Tracks.cpp revision 9cae217050aa1347d4ac5053c305754879e3f97f 
1/*
2**
3** Copyright 2012, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <math.h>
24#include <utils/Log.h>
25
26#include <private/media/AudioTrackShared.h>
27
28#include <common_time/cc_helper.h>
29#include <common_time/local_clock.h>
30
31#include "AudioMixer.h"
32#include "AudioFlinger.h"
33#include "ServiceUtilities.h"
34
35#include <media/nbaio/Pipe.h>
36#include <media/nbaio/PipeReader.h>
37
38// ----------------------------------------------------------------------------
39
40// Note: the following macro is used for extremely verbose logging message.  In
41// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
42// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
43// are so verbose that we want to suppress them even when we have ALOG_ASSERT
44// turned on.  Do not uncomment the #def below unless you really know what you
45// are doing and want to see all of the extremely verbose messages.
46//#define VERY_VERY_VERBOSE_LOGGING
47#ifdef VERY_VERY_VERBOSE_LOGGING
48#define ALOGVV ALOGV
49#else
50#define ALOGVV(a...) do { } while(0)
51#endif
52
53namespace android {
54
55// ----------------------------------------------------------------------------
56//      TrackBase
57// ----------------------------------------------------------------------------
58
59static volatile int32_t nextTrackId = 55;
60
61// TrackBase constructor must be called with AudioFlinger::mLock held
62AudioFlinger::ThreadBase::TrackBase::TrackBase(
63            ThreadBase *thread,
64            const sp<Client>& client,
65            uint32_t sampleRate,
66            audio_format_t format,
67            audio_channel_mask_t channelMask,
68            size_t frameCount,
69            const sp<IMemory>& sharedBuffer,
70            int sessionId,
71            int clientUid,
72            bool isOut)
73    :   RefBase(),
74        mThread(thread),
75        mClient(client),
76        mCblk(NULL),
77        // mBuffer
78        mState(IDLE),
79        mSampleRate(sampleRate),
80        mFormat(format),
81        mChannelMask(channelMask),
82        mChannelCount(popcount(channelMask)),
83        mFrameSize(audio_is_linear_pcm(format) ?
84                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
85        mFrameCount(frameCount),
86        mSessionId(sessionId),
87        mIsOut(isOut),
88        mServerProxy(NULL),
89        mId(android_atomic_inc(&nextTrackId)),
90        mTerminated(false)
91{
92    // if the caller is us, trust the specified uid
93    if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
94        int newclientUid = IPCThreadState::self()->getCallingUid();
95        if (clientUid != -1 && clientUid != newclientUid) {
96            ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
97        }
98        clientUid = newclientUid;
99    }
100    // clientUid contains the uid of the app that is responsible for this track, so we can blame
101    // battery usage on it.
102    mUid = clientUid;
103
104    // client == 0 implies sharedBuffer == 0
105    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
106
107    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
108            sharedBuffer->size());
109
110    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
111    size_t size = sizeof(audio_track_cblk_t);
112    size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
113    if (sharedBuffer == 0) {
114        size += bufferSize;
115    }
116
117    if (client != 0) {
118        mCblkMemory = client->heap()->allocate(size);
119        if (mCblkMemory != 0) {
120            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
121            // can't assume mCblk != NULL
122        } else {
123            ALOGE("not enough memory for AudioTrack size=%u", size);
124            client->heap()->dump("AudioTrack");
125            return;
126        }
127    } else {
128        // this syntax avoids calling the audio_track_cblk_t constructor twice
129        mCblk = (audio_track_cblk_t *) new uint8_t[size];
130        // assume mCblk != NULL
131    }
132
133    // construct the shared structure in-place.
134    if (mCblk != NULL) {
135        new(mCblk) audio_track_cblk_t();
136        // clear all buffers
137        mCblk->frameCount_ = frameCount;
138        if (sharedBuffer == 0) {
139            mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
140            memset(mBuffer, 0, bufferSize);
141        } else {
142            mBuffer = sharedBuffer->pointer();
143#if 0
144            mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
145#endif
146        }
147
148#ifdef TEE_SINK
149        if (mTeeSinkTrackEnabled) {
150            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
151            if (pipeFormat != Format_Invalid) {
152                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
153                size_t numCounterOffers = 0;
154                const NBAIO_Format offers[1] = {pipeFormat};
155                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
156                ALOG_ASSERT(index == 0);
157                PipeReader *pipeReader = new PipeReader(*pipe);
158                numCounterOffers = 0;
159                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
160                ALOG_ASSERT(index == 0);
161                mTeeSink = pipe;
162                mTeeSource = pipeReader;
163            }
164        }
165#endif
166
167    }
168}
169
170AudioFlinger::ThreadBase::TrackBase::~TrackBase()
171{
172#ifdef TEE_SINK
173    dumpTee(-1, mTeeSource, mId);
174#endif
175    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
176    delete mServerProxy;
177    if (mCblk != NULL) {
178        if (mClient == 0) {
179            delete mCblk;
180        } else {
181            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
182        }
183    }
184    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
185    if (mClient != 0) {
186        // Client destructor must run with AudioFlinger mutex locked
187        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
188        // If the client's reference count drops to zero, the associated destructor
189        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
190        // relying on the automatic clear() at end of scope.
191        mClient.clear();
192    }
193}
194
195// AudioBufferProvider interface
196// getNextBuffer() = 0;
197// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
198void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
199{
200#ifdef TEE_SINK
201    if (mTeeSink != 0) {
202        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
203    }
204#endif
205
206    ServerProxy::Buffer buf;
207    buf.mFrameCount = buffer->frameCount;
208    buf.mRaw = buffer->raw;
209    buffer->frameCount = 0;
210    buffer->raw = NULL;
211    mServerProxy->releaseBuffer(&buf);
212}
213
214status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
215{
216    mSyncEvents.add(event);
217    return NO_ERROR;
218}
219
220// ----------------------------------------------------------------------------
221//      Playback
222// ----------------------------------------------------------------------------
223
224AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
225    : BnAudioTrack(),
226      mTrack(track)
227{
228}
229
230AudioFlinger::TrackHandle::~TrackHandle() {
231    // just stop the track on deletion, associated resources
232    // will be freed from the main thread once all pending buffers have
233    // been played. Unless it's not in the active track list, in which
234    // case we free everything now...
