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ara-mdk / platform / frameworks / av / 15e5c9f49ad92ce985c9d0c2fe2767df2833b12c / . / services / audioflinger / Tracks.cpp
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/*
**
** Copyright 2012, 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 "AudioFlinger"
//#define LOG_NDEBUG 0
#include <math.h>
#include <cutils/compiler.h>
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
#include <common_time/cc_helper.h>
#include <common_time/local_clock.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#include "ServiceUtilities.h"
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
// ----------------------------------------------------------------------------
// Note: the following macro is used for extremely verbose logging message. In
// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
// 0; but one side effect of this is to turn all LOGV's as well. Some messages
// are so verbose that we want to suppress them even when we have ALOG_ASSERT
// turned on. Do not uncomment the #def below unless you really know what you
// are doing and want to see all of the extremely verbose messages.
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
namespace android {
// ----------------------------------------------------------------------------
// TrackBase
// ----------------------------------------------------------------------------
static volatile int32_t nextTrackId = 55;
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
bool isOut)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(NULL),
// mBuffer
// mBufferEnd
mStepCount(0),
mState(IDLE),
mSampleRate(sampleRate),
mFormat(format),
mChannelMask(channelMask),
mChannelCount(popcount(channelMask)),
mFrameSize(audio_is_linear_pcm(format) ?
mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
mFrameCount(frameCount),
mStepServerFailed(false),
mSessionId(sessionId),
mIsOut(isOut),
mServerProxy(NULL),
mId(android_atomic_inc(&nextTrackId))
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
sharedBuffer->size());
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = frameCount * mFrameSize;
if (sharedBuffer == 0) {
size += bufferSize;
}
if (client != 0) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
// can't assume mCblk != NULL
} else {
ALOGE("not enough memory for AudioTrack size=%u", size);
client->heap()->dump("AudioTrack");
return;
}
} else {
// this syntax avoids calling the audio_track_cblk_t constructor twice
mCblk = (audio_track_cblk_t *) new uint8_t[size];
// assume mCblk != NULL
}
// construct the shared structure in-place.
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount_ = frameCount;
// uncomment the following lines to quickly test 32-bit wraparound
// mCblk->user = 0xffff0000;
// mCblk->server = 0xffff0000;
// mCblk->userBase = 0xffff0000;
// mCblk->serverBase = 0xffff0000;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer (other flags are cleared)
mCblk->flags = CBLK_UNDERRUN;
} else {
mBuffer = sharedBuffer->pointer();
}
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
mServerProxy = new ServerProxy(mCblk, mBuffer, frameCount, mFrameSize, isOut);
#ifdef TEE_SINK
if (mTeeSinkTrackEnabled) {
NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
if (pipeFormat != Format_Invalid) {
Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {pipeFormat};
ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
PipeReader *pipeReader = new PipeReader(*pipe);
numCounterOffers = 0;
index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
mTeeSink = pipe;
mTeeSource = pipeReader;
}
}
#endif
}
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
#ifdef TEE_SINK
dumpTee(-1, mTeeSource, mId);
#endif
// delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
delete mServerProxy;
if (mCblk != NULL) {
if (mClient == 0) {
delete mCblk;
} else {
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
}
}
mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
if (mClient != 0) {
// Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
// If the client's reference count drops to zero, the associated destructor
// must run with AudioFlinger lock held. Thus the explicit clear() rather than
// relying on the automatic clear() at end of scope.
mClient.clear();
}
}
// AudioBufferProvider interface
// getNextBuffer() = 0;
// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
#ifdef TEE_SINK
if (mTeeSink != 0) {
(void) mTeeSink->write(buffer->raw, buffer->frameCount);
}
#endif
buffer->raw = NULL;
mStepCount = buffer->frameCount;
// FIXME See note at getNextBuffer()
(void) step(); // ignore return value of step()
buffer->frameCount = 0;
}
bool AudioFlinger::ThreadBase::TrackBase::step() {
bool result = mServerProxy->step(mStepCount);
if (!result) {
ALOGV("stepServer failed acquiring cblk mutex");
mStepServerFailed = true;
}
return result;
}
void AudioFlinger::ThreadBase::TrackBase::reset() {
audio_track_cblk_t* cblk = this->cblk();
cblk->user = 0;
cblk->server = 0;
cblk->userBase = 0;
cblk->serverBase = 0;
mStepServerFailed = false;
ALOGV("TrackBase::reset");
}
uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return mServerProxy->getSampleRate();
}
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
int8_t *bufferEnd = bufferStart + frames * mFrameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
"TrackBase::getBuffer buffer out of range:\n"
" start: %p, end %p , mBuffer %p mBufferEnd %p\n"
" server %u, serverBase %u, user %u, userBase %u, frameSize %u",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
return bufferStart;
}
status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
{
mSyncEvents.add(event);
return NO_ERROR;
}
// ----------------------------------------------------------------------------
// Playback
// ----------------------------------------------------------------------------
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
}
AudioFlinger::TrackHandle::~TrackHandle() {
// just stop the track on deletion, associated resources
// will be freed from the main thread once all pending buffers have
// been played. Unless it's not in the active track list, in which
// case we free everything now...
