/*
**
** 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 "Configuration.h"
#include <linux/futex.h>
#include <math.h>
#include <sys/syscall.h>
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#include "ServiceUtilities.h"
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
#include <audio_utils/minifloat.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
// TODO move to a common header (Also shared with AudioTrack.cpp)
#define NANOS_PER_SECOND 1000000000
#define TIME_TO_NANOS(time) ((uint64_t)time.tv_sec * NANOS_PER_SECOND + time.tv_nsec)
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,
void *buffer,
audio_session_t sessionId,
int clientUid,
IAudioFlinger::track_flags_t flags,
bool isOut,
alloc_type alloc,
track_type type)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(NULL),
// mBuffer
mState(IDLE),
mSampleRate(sampleRate),
mFormat(format),
mChannelMask(channelMask),
mChannelCount(isOut ?
audio_channel_count_from_out_mask(channelMask) :
audio_channel_count_from_in_mask(channelMask)),
mFrameSize(audio_has_proportional_frames(format) ?
mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
mFrameCount(frameCount),
mSessionId(sessionId),
mFlags(flags),
mIsOut(isOut),
mServerProxy(NULL),
mId(android_atomic_inc(&nextTrackId)),
mTerminated(false),
mType(type),
mThreadIoHandle(thread->id())
{
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
if (!isTrustedCallingUid(callingUid) || clientUid == -1) {
ALOGW_IF(clientUid != -1 && clientUid != (int)callingUid,
"%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
clientUid = (int)callingUid;
}
// clientUid contains the uid of the app that is responsible for this track, so we can blame
// battery usage on it.
mUid = clientUid;
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize;
if (buffer == NULL && alloc == ALLOC_CBLK) {
size += bufferSize;
}
if (client != 0) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory == 0 ||
(mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
ALOGE("not enough memory for AudioTrack size=%zu", size);
client->heap()->dump("AudioTrack");
mCblkMemory.clear();
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();
switch (alloc) {
case ALLOC_READONLY: {
const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
if (roHeap == 0 ||
(mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
(mBuffer = mBufferMemory->pointer()) == NULL) {
ALOGE("not enough memory for read-only buffer size=%zu", bufferSize);
if (roHeap != 0) {
roHeap->dump("buffer");
}
mCblkMemory.clear();
mBufferMemory.clear();
return;
}
memset(mBuffer, 0, bufferSize);
} break;
case ALLOC_PIPE:
mBufferMemory = thread->pipeMemory();
// mBuffer is the virtual address as seen from current process (mediaserver),
// and should normally be coming from mBufferMemory->pointer().
// However in this case the TrackBase does not reference the buffer directly.
// It should references the buffer via the pipe.
// Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
mBuffer = NULL;
break;
case ALLOC_CBLK:
// clear all buffers
if (buffer == NULL) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
} else {
mBuffer = buffer;
#if 0
mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic
#endif
}
break;
case ALLOC_LOCAL:
mBuffer = calloc(1, bufferSize);
break;
case ALLOC_NONE:
mBuffer = buffer;
break;
}
#ifdef TEE_SINK
if (mTeeSinkTrackEnabled) {
NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat);
if (Format_isValid(pipeFormat)) {
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
}
}
status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const
{
status_t status;
if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) {
status = cblk() != NULL ? NO_ERROR : NO_MEMORY;
} else {
status = getCblk() != 0 ? NO_ERROR : NO_MEMORY;
}
return status;
}
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 client mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
// 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();
}
// flush the binder command buffer
IPCThreadState::self()->flushCommands();
}
// AudioBufferProvider interface
// getNextBuffer() = 0;
// This implementation of releaseBuffer() is used by Track and RecordTrack
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
#ifdef TEE_SINK
if (mTeeSink != 0) {
(void) mTeeSink->write(buffer->raw, buffer->frameCount);
}
#endif
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
buffer->frameCount = 0;
buffer->raw = NULL;
mServerProxy->releaseBuffer(&buf);
}
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::setParameters(const String8& keyValuePairs) {
return mTrack->setParameters(keyValuePairs);
}
status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
{
return mTrack->getTimestamp(timestamp);
}
void AudioFlinger::TrackHandle::signal()
{
return mTrack->signal();
}
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,
void *buffer,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
int uid,
IAudioFlinger::track_flags_t flags,
track_type type)
: TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
(sharedBuffer != 0) ? sharedBuffer->pointer() : buffer,
sessionId, uid, flags, true /*isOut*/,
(type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
type),
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),
mFrameMap(16 /* sink-frame-to-track-frame map memory */),
// mSinkTimestamp
mFastIndex(-1),
mCachedVolume(1.0),
mIsInvalid(false),
mAudioTrackServerProxy(NULL),
mResumeToStopping(false),
mFlushHwPending(false)
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(),
sharedBuffer->size());
if (mCblk == NULL) {
return;
}
if (sharedBuffer == 0) {
mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack(), sampleRate);
} else {
mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize);
}
mServerProxy = mAudioTrackServerProxy;
mName = thread->getTrackName_l(channelMask, format, sessionId);
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) {
// FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential
// race with setSyncEvent(). However, if we call it, we cannot properly start
// static fast tracks (SoundPool) immediately after stopping.
//mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
int i = __builtin_ctz(thread->mFastTrackAvailMask);
ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
// 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;
thread->mFastTrackAvailMask &= ~(1 << i);
}
}
AudioFlinger::PlaybackThread::Track::~Track()
{
ALOGV("PlaybackThread::Track destructor");
// The destructor would clear mSharedBuffer,
// but it will not push the decremented reference count,
// leaving the client's IMemory dangling indefinitely.
// This prevents that leak.
if (mSharedBuffer != 0) {
mSharedBuffer.clear();
}
}
status_t AudioFlinger::PlaybackThread::Track::initCheck() const
{
status_t status = TrackBase::initCheck();
if (status == NO_ERROR && mName < 0) {
status = NO_MEMORY;
}
return status;
}
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
bool wasActive = false;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
wasActive = playbackThread->destroyTrack_l(this);
}
if (isExternalTrack() && !wasActive) {
AudioSystem::releaseOutput(mThreadIoHandle, mStreamType, mSessionId);
}
}
}
/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
{
result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate "
"L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n");
}
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
{
gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
if (isFastTrack()) {
sprintf(buffer, " F %2d", mFastIndex);
} else if (mName >= AudioMixer::TRACK0) {
sprintf(buffer, " %4d", mName - AudioMixer::TRACK0);
} else {
sprintf(buffer, " none");
}
track_state state = mState;
char stateChar;
if (isTerminated()) {
stateChar = 'T';
} else {
switch (state) {
case IDLE:
stateChar = 'I';
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[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g "
"%08X %p %p 0x%03X %9u%c\n",
active ? "yes" : "no",
(mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
mChannelMask,
mSessionId,
mFrameCount,
stateChar,
mFillingUpStatus,
mAudioTrackServerProxy->getSampleRate(),
20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
mCblk->mServer,
mMainBuffer,
mAuxBuffer,
mCblk->mFlags,
mAudioTrackServerProxy->getUnderrunFrames(),
nowInUnderrun);
}
uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
return mAudioTrackServerProxy->getSampleRate();
}
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
size_t desiredFrames = buffer->frameCount;
buf.mFrameCount = desiredFrames;
status_t status = mServerProxy->obtainBuffer(&buf);
buffer->frameCount = buf.mFrameCount;
buffer->raw = buf.mRaw;
if (buf.mFrameCount == 0) {
mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
} else {
mAudioTrackServerProxy->tallyUnderrunFrames(0);
}
return status;
}
// releaseBuffer() is not overridden
// ExtendedAudioBufferProvider interface
// framesReady() may return an approximation of the number of frames if called
// from a different thread than the one calling Proxy->obtainBuffer() and
// Proxy->releaseBuffer(). Also note there is no mutual exclusion in the
// AudioTrackServerProxy so be especially careful calling with FastTracks.
size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
if (mSharedBuffer != 0 && (isStopped() || isStopping())) {
// Static tracks return zero frames immediately upon stopping (for FastTracks).
