/*
* Copyright (C) 2015 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 "BufferProvider"
//#define LOG_NDEBUG 0
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
#include <external/sonic/sonic.h>
#include <media/audiohal/EffectBufferHalInterface.h>
#include <media/audiohal/EffectHalInterface.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include <media/AudioResamplerPublic.h>
#include <media/BufferProviders.h>
#include <system/audio_effects/effect_downmix.h>
#include <utils/Log.h>
#ifndef ARRAY_SIZE
#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
#endif
namespace android {
// ----------------------------------------------------------------------------
template <typename T>
static inline T min(const T& a, const T& b)
{
return a < b ? a : b;
}
CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize,
size_t outputFrameSize, size_t bufferFrameCount) :
mInputFrameSize(inputFrameSize),
mOutputFrameSize(outputFrameSize),
mLocalBufferFrameCount(bufferFrameCount),
mLocalBufferData(NULL),
mConsumed(0)
{
ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this,
inputFrameSize, outputFrameSize, bufferFrameCount);
LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0,
"Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)",
inputFrameSize, outputFrameSize);
if (mLocalBufferFrameCount) {
(void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize);
}
mBuffer.frameCount = 0;
}
CopyBufferProvider::~CopyBufferProvider()
{
ALOGV("~CopyBufferProvider(%p)", this);
if (mBuffer.frameCount != 0) {
mTrackBufferProvider->releaseBuffer(&mBuffer);
}
free(mLocalBufferData);
}
status_t CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer)
{
//ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu))",
// this, pBuffer, pBuffer->frameCount);
if (mLocalBufferFrameCount == 0) {
status_t res = mTrackBufferProvider->getNextBuffer(pBuffer);
if (res == OK) {
copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
}
return res;
}
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = pBuffer->frameCount;
status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
// At one time an upstream buffer provider had
// res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
//
// By API spec, if res != OK, then mBuffer.frameCount == 0.
// but there may be improper implementations.
ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
pBuffer->raw = NULL;
pBuffer->frameCount = 0;
return res;
}
mConsumed = 0;
}
ALOG_ASSERT(mConsumed < mBuffer.frameCount);
size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed);
count = min(count, pBuffer->frameCount);
pBuffer->raw = mLocalBufferData;
pBuffer->frameCount = count;
copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize,
pBuffer->frameCount);
return OK;
}
void CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
{
//ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))",
// this, pBuffer, pBuffer->frameCount);
if (mLocalBufferFrameCount == 0) {
mTrackBufferProvider->releaseBuffer(pBuffer);
return;
}
// LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
mTrackBufferProvider->releaseBuffer(&mBuffer);
ALOG_ASSERT(mBuffer.frameCount == 0);
}
pBuffer->raw = NULL;
pBuffer->frameCount = 0;
}
void CopyBufferProvider::reset()
{
if (mBuffer.frameCount != 0) {
mTrackBufferProvider->releaseBuffer(&mBuffer);
}
mConsumed = 0;
}
DownmixerBufferProvider::DownmixerBufferProvider(
audio_channel_mask_t inputChannelMask,
audio_channel_mask_t outputChannelMask, audio_format_t format,
uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) :
CopyBufferProvider(
audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
bufferFrameCount) // set bufferFrameCount to 0 to do in-place
{
ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d %d)",
this, inputChannelMask, outputChannelMask, format,
sampleRate, sessionId, (int)bufferFrameCount);
if (!sIsMultichannelCapable) {
ALOGE("DownmixerBufferProvider() error: not multichannel capable");
return;
}
mEffectsFactory = EffectsFactoryHalInterface::create();
if (mEffectsFactory == 0) {
ALOGE("DownmixerBufferProvider() error: could not obtain the effects factory");
return;
}
if (mEffectsFactory->createEffect(&sDwnmFxDesc.uuid,
sessionId,
SESSION_ID_INVALID_AND_IGNORED,
&mDownmixInterface) != 0) {
ALOGE("DownmixerBufferProvider() error creating downmixer effect");
mDownmixInterface.clear();
mEffectsFactory.clear();
return;
}
// channel input configuration will be overridden per-track