mTrack->destroy();
}
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
return mTrack->getCblk();
}
status_t AudioFlinger::TrackHandle::start() {
return mTrack->start();
}
void AudioFlinger::TrackHandle::stop() {
mTrack->stop();
}
void AudioFlinger::TrackHandle::flush() {
mTrack->flush();
}
void AudioFlinger::TrackHandle::pause() {
mTrack->pause();
}
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
}
status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
sp<IMemory>* buffer) {
if (!mTrack->isTimedTrack())
return INVALID_OPERATION;
PlaybackThread::TimedTrack* tt =
reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
return tt->allocateTimedBuffer(size, buffer);
}
status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
int64_t pts) {
if (!mTrack->isTimedTrack())
return INVALID_OPERATION;
PlaybackThread::TimedTrack* tt =
reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
return tt->queueTimedBuffer(buffer, pts);
}
status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
const LinearTransform& xform, int target) {
if (!mTrack->isTimedTrack())
return INVALID_OPERATION;
PlaybackThread::TimedTrack* tt =
reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
return tt->setMediaTimeTransform(
xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
}
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioTrack::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t flags)
: TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
sessionId, true /*isOut*/),
mFillingUpStatus(FS_INVALID),
// mRetryCount initialized later when needed
mSharedBuffer(sharedBuffer),
mStreamType(streamType),
mName(-1), // see note below
mMainBuffer(thread->mixBuffer()),
mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false),
mPresentationCompleteFrames(0),
mFlags(flags),
mFastIndex(-1),
mUnderrunCount(0),
mCachedVolume(1.0),
mIsInvalid(false)
{
if (mCblk != NULL) {
// to avoid leaking a track name, do not allocate one unless there is an mCblk
mName = thread->getTrackName_l(channelMask, sessionId);
mCblk->mName = mName;
if (mName < 0) {
ALOGE("no more track names available");
return;
}
// only allocate a fast track index if we were able to allocate a normal track name
if (flags & IAudioFlinger::TRACK_FAST) {
ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
int i = __builtin_ctz(thread->mFastTrackAvailMask);
ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
// FIXME This is too eager. We allocate a fast track index before the
// fast track becomes active. Since fast tracks are a scarce resource,
// this means we are potentially denying other more important fast tracks from
// being created. It would be better to allocate the index dynamically.
mFastIndex = i;
mCblk->mName = i;
// Read the initial underruns because this field is never cleared by the fast mixer
mObservedUnderruns = thread->getFastTrackUnderruns(i);
thread->mFastTrackAvailMask &= ~(1 << i);
}
}
ALOGV("Track constructor name %d, calling pid %d", mName,
IPCThreadState::self()->getCallingPid());
}
AudioFlinger::PlaybackThread::Track::~Track()
{
ALOGV("PlaybackThread::Track destructor");
}
void AudioFlinger::PlaybackThread::Track::destroy()
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
// destructor is called. As the destructor needs to lock mLock,
// we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
// On the other hand, as long as Track::destroy() is only called by
// TrackHandle destructor, the TrackHandle still holds a strong ref on
// this Track with its member mTrack.
