Tracks.cpp revision 3ab368e0810d894dcbc0971350c095049478a055 
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 %6zu %1c %1d %5u %5.2g %5.2g  "
491                                 "%08X %p %p 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            mMainBuffer,
505            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            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
856        }
857        status = playbackThread->attachAuxEffect(this, EffectId);
858    }
859    return status;
860}
861
862void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
863{
864    mAuxEffectId = EffectId;
865    mAuxBuffer = buffer;
866}
867
868bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
869                                                         size_t audioHalFrames)
870{
871    // a track is considered presented when the total number of frames written to audio HAL
872    // corresponds to the number of frames written when presentationComplete() is called for the
873    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
874    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
875    // to detect when all frames have been played. In this case framesWritten isn't
876    // useful because it doesn't always reflect whether there is data in the h/w
877    // buffers, particularly if a track has been paused and resumed during draining
878    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
879                      mPresentationCompleteFrames, framesWritten);
880    if (mPresentationCompleteFrames == 0) {
881        mPresentationCompleteFrames = framesWritten + audioHalFrames;
882        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
883                  mPresentationCompleteFrames, audioHalFrames);
884    }
885
886    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
887        ALOGV("presentationComplete() session %d complete: framesWritten %d",
888                  mSessionId, framesWritten);
889        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
890        mAudioTrackServerProxy->setStreamEndDone();
891        return true;
892    }
893    return false;
894}
895
896void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
897{
898    for (size_t i = 0; i < mSyncEvents.size(); i++) {
899        if (mSyncEvents[i]->type() == type) {
900            mSyncEvents[i]->trigger();
901            mSyncEvents.removeAt(i);
902            i--;
903        }
904    }
905}
906
907// implement VolumeBufferProvider interface
908
909uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
910{
911    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
912    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
913    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
914    uint32_t vl = vlr & 0xFFFF;
915    uint32_t vr = vlr >> 16;
916    // track volumes come from shared memory, so can't be trusted and must be clamped
917    if (vl > MAX_GAIN_INT) {
918        vl = MAX_GAIN_INT;
919    }
920    if (vr > MAX_GAIN_INT) {
921        vr = MAX_GAIN_INT;
922    }
923    // now apply the cached master volume and stream type volume;
924    // this is trusted but lacks any synchronization or barrier so may be stale
925    float v = mCachedVolume;
926    vl *= v;
927    vr *= v;
928    // re-combine into U4.16
929    vlr = (vr << 16) | (vl & 0xFFFF);
930    // FIXME look at mute, pause, and stop flags
931    return vlr;
932}
933
934status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
935{
936    if (isTerminated() || mState == PAUSED ||
937            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
938                                      (mState == STOPPED)))) {
939        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
940              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
941        event->cancel();
942        return INVALID_OPERATION;
943    }
944    (void) TrackBase::setSyncEvent(event);
945    return NO_ERROR;
946}
947
948void AudioFlinger::PlaybackThread::Track::invalidate()
949{
950    // FIXME should use proxy, and needs work
951    audio_track_cblk_t* cblk = mCblk;
952    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
953    android_atomic_release_store(0x40000000, &cblk->mFutex);
954    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
955    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
956    mIsInvalid = true;
957}
958
959void AudioFlinger::PlaybackThread::Track::signal()
960{
961    sp<ThreadBase> thread = mThread.promote();
962    if (thread != 0) {
963        PlaybackThread *t = (PlaybackThread *)thread.get();
964        Mutex::Autolock _l(t->mLock);
965        t->broadcast_l();
966    }
967}
968
969// ----------------------------------------------------------------------------
970
971sp<AudioFlinger::PlaybackThread::TimedTrack>
972AudioFlinger::PlaybackThread::TimedTrack::create(
973            PlaybackThread *thread,
