necessary.  Here is a sample; alter the names:
-
-  Yoyodyne, Inc., hereby disclaims all copyright interest in the
-  library `Frob' (a library for tweaking knobs) written by James Random Hacker.
-
-  <signature of Ty Coon>, 1 April 1990
-  Ty Coon, President of Vice
-
-That's all there is to it!
-
+                     END OF TERMS AND CONDITIONS

ChangeLog

@@ -0,0 +1,372 @@
+openal-soft-1.19.1:
+
+    Implemented capture support for the SoundIO backend.
+
+    Fixed source buffer queues potentially not playing properly when a queue
+    entry completes.
+
+    Fixed possible unexpected failures when generating auxiliary effect slots.
+
+    Fixed a crash with certain reverb or device settings.
+
+    Fixed OpenSL capture.
+
+    Improved output limiter response, better ensuring the sample amplitude is
+    clamped for output.
+
+openal-soft-1.19.0:
+
+    Implemented the ALC_SOFT_device_clock extension.
+
+    Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
+
+    Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
+
+    Fixed compiling on NetBSD.
+
+    Fixed the reverb effect's density scale and panning parameters.
+
+    Fixed use of the WASAPI backend with certain games, which caused odd COM
+    initialization errors.
+
+    Increased the number of virtual channels for decoding Ambisonics to HRTF
+    output.
+
+    Changed 32-bit x86 builds to use SSE2 math by default for performance.
+    Build-time options are available to use just SSE1 or x87 instead.
+
+    Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
+
+    Renamed the MMDevAPI backend to WASAPI.
+
+    Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
+    has been updated to 24-bit.
+
+    Added a 24- to 48-point band-limited Sinc resampler.
+
+    Added an SDL2 playback backend. Disabled by default to avoid a dependency
+    on SDL2.
+
+    Improved the performance and quality of the Chorus and Flanger effects.
+
+    Improved the efficiency of the band-limited Sinc resampler.
+
+    Improved the Sinc resampler's transition band to avoid over-attenuating
+    higher frequencies.
+
+    Improved the performance of some filter operations.
+
+    Improved the efficiency of object ID lookups.
+
+    Improved the efficienty of internal voice/source synchronization.
+
+    Improved AL call error logging with contextualized messages.
+
+    Removed the reverb effect's modulation stage. Due to the lack of reference
+    for its intended behavior and strength.
+
+openal-soft-1.18.2:
+
+    Fixed resetting the FPU rounding mode after certain function calls on
+    Windows.
+
+    Fixed use of SSE intrinsics when building with Clang on Windows.
+
+    Fixed a crash with the JACK backend when using JACK1.
+
+    Fixed use of pthread_setnane_np on NetBSD.
+
+    Fixed building on FreeBSD with an older freebsd-lib.
+
+    OSS now links with libossaudio if found at build time (for NetBSD).
+
+openal-soft-1.18.1:
+
+    Fixed an issue where resuming a source might not restart playing it.
+
+    Fixed PulseAudio playback when the configured stream length is much less
+    than the requested length.
+
+    Fixed MMDevAPI capture with sample rates not matching the backing device.
+
+    Fixed int32 output for the Wave Writer.
+
+    Fixed enumeration of OSS devices that are missing device files.
+
+    Added correct retrieval of the executable's path on FreeBSD.
+
+    Added a config option to specify the dithering depth.
+
+    Added a 5.1 decoder preset that excludes front-center output.
+
+openal-soft-1.18.0:
+
+    Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
+
+    Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
+    AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
+
+    Implemented 3D processing for some effects. Currently implemented for
+    Reverb, Compressor, Equalizer, and Ring Modulator.
+
+    Implemented 2-channel UHJ output encoding. This needs to be enabled with a
+    config option to be used.
+
+    Implemented dual-band processing for high-quality ambisonic decoding.
+
+    Implemented distance-compensation for surround sound output.
+
+    Implemented near-field emulation and compensation with ambisonic rendering.
+    Currently only applies when using the high-quality ambisonic decoder or
+    ambisonic output, with appropriate config options.
+
+    Implemented an output limiter to reduce the amount of distortion from
+    clipping.
+
+    Implemented dithering for 8-bit and 16-bit output.
+
+    Implemented a config option to select a preferred HRTF.
+
+    Implemented a run-time check for NEON extensions using /proc/cpuinfo.
+
+    Implemented experimental capture support for the OpenSL backend.
+
+    Fixed building on compilers with NEON support but don't default to having
+    NEON enabled.
+
+    Fixed support for JACK on Windows.
+
+    Fixed starting a source while alcSuspendContext is in effect.
+
+    Fixed detection of headsets as headphones, with MMDevAPI.
+
+    Added support for AmbDec config files, for custom ambisonic decoder
+    configurations. Version 3 files only.
+
+    Added backend-specific options to alsoft-config.
+
+    Added first-, second-, and third-order ambisonic output formats. Currently
+    only works with backends that don't rely on channel labels, like JACK,
+    ALSA, and OSS.
+
+    Added a build option to embed the default HRTFs into the lib.
+
+    Added AmbDec presets to enable high-quality ambisonic decoding.
+
+    Added an AmbDec preset for 3D7.1 speaker setups.
+
+    Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
+    the provided ambdec presets.
+
+    Added the ability for MMDevAPI to open devices given a Device ID or GUID
+    string.
+
+    Added an option to the example apps to open a specific device.
+
+    Increased the maximum auxiliary send limit to 16 (up from 4). Requires
+    requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
+    attribute.
+
+    Increased the default auxiliary effect slot count to 64 (up from 4).
+
+    Reduced the default period count to 3 (down from 4).
+
+    Slightly improved automatic naming for enumerated HRTFs.
+
+    Improved B-Format decoding with HRTF output.