sp<Track> keep(this);
{ // scope for mLock
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
#endif
}
AudioSystem::releaseOutput(thread->id());
}
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->destroyTrack_l(this);
}
}
}
/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
{
result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S F SRate "
"L dB R dB Server User Main buf Aux Buf Flags Underruns\n");
}
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
uint32_t vlr = mServerProxy->getVolumeLR();
if (isFastTrack()) {
sprintf(buffer, " F %2d", mFastIndex);
} else {
sprintf(buffer, " %4d", mName - AudioMixer::TRACK0);
}
track_state state = mState;
char stateChar;
switch (state) {
case IDLE:
stateChar = 'I';
break;
case TERMINATED:
stateChar = 'T';
break;
case STOPPING_1:
stateChar = 's';
break;
case STOPPING_2:
stateChar = '5';
break;
case STOPPED:
stateChar = 'S';
break;
case RESUMING:
stateChar = 'R';
break;
case ACTIVE:
stateChar = 'A';
break;
case PAUSING:
stateChar = 'p';
break;
case PAUSED:
stateChar = 'P';
break;
case FLUSHED:
stateChar = 'F';
break;
default:
stateChar = '?';
break;
}
char nowInUnderrun;
switch (mObservedUnderruns.mBitFields.mMostRecent) {
case UNDERRUN_FULL:
nowInUnderrun = ' ';
break;
case UNDERRUN_PARTIAL:
nowInUnderrun = '<';
break;
case UNDERRUN_EMPTY:
nowInUnderrun = '*';
break;
default:
nowInUnderrun = '?';
break;
}
snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %5u %5.2g %5.2g "
"0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
mChannelMask,
mSessionId,
mStepCount,
mFrameCount,
stateChar,
mFillingUpStatus,
mServerProxy->getSampleRate(),
20.0 * log10((vlr & 0xFFFF) / 4096.0),
20.0 * log10((vlr >> 16) / 4096.0),
mCblk->server,
mCblk->user,
(int)mMainBuffer,
(int)mAuxBuffer,
mCblk->flags,
mUnderrunCount,
nowInUnderrun);
}
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
AudioBufferProvider::Buffer* buffer, int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mStepServerFailed) {
// FIXME When called by fast mixer, this takes a mutex with tryLock().
// Since the fast mixer is higher priority than client callback thread,
// it does not result in priority inversion for client.
// But a non-blocking solution would be preferable to avoid
// fast mixer being unable to tryLock(), and
// to avoid the extra context switches if the client wakes up,
// discovers the mutex is locked, then has to wait for fast mixer to unlock.
if (!step()) goto getNextBuffer_exit;
ALOGV("stepServer recovered");
mStepServerFailed = false;
}
// FIXME Same as above
framesReady = mServerProxy->framesReady();
if (CC_LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + mFrameCount;
bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
if (framesReq > framesReady) {
framesReq = framesReady;
}
if (framesReq > bufferEnd - s) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = NULL;
buffer->frameCount = 0;
ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
return NOT_ENOUGH_DATA;
}
// Note that framesReady() takes a mutex on the control block using tryLock().
// This could result in priority inversion if framesReady() is called by the normal mixer,
// as the normal mixer thread runs at lower
// priority than the client's callback thread: there is a short window within framesReady()
// during which the normal mixer could be preempted, and the client callback would block.
// Another problem can occur if framesReady() is called by the fast mixer:
// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
return mServerProxy->framesReady();
}
// Don't call for fast tracks; the framesReady() could result in priority inversion
bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
return true;
}
if (framesReady() >= mFrameCount ||
(mCblk->flags & CBLK_FORCEREADY)) {
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
return true;
}
return false;
}
status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
int triggerSession)
{
status_t status = NO_ERROR;
ALOGV("start(%d), calling pid %d session %d",
mName, IPCThreadState::self()->getCallingPid(), mSessionId);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
track_state state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (state == PAUSED) {
mState = TrackBase::RESUMING;
ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
} else {
mState = TrackBase::ACTIVE;
ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
}
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
if (status == NO_ERROR) {
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
}
#endif
}
if (status == NO_ERROR) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->addTrack_l(this);
} else {
mState = state;
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
}
} else {
status = BAD_VALUE;
}
return status;
}
void AudioFlinger::PlaybackThread::Track::stop()
{
ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
track_state state = mState;
if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
// If the track is not active (PAUSED and buffers full), flush buffers
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
mState = STOPPED;
} else if (!isFastTrack()) {
mState = STOPPED;
} else {
// prepareTracks_l() will set state to STOPPING_2 after next underrun,
// and then to STOPPED and reset() when presentation is complete
mState = STOPPING_1;
}
ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
playbackThread);
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
#endif
}
}
}
void AudioFlinger::PlaybackThread::Track::pause()
{
ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
if (mState == ACTIVE || mState == RESUMING) {
mState = PAUSING;
ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
#endif
}
}
}
}
void AudioFlinger::PlaybackThread::Track::flush()
{
ALOGV("flush(%d)", mName);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED &&
mState != PAUSING && mState != IDLE && mState != FLUSHED) {
return;
}
// No point remaining in PAUSED state after a flush => go to
// FLUSHED state
mState = FLUSHED;
// do not reset the track if it is still in the process of being stopped or paused.
// this will be done by prepareTracks_l() when the track is stopped.
