/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
#include "Configuration.h"
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
#include <sys/types.h>
#include <utils/Errors.h>
#include <utils/Log.h>
#include <cutils/bitops.h>
#include <cutils/compiler.h>
#include <utils/Debug.h>
#include <system/audio.h>
#include <audio_utils/primitives.h>
#include <common_time/local_clock.h>
#include <common_time/cc_helper.h>
#include <media/EffectsFactoryApi.h>
#include "AudioMixer.h"
namespace android {
// ----------------------------------------------------------------------------
AudioMixer::DownmixerBufferProvider::DownmixerBufferProvider() : AudioBufferProvider(),
mTrackBufferProvider(NULL), mDownmixHandle(NULL)
{
}
AudioMixer::DownmixerBufferProvider::~DownmixerBufferProvider()
{
ALOGV("AudioMixer deleting DownmixerBufferProvider (%p)", this);
EffectRelease(mDownmixHandle);
}
status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
int64_t pts) {
//ALOGV("DownmixerBufferProvider::getNextBuffer()");
if (this->mTrackBufferProvider != NULL) {
status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
if (res == OK) {
mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount;
mDownmixConfig.inputCfg.buffer.raw = pBuffer->raw;
mDownmixConfig.outputCfg.buffer.frameCount = pBuffer->frameCount;
mDownmixConfig.outputCfg.buffer.raw = mDownmixConfig.inputCfg.buffer.raw;
// in-place so overwrite the buffer contents, has been set in prepareTrackForDownmix()
//mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
res = (*mDownmixHandle)->process(mDownmixHandle,
&mDownmixConfig.inputCfg.buffer, &mDownmixConfig.outputCfg.buffer);
//ALOGV("getNextBuffer is downmixing");
}
return res;
} else {
ALOGE("DownmixerBufferProvider::getNextBuffer() error: NULL track buffer provider");
return NO_INIT;
}
}
void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
//ALOGV("DownmixerBufferProvider::releaseBuffer()");
if (this->mTrackBufferProvider != NULL) {
mTrackBufferProvider->releaseBuffer(pBuffer);
} else {
ALOGE("DownmixerBufferProvider::releaseBuffer() error: NULL track buffer provider");
}
}
// ----------------------------------------------------------------------------
bool AudioMixer::isMultichannelCapable = false;
effect_descriptor_t AudioMixer::dwnmFxDesc;
// Ensure mConfiguredNames bitmask is initialized properly on all architectures.
// The value of 1 << x is undefined in C when x >= 32.
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)
: mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1),
mSampleRate(sampleRate)
{
// AudioMixer is not yet capable of multi-channel beyond stereo
COMPILE_TIME_ASSERT_FUNCTION_SCOPE(2 == MAX_NUM_CHANNELS);
ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u",
maxNumTracks, MAX_NUM_TRACKS);
// AudioMixer is not yet capable of more than 32 active track inputs
ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS);
// AudioMixer is not yet capable of multi-channel output beyond stereo
ALOG_ASSERT(2 == MAX_NUM_CHANNELS, "bad MAX_NUM_CHANNELS %d", MAX_NUM_CHANNELS);
LocalClock lc;
pthread_once(&sOnceControl, &sInitRoutine);
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
mState.hook = process__nop;
mState.outputTemp = NULL;
mState.resampleTemp = NULL;
mState.mLog = &mDummyLog;
// mState.reserved
// FIXME Most of the following initialization is probably redundant since
// tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0
// and mTrackNames is initially 0. However, leave it here until that's verified.