+
+    Improved internal property handling for better batching behavior.
+
+    Improved performance of certain filter uses.
+
+    Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
+    extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
+
+openal-soft-1.17.2:
+
+    Implemented device enumeration for OSSv4.
+
+    Fixed building on OSX.
+
+    Fixed building on non-Windows systems without POSIX-2008.
+
+    Fixed Dedicated Dialog and Dedicated LFE effect output.
+
+    Added a build option to override the share install dir.
+
+    Added a build option to static-link libgcc for MinGW.
+
+openal-soft-1.17.1:
+
+    Fixed building with JACK and without PulseAudio.
+
+    Fixed building on FreeBSD.
+
+    Fixed the ALSA backend's allow-resampler option.
+
+    Fixed handling of inexact ALSA period counts.
+
+    Altered device naming scheme on Windows backends to better match other
+    drivers.
+
+    Updated the CoreAudio backend to use the AudioComponent API. This clears up
+    deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
+
+openal-soft-1.17.0:
+
+    Implemented a JACK playback backend.
+
+    Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
+
+    Implemented the ALC_SOFT_HRTF extension.
+
+    Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
+
+    Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
+    24-point Sinc resampling, and performs anti-aliasing.
+
+    Implemented B-Format output support for the wave file writer. This creates
+    FuMa-style first-order Ambisonics wave files (AMB format).
+
+    Implemented a stereo-mode config option for treating stereo modes as either
+    speakers or headphones.
+
+    Implemented per-device configuration options.
+
+    Fixed handling of PulseAudio and MMDevAPI devices that have identical
+    descriptions.
+
+    Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
+
+    Fixed logging of Unicode characters on Windows.
+
+    Fixed 5.1 surround sound channels. By default it will now use the side
+    channels for the surround output. A configuration using rear channels is
+    still available.
+
+    Fixed the QSA backend potentially altering the capture format.
+
+    Fixed detecting MMDevAPI's default device.
+
+    Fixed returning the default capture device name.
+
+    Fixed mixing property calculations when deferring context updates.
+
+    Altered the behavior of alcSuspendContext and alcProcessContext to better
+    match certain Windows drivers.
+
+    Altered the panning algorithm, utilizing Ambisonics for better side and
+    back positioning cues with surround sound output.
+
+    Improved support for certain older Windows apps.
+
+    Improved the alffplay example to support surround sound streams.
+
+    Improved support for building as a sub-project.
+
+    Added an HRTF playback example.
+
+    Added a tone generator output test.
+
+    Added a toolchain to help with cross-compiling to Android.
+
+openal-soft-1.16.0:
+
+    Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
+    effects.
+
+    Implemented high-pass and band-pass EFX filters.
+
+    Implemented the high-pass filter for the EAXReverb effect.
+
+    Implemented SSE2 and SSE4.1 linear resamplers.
+
+    Implemented Neon-enhanced non-HRTF mixers.
+
+    Implemented a QSA backend, for QNX.
+
+    Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
+    AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
+    extensions.
+
+    Fixed resetting mmdevapi backend devices.
+
+    Fixed clamping when converting 32-bit float samples to integer.
+
+    Fixed modulation range in the Modulator effect.
+
+    Several fixes for the OpenSL playback backend.
+
+    Fixed device specifier names that have Unicode characters on Windows.
+
+    Added support for filenames and paths with Unicode (UTF-8) characters on
+    Windows.
+
+    Added support for alsoft.conf config files found in XDG Base Directory
+    Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
+    defaults) on non-Windows systems.
+
+    Added a GUI configuration utility (requires Qt 4.8).
+
+    Added support for environment variable expansion in config options (not
+    keys or section names).
+
+    Added an example that uses SDL2 and ffmpeg.
+
+    Modified examples to use SDL_sound.
+
+    Modified CMake config option names for better sorting.
+
+    HRTF data sets specified in the hrtf_tables config option may now be
+    relative or absolute filenames.
+
+    Made the default HRTF data set an external file, and added a data set for
+    48khz playback in addition to 44.1khz.
+
+    Added support for C11 atomic methods.
+
+    Improved support for some non-GNU build systems.
+
+openal-soft-1.15.1:
+
+    Fixed a regression with retrieving the source's AL_GAIN property.
+
+openal-soft-1.15:
+
+    Fixed device enumeration with the OSS backend.
+
+    Reorganized internal mixing logic, so unneeded steps can potentially be
+    skipped for better performance.
+
+    Removed the lookup table for calculating the mixing pans. The panning is
+    now calculated directly for better precision.
+
+    Improved the panning of stereo source channels when using stereo output.
+
+    Improved source filter quality on send paths.
+
+    Added a config option to allow PulseAudio to move streams between devices.
+
+    The PulseAudio backend will now attempt to spawn a server by default.
+
+    Added a workaround for a DirectSound bug relating to float32 output.
+
+    Added SSE-based mixers, for HRTF and non-HRTF mixing.
+
+    Added support for the new AL_SOFT_source_latency extension.
+
+    Improved ALSA capture by avoiding an extra buffer when using sizes
+    supported by the underlying device.
+
+    Improved the makehrtf utility to support new options and input formats.
+
+    Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
+    the header includes can optionally be omitted.
+
+    Added a couple example code programs to show how to apply reverb, and
+    retrieve latency.
+
+    The configuration sample is now installed into the share/openal/ directory
+    instead of /etc/openal.
+
+    The configuration sample now gets installed by default.