// prepareTracks_l() will see mState == FLUSHED, then
// remove from active track list, reset(), and trigger presentation complete
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
}
}
void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
if (!mResetDone) {
TrackBase::reset();
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
mFillingUpStatus = FS_FILLING;
mResetDone = true;
if (mState == FLUSHED) {
mState = IDLE;
}
}
}
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
{
status_t status = DEAD_OBJECT;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
sp<AudioFlinger> af = mClient->audioFlinger();
Mutex::Autolock _l(af->mLock);
sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
Mutex::Autolock _dl(playbackThread->mLock);
Mutex::Autolock _sl(srcThread->mLock);
sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (chain == 0) {
return INVALID_OPERATION;
}
sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
if (effect == 0) {
return INVALID_OPERATION;
}
srcThread->removeEffect_l(effect);
playbackThread->addEffect_l(effect);
// removeEffect_l() has stopped the effect if it was active so it must be restarted
if (effect->state() == EffectModule::ACTIVE ||
effect->state() == EffectModule::STOPPING) {
effect->start();
}
sp<EffectChain> dstChain = effect->chain().promote();
if (dstChain == 0) {
srcThread->addEffect_l(effect);
return INVALID_OPERATION;
}
AudioSystem::unregisterEffect(effect->id());
AudioSystem::registerEffect(&effect->desc(),
srcThread->id(),
dstChain->strategy(),
AUDIO_SESSION_OUTPUT_MIX,
effect->id());
}
status = playbackThread->attachAuxEffect(this, EffectId);
}
return status;
}
void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
{
mAuxEffectId = EffectId;
mAuxBuffer = buffer;
}
bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
size_t audioHalFrames)
{
// a track is considered presented when the total number of frames written to audio HAL
// corresponds to the number of frames written when presentationComplete() is called for the
// first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
if (mPresentationCompleteFrames == 0) {
mPresentationCompleteFrames = framesWritten + audioHalFrames;
ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
mPresentationCompleteFrames, audioHalFrames);
}
if (framesWritten >= mPresentationCompleteFrames) {
ALOGV("presentationComplete() session %d complete: framesWritten %d",
mSessionId, framesWritten);
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
return true;
}
return false;
}
void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
{
for (int i = 0; i < (int)mSyncEvents.size(); i++) {
if (mSyncEvents[i]->type() == type) {
mSyncEvents[i]->trigger();
mSyncEvents.removeAt(i);
i--;
}
}
}
// implement VolumeBufferProvider interface
uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
{
// called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
uint32_t vlr = mServerProxy->getVolumeLR();
uint32_t vl = vlr & 0xFFFF;
uint32_t vr = vlr >> 16;
// track volumes come from shared memory, so can't be trusted and must be clamped
if (vl > MAX_GAIN_INT) {
vl = MAX_GAIN_INT;
}
if (vr > MAX_GAIN_INT) {
vr = MAX_GAIN_INT;
}
// now apply the cached master volume and stream type volume;
// this is trusted but lacks any synchronization or barrier so may be stale
float v = mCachedVolume;
vl *= v;
vr *= v;
// re-combine into U4.16
vlr = (vr << 16) | (vl & 0xFFFF);
// FIXME look at mute, pause, and stop flags
return vlr;
}
status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
{
if (mState == TERMINATED || mState == PAUSED ||
((framesReady() == 0) && ((mSharedBuffer != 0) ||
(mState == STOPPED)))) {
ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
event->cancel();
return INVALID_OPERATION;
}
(void) TrackBase::setSyncEvent(event);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::Track::invalidate()
{
// FIXME should use proxy
android_atomic_or(CBLK_INVALID, &mCblk->flags);
mCblk->cv.signal();
mIsInvalid = true;
}
// ----------------------------------------------------------------------------
sp<AudioFlinger::PlaybackThread::TimedTrack>
AudioFlinger::PlaybackThread::TimedTrack::create(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId) {
if (!client->reserveTimedTrack())
return 0;
return new TimedTrack(
thread, client, streamType, sampleRate, format, channelMask, frameCount,
sharedBuffer, sessionId);
}
AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
: Track(thread, client, streamType, sampleRate, format, channelMask,
frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
mQueueHeadInFlight(false),
mTrimQueueHeadOnRelease(false),
mFramesPendingInQueue(0),
mTimedSilenceBuffer(NULL),
mTimedSilenceBufferSize(0),
mTimedAudioOutputOnTime(false),
mMediaTimeTransformValid(false)
{
LocalClock lc;
mLocalTimeFreq = lc.getLocalFreq();
mLocalTimeToSampleTransform.a_zero = 0;
mLocalTimeToSampleTransform.b_zero = 0;
mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
&mLocalTimeToSampleTransform.a_to_b_denom);