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->resampler = NULL;
t->downmixerBufferProvider = NULL;
t++;
}
// find multichannel downmix effect if we have to play multichannel content
uint32_t numEffects = 0;
int ret = EffectQueryNumberEffects(&numEffects);
if (ret != 0) {
ALOGE("AudioMixer() error %d querying number of effects", ret);
return;
}
ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
for (uint32_t i = 0 ; i < numEffects ; i++) {
if (EffectQueryEffect(i, &dwnmFxDesc) == 0) {
ALOGV("effect %d is called %s", i, dwnmFxDesc.name);
if (memcmp(&dwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
ALOGI("found effect \"%s\" from %s",
dwnmFxDesc.name, dwnmFxDesc.implementor);
isMultichannelCapable = true;
break;
}
}
}
ALOGE_IF(!isMultichannelCapable, "unable to find downmix effect");
}
AudioMixer::~AudioMixer()
{
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
delete t->resampler;
delete t->downmixerBufferProvider;
t++;
}
delete [] mState.outputTemp;
delete [] mState.resampleTemp;
}
void AudioMixer::setLog(NBLog::Writer *log)
{
mState.mLog = log;
}
int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
{
uint32_t names = (~mTrackNames) & mConfiguredNames;
if (names != 0) {
int n = __builtin_ctz(names);
ALOGV("add track (%d)", n);
mTrackNames |= 1 << n;
// assume default parameters for the track, except where noted below
track_t* t = &mState.tracks[n];
t->needs = 0;
t->volume[0] = UNITY_GAIN;
t->volume[1] = UNITY_GAIN;
// no initialization needed
// t->prevVolume[0]
// t->prevVolume[1]
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
t->auxLevel = 0;
t->auxInc = 0;
// no initialization needed
// t->prevAuxLevel
// t->frameCount
t->channelCount = 2;
t->enabled = false;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->sessionId = sessionId;
// setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
t->bufferProvider = NULL;
t->buffer.raw = NULL;
// no initialization needed
// t->buffer.frameCount
t->hook = NULL;
t->in = NULL;
t->resampler = NULL;
t->sampleRate = mSampleRate;
// setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t->downmixerBufferProvider = NULL;
status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
if (status == OK) {
return TRACK0 + n;
}
ALOGE("AudioMixer::getTrackName(0x%x) failed, error preparing track for downmix",
channelMask);
}
return -1;
}
void AudioMixer::invalidateState(uint32_t mask)
{
if (mask) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
}
status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask)
{
uint32_t channelCount = popcount(mask);
ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
status_t status = OK;
if (channelCount > MAX_NUM_CHANNELS) {
pTrack->channelMask = mask;
pTrack->channelCount = channelCount;
ALOGV("initTrackDownmix(track=%d, mask=0x%x) calls prepareTrackForDownmix()",
trackNum, mask);
status = prepareTrackForDownmix(pTrack, trackNum);
} else {
unprepareTrackForDownmix(pTrack, trackNum);
}
return status;
}
void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName) {
ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
if (pTrack->downmixerBufferProvider != NULL) {
// this track had previously been configured with a downmixer, delete it
ALOGV(" deleting old downmixer");
pTrack->bufferProvider = pTrack->downmixerBufferProvider->mTrackBufferProvider;
delete pTrack->downmixerBufferProvider;
pTrack->downmixerBufferProvider = NULL;
} else {
ALOGV(" nothing to do, no downmixer to delete");
}
}
status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName)
{
ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask);
// discard the previous downmixer if there was one
unprepareTrackForDownmix(pTrack, trackName);
DownmixerBufferProvider* pDbp = new DownmixerBufferProvider();
int32_t status;
if (!isMultichannelCapable) {
ALOGE("prepareTrackForDownmix(%d) fails: mixer doesn't support multichannel content",
trackName);
goto noDownmixForActiveTrack;
}
if (EffectCreate(&dwnmFxDesc.uuid,
pTrack->sessionId /*sessionId*/, -2 /*ioId not relevant here, using random value*/,
&pDbp->mDownmixHandle/*pHandle*/) != 0) {
ALOGE("prepareTrackForDownmix(%d) fails: error creating downmixer effect", trackName);
goto noDownmixForActiveTrack;
}
// channel input configuration will be overridden per-track
pDbp->mDownmixConfig.inputCfg.channels = pTrack->channelMask;
pDbp->mDownmixConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
pDbp->mDownmixConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
pDbp->mDownmixConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