// The remainder of the buffer is not drained.
return 0;
}
return mAudioTrackServerProxy->framesReady();
}
int64_t AudioFlinger::PlaybackThread::Track::framesReleased() const
{
return mAudioTrackServerProxy->framesReleased();
}
void AudioFlinger::PlaybackThread::Track::onTimestamp(const ExtendedTimestamp ×tamp)
{
// This call comes from a FastTrack and should be kept lockless.
// The server side frames are already translated to client frames.
mAudioTrackServerProxy->setTimestamp(timestamp);
// We do not set drained here, as FastTrack timestamp may not go to very last frame.
}
// 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 (isStopping()) {
if (framesReady() > 0) {
mFillingUpStatus = FS_FILLED;
}
return true;
}
if (framesReady() >= mServerProxy->getBufferSizeInFrames() ||
(mCblk->mFlags & CBLK_FORCEREADY)) {
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
return true;
}
return false;
}
status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
audio_session_t triggerSession __unused)
{
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) {
if (isOffloaded()) {
Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
Mutex::Autolock _lth(thread->mLock);
sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
(ec != 0 && ec->isNonOffloadableEnabled())) {
invalidate();
return PERMISSION_DENIED;
}
}
Mutex::Autolock _lth(thread->mLock);
track_state state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
// initial state-stopping. next state-pausing.
// What if resume is called ?
if (state == PAUSED || state == PAUSING) {
if (mResumeToStopping) {
// happened we need to resume to STOPPING_1
mState = TrackBase::STOPPING_1;
ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
} else {
mState = TrackBase::RESUMING;
ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
}
} else {
mState = TrackBase::ACTIVE;
ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
}
// states to reset position info for non-offloaded/direct tracks
if (!isOffloaded() && !isDirect()
&& (state == IDLE || state == STOPPED || state == FLUSHED)) {
mFrameMap.reset();
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (isFastTrack()) {
// refresh fast track underruns on start because that field is never cleared
// by the fast mixer; furthermore, the same track can be recycled, i.e. start
// after stop.
mObservedUnderruns = playbackThread->getFastTrackUnderruns(mFastIndex);
}
status = playbackThread->addTrack_l(this);
if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
// restore previous state if start was rejected by policy manager
if (status == PERMISSION_DENIED) {
mState = state;
}
}
// track was already in the active list, not a problem
if (status == ALREADY_EXISTS) {
status = NO_ERROR;
} else {
// Acknowledge any pending flush(), so that subsequent new data isn't discarded.
// It is usually unsafe to access the server proxy from a binder thread.
// But in this case we know the mixer thread (whether normal mixer or fast mixer)
// isn't looking at this track yet: we still hold the normal mixer thread lock,
// and for fast tracks the track is not yet in the fast mixer thread's active set.
// For static tracks, this is used to acknowledge change in position or loop.
ServerProxy::Buffer buffer;
buffer.mFrameCount = 1;
(void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
}
} 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() && !isOffloaded() && !isDirect()) {
mState = STOPPED;
} else {
// For fast tracks prepareTracks_l() will set state to STOPPING_2
// presentation is complete
// For an offloaded track this starts a drain and state will
// move to STOPPING_2 when drain completes and then STOPPED
mState = STOPPING_1;
if (isOffloaded()) {
mRetryCount = PlaybackThread::kMaxTrackStopRetriesOffload;
}
}
playbackThread->broadcast_l();
ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
playbackThread);
}
}
}
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);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
switch (mState) {
case STOPPING_1:
case STOPPING_2:
if (!isOffloaded()) {
/* nothing to do if track is not offloaded */
break;
}
// Offloaded track was draining, we need to carry on draining when resumed
mResumeToStopping = true;
// fall through...