mMediaTimeToSampleTransform.a_zero = 0;
mMediaTimeToSampleTransform.b_zero = 0;
mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
&mMediaTimeToSampleTransform.a_to_b_denom);
}
AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
mClient->releaseTimedTrack();
delete [] mTimedSilenceBuffer;
}
status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
size_t size, sp<IMemory>* buffer) {
Mutex::Autolock _l(mTimedBufferQueueLock);
trimTimedBufferQueue_l();
// lazily initialize the shared memory heap for timed buffers
if (mTimedMemoryDealer == NULL) {
const int kTimedBufferHeapSize = 512 << 10;
mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
"AudioFlingerTimed");
if (mTimedMemoryDealer == NULL)
return NO_MEMORY;
}
sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
if (newBuffer == NULL) {
newBuffer = mTimedMemoryDealer->allocate(size);
if (newBuffer == NULL)
return NO_MEMORY;
}
*buffer = newBuffer;
return NO_ERROR;
}
// caller must hold mTimedBufferQueueLock
void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
int64_t mediaTimeNow;
{
Mutex::Autolock mttLock(mMediaTimeTransformLock);
if (!mMediaTimeTransformValid)
return;
int64_t targetTimeNow;
status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
? mCCHelper.getCommonTime(&targetTimeNow)
: mCCHelper.getLocalTime(&targetTimeNow);
if (OK != res)
return;
if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
&mediaTimeNow)) {
return;
}
}
size_t trimEnd;
for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
int64_t bufEnd;
if ((trimEnd + 1) < mTimedBufferQueue.size()) {
// We have a next buffer. Just use its PTS as the PTS of the frame
// following the last frame in this buffer. If the stream is sparse
// (ie, there are deliberate gaps left in the stream which should be
// filled with silence by the TimedAudioTrack), then this can result
// in one extra buffer being left un-trimmed when it could have
// been. In general, this is not typical, and we would rather
// optimized away the TS calculation below for the more common case
// where PTSes are contiguous.
bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
} else {
// We have no next buffer. Compute the PTS of the frame following
// the last frame in this buffer by computing the duration of of
// this frame in media time units and adding it to the PTS of the
// buffer.
int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
/ mFrameSize;
if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
&bufEnd)) {
ALOGE("Failed to convert frame count of %lld to media time"
" duration" " (scale factor %d/%u) in %s",
frameCount,
mMediaTimeToSampleTransform.a_to_b_numer,
mMediaTimeToSampleTransform.a_to_b_denom,
__PRETTY_FUNCTION__);
break;
}
bufEnd += mTimedBufferQueue[trimEnd].pts();
}
if (bufEnd > mediaTimeNow)
break;
// Is the buffer we want to use in the middle of a mix operation right
// now? If so, don't actually trim it. Just wait for the releaseBuffer
// from the mixer which should be coming back shortly.
if (!trimEnd && mQueueHeadInFlight) {
mTrimQueueHeadOnRelease = true;
}
}
size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
if (trimStart < trimEnd) {
// Update the bookkeeping for framesReady()
for (size_t i = trimStart; i < trimEnd; ++i) {
updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
}
// Now actually remove the buffers from the queue.
mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
}
}
void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
const char* logTag) {
ALOG_ASSERT(mTimedBufferQueue.size() > 0,
"%s called (reason \"%s\"), but timed buffer queue has no"
" elements to trim.", __FUNCTION__, logTag);
updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
mTimedBufferQueue.removeAt(0);
}
void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
const TimedBuffer& buf,
const char* logTag) {
uint32_t bufBytes = buf.buffer()->size();
uint32_t consumedAlready = buf.position();
ALOG_ASSERT(consumedAlready <= bufBytes,
"Bad bookkeeping while updating frames pending. Timed buffer is"
" only %u bytes long, but claims to have consumed %u"
" bytes. (update reason: \"%s\")",
bufBytes, consumedAlready, logTag);
uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
"Bad bookkeeping while updating frames pending. Should have at"
" least %u queued frames, but we think we have only %u. (update"
" reason: \"%s\")",
bufFrames, mFramesPendingInQueue, logTag);
mFramesPendingInQueue -= bufFrames;
}
status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
const sp<IMemory>& buffer, int64_t pts) {
{
Mutex::Autolock mttLock(mMediaTimeTransformLock);
if (!mMediaTimeTransformValid)
return INVALID_OPERATION;
}
Mutex::Autolock _l(mTimedBufferQueueLock);
uint32_t bufFrames = buffer->size() / mFrameSize;
mFramesPendingInQueue += bufFrames;
mTimedBufferQueue.add(TimedBuffer(buffer, pts));
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
target);
if (!(target == TimedAudioTrack::LOCAL_TIME ||
target == TimedAudioTrack::COMMON_TIME)) {
return BAD_VALUE;
}
Mutex::Autolock lock(mMediaTimeTransformLock);