235    mTrack->destroy();
236}
237
238sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
239    return mTrack->getCblk();
240}
241
242status_t AudioFlinger::TrackHandle::start() {
243    return mTrack->start();
244}
245
246void AudioFlinger::TrackHandle::stop() {
247    mTrack->stop();
248}
249
250void AudioFlinger::TrackHandle::flush() {
251    mTrack->flush();
252}
253
254void AudioFlinger::TrackHandle::pause() {
255    mTrack->pause();
256}
257
258status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
259{
260    return mTrack->attachAuxEffect(EffectId);
261}
262
263status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
264                                                         sp<IMemory>* buffer) {
265    if (!mTrack->isTimedTrack())
266        return INVALID_OPERATION;
267
268    PlaybackThread::TimedTrack* tt =
269            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
270    return tt->allocateTimedBuffer(size, buffer);
271}
272
273status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
274                                                     int64_t pts) {
275    if (!mTrack->isTimedTrack())
276        return INVALID_OPERATION;
277
278    PlaybackThread::TimedTrack* tt =
279            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
280    return tt->queueTimedBuffer(buffer, pts);
281}
282
283status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
284    const LinearTransform& xform, int target) {
285
286    if (!mTrack->isTimedTrack())
287        return INVALID_OPERATION;
288
289    PlaybackThread::TimedTrack* tt =
290            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
291    return tt->setMediaTimeTransform(
292        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
293}
294
295status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
296    return mTrack->setParameters(keyValuePairs);
297}
298
299status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
300{
301    return mTrack->getTimestamp(timestamp);
302}
303
304
305void AudioFlinger::TrackHandle::signal()
306{
307    return mTrack->signal();
308}
309
310status_t AudioFlinger::TrackHandle::onTransact(
311    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
312{
313    return BnAudioTrack::onTransact(code, data, reply, flags);
314}
315
316// ----------------------------------------------------------------------------
317
318// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
319AudioFlinger::PlaybackThread::Track::Track(
320            PlaybackThread *thread,
321            const sp<Client>& client,
322            audio_stream_type_t streamType,
323            uint32_t sampleRate,
324            audio_format_t format,
325            audio_channel_mask_t channelMask,
326            size_t frameCount,
327            const sp<IMemory>& sharedBuffer,
328            int sessionId,
329            int uid,
330            IAudioFlinger::track_flags_t flags)
331    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
332            sessionId, uid, true /*isOut*/),
333    mFillingUpStatus(FS_INVALID),
334    // mRetryCount initialized later when needed
335    mSharedBuffer(sharedBuffer),
336    mStreamType(streamType),
337    mName(-1),  // see note below
338    mMainBuffer(thread->mixBuffer()),
339    mAuxBuffer(NULL),
340    mAuxEffectId(0), mHasVolumeController(false),
341    mPresentationCompleteFrames(0),
342    mFlags(flags),
343    mFastIndex(-1),
344    mCachedVolume(1.0),
345    mIsInvalid(false),
346    mAudioTrackServerProxy(NULL),
347    mResumeToStopping(false)
348{
349    if (mCblk != NULL) {
350        if (sharedBuffer == 0) {
351            mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
352                    mFrameSize);
353        } else {
354            mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
355                    mFrameSize);
356        }
357        mServerProxy = mAudioTrackServerProxy;
358        // to avoid leaking a track name, do not allocate one unless there is an mCblk
359        mName = thread->getTrackName_l(channelMask, sessionId);
360        if (mName < 0) {
361            ALOGE("no more track names available");
362            return;
363        }
364        // only allocate a fast track index if we were able to allocate a normal track name
365        if (flags & IAudioFlinger::TRACK_FAST) {
366            mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
367            ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
368            int i = __builtin_ctz(thread->mFastTrackAvailMask);
369            ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
370            // FIXME This is too eager.  We allocate a fast track index before the
371            //       fast track becomes active.  Since fast tracks are a scarce resource,
372            //       this means we are potentially denying other more important fast tracks from
373            //       being created.  It would be better to allocate the index dynamically.
374            mFastIndex = i;
375            // Read the initial underruns because this field is never cleared by the fast mixer
376            mObservedUnderruns = thread->getFastTrackUnderruns(i);
377            thread->mFastTrackAvailMask &= ~(1 << i);
378        }
379    }
380    ALOGV("Track constructor name %d, calling pid %d", mName,
381            IPCThreadState::self()->getCallingPid());
382}
383
384AudioFlinger::PlaybackThread::Track::~Track()
385{
386    ALOGV("PlaybackThread::Track destructor");
387
388    // The destructor would clear mSharedBuffer,
389    // but it will not push the decremented reference count,
390    // leaving the client's IMemory dangling indefinitely.
391    // This prevents that leak.
392    if (mSharedBuffer != 0) {
393        mSharedBuffer.clear();
394        // flush the binder command buffer
395        IPCThreadState::self()->flushCommands();
396    }
397}
398
399void AudioFlinger::PlaybackThread::Track::destroy()
400{
401    // NOTE: destroyTrack_l() can remove a strong reference to this Track
402    // by removing it from mTracks vector, so there is a risk that this Tracks's
403    // destructor is called. As the destructor needs to lock mLock,
404    // we must acquire a strong reference on this Track before locking mLock
405    // here so that the destructor is called only when exiting this function.
406    // On the other hand, as long as Track::destroy() is only called by
407    // TrackHandle destructor, the TrackHandle still holds a strong ref on
408    // this Track with its member mTrack.
409    sp<Track> keep(this);
410    { // scope for mLock
411        sp<ThreadBase> thread = mThread.promote();
412        if (thread != 0) {
413            Mutex::Autolock _l(thread->mLock);
414            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
415            bool wasActive = playbackThread->destroyTrack_l(this);
416            if (!isOutputTrack() && !wasActive) {
417                AudioSystem::releaseOutput(thread->id());
418            }
419        }
420    }
421}
422
423/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
424{
425    result.append("   Name Client Type      Fmt Chn mask Session fCount S F SRate  "
426                  "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
427}
428
429void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
430{
431    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
432    if (isFastTrack()) {
433        sprintf(buffer, "   F %2d", mFastIndex);
434    } else {
435        sprintf(buffer, "   %4d", mName - AudioMixer::TRACK0);
436    }
437    track_state state = mState;
438    char stateChar;
439    if (isTerminated()) {
440        stateChar = 'T';
441    } else {
442        switch (state) {
443        case IDLE:
444            stateChar = 'I';
445            break;
446        case STOPPING_1:
447            stateChar = 's';
448            break;
449        case STOPPING_2:
450            stateChar = '5';
451            break;
452        case STOPPED:
453            stateChar = 'S';
454            break;
455        case RESUMING:
456            stateChar = 'R';
457            break;
458        case ACTIVE:
459            stateChar = 'A';
460            break;
461        case PAUSING:
462            stateChar = 'p';
463            break;
464        case PAUSED:
465            stateChar = 'P';
466            break;
467        case FLUSHED:
468            stateChar = 'F';
469            break;
470        default:
471            stateChar = '?';
472            break;
473        }
474    }
475    char nowInUnderrun;
476    switch (mObservedUnderruns.mBitFields.mMostRecent) {
477    case UNDERRUN_FULL:
478        nowInUnderrun = ' ';
479        break;
480    case UNDERRUN_PARTIAL:
481        nowInUnderrun = '<';
482        break;
483    case UNDERRUN_EMPTY:
484        nowInUnderrun = '*';
485        break;
486    default:
487        nowInUnderrun = '?';
488        break;
489    }
490    snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g  "
491                                 "%08X %08X %08X 0x%03X %9u%c\n",
492            (mClient == 0) ? getpid_cached : mClient->pid(),
493            mStreamType,
494            mFormat,
495            mChannelMask,
496            mSessionId,
497            mFrameCount,
498            stateChar,
499            mFillingUpStatus,
500            mAudioTrackServerProxy->getSampleRate(),
501            20.0 * log10((vlr & 0xFFFF) / 4096.0),
502            20.0 * log10((vlr >> 16) / 4096.0),
503            mCblk->mServer,
504            (int)mMainBuffer,
505            (int)mAuxBuffer,
506            mCblk->mFlags,
507            mAudioTrackServerProxy->getUnderrunFrames(),
508            nowInUnderrun);
509}
510
511uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
512    return mAudioTrackServerProxy->getSampleRate();
513}
514
515// AudioBufferProvider interface
516status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
517        AudioBufferProvider::Buffer* buffer, int64_t pts)
518{
519    ServerProxy::Buffer buf;
520    size_t desiredFrames = buffer->frameCount;
521    buf.mFrameCount = desiredFrames;
522    status_t status = mServerProxy->obtainBuffer(&buf);
523    buffer->frameCount = buf.mFrameCount;
524    buffer->raw = buf.mRaw;
525    if (buf.mFrameCount == 0) {
526        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
527    }
528    return status;
529}
530
531// releaseBuffer() is not overridden
532
533// ExtendedAudioBufferProvider interface
534
535// Note that framesReady() takes a mutex on the control block using tryLock().