OpenAL32/Include/alAuxEffectSlot.h

@@ -4,55 +4,179 @@
 #include "alMain.h"
 #include "alEffect.h"
 
+#include "atomic.h"
+#include "align.h"
+
 #ifdef __cplusplus
 extern "C" {
 #endif
 
+struct ALeffectStateVtable;
+struct ALeffectslot;
+
 typedef struct ALeffectState {
-    ALvoid (*Destroy)(struct ALeffectState *State);
-    ALboolean (*DeviceUpdate)(struct ALeffectState *State, ALCdevice *Device);
-    ALvoid (*Update)(struct ALeffectState *State, ALCdevice *Device, const struct ALeffectslot *Slot);
-    ALvoid (*Process)(struct ALeffectState *State, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE]);
+    RefCount Ref;
+    const struct ALeffectStateVtable *vtbl;
+
+    ALfloat (*OutBuffer)[BUFFERSIZE];
+    ALsizei OutChannels;
 } ALeffectState;
 
+void ALeffectState_Construct(ALeffectState *state);
+void ALeffectState_Destruct(ALeffectState *state);
 
-typedef struct ALeffectslot
-{
-    ALeffect effect;
+struct ALeffectStateVtable {
+    void (*const Destruct)(ALeffectState *state);
 
-    volatile ALfloat   Gain;
-    volatile ALboolean AuxSendAuto;
+    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);
 
-    volatile ALenum NeedsUpdate;
-    ALeffectState *EffectState;
+    void (*const Delete)(void *ptr);
+};
 
-    ALIGN(16) ALfloat WetBuffer[1][BUFFERSIZE];
+/* Small hack to use a pointer-to-array types as a normal argument type.
+ * Shouldn't be used directly.
+ */
+typedef ALfloat ALfloatBUFFERSIZE[BUFFERSIZE];
 
-    ALfloat ClickRemoval[1];
-    ALfloat PendingClicks[1];
+#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);
 