case ACTIVE:
case RESUMING:
mState = PAUSING;
ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
playbackThread->broadcast_l();
break;
default:
break;
}
}
}
void AudioFlinger::PlaybackThread::Track::flush()
{
ALOGV("flush(%d)", mName);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (isOffloaded()) {
// If offloaded we allow flush during any state except terminated
// and keep the track active to avoid problems if user is seeking
// rapidly and underlying hardware has a significant delay handling
// a pause
if (isTerminated()) {
return;
}
ALOGV("flush: offload flush");
reset();
if (mState == STOPPING_1 || mState == STOPPING_2) {
ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
mState = ACTIVE;
}
mFlushHwPending = true;
mResumeToStopping = false;
} else {
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
if (isDirect()) {
mFlushHwPending = true;
}
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
}
// Prevent flush being lost if the track is flushed and then resumed
// before mixer thread can run. This is important when offloading
// because the hardware buffer could hold a large amount of audio
playbackThread->broadcast_l();
}
}
// must be called with thread lock held
void AudioFlinger::PlaybackThread::Track::flushAck()
{
if (!isOffloaded() && !isDirect())
return;
mFlushHwPending = false;
}
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) {
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
mFillingUpStatus = FS_FILLING;
mResetDone = true;
if (mState == FLUSHED) {
mState = IDLE;
}
}
}
status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
{
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
ALOGE("thread is dead");
return FAILED_TRANSACTION;
} else if ((thread->type() == ThreadBase::DIRECT) ||
(thread->type() == ThreadBase::OFFLOAD)) {
return thread->setParameters(keyValuePairs);
} else {
return PERMISSION_DENIED;
}
}
status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
{
if (!isOffloaded() && !isDirect()) {
return INVALID_OPERATION; // normal tracks handled through SSQ
}
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return INVALID_OPERATION;
}
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
return playbackThread->getTimestamp_l(timestamp);
}
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);
status = playbackThread->addEffect_l(effect);
if (status != NO_ERROR) {
srcThread->addEffect_l(effect);
return INVALID_OPERATION;
}
// 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());
AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
}
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(
int64_t framesWritten, size_t audioHalFrames)
{
// TODO: improve this based on FrameMap if it exists, to ensure full drain.
// This assists in proper timestamp computation as well as wakelock management.
// 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.
// For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
// to detect when all frames have been played. In this case framesWritten isn't
// useful because it doesn't always reflect whether there is data in the h/w
// buffers, particularly if a track has been paused and resumed during draining
ALOGV("presentationComplete() mPresentationCompleteFrames %lld framesWritten %lld",
(long long)mPresentationCompleteFrames, (long long)framesWritten);
if (mPresentationCompleteFrames == 0) {
mPresentationCompleteFrames = framesWritten + audioHalFrames;
ALOGV("presentationComplete() reset: mPresentationCompleteFrames %lld audioHalFrames %zu",
(long long)mPresentationCompleteFrames, audioHalFrames);
}
bool complete;
if (isOffloaded()) {
complete = true;
} else if (isDirect() || isFastTrack()) { // these do not go through linear map
complete = framesWritten >= (int64_t) mPresentationCompleteFrames;
} else { // Normal tracks, OutputTracks, and PatchTracks
complete = framesWritten >= (int64_t) mPresentationCompleteFrames
&& mAudioTrackServerProxy->isDrained();
}
if (complete) {
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
mAudioTrackServerProxy->setStreamEndDone();
return true;
}
return false;
}
void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
{
for (size_t i = 0; i < mSyncEvents.size(); i++) {
if (mSyncEvents[i]->type() == type) {
mSyncEvents[i]->trigger();
mSyncEvents.removeAt(i);
i--;
}
}
}
// implement VolumeBufferProvider interface
gain_minifloat_packed_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));
gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
if (vl > GAIN_FLOAT_UNITY) {
vl = GAIN_FLOAT_UNITY;
}
if (vr > GAIN_FLOAT_UNITY) {
vr = GAIN_FLOAT_UNITY;
}
// 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 packed minifloat
vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
// FIXME look at mute, pause, and stop flags
return vlr;
}
status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
{
if (isTerminated() || mState == PAUSED ||
((framesReady() == 0) && ((mSharedBuffer != 0) ||
(mState == STOPPED)))) {
ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %zu",
mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
event->cancel();
return INVALID_OPERATION;
}
(void) TrackBase::setSyncEvent(event);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::Track::invalidate()
{
signalClientFlag(CBLK_INVALID);
mIsInvalid = true;
}
void AudioFlinger::PlaybackThread::Track::disable()
{
signalClientFlag(CBLK_DISABLED);
}
void AudioFlinger::PlaybackThread::Track::signalClientFlag(int32_t flag)
{
// FIXME should use proxy, and needs work
audio_track_cblk_t* cblk = mCblk;
android_atomic_or(flag, &cblk->mFlags);
android_atomic_release_store(0x40000000, &cblk->mFutex);
// client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
(void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
}
void AudioFlinger::PlaybackThread::Track::signal()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
PlaybackThread *t = (PlaybackThread *)thread.get();
Mutex::Autolock _l(t->mLock);
t->broadcast_l();
}
}
//To be called with thread lock held
bool AudioFlinger::PlaybackThread::Track::isResumePending() {
if (mState == RESUMING)
return true;
/* Resume is pending if track was stopping before pause was called */
if (mState == STOPPING_1 &&
mResumeToStopping)
return true;
return false;
}
//To be called with thread lock held
void AudioFlinger::PlaybackThread::Track::resumeAck() {
if (mState == RESUMING)
mState = ACTIVE;
// Other possibility of pending resume is stopping_1 state
// Do not update the state from stopping as this prevents
// drain being called.