pDbp->mDownmixConfig.inputCfg.samplingRate = pTrack->sampleRate;
pDbp->mDownmixConfig.outputCfg.samplingRate = pTrack->sampleRate;
pDbp->mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
pDbp->mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
// input and output buffer provider, and frame count will not be used as the downmix effect
// process() function is called directly (see DownmixerBufferProvider::getNextBuffer())
pDbp->mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
pDbp->mDownmixConfig.outputCfg.mask = pDbp->mDownmixConfig.inputCfg.mask;
{// scope for local variables that are not used in goto label "noDownmixForActiveTrack"
int cmdStatus;
uint32_t replySize = sizeof(int);
// Configure and enable downmixer
status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle,
EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
&pDbp->mDownmixConfig /*pCmdData*/,
&replySize /*replySize*/, &cmdStatus /*pReplyData*/);
if ((status != 0) || (cmdStatus != 0)) {
ALOGE("error %d while configuring downmixer for track %d", status, trackName);
goto noDownmixForActiveTrack;
}
replySize = sizeof(int);
status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle,
EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
&replySize /*replySize*/, &cmdStatus /*pReplyData*/);
if ((status != 0) || (cmdStatus != 0)) {
ALOGE("error %d while enabling downmixer for track %d", status, trackName);
goto noDownmixForActiveTrack;
}
// Set downmix type
// parameter size rounded for padding on 32bit boundary
const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int);
const int downmixParamSize =
sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t);
effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize);
param->psize = sizeof(downmix_params_t);
const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE;
memcpy(param->data, &downmixParam, param->psize);
const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD;
param->vsize = sizeof(downmix_type_t);
memcpy(param->data + psizePadded, &downmixType, param->vsize);
status = (*pDbp->mDownmixHandle)->command(pDbp->mDownmixHandle,
EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize/* cmdSize */,
param /*pCmndData*/, &replySize /*replySize*/, &cmdStatus /*pReplyData*/);
free(param);
if ((status != 0) || (cmdStatus != 0)) {
ALOGE("error %d while setting downmix type for track %d", status, trackName);
goto noDownmixForActiveTrack;
} else {
ALOGV("downmix type set to %d for track %d", (int) downmixType, trackName);
}
}// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack"
// initialization successful:
// - keep track of the real buffer provider in case it was set before
pDbp->mTrackBufferProvider = pTrack->bufferProvider;
// - we'll use the downmix effect integrated inside this
// track's buffer provider, and we'll use it as the track's buffer provider
pTrack->downmixerBufferProvider = pDbp;
pTrack->bufferProvider = pDbp;
return NO_ERROR;
noDownmixForActiveTrack:
delete pDbp;
pTrack->downmixerBufferProvider = NULL;
return NO_INIT;
}
void AudioMixer::deleteTrackName(int name)
{
ALOGV("AudioMixer::deleteTrackName(%d)", name);
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
ALOGV("deleteTrackName(%d)", name);
track_t& track(mState.tracks[ name ]);
if (track.enabled) {
track.enabled = false;
invalidateState(1<<name);
}
// delete the resampler
delete track.resampler;
track.resampler = NULL;
// delete the downmixer
unprepareTrackForDownmix(&mState.tracks[name], name);
mTrackNames &= ~(1<<name);
}
void AudioMixer::enable(int name)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
track_t& track = mState.tracks[name];
if (!track.enabled) {
track.enabled = true;
ALOGV("enable(%d)", name);
invalidateState(1 << name);
}
}
void AudioMixer::disable(int name)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
track_t& track = mState.tracks[name];
if (track.enabled) {
track.enabled = false;
ALOGV("disable(%d)", name);
invalidateState(1 << name);
}
}
void AudioMixer::setParameter(int name, int target, int param, void *value)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
track_t& track = mState.tracks[name];
int valueInt = (int)value;
int32_t *valueBuf = (int32_t *)value;
switch (target) {
case TRACK:
switch (param) {
case CHANNEL_MASK: {
audio_channel_mask_t mask = (audio_channel_mask_t) value;
if (track.channelMask != mask) {
uint32_t channelCount = popcount(mask);
ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
track.channelMask = mask;
track.channelCount = channelCount;
// the mask has changed, does this track need a downmixer?