mMediaTimeTransform = xform;
mMediaTimeTransformTarget = target;
mMediaTimeTransformValid = true;
return NO_ERROR;
}
#define min(a, b) ((a) < (b) ? (a) : (b))
// implementation of getNextBuffer for tracks whose buffers have timestamps
status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
AudioBufferProvider::Buffer* buffer, int64_t pts)
{
if (pts == AudioBufferProvider::kInvalidPTS) {
buffer->raw = NULL;
buffer->frameCount = 0;
mTimedAudioOutputOnTime = false;
return INVALID_OPERATION;
}
Mutex::Autolock _l(mTimedBufferQueueLock);
ALOG_ASSERT(!mQueueHeadInFlight,
"getNextBuffer called without releaseBuffer!");
while (true) {
// if we have no timed buffers, then fail
if (mTimedBufferQueue.isEmpty()) {
buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
// calculate the PTS of the head of the timed buffer queue expressed in
// local time
int64_t headLocalPTS;
{
Mutex::Autolock mttLock(mMediaTimeTransformLock);
ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
if (mMediaTimeTransform.a_to_b_denom == 0) {
// the transform represents a pause, so yield silence
timedYieldSilence_l(buffer->frameCount, buffer);
return NO_ERROR;
}
int64_t transformedPTS;
if (!mMediaTimeTransform.doForwardTransform(head.pts(),
&transformedPTS)) {
// the transform failed. this shouldn't happen, but if it does
// then just drop this buffer
ALOGW("timedGetNextBuffer transform failed");
buffer->raw = NULL;
buffer->frameCount = 0;
trimTimedBufferQueueHead_l("getNextBuffer; no transform");
return NO_ERROR;
}
if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
&headLocalPTS)) {
buffer->raw = NULL;
buffer->frameCount = 0;
return INVALID_OPERATION;
}
} else {
headLocalPTS = transformedPTS;
}
}
// adjust the head buffer's PTS to reflect the portion of the head buffer
// that has already been consumed
int64_t effectivePTS = headLocalPTS +
((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
// Calculate the delta in samples between the head of the input buffer
// queue and the start of the next output buffer that will be written.
// If the transformation fails because of over or underflow, it means
// that the sample's position in the output stream is so far out of
// whack that it should just be dropped.
int64_t sampleDelta;
if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
ALOGV("*** head buffer is too far from PTS: dropped buffer");
trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
" mix");
continue;
}
if (!mLocalTimeToSampleTransform.doForwardTransform(
(effectivePTS - pts) << 32, &sampleDelta)) {
ALOGV("*** too late during sample rate transform: dropped buffer");
trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
continue;
}
ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
" sampleDelta=[%d.%08x]",
head.pts(), head.position(), pts,
static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
+ (sampleDelta >> 32)),
static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
// if the delta between the ideal placement for the next input sample and
// the current output position is within this threshold, then we will
// concatenate the next input samples to the previous output
const int64_t kSampleContinuityThreshold =
(static_cast<int64_t>(sampleRate()) << 32) / 250;
// if this is the first buffer of audio that we're emitting from this track
// then it should be almost exactly on time.
const int64_t kSampleStartupThreshold = 1LL << 32;
if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
(!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
// the next input is close enough to being on time, so concatenate it
// with the last output
timedYieldSamples_l(buffer);
ALOGVV("*** on time: head.pos=%d frameCount=%u",
head.position(), buffer->frameCount);
return NO_ERROR;
}
// Looks like our output is not on time. Reset our on timed status.
// Next time we mix samples from our input queue, then should be within
// the StartupThreshold.
mTimedAudioOutputOnTime = false;
if (sampleDelta > 0) {
// the gap between the current output position and the proper start of
// the next input sample is too big, so fill it with silence
uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
timedYieldSilence_l(framesUntilNextInput, buffer);
ALOGV("*** silence: frameCount=%u", buffer->frameCount);
return NO_ERROR;
} else {
// the next input sample is late
uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
size_t onTimeSamplePosition =
head.position() + lateFrames * mFrameSize;
if (onTimeSamplePosition > head.buffer()->size()) {
// all the remaining samples in the head are too late, so
// drop it and move on
ALOGV("*** too late: dropped buffer");
trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
continue;
} else {
// skip over the late samples
head.setPosition(onTimeSamplePosition);
// yield the available samples
timedYieldSamples_l(buffer);
ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
return NO_ERROR;
}
}
}
}
// Yield samples from the timed buffer queue head up to the given output
// buffer's capacity.