mDownmixConfig.inputCfg.channels = inputChannelMask; // FIXME: Should be bits
mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits
mDownmixConfig.inputCfg.format = format;
mDownmixConfig.outputCfg.format = format;
mDownmixConfig.inputCfg.samplingRate = sampleRate;
mDownmixConfig.outputCfg.samplingRate = sampleRate;
mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
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())
mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask;
mInFrameSize =
audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask);
mOutFrameSize =
audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask);
status_t status;
status = EffectBufferHalInterface::mirror(
nullptr, mInFrameSize * bufferFrameCount, &mInBuffer);
if (status != 0) {
ALOGE("DownmixerBufferProvider() error %d while creating input buffer", status);
mDownmixInterface.clear();
mEffectsFactory.clear();
return;
}
status = EffectBufferHalInterface::mirror(
nullptr, mOutFrameSize * bufferFrameCount, &mOutBuffer);
if (status != 0) {
ALOGE("DownmixerBufferProvider() error %d while creating output buffer", status);
mInBuffer.clear();
mDownmixInterface.clear();
mEffectsFactory.clear();
return;
}
mDownmixInterface->setInBuffer(mInBuffer);
mDownmixInterface->setOutBuffer(mOutBuffer);
int cmdStatus;
uint32_t replySize = sizeof(int);
// Configure downmixer
status = mDownmixInterface->command(
EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
&mDownmixConfig /*pCmdData*/,
&replySize, &cmdStatus /*pReplyData*/);
if (status != 0 || cmdStatus != 0) {
ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer",
status, cmdStatus);
mOutBuffer.clear();
mInBuffer.clear();
mDownmixInterface.clear();
mEffectsFactory.clear();
return;
}
// Enable downmixer
replySize = sizeof(int);
status = mDownmixInterface->command(
EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
&replySize, &cmdStatus /*pReplyData*/);
if (status != 0 || cmdStatus != 0) {
ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer",
status, cmdStatus);
mOutBuffer.clear();
mInBuffer.clear();
mDownmixInterface.clear();
mEffectsFactory.clear();
return;
}
// 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);
replySize = sizeof(int);
status = mDownmixInterface->command(
EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */,
param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/);
free(param);
if (status != 0 || cmdStatus != 0) {
ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type",
status, cmdStatus);
mOutBuffer.clear();
mInBuffer.clear();
mDownmixInterface.clear();
mEffectsFactory.clear();
return;
}
ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType);
}
DownmixerBufferProvider::~DownmixerBufferProvider()
{
ALOGV("~DownmixerBufferProvider (%p)", this);
if (mDownmixInterface != 0) {
mDownmixInterface->close();
}
}
void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
{
mInBuffer->setExternalData(const_cast<void*>(src));
mInBuffer->setFrameCount(frames);
mInBuffer->update(mInFrameSize * frames);
mOutBuffer->setFrameCount(frames);
mOutBuffer->setExternalData(dst);
if (dst != src) {
// Downmix may be accumulating, need to populate the output buffer
// with the dst data.
mOutBuffer->update(mOutFrameSize * frames);
}
// may be in-place if src == dst.
status_t res = mDownmixInterface->process();
if (res == OK) {
mOutBuffer->commit(mOutFrameSize * frames);
} else {
ALOGE("DownmixBufferProvider error %d", res);
}
}
/* call once in a pthread_once handler. */
/*static*/ status_t DownmixerBufferProvider::init()
{
// find multichannel downmix effect if we have to play multichannel content
sp<EffectsFactoryHalInterface> effectsFactory = EffectsFactoryHalInterface::create();
if (effectsFactory == 0) {
ALOGE("AudioMixer() error: could not obtain the effects factory");
return NO_INIT;
}
uint32_t numEffects = 0;
int ret = effectsFactory->queryNumberEffects(&numEffects);
if (ret != 0) {
ALOGE("AudioMixer() error %d querying number of effects", ret);
return NO_INIT;
}
ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
for (uint32_t i = 0 ; i < numEffects ; i++) {
if (effectsFactory->getDescriptor(i, &sDwnmFxDesc) == 0) {
ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
ALOGI("found effect \"%s\" from %s",
sDwnmFxDesc.name, sDwnmFxDesc.implementor);
sIsMultichannelCapable = true;
break;
}
}
}
ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
return NO_INIT;
}
/*static*/ bool DownmixerBufferProvider::sIsMultichannelCapable = false;