974            const sp<Client>& client,
975            audio_stream_type_t streamType,
976            uint32_t sampleRate,
977            audio_format_t format,
978            audio_channel_mask_t channelMask,
979            size_t frameCount,
980            const sp<IMemory>& sharedBuffer,
981            int sessionId,
982            int uid) {
983    if (!client->reserveTimedTrack())
984        return 0;
985
986    return new TimedTrack(
987        thread, client, streamType, sampleRate, format, channelMask, frameCount,
988        sharedBuffer, sessionId, uid);
989}
990
991AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
992            PlaybackThread *thread,
993            const sp<Client>& client,
994            audio_stream_type_t streamType,
995            uint32_t sampleRate,
996            audio_format_t format,
997            audio_channel_mask_t channelMask,
998            size_t frameCount,
999            const sp<IMemory>& sharedBuffer,
1000            int sessionId,
1001            int uid)
1002    : Track(thread, client, streamType, sampleRate, format, channelMask,
1003            frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
1004      mQueueHeadInFlight(false),
1005      mTrimQueueHeadOnRelease(false),
1006      mFramesPendingInQueue(0),
1007      mTimedSilenceBuffer(NULL),
1008      mTimedSilenceBufferSize(0),
1009      mTimedAudioOutputOnTime(false),
1010      mMediaTimeTransformValid(false)
1011{
1012    LocalClock lc;
1013    mLocalTimeFreq = lc.getLocalFreq();
1014
1015    mLocalTimeToSampleTransform.a_zero = 0;
1016    mLocalTimeToSampleTransform.b_zero = 0;
1017    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1018    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1019    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1020                            &mLocalTimeToSampleTransform.a_to_b_denom);
1021
1022    mMediaTimeToSampleTransform.a_zero = 0;
1023    mMediaTimeToSampleTransform.b_zero = 0;
1024    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1025    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1026    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1027                            &mMediaTimeToSampleTransform.a_to_b_denom);
1028}
1029
1030AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1031    mClient->releaseTimedTrack();
1032    delete [] mTimedSilenceBuffer;
1033}
1034
1035status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1036    size_t size, sp<IMemory>* buffer) {
1037
1038    Mutex::Autolock _l(mTimedBufferQueueLock);
1039
1040    trimTimedBufferQueue_l();
1041
1042    // lazily initialize the shared memory heap for timed buffers
1043    if (mTimedMemoryDealer == NULL) {
1044        const int kTimedBufferHeapSize = 512 << 10;
1045
1046        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1047                                              "AudioFlingerTimed");
1048        if (mTimedMemoryDealer == NULL)
1049            return NO_MEMORY;
1050    }
1051
1052    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1053    if (newBuffer == NULL) {
1054        newBuffer = mTimedMemoryDealer->allocate(size);
1055        if (newBuffer == NULL)
1056            return NO_MEMORY;
1057    }
1058
1059    *buffer = newBuffer;
1060    return NO_ERROR;
1061}
1062
1063// caller must hold mTimedBufferQueueLock
1064void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1065    int64_t mediaTimeNow;
1066    {
1067        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1068        if (!mMediaTimeTransformValid)
1069            return;
1070
1071        int64_t targetTimeNow;
1072        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1073            ? mCCHelper.getCommonTime(&targetTimeNow)
1074            : mCCHelper.getLocalTime(&targetTimeNow);
1075
1076        if (OK != res)
1077            return;
1078
1079        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1080                                                    &mediaTimeNow)) {
1081            return;
1082        }
1083    }
1084
1085    size_t trimEnd;
1086    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1087        int64_t bufEnd;
1088
1089        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1090            // We have a next buffer.  Just use its PTS as the PTS of the frame
1091            // following the last frame in this buffer.  If the stream is sparse
1092            // (ie, there are deliberate gaps left in the stream which should be
1093            // filled with silence by the TimedAudioTrack), then this can result
1094            // in one extra buffer being left un-trimmed when it could have
1095            // been.  In general, this is not typical, and we would rather
1096            // optimized away the TS calculation below for the more common case
1097            // where PTSes are contiguous.