if (mState == STOPPING_1) {
mResumeToStopping = false;
}
}
//To be called with thread lock held
void AudioFlinger::PlaybackThread::Track::updateTrackFrameInfo(
int64_t trackFramesReleased, int64_t sinkFramesWritten,
const ExtendedTimestamp &timeStamp) {
//update frame map
mFrameMap.push(trackFramesReleased, sinkFramesWritten);
// adjust server times and set drained state.
//
// Our timestamps are only updated when the track is on the Thread active list.
// We need to ensure that tracks are not removed before full drain.
ExtendedTimestamp local = timeStamp;
bool checked = false;
for (int i = ExtendedTimestamp::LOCATION_MAX - 1;
i >= ExtendedTimestamp::LOCATION_SERVER; --i) {
// Lookup the track frame corresponding to the sink frame position.
if (local.mTimeNs[i] > 0) {
local.mPosition[i] = mFrameMap.findX(local.mPosition[i]);
// check drain state from the latest stage in the pipeline.
if (!checked && i <= ExtendedTimestamp::LOCATION_KERNEL) {
mAudioTrackServerProxy->setDrained(
local.mPosition[i] >= mAudioTrackServerProxy->framesReleased());
checked = true;
}
}
}
if (!checked) { // no server info, assume drained.
mAudioTrackServerProxy->setDrained(true);
}
// Set correction for flushed frames that are not accounted for in released.
local.mFlushed = mAudioTrackServerProxy->framesFlushed();
mServerProxy->setTimestamp(local);
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
PlaybackThread *playbackThread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
int uid)
: Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
sampleRate, format, channelMask, frameCount,
NULL, 0, AUDIO_SESSION_NONE, uid, IAudioFlinger::TRACK_DEFAULT,
TYPE_OUTPUT),
mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
{
if (mCblk != NULL) {
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
"frameCount %zu, mChannelMask 0x%08x",
mCblk, mBuffer,
frameCount, mChannelMask);
// 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,
true /*clientInServer*/);
mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
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,
audio_session_t 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(void* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.raw = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
(void) start();
}
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();
status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
if (status != NO_ERROR && status != NOT_ENOUGH_DATA) {
ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
mThread.unsafe_get(), status);
outputBufferFull = true;
break;
}
uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
if (waitTimeLeftMs >= waitTimeMs) {
waitTimeLeftMs -= waitTimeMs;
} else {
waitTimeLeftMs = 0;
}
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
continue;
}
}
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize);
Proxy::Buffer buf;
buf.mFrameCount = outFrames;
buf.mRaw = NULL;
mClientProxy->releaseBuffer(&buf);
restartIfDisabled();
pInBuffer->frameCount -= outFrames;
pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize;
mOutBuffer.frameCount -= outFrames;
mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
free(pInBuffer->mBuffer);
delete pInBuffer;
ALOGV("OutputTrack::write() %p thread %p released overflow buffer %zu", 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 = malloc(inBuffer.frameCount * mFrameSize);
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->raw = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
mBufferQueue.add(pInBuffer);
ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %zu", 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:
// We rely on stop() to set the appropriate flags to allow the remaining frames to play out.