//
// Caller must hold mTimedBufferQueueLock
void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
AudioBufferProvider::Buffer* buffer) {
const TimedBuffer& head = mTimedBufferQueue[0];
buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
head.position());
uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
mFrameSize);
size_t framesRequested = buffer->frameCount;
buffer->frameCount = min(framesLeftInHead, framesRequested);
mQueueHeadInFlight = true;
mTimedAudioOutputOnTime = true;
}
// Yield samples of silence up to the given output buffer's capacity
//
// Caller must hold mTimedBufferQueueLock
void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
// lazily allocate a buffer filled with silence
if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
delete [] mTimedSilenceBuffer;
mTimedSilenceBufferSize = numFrames * mFrameSize;
mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
}
buffer->raw = mTimedSilenceBuffer;
size_t framesRequested = buffer->frameCount;
buffer->frameCount = min(numFrames, framesRequested);
mTimedAudioOutputOnTime = false;
}
// AudioBufferProvider interface
void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
AudioBufferProvider::Buffer* buffer) {
Mutex::Autolock _l(mTimedBufferQueueLock);
// If the buffer which was just released is part of the buffer at the head
// of the queue, be sure to update the amt of the buffer which has been
// consumed. If the buffer being returned is not part of the head of the
// queue, its either because the buffer is part of the silence buffer, or
// because the head of the timed queue was trimmed after the mixer called
// getNextBuffer but before the mixer called releaseBuffer.
if (buffer->raw == mTimedSilenceBuffer) {
ALOG_ASSERT(!mQueueHeadInFlight,
"Queue head in flight during release of silence buffer!");
goto done;
}
ALOG_ASSERT(mQueueHeadInFlight,
"TimedTrack::releaseBuffer of non-silence buffer, but no queue"
" head in flight.");
if (mTimedBufferQueue.size()) {
TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
void* start = head.buffer()->pointer();
void* end = reinterpret_cast<void*>(
reinterpret_cast<uint8_t*>(head.buffer()->pointer())
+ head.buffer()->size());
ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
"released buffer not within the head of the timed buffer"
" queue; qHead = [%p, %p], released buffer = %p",
start, end, buffer->raw);
head.setPosition(head.position() +
(buffer->frameCount * mFrameSize));
mQueueHeadInFlight = false;
ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
"Bad bookkeeping during releaseBuffer! Should have at"
" least %u queued frames, but we think we have only %u",
buffer->frameCount, mFramesPendingInQueue);
mFramesPendingInQueue -= buffer->frameCount;
if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
|| mTrimQueueHeadOnRelease) {
trimTimedBufferQueueHead_l("releaseBuffer");
mTrimQueueHeadOnRelease = false;
}
} else {
LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
" buffers in the timed buffer queue");
}
done:
buffer->raw = 0;
buffer->frameCount = 0;
}
size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
Mutex::Autolock _l(mTimedBufferQueueLock);
return mFramesPendingInQueue;
}
AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
: mPTS(0), mPosition(0) {}
AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
const sp<IMemory>& buffer, int64_t pts)
: mBuffer(buffer), mPTS(pts), mPosition(0) {}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
PlaybackThread *playbackThread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount)
: Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
NULL, 0, IAudioFlinger::TRACK_DEFAULT),
mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
{
if (mCblk != NULL) {
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
"mCblk->frameCount_ %u, mChannelMask 0x%08x mBufferEnd %p",
mCblk, mBuffer,
mCblk->frameCount_, mChannelMask, mBufferEnd);
// since client and server are in the same process,
// the buffer has the same virtual address on both sides
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
mClientProxy->setSendLevel(0.0);
mClientProxy->setSampleRate(sampleRate);
} else {
ALOGW("Error creating output track on thread %p", playbackThread);
}
}
AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
{
clearBufferQueue();
delete mClientProxy;
// superclass destructor will now delete the server proxy and shared memory both refer to
}
status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
int triggerSession)
{
status_t status = Track::start(event, triggerSession);
if (status != NO_ERROR) {
return status;
}
mActive = true;
mRetryCount = 127;
return status;
}
void AudioFlinger::PlaybackThread::OutputTrack::stop()
{
Track::stop();
clearBufferQueue();
mOutBuffer.frameCount = 0;
mActive = false;
}
bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
uint32_t channelCount = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
start();
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
MixerThread *mixerThread = (MixerThread *)thread.get();
if (mFrameCount > frames) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
uint32_t startFrames = (mFrameCount - frames);
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
pInBuffer->frameCount = startFrames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else {
ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
}
}
}
}
while (waitTimeLeftMs) {
// First write pending buffers, then new data
if (mBufferQueue.size()) {
pInBuffer = mBufferQueue.itemAt(0);
} else {
pInBuffer = &inBuffer;
}
if (pInBuffer->frameCount == 0) {
break;
}
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
mThread.unsafe_get());
outputBufferFull = true;
break;
}
uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
if (waitTimeLeftMs >= waitTimeMs) {
waitTimeLeftMs -= waitTimeMs;
} else {
waitTimeLeftMs = 0;
}
}
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
mClientProxy->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channelCount;
mOutBuffer.frameCount -= outFrames;
mOutBuffer.i16 += outFrames * channelCount;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
delete [] pInBuffer->mBuffer;
delete pInBuffer;
ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
} else {
break;
}
}
}
// If we could not write all frames, allocate a buffer and queue it for next time.