/*static*/ effect_descriptor_t DownmixerBufferProvider::sDwnmFxDesc;
RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask,
audio_channel_mask_t outputChannelMask, audio_format_t format,
size_t bufferFrameCount) :
CopyBufferProvider(
audio_bytes_per_sample(format)
* audio_channel_count_from_out_mask(inputChannelMask),
audio_bytes_per_sample(format)
* audio_channel_count_from_out_mask(outputChannelMask),
bufferFrameCount),
mFormat(format),
mSampleSize(audio_bytes_per_sample(format)),
mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)),
mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask))
{
ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
this, format, inputChannelMask, outputChannelMask,
mInputChannels, mOutputChannels);
(void) memcpy_by_index_array_initialization_from_channel_mask(
mIdxAry, ARRAY_SIZE(mIdxAry), outputChannelMask, inputChannelMask);
}
void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
{
memcpy_by_index_array(dst, mOutputChannels,
src, mInputChannels, mIdxAry, mSampleSize, frames);
}
ReformatBufferProvider::ReformatBufferProvider(int32_t channelCount,
audio_format_t inputFormat, audio_format_t outputFormat,
size_t bufferFrameCount) :
CopyBufferProvider(
channelCount * audio_bytes_per_sample(inputFormat),
channelCount * audio_bytes_per_sample(outputFormat),
bufferFrameCount),
mChannelCount(channelCount),
mInputFormat(inputFormat),
mOutputFormat(outputFormat)
{
ALOGV("ReformatBufferProvider(%p)(%u, %#x, %#x)",
this, channelCount, inputFormat, outputFormat);
}
void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
{
memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount);
}
TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) :
mChannelCount(channelCount),
mFormat(format),
mSampleRate(sampleRate),
mFrameSize(channelCount * audio_bytes_per_sample(format)),
mLocalBufferFrameCount(0),
mLocalBufferData(NULL),
mRemaining(0),
mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),
mFallbackFailErrorShown(false),
mAudioPlaybackRateValid(false)
{
LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,
"TimestretchBufferProvider can't allocate Sonic stream");
setPlaybackRate(playbackRate);
ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f %d %d)",
this, channelCount, format, sampleRate, playbackRate.mSpeed,
playbackRate.mPitch, playbackRate.mStretchMode, playbackRate.mFallbackMode);
mBuffer.frameCount = 0;
}
TimestretchBufferProvider::~TimestretchBufferProvider()
{
ALOGV("~TimestretchBufferProvider(%p)", this);
sonicDestroyStream(mSonicStream);
if (mBuffer.frameCount != 0) {
mTrackBufferProvider->releaseBuffer(&mBuffer);
}
free(mLocalBufferData);
}
status_t TimestretchBufferProvider::getNextBuffer(
AudioBufferProvider::Buffer *pBuffer)
{
ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu))",
this, pBuffer, pBuffer->frameCount);
// BYPASS
//return mTrackBufferProvider->getNextBuffer(pBuffer);
// check if previously processed data is sufficient.
if (pBuffer->frameCount <= mRemaining) {
ALOGV("previous sufficient");
pBuffer->raw = mLocalBufferData;
return OK;
}
// do we need to resize our buffer?
if (pBuffer->frameCount > mLocalBufferFrameCount) {
void *newmem;
if (posix_memalign(&newmem, 32, pBuffer->frameCount * mFrameSize) == OK) {
if (mRemaining != 0) {
memcpy(newmem, mLocalBufferData, mRemaining * mFrameSize);
}
free(mLocalBufferData);
mLocalBufferData = newmem;
mLocalBufferFrameCount = pBuffer->frameCount;
}
}
// need to fetch more data
const size_t outputDesired = pBuffer->frameCount - mRemaining;
size_t dstAvailable;
do {
mBuffer.frameCount = mPlaybackRate.mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
? outputDesired : outputDesired * mPlaybackRate.mSpeed + 1;
status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
ALOGV("upstream provider cannot provide data");
if (mRemaining == 0) {
pBuffer->raw = NULL;
pBuffer->frameCount = 0;
return res;
} else { // return partial count
pBuffer->raw = mLocalBufferData;
pBuffer->frameCount = mRemaining;
return OK;
}
}
// time-stretch the data
dstAvailable = min(mLocalBufferFrameCount - mRemaining, outputDesired);
size_t srcAvailable = mBuffer.frameCount;
processFrames((uint8_t*)mLocalBufferData + mRemaining * mFrameSize, &dstAvailable,
mBuffer.raw, &srcAvailable);
// release all data consumed
mBuffer.frameCount = srcAvailable;
mTrackBufferProvider->releaseBuffer(&mBuffer);
} while (dstAvailable == 0); // try until we get output data or upstream provider fails.