536// This could result in priority inversion if framesReady() is called by the normal mixer,
537// as the normal mixer thread runs at lower
538// priority than the client's callback thread:  there is a short window within framesReady()
539// during which the normal mixer could be preempted, and the client callback would block.
540// Another problem can occur if framesReady() is called by the fast mixer:
541// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
542// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
543size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
544    return mAudioTrackServerProxy->framesReady();
545}
546
547size_t AudioFlinger::PlaybackThread::Track::framesReleased() const
548{
549    return mAudioTrackServerProxy->framesReleased();
550}
551
552// Don't call for fast tracks; the framesReady() could result in priority inversion
553bool AudioFlinger::PlaybackThread::Track::isReady() const {
554    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
555        return true;
556    }
557
558    if (framesReady() >= mFrameCount ||
559            (mCblk->mFlags & CBLK_FORCEREADY)) {
560        mFillingUpStatus = FS_FILLED;
561        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
562        return true;
563    }
564    return false;
565}
566
567status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
568                                                    int triggerSession)
569{
570    status_t status = NO_ERROR;
571    ALOGV("start(%d), calling pid %d session %d",
572            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
573
574    sp<ThreadBase> thread = mThread.promote();
575    if (thread != 0) {
576        if (isOffloaded()) {
577            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
578            Mutex::Autolock _lth(thread->mLock);
579            sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
580            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
581                    (ec != 0 && ec->isNonOffloadableEnabled())) {
582                invalidate();
583                return PERMISSION_DENIED;
584            }
585        }
586        Mutex::Autolock _lth(thread->mLock);
587        track_state state = mState;
588        // here the track could be either new, or restarted
589        // in both cases "unstop" the track
590
591        if (state == PAUSED) {
592            if (mResumeToStopping) {
593                // happened we need to resume to STOPPING_1
594                mState = TrackBase::STOPPING_1;
595                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
596            } else {
597                mState = TrackBase::RESUMING;
598                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
599            }
600        } else {
601            mState = TrackBase::ACTIVE;
602            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
603        }
604
605        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
606        status = playbackThread->addTrack_l(this);
607        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
608            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
609            //  restore previous state if start was rejected by policy manager
610            if (status == PERMISSION_DENIED) {
611                mState = state;
612            }
613        }
614        // track was already in the active list, not a problem
615        if (status == ALREADY_EXISTS) {
616            status = NO_ERROR;
617        } else {
618            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
619            // It is usually unsafe to access the server proxy from a binder thread.
620            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
621            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
622            // and for fast tracks the track is not yet in the fast mixer thread's active set.
623            ServerProxy::Buffer buffer;
624            buffer.mFrameCount = 1;
625            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
626        }
627    } else {
628        status = BAD_VALUE;
629    }
630    return status;
631}
632
633void AudioFlinger::PlaybackThread::Track::stop()
634{
635    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
636    sp<ThreadBase> thread = mThread.promote();
637    if (thread != 0) {
638        Mutex::Autolock _l(thread->mLock);
639        track_state state = mState;
640        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
641            // If the track is not active (PAUSED and buffers full), flush buffers
642            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
643            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
644                reset();
645                mState = STOPPED;
646            } else if (!isFastTrack() && !isOffloaded()) {
647                mState = STOPPED;
648            } else {
649                // For fast tracks prepareTracks_l() will set state to STOPPING_2
650                // presentation is complete
651                // For an offloaded track this starts a drain and state will
652                // move to STOPPING_2 when drain completes and then STOPPED
653                mState = STOPPING_1;
654            }
655            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
656                    playbackThread);
657        }
658    }
659}
660
661void AudioFlinger::PlaybackThread::Track::pause()
662{
663    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
664    sp<ThreadBase> thread = mThread.promote();
665    if (thread != 0) {
666        Mutex::Autolock _l(thread->mLock);
667        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
668        switch (mState) {
669        case STOPPING_1:
670        case STOPPING_2:
671            if (!isOffloaded()) {
672                /* nothing to do if track is not offloaded */
673                break;
674            }
675
676            // Offloaded track was draining, we need to carry on draining when resumed
677            mResumeToStopping = true;
678            // fall through...
679        case ACTIVE:
680        case RESUMING:
681            mState = PAUSING;
682            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
683            playbackThread->broadcast_l();
684            break;
685
686        default:
687            break;
688        }
689    }
690}
691
692void AudioFlinger::PlaybackThread::Track::flush()
693{
694    ALOGV("flush(%d)", mName);
695    sp<ThreadBase> thread = mThread.promote();
696    if (thread != 0) {
697        Mutex::Autolock _l(thread->mLock);
698        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
699
700        if (isOffloaded()) {
701            // If offloaded we allow flush during any state except terminated
702            // and keep the track active to avoid problems if user is seeking
703            // rapidly and underlying hardware has a significant delay handling
704            // a pause
705            if (isTerminated()) {
706                return;
707            }
708
709            ALOGV("flush: offload flush");
710            reset();
711
712            if (mState == STOPPING_1 || mState == STOPPING_2) {
713                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
714                mState = ACTIVE;
715            }
716
717            if (mState == ACTIVE) {
718                ALOGV("flush called in active state, resetting buffer time out retry count");
719                mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
720            }
721
722            mResumeToStopping = false;
723        } else {
724            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
725                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
726                return;
727            }
728            // No point remaining in PAUSED state after a flush => go to
729            // FLUSHED state
730            mState = FLUSHED;
731            // do not reset the track if it is still in the process of being stopped or paused.
732            // this will be done by prepareTracks_l() when the track is stopped.