-ALeffectState *NoneCreate(void);
-ALeffectState *ReverbCreate(void);
-ALeffectState *EchoCreate(void);
-ALeffectState *ModulatorCreate(void);
-ALeffectState *DedicatedCreate(void);
 
-#define ALeffectState_Destroy(a)        ((a)->Destroy((a)))
-#define ALeffectState_DeviceUpdate(a,b) ((a)->DeviceUpdate((a),(b)))
-#define ALeffectState_Update(a,b,c)     ((a)->Update((a),(b),(c)))
-#define ALeffectState_Process(a,b,c,d)  ((a)->Process((a),(b),(c),(d)))
+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);
 
-ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect);
+EffectStateFactory *DedicatedStateFactory_getFactory(void);
+
+
+ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect);
+
+void ALeffectState_DecRef(ALeffectState *state);
 
 #ifdef __cplusplus
 }

OpenAL32/Include/alBuffer.h

@@ -1,7 +1,13 @@
 #ifndef _AL_BUFFER_H_
 #define _AL_BUFFER_H_
 
-#include "alMain.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" {
@@ -9,34 +15,30 @@ extern "C" {
 
 /* User formats */
 enum UserFmtType {
-    UserFmtByte   = AL_BYTE_SOFT,
-    UserFmtUByte  = AL_UNSIGNED_BYTE_SOFT,
-    UserFmtShort  = AL_SHORT_SOFT,
-    UserFmtUShort = AL_UNSIGNED_SHORT_SOFT,
-    UserFmtInt    = AL_INT_SOFT,
-    UserFmtUInt   = AL_UNSIGNED_INT_SOFT,
-    UserFmtFloat  = AL_FLOAT_SOFT,
-    UserFmtDouble = AL_DOUBLE_SOFT,
-    UserFmtByte3  = AL_BYTE3_SOFT,
-    UserFmtUByte3 = AL_UNSIGNED_BYTE3_SOFT,
+    UserFmtUByte,
+    UserFmtShort,
+    UserFmtFloat,
+    UserFmtDouble,
     UserFmtMulaw,
     UserFmtAlaw,
     UserFmtIMA4,
+    UserFmtMSADPCM,
 };
 enum UserFmtChannels {
-    UserFmtMono   = AL_MONO_SOFT,
-    UserFmtStereo = AL_STEREO_SOFT,
-    UserFmtRear   = AL_REAR_SOFT,
-    UserFmtQuad   = AL_QUAD_SOFT,
-    UserFmtX51    = AL_5POINT1_SOFT, /* (WFX order) */
-    UserFmtX61    = AL_6POINT1_SOFT, /* (WFX order) */
-    UserFmtX71    = AL_7POINT1_SOFT, /* (WFX order) */
+    UserFmtMono,
+    UserFmtStereo,
+    UserFmtRear,
+    UserFmtQuad,
+    UserFmtX51, /* (WFX order) */
+    UserFmtX61, /* (WFX order) */
+    UserFmtX71, /* (WFX order) */
+    UserFmtBFormat2D, /* WXY */
+    UserFmtBFormat3D, /* WXYZ */
 };
 
-ALuint BytesFromUserFmt(enum UserFmtType type);
-ALuint ChannelsFromUserFmt(enum UserFmtChannels chans);
-static __inline ALuint FrameSizeFromUserFmt(enum UserFmtChannels chans,
-                                            enum UserFmtType type)
+ALsizei BytesFromUserFmt(enum UserFmtType type);
+ALsizei ChannelsFromUserFmt(enum UserFmtChannels chans);
+inline ALsizei FrameSizeFromUserFmt(enum UserFmtChannels chans, enum UserFmtType type)
 {
     return ChannelsFromUserFmt(chans) * BytesFromUserFmt(type);
 }
@@ -44,9 +46,12 @@ static __inline ALuint FrameSizeFromUserFmt(enum UserFmtChannels chans,
 
 /* Storable formats */
 enum FmtType {
-    FmtByte  = UserFmtByte,
-    FmtShort = UserFmtShort,
-    FmtFloat = UserFmtFloat,
+    FmtUByte  = UserFmtUByte,
+    FmtShort  = UserFmtShort,
+    FmtFloat  = UserFmtFloat,
+    FmtDouble = UserFmtDouble,
+    FmtMulaw  = UserFmtMulaw,
+    FmtAlaw   = UserFmtAlaw,
 };
 enum FmtChannels {
     FmtMono   = UserFmtMono,
@@ -56,39 +61,47 @@ enum FmtChannels {
     FmtX51    = UserFmtX51,
     FmtX61    = UserFmtX61,
     FmtX71    = UserFmtX71,
+    FmtBFormat2D = UserFmtBFormat2D,
+    FmtBFormat3D = UserFmtBFormat3D,
 };
+#define MAX_INPUT_CHANNELS  (8)
 