initTrackDownmix(&mState.tracks[name], name, mask);
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
invalidateState(1 << name);
}
} break;
case MAIN_BUFFER:
if (track.mainBuffer != valueBuf) {
track.mainBuffer = valueBuf;
ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
invalidateState(1 << name);
}
break;
case AUX_BUFFER:
if (track.auxBuffer != valueBuf) {
track.auxBuffer = valueBuf;
ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
invalidateState(1 << name);
}
break;
case FORMAT:
ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT);
break;
// FIXME do we want to support setting the downmix type from AudioFlinger?
// for a specific track? or per mixer?
/* case DOWNMIX_TYPE:
break */
default:
LOG_FATAL("bad param");
}
break;
case RESAMPLE:
switch (param) {
case SAMPLE_RATE:
ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
uint32_t(valueInt));
invalidateState(1 << name);
}
break;
case RESET:
track.resetResampler();
invalidateState(1 << name);
break;
case REMOVE:
delete track.resampler;
track.resampler = NULL;
track.sampleRate = mSampleRate;
invalidateState(1 << name);
break;
default:
LOG_FATAL("bad param");
}
break;
case RAMP_VOLUME:
case VOLUME:
switch (param) {
case VOLUME0:
case VOLUME1:
if (track.volume[param-VOLUME0] != valueInt) {
ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
track.volume[param-VOLUME0] = valueInt;
if (target == VOLUME) {
track.prevVolume[param-VOLUME0] = valueInt << 16;
track.volumeInc[param-VOLUME0] = 0;
} else {
int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0];
int32_t volInc = d / int32_t(mState.frameCount);
track.volumeInc[param-VOLUME0] = volInc;
if (volInc == 0) {
track.prevVolume[param-VOLUME0] = valueInt << 16;
}
}
invalidateState(1 << name);
}
break;
case AUXLEVEL:
//ALOG_ASSERT(0 <= valueInt && valueInt <= MAX_GAIN_INT, "bad aux level %d", valueInt);
if (track.auxLevel != valueInt) {
ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
track.prevAuxLevel = track.auxLevel << 16;
track.auxLevel = valueInt;
if (target == VOLUME) {
track.prevAuxLevel = valueInt << 16;
track.auxInc = 0;
} else {
int32_t d = (valueInt<<16) - track.prevAuxLevel;
int32_t volInc = d / int32_t(mState.frameCount);
track.auxInc = volInc;
if (volInc == 0) {
track.prevAuxLevel = valueInt << 16;
}
}
invalidateState(1 << name);
}
break;
default:
LOG_FATAL("bad param");
}
break;
default:
LOG_FATAL("bad target");
}
}
bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
{
if (value != devSampleRate || resampler != NULL) {
if (sampleRate != value) {
sampleRate = value;
if (resampler == NULL) {
ALOGV("creating resampler from track %d Hz to device %d Hz", value, devSampleRate);
AudioResampler::src_quality quality;
// force lowest quality level resampler if use case isn't music or video
// FIXME this is flawed for dynamic sample rates, as we choose the resampler
// quality level based on the initial ratio, but that could change later.
// Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
if (!((value == 44100 && devSampleRate == 48000) ||
(value == 48000 && devSampleRate == 44100))) {
quality = AudioResampler::LOW_QUALITY;
} else {
quality = AudioResampler::DEFAULT_QUALITY;
}
resampler = AudioResampler::create(
format,
// the resampler sees the number of channels after the downmixer, if any
downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount,
devSampleRate, quality);
resampler->setLocalTimeFreq(sLocalTimeFreq);
}
return true;
}
}
return false;
}
inline
void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) {
if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
volumeInc[i] = 0;
prevVolume[i] = volume[i]<<16;
}
}
if (aux) {
if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
auxInc = 0;
prevAuxLevel = auxLevel<<16;
}
}
}
size_t AudioMixer::getUnreleasedFrames(int name) const
{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
return mState.tracks[name].getUnreleasedFrames();
}
return 0;
}
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
{
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
if (mState.tracks[name].downmixerBufferProvider != NULL) {
// update required?
if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) {
ALOGV("AudioMixer::setBufferProvider(%p) for downmix", bufferProvider);
// setting the buffer provider for a track that gets downmixed consists in:
// 1/ setting the buffer provider to the "downmix / buffer provider" wrapper
// so it's the one that gets called when the buffer provider is needed,
mState.tracks[name].bufferProvider = mState.tracks[name].downmixerBufferProvider;
// 2/ saving the buffer provider for the track so the wrapper can use it
// when it downmixes.
mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider = bufferProvider;
}
} else {
mState.tracks[name].bufferProvider = bufferProvider;
}
}
void AudioMixer::process(int64_t pts)
{
mState.hook(&mState, pts);
}
void AudioMixer::process__validate(state_t* state, int64_t pts)
{
ALOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
uint32_t changed = state->needsChanged;
state->needsChanged = 0; // clear the validation flag
// recompute which tracks are enabled / disabled
uint32_t enabled = 0;
uint32_t disabled = 0;
while (changed) {
const int i = 31 - __builtin_clz(changed);
const uint32_t mask = 1<<i;
changed &= ~mask;
track_t& t = state->tracks[i];
(t.enabled ? enabled : disabled) |= mask;
}
state->enabledTracks &= ~disabled;
state->enabledTracks |= enabled;
// compute everything we need...
int countActiveTracks = 0;
bool all16BitsStereoNoResample = true;
bool resampling = false;
bool volumeRamp = false;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
countActiveTracks++;
track_t& t = state->tracks[i];
uint32_t n = 0;
n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
n |= NEEDS_FORMAT_16;
n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED;
if (t.auxLevel != 0 && t.auxBuffer != NULL) {
n |= NEEDS_AUX_ENABLED;
}
if (t.volumeInc[0]|t.volumeInc[1]) {
volumeRamp = true;
} else if (!t.doesResample() && t.volumeRL == 0) {
n |= NEEDS_MUTE_ENABLED;
}
t.needs = n;
if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) {
t.hook = track__nop;
} else {
if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
all16BitsStereoNoResample = false;
}
if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
all16BitsStereoNoResample = false;
resampling = true;
t.hook = track__genericResample;
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix + resample", i);
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
t.hook = track__16BitsMono;
all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
t.hook = track__16BitsStereo;
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix", i);
}
}
}
}
// select the processing hooks
state->hook = process__nop;
if (countActiveTracks) {
if (resampling) {
if (!state->outputTemp) {
state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
if (!state->resampleTemp) {
state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
state->hook = process__genericResampling;
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling;
}
}
}
}
ALOGV("mixer configuration change: %d activeTracks (%08x) "
"all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
state->hook(state, pts);
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
if (countActiveTracks) {
bool allMuted = true;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
track_t& t = state->tracks[i];
if (!t.doesResample() && t.volumeRL == 0)
{
t.needs |= NEEDS_MUTE_ENABLED;
t.hook = track__nop;
} else {
allMuted = false;
}
}
if (allMuted) {
state->hook = process__nop;
} else if (all16BitsStereoNoResample) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling;
}
}
}
}
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
int32_t* temp, int32_t* aux)
{
t->resampler->setSampleRate(t->sampleRate);
// ramp gain - resample to temp buffer and scale/mix in 2nd step
if (aux != NULL) {
// always resample with unity gain when sending to auxiliary buffer to be able
// to apply send level after resampling
// TODO: modify each resampler to support aux channel?