733            // prepareTracks_l() will see mState == FLUSHED, then
734            // remove from active track list, reset(), and trigger presentation complete
735            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
736                reset();
737            }
738        }
739        // Prevent flush being lost if the track is flushed and then resumed
740        // before mixer thread can run. This is important when offloading
741        // because the hardware buffer could hold a large amount of audio
742        playbackThread->flushOutput_l();
743        playbackThread->broadcast_l();
744    }
745}
746
747void AudioFlinger::PlaybackThread::Track::reset()
748{
749    // Do not reset twice to avoid discarding data written just after a flush and before
750    // the audioflinger thread detects the track is stopped.
751    if (!mResetDone) {
752        // Force underrun condition to avoid false underrun callback until first data is
753        // written to buffer
754        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
755        mFillingUpStatus = FS_FILLING;
756        mResetDone = true;
757        if (mState == FLUSHED) {
758            mState = IDLE;
759        }
760    }
761}
762
763status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
764{
765    sp<ThreadBase> thread = mThread.promote();
766    if (thread == 0) {
767        ALOGE("thread is dead");
768        return FAILED_TRANSACTION;
769    } else if ((thread->type() == ThreadBase::DIRECT) ||
770                    (thread->type() == ThreadBase::OFFLOAD)) {
771        return thread->setParameters(keyValuePairs);
772    } else {
773        return PERMISSION_DENIED;
774    }
775}
776
777status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
778{
779    // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
780    if (isFastTrack()) {
781        return INVALID_OPERATION;
782    }
783    sp<ThreadBase> thread = mThread.promote();
784    if (thread == 0) {
785        return INVALID_OPERATION;
786    }
787    Mutex::Autolock _l(thread->mLock);
788    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
789    if (!isOffloaded()) {
790        if (!playbackThread->mLatchQValid) {
791            return INVALID_OPERATION;
792        }
793        uint32_t unpresentedFrames =
794                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
795                playbackThread->mSampleRate;
796        uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
797        if (framesWritten < unpresentedFrames) {
798            return INVALID_OPERATION;
799        }
800        timestamp.mPosition = framesWritten - unpresentedFrames;
801        timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
802        return NO_ERROR;
803    }
804
805    return playbackThread->getTimestamp_l(timestamp);
806}
807
808status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
809{
810    status_t status = DEAD_OBJECT;
811    sp<ThreadBase> thread = mThread.promote();
812    if (thread != 0) {
813        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
814        sp<AudioFlinger> af = mClient->audioFlinger();
815
816        Mutex::Autolock _l(af->mLock);
817
818        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
819
820        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
821            Mutex::Autolock _dl(playbackThread->mLock);
822            Mutex::Autolock _sl(srcThread->mLock);
823            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
824            if (chain == 0) {
825                return INVALID_OPERATION;
826            }
827
828            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
829            if (effect == 0) {
830                return INVALID_OPERATION;
831            }
832            srcThread->removeEffect_l(effect);
833            status = playbackThread->addEffect_l(effect);
834            if (status != NO_ERROR) {
835                srcThread->addEffect_l(effect);
836                return INVALID_OPERATION;
837            }
838            // removeEffect_l() has stopped the effect if it was active so it must be restarted
839            if (effect->state() == EffectModule::ACTIVE ||
840                    effect->state() == EffectModule::STOPPING) {
841                effect->start();
842            }
843
844            sp<EffectChain> dstChain = effect->chain().promote();
845            if (dstChain == 0) {
846                srcThread->addEffect_l(effect);
847                return INVALID_OPERATION;
848            }
849            AudioSystem::unregisterEffect(effect->id());
850            AudioSystem::registerEffect(&effect->desc(),
851                                        srcThread->id(),
852                                        dstChain->strategy(),
853                                        AUDIO_SESSION_OUTPUT_MIX,
854                                        effect->id());
855        }
856        status = playbackThread->attachAuxEffect(this, EffectId);
857    }
858    return status;
859}
860
861void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
862{
863    mAuxEffectId = EffectId;
864    mAuxBuffer = buffer;
865}
866
867bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
868                                                         size_t audioHalFrames)
869{
870    // a track is considered presented when the total number of frames written to audio HAL
871    // corresponds to the number of frames written when presentationComplete() is called for the
872    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
873    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
874    // to detect when all frames have been played. In this case framesWritten isn't
875    // useful because it doesn't always reflect whether there is data in the h/w
876    // buffers, particularly if a track has been paused and resumed during draining
877    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
878                      mPresentationCompleteFrames, framesWritten);
879    if (mPresentationCompleteFrames == 0) {
880        mPresentationCompleteFrames = framesWritten + audioHalFrames;
881        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
882                  mPresentationCompleteFrames, audioHalFrames);
883    }
884
885    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
886        ALOGV("presentationComplete() session %d complete: framesWritten %d",
887                  mSessionId, framesWritten);
888        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
889        mAudioTrackServerProxy->setStreamEndDone();
890        return true;
891    }
892    return false;
893}
894
895void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
896{
897    for (int i = 0; i < (int)mSyncEvents.size(); i++) {
898        if (mSyncEvents[i]->type() == type) {
899            mSyncEvents[i]->trigger();
900            mSyncEvents.removeAt(i);
901            i--;
902        }
903    }
904}
905
906// implement VolumeBufferProvider interface
907
908uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
909{
910    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
911    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
912    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
913    uint32_t vl = vlr & 0xFFFF;
914    uint32_t vr = vlr >> 16;
915    // track volumes come from shared memory, so can't be trusted and must be clamped
916    if (vl > MAX_GAIN_INT) {
917        vl = MAX_GAIN_INT;
918    }
919    if (vr > MAX_GAIN_INT) {
920        vr = MAX_GAIN_INT;
921    }
922    // now apply the cached master volume and stream type volume;
923    // this is trusted but lacks any synchronization or barrier so may be stale
924    float v = mCachedVolume;
925    vl *= v;
926    vr *= v;
927    // re-combine into U4.16
928    vlr = (vr << 16) | (vl & 0xFFFF);
929    // FIXME look at mute, pause, and stop flags
930    return vlr;
931}
932
933status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
934{
935    if (isTerminated() || mState == PAUSED ||
936            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