if (frames == 0 && mBufferQueue.size() == 0 && mActive) {
stop();
}
return outputBufferFull;
}
status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
ClientProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
struct timespec timeout;
timeout.tv_sec = waitTimeMs / 1000;
timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
buffer->frameCount = buf.mFrameCount;
buffer->raw = buf.mRaw;
return status;
}
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
for (size_t i = 0; i < size; i++) {
Buffer *pBuffer = mBufferQueue.itemAt(i);
free(pBuffer->mBuffer);
delete pBuffer;
}
mBufferQueue.clear();
}
void AudioFlinger::PlaybackThread::OutputTrack::restartIfDisabled()
{
int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
if (mActive && (flags & CBLK_DISABLED)) {
start();
}
}
AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
IAudioFlinger::track_flags_t flags)
: Track(playbackThread, NULL, streamType,
sampleRate, format, channelMask, frameCount,
buffer, 0, AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true))
{
uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) /
playbackThread->sampleRate();
mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec",
this, sampleRate,
(int)mPeerTimeout.tv_sec,
(int)(mPeerTimeout.tv_nsec / 1000000));
}
AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack()
{
}
status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
status_t status = Track::start(event, triggerSession);
if (status != NO_ERROR) {
return status;
}
android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
return status;
}
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy");
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status);
buffer->frameCount = buf.mFrameCount;
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
status = Track::getNextBuffer(buffer);
return status;
}
void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy");
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
mPeerProxy->releaseBuffer(&buf);
TrackBase::releaseBuffer(buffer);
}
status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
const struct timespec *timeOut)
{
status_t status = NO_ERROR;
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
do {
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
}
status = mProxy->obtainBuffer(buffer, timeOut);
} while ((status == NOT_ENOUGH_DATA) && (tryCounter-- > 0));
return status;
}
void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
{
mProxy->releaseBuffer(buffer);
restartIfDisabled();
android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
}
void AudioFlinger::PlaybackThread::PatchTrack::restartIfDisabled()
{
if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) {
ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting");
start();
}
}
// ----------------------------------------------------------------------------
// Record
// ----------------------------------------------------------------------------
AudioFlinger::RecordHandle::RecordHandle(
const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
}
AudioFlinger::RecordHandle::~RecordHandle() {
stop_nonvirtual();
mRecordTrack->destroy();
}
status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
audio_session_t 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 and ThreadBase::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,
void *buffer,
audio_session_t sessionId,
int uid,
IAudioFlinger::track_flags_t flags,
track_type type)
: TrackBase(thread, client, sampleRate, format,
channelMask, frameCount, buffer, sessionId, uid,
flags, false /*isOut*/,
(type == TYPE_DEFAULT) ?
((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
type),
mOverflow(false),
mFramesToDrop(0),
mResamplerBufferProvider(NULL), // initialize in case of early constructor exit
mRecordBufferConverter(NULL)
{
if (mCblk == NULL) {
return;
}
mRecordBufferConverter = new RecordBufferConverter(
thread->mChannelMask, thread->mFormat, thread->mSampleRate,
channelMask, format, sampleRate);
// Check if the RecordBufferConverter construction was successful.
// If not, don't continue with construction.
//
// NOTE: It would be extremely rare that the record track cannot be created
// for the current device, but a pending or future device change would make
// the record track configuration valid.