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
volumeRampStereo(t, out, outFrameCount, temp, aux);
}
// constant gain
else {
t->resampler->setVolume(t->volume[0], t->volume[1]);
t->resampler->resample(out, outFrameCount, t->bufferProvider);
}
}
}
void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp,
int32_t* aux)
{
}
void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux)
{
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
//ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
// ramp volume
if (CC_UNLIKELY(aux != NULL)) {
int32_t va = t->prevAuxLevel;
const int32_t vaInc = t->auxInc;
int32_t l;
int32_t r;
do {
l = (*temp++ >> 12);
r = (*temp++ >> 12);
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * r;
*aux++ += (va >> 17) * (l + r);
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevAuxLevel = va;
} else {
do {
*out++ += (vl >> 16) * (*temp++ >> 12);
*out++ += (vr >> 16) * (*temp++ >> 12);
vl += vlInc;
vr += vrInc;
} while (--frameCount);
}
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(aux != NULL);
}
void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux)
{
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
if (CC_UNLIKELY(aux != NULL)) {
const int16_t va = t->auxLevel;
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
out[0] = mulAdd(l, vl, out[0]);
int16_t a = (int16_t)(((int32_t)l + r) >> 1);
out[1] = mulAdd(r, vr, out[1]);
out += 2;
aux[0] = mulAdd(a, va, aux[0]);
aux++;
} while (--frameCount);
} else {
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(r, vr, out[1]);
out += 2;
} while (--frameCount);
}
}
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux)
{
const int16_t *in = static_cast<const int16_t *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
int32_t l;
int32_t r;
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
const int32_t vaInc = t->auxInc;
// ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
l = (int32_t)*in++;
r = (int32_t)*in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * r;
*aux++ += (va >> 17) * (l + r);
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->prevAuxLevel = va;
t->adjustVolumeRamp(true);
}
// constant gain
else {
const uint32_t vrl = t->volumeRL;
const int16_t va = (int16_t)t->auxLevel;
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
out += 2;
aux[0] = mulAdd(a, va, aux[0]);
aux++;
} while (--frameCount);
}
} else {
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
// ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
*out++ += (vl >> 16) * (int32_t) *in++;
*out++ += (vr >> 16) * (int32_t) *in++;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(false);
}
// constant gain
else {
const uint32_t vrl = t->volumeRL;
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
out += 2;
} while (--frameCount);
}
}
t->in = in;
}
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux)
{
const int16_t *in = static_cast<int16_t const *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
const int32_t vaInc = t->auxInc;
// ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
int32_t l = *in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * l;
*aux++ += (va >> 16) * l;
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->prevAuxLevel = va;
t->adjustVolumeRamp(true);
}
// constant gain
else {
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
const int16_t va = (int16_t)t->auxLevel;
do {
int16_t l = *in++;
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(l, vr, out[1]);
out += 2;
aux[0] = mulAdd(l, va, aux[0]);
aux++;
} while (--frameCount);
}
} else {
// ramp gain
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
// ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
int32_t l = *in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * l;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(false);
}
// constant gain
else {
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
do {
int16_t l = *in++;
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(l, vr, out[1]);
out += 2;
} while (--frameCount);
}
}
t->in = in;
}
// no-op case
void AudioMixer::process__nop(state_t* state, int64_t pts)
{
uint32_t e0 = state->enabledTracks;
size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
while (e0) {
// process by group of tracks with same output buffer to