937                                      (mState == STOPPED)))) {
938        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
939              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
940        event->cancel();
941        return INVALID_OPERATION;
942    }
943    (void) TrackBase::setSyncEvent(event);
944    return NO_ERROR;
945}
946
947void AudioFlinger::PlaybackThread::Track::invalidate()
948{
949    // FIXME should use proxy, and needs work
950    audio_track_cblk_t* cblk = mCblk;
951    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
952    android_atomic_release_store(0x40000000, &cblk->mFutex);
953    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
954    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
955    mIsInvalid = true;
956}
957
958void AudioFlinger::PlaybackThread::Track::signal()
959{
960    sp<ThreadBase> thread = mThread.promote();
961    if (thread != 0) {
962        PlaybackThread *t = (PlaybackThread *)thread.get();
963        Mutex::Autolock _l(t->mLock);
964        t->broadcast_l();
965    }
966}
967
968// ----------------------------------------------------------------------------
969
970sp<AudioFlinger::PlaybackThread::TimedTrack>
971AudioFlinger::PlaybackThread::TimedTrack::create(
972            PlaybackThread *thread,
973            const sp<Client>& client,
974            audio_stream_type_t streamType,
975            uint32_t sampleRate,
976            audio_format_t format,
977            audio_channel_mask_t channelMask,
978            size_t frameCount,
979            const sp<IMemory>& sharedBuffer,
980            int sessionId,
981            int uid) {
982    if (!client->reserveTimedTrack())
983        return 0;
984
985    return new TimedTrack(
986        thread, client, streamType, sampleRate, format, channelMask, frameCount,
987        sharedBuffer, sessionId, uid);
988}
989
990AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
991            PlaybackThread *thread,
992            const sp<Client>& client,
993            audio_stream_type_t streamType,
994            uint32_t sampleRate,
995            audio_format_t format,
996            audio_channel_mask_t channelMask,
997            size_t frameCount,
998            const sp<IMemory>& sharedBuffer,
999            int sessionId,
1000            int uid)
1001    : Track(thread, client, streamType, sampleRate, format, channelMask,
1002            frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
1003      mQueueHeadInFlight(false),
1004      mTrimQueueHeadOnRelease(false),
1005      mFramesPendingInQueue(0),
1006      mTimedSilenceBuffer(NULL),
1007      mTimedSilenceBufferSize(0),
1008      mTimedAudioOutputOnTime(false),
1009      mMediaTimeTransformValid(false)
1010{
1011    LocalClock lc;
1012    mLocalTimeFreq = lc.getLocalFreq();
1013
1014    mLocalTimeToSampleTransform.a_zero = 0;
1015    mLocalTimeToSampleTransform.b_zero = 0;
1016    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1017    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1018    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1019                            &mLocalTimeToSampleTransform.a_to_b_denom);
1020
1021    mMediaTimeToSampleTransform.a_zero = 0;
1022    mMediaTimeToSampleTransform.b_zero = 0;
1023    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1024    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1025    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1026                            &mMediaTimeToSampleTransform.a_to_b_denom);
1027}
1028
1029AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1030    mClient->releaseTimedTrack();
1031    delete [] mTimedSilenceBuffer;
1032}
1033
1034status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1035    size_t size, sp<IMemory>* buffer) {
1036
1037    Mutex::Autolock _l(mTimedBufferQueueLock);
1038
1039    trimTimedBufferQueue_l();
1040
1041    // lazily initialize the shared memory heap for timed buffers
1042    if (mTimedMemoryDealer == NULL) {
1043        const int kTimedBufferHeapSize = 512 << 10;
1044
1045        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1046                                              "AudioFlingerTimed");
1047        if (mTimedMemoryDealer == NULL)
1048            return NO_MEMORY;
1049    }
1050
1051    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1052    if (newBuffer == NULL) {
1053        newBuffer = mTimedMemoryDealer->allocate(size);
1054        if (newBuffer == NULL)
1055            return NO_MEMORY;
1056    }
1057
1058    *buffer = newBuffer;
1059    return NO_ERROR;
1060}
1061
1062// caller must hold mTimedBufferQueueLock
1063void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1064    int64_t mediaTimeNow;
1065    {
1066        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1067        if (!mMediaTimeTransformValid)
1068            return;
1069
1070        int64_t targetTimeNow;
1071        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1072            ? mCCHelper.getCommonTime(&targetTimeNow)
1073            : mCCHelper.getLocalTime(&targetTimeNow);
1074
1075        if (OK != res)
1076            return;
1077
1078        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1079                                                    &mediaTimeNow)) {
1080            return;
1081        }
1082    }
1083
1084    size_t trimEnd;
1085    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1086        int64_t bufEnd;
1087
1088        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1089            // We have a next buffer.  Just use its PTS as the PTS of the frame
1090            // following the last frame in this buffer.  If the stream is sparse
1091            // (ie, there are deliberate gaps left in the stream which should be
1092            // filled with silence by the TimedAudioTrack), then this can result
1093            // in one extra buffer being left un-trimmed when it could have
1094            // been.  In general, this is not typical, and we would rather
1095            // optimized away the TS calculation below for the more common case
1096            // where PTSes are contiguous.
1097            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1098        } else {
1099            // We have no next buffer.  Compute the PTS of the frame following
1100            // the last frame in this buffer by computing the duration of of
1101            // this frame in media time units and adding it to the PTS of the
1102            // buffer.
1103            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1104                               / mFrameSize;
1105
1106            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1107                                                                &bufEnd)) {
1108                ALOGE("Failed to convert frame count of %lld to media time"
1109                      " duration" " (scale factor %d/%u) in %s",
1110                      frameCount,
1111                      mMediaTimeToSampleTransform.a_to_b_numer,
1112                      mMediaTimeToSampleTransform.a_to_b_denom,
1113                      __PRETTY_FUNCTION__);
1114                break;
1115            }
1116            bufEnd += mTimedBufferQueue[trimEnd].pts();
1117        }
1118
1119        if (bufEnd > mediaTimeNow)
1120            break;
1121
1122        // Is the buffer we want to use in the middle of a mix operation right
1123        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1124        // from the mixer which should be coming back shortly.
1125        if (!trimEnd && mQueueHeadInFlight) {
1126            mTrimQueueHeadOnRelease = true;
1127        }
1128    }
1129
1130    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1131    if (trimStart < trimEnd) {
1132        // Update the bookkeeping for framesReady()
1133        for (size_t i = trimStart; i < trimEnd; ++i) {
1134            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1135        }
1136
1137        // Now actually remove the buffers from the queue.