1098            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1099        } else {
1100            // We have no next buffer.  Compute the PTS of the frame following
1101            // the last frame in this buffer by computing the duration of of
1102            // this frame in media time units and adding it to the PTS of the
1103            // buffer.
1104            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1105                               / mFrameSize;
1106
1107            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1108                                                                &bufEnd)) {
1109                ALOGE("Failed to convert frame count of %lld to media time"
1110                      " duration" " (scale factor %d/%u) in %s",
1111                      frameCount,
1112                      mMediaTimeToSampleTransform.a_to_b_numer,
1113                      mMediaTimeToSampleTransform.a_to_b_denom,
1114                      __PRETTY_FUNCTION__);
1115                break;
1116            }
1117            bufEnd += mTimedBufferQueue[trimEnd].pts();
1118        }
1119
1120        if (bufEnd > mediaTimeNow)
1121            break;
1122
1123        // Is the buffer we want to use in the middle of a mix operation right
1124        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1125        // from the mixer which should be coming back shortly.
1126        if (!trimEnd && mQueueHeadInFlight) {
1127            mTrimQueueHeadOnRelease = true;
1128        }
1129    }
1130
1131    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1132    if (trimStart < trimEnd) {
1133        // Update the bookkeeping for framesReady()
1134        for (size_t i = trimStart; i < trimEnd; ++i) {
1135            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1136        }
1137
1138        // Now actually remove the buffers from the queue.
1139        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1140    }
1141}
1142
1143void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1144        const char* logTag) {
1145    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1146                "%s called (reason \"%s\"), but timed buffer queue has no"
1147                " elements to trim.", __FUNCTION__, logTag);
1148
1149    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1150    mTimedBufferQueue.removeAt(0);
1151}
1152
1153void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1154        const TimedBuffer& buf,
1155        const char* logTag) {
1156    uint32_t bufBytes        = buf.buffer()->size();
1157    uint32_t consumedAlready = buf.position();
1158
1159    ALOG_ASSERT(consumedAlready <= bufBytes,
1160                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1161                " only %u bytes long, but claims to have consumed %u"
1162                " bytes.  (update reason: \"%s\")",
1163                bufBytes, consumedAlready, logTag);
1164
1165    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1166    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1167                "Bad bookkeeping while updating frames pending.  Should have at"
1168                " least %u queued frames, but we think we have only %u.  (update"
1169                " reason: \"%s\")",
1170                bufFrames, mFramesPendingInQueue, logTag);
1171
1172    mFramesPendingInQueue -= bufFrames;
1173}
1174
1175status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1176    const sp<IMemory>& buffer, int64_t pts) {
1177
1178    {
1179        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1180        if (!mMediaTimeTransformValid)
1181            return INVALID_OPERATION;
1182    }
1183
1184    Mutex::Autolock _l(mTimedBufferQueueLock);
1185
1186    uint32_t bufFrames = buffer->size() / mFrameSize;
1187    mFramesPendingInQueue += bufFrames;
1188    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1189
1190    return NO_ERROR;
1191}
1192
1193status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1194    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1195
1196    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1197           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1198           target);
1199
1200    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1201          target == TimedAudioTrack::COMMON_TIME)) {
1202        return BAD_VALUE;
1203    }
1204
1205    Mutex::Autolock lock(mMediaTimeTransformLock);
1206    mMediaTimeTransform = xform;
1207    mMediaTimeTransformTarget = target;
1208    mMediaTimeTransformValid = true;
1209
1210    return NO_ERROR;
1211}
1212
1213#define min(a, b) ((a) < (b) ? (a) : (b))
1214
1215// implementation of getNextBuffer for tracks whose buffers have timestamps
1216status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1217    AudioBufferProvider::Buffer* buffer, int64_t pts)
1218{
1219    if (pts == AudioBufferProvider::kInvalidPTS) {
1220        buffer->raw = NULL;
1221        buffer->frameCount = 0;
1222        mTimedAudioOutputOnTime = false;
1223        return INVALID_OPERATION;
1224    }
1225