if (inBuffer.frameCount) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->i16 = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
sizeof(int16_t));
mBufferQueue.add(pInBuffer);
ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
} else {
ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
mThread.unsafe_get(), this);
}
}
}
// Calling write() with a 0 length buffer, means that no more data will be written:
// If no more buffers are pending, fill output track buffer to make sure it is started
// by output mixer.
if (frames == 0 && mBufferQueue.size() == 0) {
if (mCblk->user < mFrameCount) {
frames = mFrameCount - mCblk->user;
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[frames * channelCount];
pInBuffer->frameCount = frames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else if (mActive) {
stop();
}
}
return outputBufferFull;
}
status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = buffer->frameCount;
ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
buffer->frameCount = 0;
size_t framesAvail;
{
Mutex::Autolock _l(cblk->lock);
// read the server count again
while (!(framesAvail = mClientProxy->framesAvailable_l())) {
if (CC_UNLIKELY(!mActive)) {
ALOGV("Not active and NO_MORE_BUFFERS");
return NO_MORE_BUFFERS;
}
status_t result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (result != NO_ERROR) {
return NO_MORE_BUFFERS;
}
}
}
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + mFrameCount;
if (framesReq > bufferEnd - u) {
framesReq = bufferEnd - u;
}
buffer->frameCount = framesReq;
buffer->raw = mClientProxy->buffer(u);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
for (size_t i = 0; i < size; i++) {
Buffer *pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
delete pBuffer;
}
mBufferQueue.clear();
}
// ----------------------------------------------------------------------------
// Record
// ----------------------------------------------------------------------------
AudioFlinger::RecordHandle::RecordHandle(
const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
}
AudioFlinger::RecordHandle::~RecordHandle() {
stop_nonvirtual();
mRecordTrack->destroy();
}
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
return mRecordTrack->getCblk();
}
status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
int triggerSession) {
ALOGV("RecordHandle::start()");
return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
}
void AudioFlinger::RecordHandle::stop() {
stop_nonvirtual();
}
void AudioFlinger::RecordHandle::stop_nonvirtual() {
ALOGV("RecordHandle::stop()");
mRecordTrack->stop();
}
status_t AudioFlinger::RecordHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioRecord::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
mOverflow(false)
{
ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
ALOGV("%s", __func__);
}
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mStepServerFailed) {
if (!step()) {
goto getNextBuffer_exit;
}
ALOGV("stepServer recovered");
mStepServerFailed = false;
}
// FIXME lock is not actually held, so overrun is possible
framesAvail = mServerProxy->framesAvailableIn_l();
if (CC_LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + mFrameCount;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
if (framesReq > bufferEnd - s) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
int triggerSession)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->start(this, event, triggerSession);
} else {
return BAD_VALUE;
}
}
void AudioFlinger::RecordThread::RecordTrack::stop()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
recordThread->mLock.lock();
bool doStop = recordThread->stop_l(this);
if (doStop) {
TrackBase::reset();
// Force overrun condition to avoid false overrun callback until first data is
// read from buffer
android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
}
recordThread->mLock.unlock();
if (doStop) {
AudioSystem::stopInput(recordThread->id());
}
}
}
void AudioFlinger::RecordThread::RecordTrack::destroy()
{
// see comments at AudioFlinger::PlaybackThread::Track::destroy()
sp<RecordTrack> keep(this);
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
if (mState == ACTIVE || mState == RESUMING) {
AudioSystem::stopInput(thread->id());
}
AudioSystem::releaseInput(thread->id());
Mutex::Autolock _l(thread->mLock);
RecordThread *recordThread = (RecordThread *) thread.get();
recordThread->destroyTrack_l(this);
}
}
}
/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
{
result.append(" Clien Fmt Chn mask Session Step S Serv User FrameCount\n");
}
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %08x %05d\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
mSessionId,
mStepCount,
mState,
mCblk->server,
mCblk->user,
mFrameCount);
}
}; // namespace android