1138        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1139    }
1140}
1141
1142void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1143        const char* logTag) {
1144    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1145                "%s called (reason \"%s\"), but timed buffer queue has no"
1146                " elements to trim.", __FUNCTION__, logTag);
1147
1148    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1149    mTimedBufferQueue.removeAt(0);
1150}
1151
1152void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1153        const TimedBuffer& buf,
1154        const char* logTag) {
1155    uint32_t bufBytes        = buf.buffer()->size();
1156    uint32_t consumedAlready = buf.position();
1157
1158    ALOG_ASSERT(consumedAlready <= bufBytes,
1159                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1160                " only %u bytes long, but claims to have consumed %u"
1161                " bytes.  (update reason: \"%s\")",
1162                bufBytes, consumedAlready, logTag);
1163
1164    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1165    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1166                "Bad bookkeeping while updating frames pending.  Should have at"
1167                " least %u queued frames, but we think we have only %u.  (update"
1168                " reason: \"%s\")",
1169                bufFrames, mFramesPendingInQueue, logTag);
1170
1171    mFramesPendingInQueue -= bufFrames;
1172}
1173
1174status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1175    const sp<IMemory>& buffer, int64_t pts) {
1176
1177    {
1178        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1179        if (!mMediaTimeTransformValid)
1180            return INVALID_OPERATION;
1181    }
1182
1183    Mutex::Autolock _l(mTimedBufferQueueLock);
1184
1185    uint32_t bufFrames = buffer->size() / mFrameSize;
1186    mFramesPendingInQueue += bufFrames;
1187    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1188
1189    return NO_ERROR;
1190}
1191
1192status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1193    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1194
1195    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1196           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1197           target);
1198
1199    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1200          target == TimedAudioTrack::COMMON_TIME)) {
1201        return BAD_VALUE;
1202    }
1203
1204    Mutex::Autolock lock(mMediaTimeTransformLock);
1205    mMediaTimeTransform = xform;
1206    mMediaTimeTransformTarget = target;
1207    mMediaTimeTransformValid = true;
1208
1209    return NO_ERROR;
1210}
1211
1212#define min(a, b) ((a) < (b) ? (a) : (b))
1213
1214// implementation of getNextBuffer for tracks whose buffers have timestamps
1215status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1216    AudioBufferProvider::Buffer* buffer, int64_t pts)
1217{
1218    if (pts == AudioBufferProvider::kInvalidPTS) {
1219        buffer->raw = NULL;
1220        buffer->frameCount = 0;
1221        mTimedAudioOutputOnTime = false;
1222        return INVALID_OPERATION;
1223    }
1224
1225    Mutex::Autolock _l(mTimedBufferQueueLock);
1226
1227    ALOG_ASSERT(!mQueueHeadInFlight,
1228                "getNextBuffer called without releaseBuffer!");
1229
1230    while (true) {
1231
1232        // if we have no timed buffers, then fail
1233        if (mTimedBufferQueue.isEmpty()) {
1234            buffer->raw = NULL;
1235            buffer->frameCount = 0;
1236            return NOT_ENOUGH_DATA;
1237        }
1238
1239        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1240
1241        // calculate the PTS of the head of the timed buffer queue expressed in
1242        // local time
1243        int64_t headLocalPTS;
1244        {
1245            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1246
1247            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1248
1249            if (mMediaTimeTransform.a_to_b_denom == 0) {
1250                // the transform represents a pause, so yield silence
1251                timedYieldSilence_l(buffer->frameCount, buffer);
1252                return NO_ERROR;
1253            }
1254
1255            int64_t transformedPTS;
1256            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1257                                                        &transformedPTS)) {
1258                // the transform failed.  this shouldn't happen, but if it does
1259                // then just drop this buffer
1260                ALOGW("timedGetNextBuffer transform failed");
1261                buffer->raw = NULL;
1262                buffer->frameCount = 0;
1263                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1264                return NO_ERROR;
1265            }
1266
1267            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1268                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1269                                                          &headLocalPTS)) {
1270                    buffer->raw = NULL;
1271                    buffer->frameCount = 0;
1272                    return INVALID_OPERATION;
1273                }
1274            } else {
1275                headLocalPTS = transformedPTS;
1276            }
1277        }
1278
1279        uint32_t sr = sampleRate();
1280
1281        // adjust the head buffer's PTS to reflect the portion of the head buffer
1282        // that has already been consumed
1283        int64_t effectivePTS = headLocalPTS +
1284                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1285
1286        // Calculate the delta in samples between the head of the input buffer
1287        // queue and the start of the next output buffer that will be written.
1288        // If the transformation fails because of over or underflow, it means
1289        // that the sample's position in the output stream is so far out of
1290        // whack that it should just be dropped.
1291        int64_t sampleDelta;
1292        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1293            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1294            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1295                                       " mix");
1296            continue;
1297        }
1298        if (!mLocalTimeToSampleTransform.doForwardTransform(
1299                (effectivePTS - pts) << 32, &sampleDelta)) {
1300            ALOGV("*** too late during sample rate transform: dropped buffer");
1301            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1302            continue;
1303        }
1304
1305        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1306               " sampleDelta=[%d.%08x]",
1307               head.pts(), head.position(), pts,
1308               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1309                   + (sampleDelta >> 32)),
1310               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1311
1312        // if the delta between the ideal placement for the next input sample and
1313        // the current output position is within this threshold, then we will
1314        // concatenate the next input samples to the previous output
1315        const int64_t kSampleContinuityThreshold =
1316                (static_cast<int64_t>(sr) << 32) / 250;
1317
1318        // if this is the first buffer of audio that we're emitting from this track
1319        // then it should be almost exactly on time.
1320        const int64_t kSampleStartupThreshold = 1LL << 32;
1321
1322        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1323           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1324            // the next input is close enough to being on time, so concatenate it
1325            // with the last output
1326            timedYieldSamples_l(buffer);
1327
1328            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1329                    head.position(), buffer->frameCount);
1330            return NO_ERROR;
1331        }
1332
1333        // Looks like our output is not on time.  Reset our on timed status.
1334        // Next time we mix samples from our input queue, then should be within
1335        // the StartupThreshold.
1336        mTimedAudioOutputOnTime = false;
1337        if (sampleDelta > 0) {
1338            // the gap between the current output position and the proper start of
1339            // the next input sample is too big, so fill it with silence
1340            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1341
1342            timedYieldSilence_l(framesUntilNextInput, buffer);
1343            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1344            return NO_ERROR;
1345        } else {
1346            // the next input sample is late
1347            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1348            size_t onTimeSamplePosition =
1349                    head.position() + lateFrames * mFrameSize;
1350
1351            if (onTimeSamplePosition > head.buffer()->size()) {
1352                // all the remaining samples in the head are too late, so
1353                // drop it and move on
1354                ALOGV("*** too late: dropped buffer");
1355                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1356                continue;
1357            } else {
1358                // skip over the late samples
1359                head.setPosition(onTimeSamplePosition);
1360
1361                // yield the available samples
1362                timedYieldSamples_l(buffer);
1363
1364                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1365                return NO_ERROR;
1366            }
1367        }
1368    }
1369}
1370
1371// Yield samples from the timed buffer queue head up to the given output
1372// buffer's capacity.
1373//
1374// Caller must hold mTimedBufferQueueLock
1375void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1376    AudioBufferProvider::Buffer* buffer) {
1377
1378    const TimedBuffer& head = mTimedBufferQueue[0];
1379
1380    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1381                   head.position());
1382
1383    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1384                                 mFrameSize);
1385    size_t framesRequested = buffer->frameCount;
1386    buffer->frameCount = min(framesLeftInHead, framesRequested);
1387
1388    mQueueHeadInFlight = true;
1389    mTimedAudioOutputOnTime = true;
1390}
1391
1392// Yield samples of silence up to the given output buffer's capacity
1393//
1394// Caller must hold mTimedBufferQueueLock
1395void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1396    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1397
1398    // lazily allocate a buffer filled with silence
1399    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1400        delete [] mTimedSilenceBuffer;
1401        mTimedSilenceBufferSize = numFrames * mFrameSize;
1402        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1403        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1404    }
1405
1406    buffer->raw = mTimedSilenceBuffer;
1407    size_t framesRequested = buffer->frameCount;
1408    buffer->frameCount = min(numFrames, framesRequested);
1409
1410    mTimedAudioOutputOnTime = false;
1411}
1412
1413// AudioBufferProvider interface
1414void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1415    AudioBufferProvider::Buffer* buffer) {
1416
1417    Mutex::Autolock _l(mTimedBufferQueueLock);
1418
1419    // If the buffer which was just released is part of the buffer at the head
1420    // of the queue, be sure to update the amt of the buffer which has been
1421    // consumed.  If the buffer being returned is not part of the head of the
1422    // queue, its either because the buffer is part of the silence buffer, or
1423    // because the head of the timed queue was trimmed after the mixer called
1424    // getNextBuffer but before the mixer called releaseBuffer.