if (mRecordBufferConverter->initCheck() != NO_ERROR) {
ALOGE("RecordTrack unable to create record buffer converter");
return;
}
mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack());
mResamplerBufferProvider = new ResamplerBufferProvider(this);
if (flags & IAudioFlinger::TRACK_FAST) {
ALOG_ASSERT(thread->mFastTrackAvail);
thread->mFastTrackAvail = false;
}
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
ALOGV("%s", __func__);
delete mRecordBufferConverter;
delete mResamplerBufferProvider;
}
status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const
{
status_t status = TrackBase::initCheck();
if (status == NO_ERROR && mServerProxy == 0) {
status = BAD_VALUE;
}
return status;
}
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
status_t status = mServerProxy->obtainBuffer(&buf);
buffer->frameCount = buf.mFrameCount;
buffer->raw = buf.mRaw;
if (buf.mFrameCount == 0) {
// FIXME also wake futex so that overrun is noticed more quickly
(void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
}
return status;
}
status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
audio_session_t 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();
if (recordThread->stop(this) && isExternalTrack()) {
AudioSystem::stopInput(mThreadIoHandle, mSessionId);
}
}
}
void AudioFlinger::RecordThread::RecordTrack::destroy()
{
// see comments at AudioFlinger::PlaybackThread::Track::destroy()
sp<RecordTrack> keep(this);
{
if (isExternalTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
AudioSystem::stopInput(mThreadIoHandle, mSessionId);
}
AudioSystem::releaseInput(mThreadIoHandle, mSessionId);
}
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
RecordThread *recordThread = (RecordThread *) thread.get();
recordThread->destroyTrack_l(this);
}
}
}
void AudioFlinger::RecordThread::RecordTrack::invalidate()
{
// FIXME should use proxy, and needs work
audio_track_cblk_t* cblk = mCblk;
android_atomic_or(CBLK_INVALID, &cblk->mFlags);
android_atomic_release_store(0x40000000, &cblk->mFutex);
// client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
(void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
}
/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
{
result.append(" Active Client Fmt Chn mask Session S Server fCount SRate\n");
}
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
{
snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n",
active ? "yes" : "no",
(mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
mSessionId,
mState,
mCblk->mServer,
mFrameCount,
mSampleRate);
}
void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
{
if (event == mSyncStartEvent) {
ssize_t framesToDrop = 0;
sp<ThreadBase> threadBase = mThread.promote();
if (threadBase != 0) {
// TODO: use actual buffer filling status instead of 2 buffers when info is available
// from audio HAL
framesToDrop = threadBase->mFrameCount * 2;
}
mFramesToDrop = framesToDrop;
}
}
void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
{
if (mSyncStartEvent != 0) {
mSyncStartEvent->cancel();
mSyncStartEvent.clear();
}
mFramesToDrop = 0;
}
void AudioFlinger::RecordThread::RecordTrack::updateTrackFrameInfo(
int64_t trackFramesReleased, int64_t sourceFramesRead,
uint32_t halSampleRate, const ExtendedTimestamp ×tamp)
{
ExtendedTimestamp local = timestamp;
// Convert HAL frames to server-side track frames at track sample rate.
// We use trackFramesReleased and sourceFramesRead as an anchor point.
for (int i = ExtendedTimestamp::LOCATION_SERVER; i < ExtendedTimestamp::LOCATION_MAX; ++i) {
if (local.mTimeNs[i] != 0) {
const int64_t relativeServerFrames = local.mPosition[i] - sourceFramesRead;
const int64_t relativeTrackFrames = relativeServerFrames
* mSampleRate / halSampleRate; // TODO: potential computation overflow
local.mPosition[i] = relativeTrackFrames + trackFramesReleased;
}
}
mServerProxy->setTimestamp(local);
}
AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
IAudioFlinger::track_flags_t flags)
: RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount,
buffer, AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true))
{
uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) /
recordThread->sampleRate();
mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec",
this, sampleRate,
(int)mPeerTimeout.tv_sec,
(int)(mPeerTimeout.tv_nsec / 1000000));
}
AudioFlinger::RecordThread::PatchRecord::~PatchRecord()
{
}
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy");
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
ALOGV_IF(status != NO_ERROR,
"PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status);
buffer->frameCount = buf.mFrameCount;
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
status = RecordTrack::getNextBuffer(buffer);
return status;
}
void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy");
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
mPeerProxy->releaseBuffer(&buf);
TrackBase::releaseBuffer(buffer);
}
status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer,
const struct timespec *timeOut)
{
return mProxy->obtainBuffer(buffer, timeOut);
}
void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer)
{
mProxy->releaseBuffer(buffer);
}
} // namespace android