// avoid multiple memset() on same buffer
uint32_t e1 = e0, e2 = e0;
int i = 31 - __builtin_clz(e1);
{
track_t& t1 = state->tracks[i];
e2 &= ~(1<<i);
while (e2) {
i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t2 = state->tracks[i];
if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<i);
}
}
e0 &= ~(e1);
memset(t1.mainBuffer, 0, bufSize);
}
while (e1) {
i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
{
track_t& t3 = state->tracks[i];
size_t outFrames = state->frameCount;
while (outFrames) {
t3.buffer.frameCount = outFrames;
int64_t outputPTS = calculateOutputPTS(
t3, pts, state->frameCount - outFrames);
t3.bufferProvider->getNextBuffer(&t3.buffer, outputPTS);
if (t3.buffer.raw == NULL) break;
outFrames -= t3.buffer.frameCount;
t3.bufferProvider->releaseBuffer(&t3.buffer);
}
}
}
}
}
// generic code without resampling
void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
// acquire each track's buffer
uint32_t enabledTracks = state->enabledTracks;
uint32_t e0 = enabledTracks;
while (e0) {
const int i = 31 - __builtin_clz(e0);
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
}
e0 = enabledTracks;
while (e0) {
// process by group of tracks with same output buffer to
// optimize cache use
uint32_t e1 = e0, e2 = e0;
int j = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[j];
e2 &= ~(1<<j);
while (e2) {
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
// this assumes output 16 bits stereo, no resampling
int32_t *out = t1.mainBuffer;
size_t numFrames = 0;
do {
memset(outTemp, 0, sizeof(outTemp));
e2 = e1;
while (e2) {
const int i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t = state->tracks[i];
size_t outFrames = BLOCKSIZE;
int32_t *aux = NULL;
if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
e1 &= ~(1<<i);
break;
}
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
if (inFrames) {
t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
state->resampleTemp, aux);
t.frameCount -= inFrames;
outFrames -= inFrames;
if (CC_UNLIKELY(aux != NULL)) {
aux += inFrames;
}
}
if (t.frameCount == 0 && outFrames) {
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) -
(BLOCKSIZE - outFrames);
int64_t outputPTS = calculateOutputPTS(
t, pts, numFrames + (BLOCKSIZE - outFrames));
t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
e1 &= ~(1<<i);
break;
}
t.frameCount = t.buffer.frameCount;
}
}
}
ditherAndClamp(out, outTemp, BLOCKSIZE);
out += BLOCKSIZE;
numFrames += BLOCKSIZE;
} while (numFrames < state->frameCount);
}
// release each track's buffer
e0 = enabledTracks;
while (e0) {
const int i = 31 - __builtin_clz(e0);
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.bufferProvider->releaseBuffer(&t.buffer);
}
}
// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
// this const just means that local variable outTemp doesn't change
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
size_t numFrames = state->frameCount;
uint32_t e0 = state->enabledTracks;
while (e0) {
// process by group of tracks with same output buffer
// to optimize cache use
uint32_t e1 = e0, e2 = e0;
int j = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[j];
e2 &= ~(1<<j);
while (e2) {
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
int32_t *out = t1.mainBuffer;
memset(outTemp, 0, size);
while (e1) {
const int i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer;
}
// this is a little goofy, on the resampling case we don't
// acquire/release the buffers because it's done by
// the resampler.
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
t.resampler->setPTS(pts);
t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
size_t outFrames = 0;
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) break;
if (CC_UNLIKELY(aux != NULL)) {
aux += outFrames;
}
t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount,
state->resampleTemp, aux);
outFrames += t.buffer.frameCount;
t.bufferProvider->releaseBuffer(&t.buffer);
}
}
}
ditherAndClamp(out, outTemp, numFrames);
}
}
// one track, 16 bits stereo without resampling is the most common case
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts)
{
// This method is only called when state->enabledTracks has exactly
// one bit set. The asserts below would verify this, but are commented out
// since the whole point of this method is to optimize performance.
//ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled");
const int i = 31 - __builtin_clz(state->enabledTracks);
//ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
const track_t& t = state->tracks[i];
AudioBufferProvider::Buffer& b(t.buffer);
int32_t* out = t.mainBuffer;
size_t numFrames = state->frameCount;
const int16_t vl = t.volume[0];
const int16_t vr = t.volume[1];
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
t.bufferProvider->getNextBuffer(&b, outputPTS);
const int16_t *in = b.i16;
// in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (in == NULL || ((unsigned long)in & 3)) {
memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t));
ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: "
"buffer %p track %d, channels %d, needs %08x",
in, i, t.channelCount, t.needs);
return;
}
size_t outFrames = b.frameCount;
if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
// volume is boosted, so we might need to clamp even though
// we process only one track.
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
// clamping...
l = clamp16(l);
r = clamp16(r);
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
} else {
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
}
numFrames -= b.frameCount;
t.bufferProvider->releaseBuffer(&b);
}
}
#if 0
// 2 tracks is also a common case
// NEVER used in current implementation of process__validate()
// only use if the 2 tracks have the same output buffer
void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
int64_t pts)
{
int i;
uint32_t en = state->enabledTracks;
i = 31 - __builtin_clz(en);
const track_t& t0 = state->tracks[i];
AudioBufferProvider::Buffer& b0(t0.buffer);
en &= ~(1<<i);
i = 31 - __builtin_clz(en);
const track_t& t1 = state->tracks[i];
AudioBufferProvider::Buffer& b1(t1.buffer);
const int16_t *in0;
const int16_t vl0 = t0.volume[0];
const int16_t vr0 = t0.volume[1];
size_t frameCount0 = 0;
const int16_t *in1;
const int16_t vl1 = t1.volume[0];
const int16_t vr1 = t1.volume[1];
size_t frameCount1 = 0;
//FIXME: only works if two tracks use same buffer
int32_t* out = t0.mainBuffer;
size_t numFrames = state->frameCount;
const int16_t *buff = NULL;
while (numFrames) {
if (frameCount0 == 0) {
b0.frameCount = numFrames;
int64_t outputPTS = calculateOutputPTS(t0, pts,
out - t0.mainBuffer);
t0.bufferProvider->getNextBuffer(&b0, outputPTS);
if (b0.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
}
in0 = buff;
b0.frameCount = numFrames;
} else {
in0 = b0.i16;
}
frameCount0 = b0.frameCount;
}
if (frameCount1 == 0) {
b1.frameCount = numFrames;
int64_t outputPTS = calculateOutputPTS(t1, pts,
out - t0.mainBuffer);
t1.bufferProvider->getNextBuffer(&b1, outputPTS);
if (b1.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
}
in1 = buff;
b1.frameCount = numFrames;
} else {
in1 = b1.i16;
}
frameCount1 = b1.frameCount;
}
size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1;
numFrames -= outFrames;
frameCount0 -= outFrames;
frameCount1 -= outFrames;
do {
int32_t l0 = *in0++;
int32_t r0 = *in0++;
l0 = mul(l0, vl0);
r0 = mul(r0, vr0);
int32_t l = *in1++;
int32_t r = *in1++;
l = mulAdd(l, vl1, l0) >> 12;
r = mulAdd(r, vr1, r0) >> 12;
// clamping...
l = clamp16(l);
r = clamp16(r);
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
if (frameCount0 == 0) {
t0.bufferProvider->releaseBuffer(&b0);
}
if (frameCount1 == 0) {
t1.bufferProvider->releaseBuffer(&b1);
}
}
delete [] buff;
}
#endif
int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
int outputFrameIndex)
{
if (AudioBufferProvider::kInvalidPTS == basePTS)
return AudioBufferProvider::kInvalidPTS;
return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
}
/*static*/ uint64_t AudioMixer::sLocalTimeFreq;
/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
/*static*/ void AudioMixer::sInitRoutine()
{
LocalClock lc;
sLocalTimeFreq = lc.getLocalFreq();
}
// ----------------------------------------------------------------------------
}; // namespace android