1425    if (buffer->raw == mTimedSilenceBuffer) {
1426        ALOG_ASSERT(!mQueueHeadInFlight,
1427                    "Queue head in flight during release of silence buffer!");
1428        goto done;
1429    }
1430
1431    ALOG_ASSERT(mQueueHeadInFlight,
1432                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1433                " head in flight.");
1434
1435    if (mTimedBufferQueue.size()) {
1436        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1437
1438        void* start = head.buffer()->pointer();
1439        void* end   = reinterpret_cast<void*>(
1440                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1441                        + head.buffer()->size());
1442
1443        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1444                    "released buffer not within the head of the timed buffer"
1445                    " queue; qHead = [%p, %p], released buffer = %p",
1446                    start, end, buffer->raw);
1447
1448        head.setPosition(head.position() +
1449                (buffer->frameCount * mFrameSize));
1450        mQueueHeadInFlight = false;
1451
1452        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1453                    "Bad bookkeeping during releaseBuffer!  Should have at"
1454                    " least %u queued frames, but we think we have only %u",
1455                    buffer->frameCount, mFramesPendingInQueue);
1456
1457        mFramesPendingInQueue -= buffer->frameCount;
1458
1459        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1460            || mTrimQueueHeadOnRelease) {
1461            trimTimedBufferQueueHead_l("releaseBuffer");
1462            mTrimQueueHeadOnRelease = false;
1463        }
1464    } else {
1465        LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1466                  " buffers in the timed buffer queue");
1467    }
1468
1469done:
1470    buffer->raw = 0;
1471    buffer->frameCount = 0;
1472}
1473
1474size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1475    Mutex::Autolock _l(mTimedBufferQueueLock);
1476    return mFramesPendingInQueue;
1477}
1478
1479AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1480        : mPTS(0), mPosition(0) {}
1481
1482AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1483    const sp<IMemory>& buffer, int64_t pts)
1484        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1485
1486
1487// ----------------------------------------------------------------------------
1488
1489AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1490            PlaybackThread *playbackThread,
1491            DuplicatingThread *sourceThread,
1492            uint32_t sampleRate,
1493            audio_format_t format,
1494            audio_channel_mask_t channelMask,
1495            size_t frameCount,
1496            int uid)
1497    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1498                NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
1499    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1500{
1501
1502    if (mCblk != NULL) {
1503        mOutBuffer.frameCount = 0;
1504        playbackThread->mTracks.add(this);
1505        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1506                "mCblk->frameCount_ %u, mChannelMask 0x%08x",
1507                mCblk, mBuffer,
1508                mCblk->frameCount_, mChannelMask);
1509        // since client and server are in the same process,
1510        // the buffer has the same virtual address on both sides
1511        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1512        mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
1513        mClientProxy->setSendLevel(0.0);
1514        mClientProxy->setSampleRate(sampleRate);
1515        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1516                true /*clientInServer*/);
1517    } else {
1518        ALOGW("Error creating output track on thread %p", playbackThread);
1519    }
1520}
1521
1522AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1523{
1524    clearBufferQueue();
1525    delete mClientProxy;
1526    // superclass destructor will now delete the server proxy and shared memory both refer to
1527}
1528
1529status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1530                                                          int triggerSession)
1531{
1532    status_t status = Track::start(event, triggerSession);
1533    if (status != NO_ERROR) {
1534        return status;
1535    }
1536
1537    mActive = true;
1538    mRetryCount = 127;
1539    return status;
1540}
1541
1542void AudioFlinger::PlaybackThread::OutputTrack::stop()
1543{
1544    Track::stop();
1545    clearBufferQueue();
1546    mOutBuffer.frameCount = 0;
1547    mActive = false;
1548}
1549
1550bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1551{
1552    Buffer *pInBuffer;
1553    Buffer inBuffer;
1554    uint32_t channelCount = mChannelCount;
1555    bool outputBufferFull = false;
1556    inBuffer.frameCount = frames;
1557    inBuffer.i16 = data;
1558
1559    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1560
1561    if (!mActive && frames != 0) {
1562        start();
1563        sp<ThreadBase> thread = mThread.promote();
1564        if (thread != 0) {
1565            MixerThread *mixerThread = (MixerThread *)thread.get();
1566            if (mFrameCount > frames) {
1567                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1568                    uint32_t startFrames = (mFrameCount - frames);
1569                    pInBuffer = new Buffer;
1570                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1571                    pInBuffer->frameCount = startFrames;
1572                    pInBuffer->i16 = pInBuffer->mBuffer;
1573                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1574                    mBufferQueue.add(pInBuffer);
1575                } else {
1576                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1577                }
1578            }
1579        }
1580    }
1581
1582    while (waitTimeLeftMs) {
1583        // First write pending buffers, then new data
1584        if (mBufferQueue.size()) {
1585            pInBuffer = mBufferQueue.itemAt(0);
1586        } else {
1587            pInBuffer = &inBuffer;
1588        }
1589
1590        if (pInBuffer->frameCount == 0) {
1591            break;
1592        }
1593
1594        if (mOutBuffer.frameCount == 0) {
1595            mOutBuffer.frameCount = pInBuffer->frameCount;
1596            nsecs_t startTime = systemTime();
1597            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1598            if (status != NO_ERROR) {
1599                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1600                        mThread.unsafe_get(), status);
1601                outputBufferFull = true;
1602                break;
1603            }
1604            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1605            if (waitTimeLeftMs >= waitTimeMs) {
1606                waitTimeLeftMs -= waitTimeMs;
1607            } else {
1608                waitTimeLeftMs = 0;
1609            }
1610        }
1611
1612        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1613                pInBuffer->frameCount;
1614        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1615        Proxy::Buffer buf;
1616        buf.mFrameCount = outFrames;
1617        buf.mRaw = NULL;
1618        mClientProxy->releaseBuffer(&buf);
1619        pInBuffer->frameCount -= outFrames;
1620        pInBuffer->i16 += outFrames * channelCount;
1621        mOutBuffer.frameCount -= outFrames;
1622        mOutBuffer.i16 += outFrames * channelCount;
1623
1624        if (pInBuffer->frameCount == 0) {
1625            if (mBufferQueue.size()) {
1626                mBufferQueue.removeAt(0);
1627                delete [] pInBuffer->mBuffer;
1628                delete pInBuffer;
1629                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1630                        mThread.unsafe_get(), mBufferQueue.size());
1631            } else {
1632                break;
1633            }
1634        }
1635    }
1636
1637    // If we could not write all frames, allocate a buffer and queue it for next time.