1226    Mutex::Autolock _l(mTimedBufferQueueLock);
1227
1228    ALOG_ASSERT(!mQueueHeadInFlight,
1229                "getNextBuffer called without releaseBuffer!");
1230
1231    while (true) {
1232
1233        // if we have no timed buffers, then fail
1234        if (mTimedBufferQueue.isEmpty()) {
1235            buffer->raw = NULL;
1236            buffer->frameCount = 0;
1237            return NOT_ENOUGH_DATA;
1238        }
1239
1240        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1241
1242        // calculate the PTS of the head of the timed buffer queue expressed in
1243        // local time
1244        int64_t headLocalPTS;
1245        {
1246            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1247
1248            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1249
1250            if (mMediaTimeTransform.a_to_b_denom == 0) {
1251                // the transform represents a pause, so yield silence
1252                timedYieldSilence_l(buffer->frameCount, buffer);
1253                return NO_ERROR;
1254            }
1255
1256            int64_t transformedPTS;
1257            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1258                                                        &transformedPTS)) {
1259                // the transform failed.  this shouldn't happen, but if it does
1260                // then just drop this buffer
1261                ALOGW("timedGetNextBuffer transform failed");
1262                buffer->raw = NULL;
1263                buffer->frameCount = 0;
1264                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1265                return NO_ERROR;
1266            }
1267
1268            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1269                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1270                                                          &headLocalPTS)) {
1271                    buffer->raw = NULL;
1272                    buffer->frameCount = 0;
1273                    return INVALID_OPERATION;
1274                }
1275            } else {
1276                headLocalPTS = transformedPTS;
1277            }
1278        }
1279
1280        uint32_t sr = sampleRate();
1281
1282        // adjust the head buffer's PTS to reflect the portion of the head buffer
1283        // that has already been consumed
1284        int64_t effectivePTS = headLocalPTS +
1285                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1286
1287        // Calculate the delta in samples between the head of the input buffer
1288        // queue and the start of the next output buffer that will be written.
1289        // If the transformation fails because of over or underflow, it means
1290        // that the sample's position in the output stream is so far out of
1291        // whack that it should just be dropped.
1292        int64_t sampleDelta;
1293        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1294            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1295            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1296                                       " mix");
1297            continue;
1298        }
1299        if (!mLocalTimeToSampleTransform.doForwardTransform(
1300                (effectivePTS - pts) << 32, &sampleDelta)) {
1301            ALOGV("*** too late during sample rate transform: dropped buffer");
1302            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1303            continue;
1304        }
1305
1306        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1307               " sampleDelta=[%d.%08x]",
1308               head.pts(), head.position(), pts,
1309               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1310                   + (sampleDelta >> 32)),
1311               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1312
1313        // if the delta between the ideal placement for the next input sample and
1314        // the current output position is within this threshold, then we will
1315        // concatenate the next input samples to the previous output
1316        const int64_t kSampleContinuityThreshold =
1317                (static_cast<int64_t>(sr) << 32) / 250;
1318
1319        // if this is the first buffer of audio that we're emitting from this track
1320        // then it should be almost exactly on time.
1321        const int64_t kSampleStartupThreshold = 1LL << 32;
1322
1323        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1324           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1325            // the next input is close enough to being on time, so concatenate it
1326            // with the last output
1327            timedYieldSamples_l(buffer);
1328
1329            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1330                    head.position(), buffer->frameCount);
1331            return NO_ERROR;
1332        }
1333
1334        // Looks like our output is not on time.  Reset our on timed status.
1335        // Next time we mix samples from our input queue, then should be within
1336        // the StartupThreshold.