-ALuint BytesFromFmt(enum FmtType type);
-ALuint ChannelsFromFmt(enum FmtChannels chans);
-static __inline ALuint FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type)
+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
-{
+typedef struct ALbuffer {
     ALvoid  *data;
 
-    ALsizei  Frequency;
-    ALenum   Format;
-    ALsizei  SampleLen;
+    ALsizei Frequency;
+    ALbitfieldSOFT Access;
+    ALsizei SampleLen;
 
     enum FmtChannels FmtChannels;
     enum FmtType     FmtType;
+    ALsizei BytesAlloc;
 
-    enum UserFmtChannels OriginalChannels;
-    enum UserFmtType     OriginalType;
-    ALsizei              OriginalSize;
+    enum UserFmtType OriginalType;
+    ALsizei OriginalSize;
+    ALsizei OriginalAlign;
 
-    ALsizei  LoopStart;
-    ALsizei  LoopEnd;
+    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;
 
-    RWLock lock;
-
     /* Self ID */
     ALuint id;
 } ALbuffer;

OpenAL32/Include/alEffect.h

@@ -7,25 +7,75 @@
 extern "C" {
 #endif
 
+struct ALeffect;
+
 enum {
-    EAXREVERB = 0,
-    REVERB,
-    ECHO,
-    MODULATOR,
-    DEDICATED,
+    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;
-extern ALboolean EmulateEAXReverb;
 
-typedef struct ALeffect
-{
-    // Effect type (AL_EFFECT_NULL, ...)
-    ALenum type;
+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;
@@ -55,6 +105,34 @@ typedef struct ALeffect
         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;
@@ -65,47 +143,68 @@ typedef struct ALeffect
         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;
 
-    void (*SetParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
-    void (*SetParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
-    void (*SetParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
-    void (*SetParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals);
+typedef struct ALeffect {
+    // Effect type (AL_EFFECT_NULL, ...)
+    ALenum type;
 
-    void (*GetParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *val);
-    void (*GetParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals);
-    void (*GetParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
-    void (*GetParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
+    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))
 
-#define ALeffect_SetParami(x, c, p, v)  ((x)->SetParami((x),(c),(p),(v)))
-#define ALeffect_SetParamiv(x, c, p, v) ((x)->SetParamiv((x),(c),(p),(v)))
-#define ALeffect_SetParamf(x, c, p, v)  ((x)->SetParamf((x),(c),(p),(v)))
-#define ALeffect_SetParamfv(x, c, p, v) ((x)->SetParamfv((x),(c),(p),(v)))
-
-#define ALeffect_GetParami(x, c, p, v)  ((x)->GetParami((x),(c),(p),(v)))
-#define ALeffect_GetParamiv(x, c, p, v) ((x)->GetParamiv((x),(c),(p),(v)))
-#define ALeffect_GetParamf(x, c, p, v)  ((x)->GetParamf((x),(c),(p),(v)))
-#define ALeffect_GetParamfv(x, c, p, v) ((x)->GetParamfv((x),(c),(p),(v)))
-
-static __inline ALboolean IsReverbEffect(ALenum type)
+inline ALboolean IsReverbEffect(ALenum type)
 { return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; }
 
-ALenum InitEffect(ALeffect *effect);
-ALvoid ReleaseALEffects(ALCdevice *device);
+void InitEffect(ALeffect *effect);
+void ReleaseALEffects(ALCdevice *device);
 
-ALvoid LoadReverbPreset(const char *name, ALeffect *effect);
+void LoadReverbPreset(const char *name, ALeffect *effect);
 
 #ifdef __cplusplus
 }

OpenAL32/Include/alError.h

@@ -2,6 +2,7 @@
 #define _AL_ERROR_H_
 
 #include "alMain.h"
+#include "logging.h"
 
 #ifdef __cplusplus
 extern "C" {
@@ -9,7 +10,17 @@ extern "C" {
 
 extern ALboolean TrapALError;
 
-ALvoid alSetError(ALCcontext *Context, ALenum errorCode);
+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
 }