1638    if (inBuffer.frameCount) {
1639        sp<ThreadBase> thread = mThread.promote();
1640        if (thread != 0 && !thread->standby()) {
1641            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1642                pInBuffer = new Buffer;
1643                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1644                pInBuffer->frameCount = inBuffer.frameCount;
1645                pInBuffer->i16 = pInBuffer->mBuffer;
1646                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1647                        sizeof(int16_t));
1648                mBufferQueue.add(pInBuffer);
1649                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1650                        mThread.unsafe_get(), mBufferQueue.size());
1651            } else {
1652                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1653                        mThread.unsafe_get(), this);
1654            }
1655        }
1656    }
1657
1658    // Calling write() with a 0 length buffer, means that no more data will be written:
1659    // If no more buffers are pending, fill output track buffer to make sure it is started
1660    // by output mixer.
1661    if (frames == 0 && mBufferQueue.size() == 0) {
1662        // FIXME borken, replace by getting framesReady() from proxy
1663        size_t user = 0;    // was mCblk->user
1664        if (user < mFrameCount) {
1665            frames = mFrameCount - user;
1666            pInBuffer = new Buffer;
1667            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1668            pInBuffer->frameCount = frames;
1669            pInBuffer->i16 = pInBuffer->mBuffer;
1670            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1671            mBufferQueue.add(pInBuffer);
1672        } else if (mActive) {
1673            stop();
1674        }
1675    }
1676
1677    return outputBufferFull;
1678}
1679
1680status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1681        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1682{
1683    ClientProxy::Buffer buf;
1684    buf.mFrameCount = buffer->frameCount;
1685    struct timespec timeout;
1686    timeout.tv_sec = waitTimeMs / 1000;
1687    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1688    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1689    buffer->frameCount = buf.mFrameCount;
1690    buffer->raw = buf.mRaw;
1691    return status;
1692}
1693
1694void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1695{
1696    size_t size = mBufferQueue.size();
1697
1698    for (size_t i = 0; i < size; i++) {
1699        Buffer *pBuffer = mBufferQueue.itemAt(i);
1700        delete [] pBuffer->mBuffer;
1701        delete pBuffer;
1702    }
1703    mBufferQueue.clear();
1704}
1705
1706
1707// ----------------------------------------------------------------------------
1708//      Record
1709// ----------------------------------------------------------------------------
1710
1711AudioFlinger::RecordHandle::RecordHandle(
1712        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1713    : BnAudioRecord(),
1714    mRecordTrack(recordTrack)
1715{
1716}
1717
1718AudioFlinger::RecordHandle::~RecordHandle() {
1719    stop_nonvirtual();
1720    mRecordTrack->destroy();
1721}
1722
1723sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
1724    return mRecordTrack->getCblk();
1725}
1726
1727status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1728        int triggerSession) {
1729    ALOGV("RecordHandle::start()");
1730    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1731}
1732
1733void AudioFlinger::RecordHandle::stop() {
1734    stop_nonvirtual();
1735}
1736
1737void AudioFlinger::RecordHandle::stop_nonvirtual() {
1738    ALOGV("RecordHandle::stop()");
1739    mRecordTrack->stop();
1740}
1741
1742status_t AudioFlinger::RecordHandle::onTransact(
1743    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1744{
1745    return BnAudioRecord::onTransact(code, data, reply, flags);
1746}
1747
1748// ----------------------------------------------------------------------------
1749
1750// RecordTrack constructor must be called with AudioFlinger::mLock held
1751AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1752            RecordThread *thread,
1753            const sp<Client>& client,
1754            uint32_t sampleRate,
1755            audio_format_t format,
1756            audio_channel_mask_t channelMask,
1757            size_t frameCount,
1758            int sessionId,
1759            int uid)
1760    :   TrackBase(thread, client, sampleRate, format,
1761                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/),
1762        mOverflow(false)
1763{
1764    ALOGV("RecordTrack constructor");
1765    if (mCblk != NULL) {
1766        mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
1767                mFrameSize);
1768        mServerProxy = mAudioRecordServerProxy;
1769    }
1770}
1771
1772AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1773{
1774    ALOGV("%s", __func__);
1775}
1776
1777// AudioBufferProvider interface
1778status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1779        int64_t pts)
1780{
1781    ServerProxy::Buffer buf;
1782    buf.mFrameCount = buffer->frameCount;
1783    status_t status = mServerProxy->obtainBuffer(&buf);
1784    buffer->frameCount = buf.mFrameCount;
1785    buffer->raw = buf.mRaw;
1786    if (buf.mFrameCount == 0) {
1787        // FIXME also wake futex so that overrun is noticed more quickly
1788        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1789    }
1790    return status;
1791}
1792
1793status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1794                                                        int triggerSession)
1795{
1796    sp<ThreadBase> thread = mThread.promote();
1797    if (thread != 0) {
1798        RecordThread *recordThread = (RecordThread *)thread.get();
1799        return recordThread->start(this, event, triggerSession);
1800    } else {
1801        return BAD_VALUE;
1802    }
1803}
1804
1805void AudioFlinger::RecordThread::RecordTrack::stop()
1806{
1807    sp<ThreadBase> thread = mThread.promote();
1808    if (thread != 0) {
1809        RecordThread *recordThread = (RecordThread *)thread.get();
1810        if (recordThread->stop(this)) {
1811            AudioSystem::stopInput(recordThread->id());
1812        }
1813    }
1814}
1815
1816void AudioFlinger::RecordThread::RecordTrack::destroy()
1817{
1818    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1819    sp<RecordTrack> keep(this);
1820    {
1821        sp<ThreadBase> thread = mThread.promote();
1822        if (thread != 0) {
1823            if (mState == ACTIVE || mState == RESUMING) {
1824                AudioSystem::stopInput(thread->id());
1825            }
1826            AudioSystem::releaseInput(thread->id());
1827            Mutex::Autolock _l(thread->mLock);
1828            RecordThread *recordThread = (RecordThread *) thread.get();
1829            recordThread->destroyTrack_l(this);
1830        }
1831    }
1832}
1833
1834void AudioFlinger::RecordThread::RecordTrack::invalidate()
1835{
1836    // FIXME should use proxy, and needs work
1837    audio_track_cblk_t* cblk = mCblk;
1838    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1839    android_atomic_release_store(0x40000000, &cblk->mFutex);
1840    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1841    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1842}
1843
1844
1845/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1846{
1847    result.append("Client Fmt Chn mask Session S   Server fCount\n");
1848}
1849
1850void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
1851{
1852    snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n",
1853            (mClient == 0) ? getpid_cached : mClient->pid(),
1854            mFormat,
1855            mChannelMask,
1856            mSessionId,
1857            mState,
1858            mCblk->mServer,
1859            mFrameCount);
1860}
1861
1862}; // namespace android
1863