// update buffer vars with the actual data processed and return with buffer
mRemaining += dstAvailable;
pBuffer->raw = mLocalBufferData;
pBuffer->frameCount = mRemaining;
return OK;
}
void TimestretchBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
{
ALOGV("TimestretchBufferProvider(%p)::releaseBuffer(%p (%zu))",
this, pBuffer, pBuffer->frameCount);
// BYPASS
//return mTrackBufferProvider->releaseBuffer(pBuffer);
// LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
if (pBuffer->frameCount < mRemaining) {
memcpy(mLocalBufferData,
(uint8_t*)mLocalBufferData + pBuffer->frameCount * mFrameSize,
(mRemaining - pBuffer->frameCount) * mFrameSize);
mRemaining -= pBuffer->frameCount;
} else if (pBuffer->frameCount == mRemaining) {
mRemaining = 0;
} else {
LOG_ALWAYS_FATAL("Releasing more frames(%zu) than available(%zu)",
pBuffer->frameCount, mRemaining);
}
pBuffer->raw = NULL;
pBuffer->frameCount = 0;
}
void TimestretchBufferProvider::reset()
{
mRemaining = 0;
}
status_t TimestretchBufferProvider::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
mPlaybackRate = playbackRate;
mFallbackFailErrorShown = false;
sonicSetSpeed(mSonicStream, mPlaybackRate.mSpeed);
//TODO: pitch is ignored for now
//TODO: optimize: if parameters are the same, don't do any extra computation.
mAudioPlaybackRateValid = isAudioPlaybackRateValid(mPlaybackRate);
return OK;
}
void TimestretchBufferProvider::processFrames(void *dstBuffer, size_t *dstFrames,
const void *srcBuffer, size_t *srcFrames)
{
ALOGV("processFrames(%zu %zu) remaining(%zu)", *dstFrames, *srcFrames, mRemaining);
// Note dstFrames is the required number of frames.
if (!mAudioPlaybackRateValid) {
//fallback mode
// Ensure consumption from src is as expected.
// TODO: add logic to track "very accurate" consumption related to speed, original sampling
// rate, actual frames processed.
const size_t targetSrc = *dstFrames * mPlaybackRate.mSpeed;
if (*srcFrames < targetSrc) { // limit dst frames to that possible
*dstFrames = *srcFrames / mPlaybackRate.mSpeed;
} else if (*srcFrames > targetSrc + 1) {
*srcFrames = targetSrc + 1;
}
if (*dstFrames > 0) {
switch(mPlaybackRate.mFallbackMode) {
case AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT:
if (*dstFrames <= *srcFrames) {
size_t copySize = mFrameSize * *dstFrames;
memcpy(dstBuffer, srcBuffer, copySize);
} else {
// cyclically repeat the source.
for (size_t count = 0; count < *dstFrames; count += *srcFrames) {
size_t remaining = min(*srcFrames, *dstFrames - count);
memcpy((uint8_t*)dstBuffer + mFrameSize * count,
srcBuffer, mFrameSize * remaining);
}
}
break;
case AUDIO_TIMESTRETCH_FALLBACK_DEFAULT:
case AUDIO_TIMESTRETCH_FALLBACK_MUTE:
memset(dstBuffer,0, mFrameSize * *dstFrames);
break;
case AUDIO_TIMESTRETCH_FALLBACK_FAIL:
default:
if(!mFallbackFailErrorShown) {
ALOGE("invalid parameters in TimestretchBufferProvider fallbackMode:%d",
mPlaybackRate.mFallbackMode);
mFallbackFailErrorShown = true;
}
break;
}
}
} else {
switch (mFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
if (sonicWriteFloatToStream(mSonicStream, (float*)srcBuffer, *srcFrames) != 1) {
ALOGE("sonicWriteFloatToStream cannot realloc");
*srcFrames = 0; // cannot consume all of srcBuffer
}
*dstFrames = sonicReadFloatFromStream(mSonicStream, (float*)dstBuffer, *dstFrames);
break;
case AUDIO_FORMAT_PCM_16_BIT:
if (sonicWriteShortToStream(mSonicStream, (short*)srcBuffer, *srcFrames) != 1) {
ALOGE("sonicWriteShortToStream cannot realloc");
*srcFrames = 0; // cannot consume all of srcBuffer
}
*dstFrames = sonicReadShortFromStream(mSonicStream, (short*)dstBuffer, *dstFrames);
break;
default:
// could also be caught on construction
LOG_ALWAYS_FATAL("invalid format %#x for TimestretchBufferProvider", mFormat);
}
}
}
// ----------------------------------------------------------------------------
} // namespace android