1337        mTimedAudioOutputOnTime = false;
1338        if (sampleDelta > 0) {
1339            // the gap between the current output position and the proper start of
1340            // the next input sample is too big, so fill it with silence
1341            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1342
1343            timedYieldSilence_l(framesUntilNextInput, buffer);
1344            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1345            return NO_ERROR;
1346        } else {
1347            // the next input sample is late
1348            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1349            size_t onTimeSamplePosition =
1350                    head.position() + lateFrames * mFrameSize;
1351
1352            if (onTimeSamplePosition > head.buffer()->size()) {
1353                // all the remaining samples in the head are too late, so
1354                // drop it and move on
1355                ALOGV("*** too late: dropped buffer");
1356                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1357                continue;
1358            } else {
1359                // skip over the late samples
1360                head.setPosition(onTimeSamplePosition);
1361
1362                // yield the available samples
1363                timedYieldSamples_l(buffer);
1364
1365                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1366                return NO_ERROR;
1367            }
1368        }
1369    }
1370}
1371
1372// Yield samples from the timed buffer queue head up to the given output
1373// buffer's capacity.
1374//
1375// Caller must hold mTimedBufferQueueLock
1376void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1377    AudioBufferProvider::Buffer* buffer) {
1378
1379    const TimedBuffer& head = mTimedBufferQueue[0];
1380
1381    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1382                   head.position());
1383
1384    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1385                                 mFrameSize);
1386    size_t framesRequested = buffer->frameCount;
1387    buffer->frameCount = min(framesLeftInHead, framesRequested);
1388
1389    mQueueHeadInFlight = true;
1390    mTimedAudioOutputOnTime = true;
1391}
1392
1393// Yield samples of silence up to the given output buffer's capacity
1394//
1395// Caller must hold mTimedBufferQueueLock
1396void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1397    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1398
1399    // lazily allocate a buffer filled with silence
1400    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1401        delete [] mTimedSilenceBuffer;
1402        mTimedSilenceBufferSize = numFrames * mFrameSize;
1403        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1404        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1405    }
1406
1407    buffer->raw = mTimedSilenceBuffer;
1408    size_t framesRequested = buffer->frameCount;
1409    buffer->frameCount = min(numFrames, framesRequested);
1410
1411    mTimedAudioOutputOnTime = false;
1412}
1413
1414// AudioBufferProvider interface
1415void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1416    AudioBufferProvider::Buffer* buffer) {
1417
1418    Mutex::Autolock _l(mTimedBufferQueueLock);
1419
1420    // If the buffer which was just released is part of the buffer at the head
1421    // of the queue, be sure to update the amt of the buffer which has been
1422    // consumed.  If the buffer being returned is not part of the head of the
1423    // queue, its either because the buffer is part of the silence buffer, or
1424    // because the head of the timed queue was trimmed after the mixer called
1425    // getNextBuffer but before the mixer called releaseBuffer.
1426    if (buffer->raw == mTimedSilenceBuffer) {
1427        ALOG_ASSERT(!mQueueHeadInFlight,
1428                    "Queue head in flight during release of silence buffer!");
1429        goto done;
1430    }
1431
1432    ALOG_ASSERT(mQueueHeadInFlight,
1433                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1434                " head in flight.");
1435
1436    if (mTimedBufferQueue.size()) {
1437        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1438
1439        void* start = head.buffer()->pointer();
1440        void* end   = reinterpret_cast<void*>(
1441                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1442                        + head.buffer()->size());
1443
1444        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1445                    "released buffer not within the head of the timed buffer"
1446                    " queue; qHead = [%p, %p], released buffer = %p",
1447                    start, end, buffer->raw);
1448
1449        head.setPosition(head.position() +
1450                (buffer->frameCount * mFrameSize));
1451        mQueueHeadInFlight = false;
1452
1453        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1454                    "Bad bookkeeping during releaseBuffer!  Should have at"
1455                    " least %u queued frames, but we think we have only %u",
1456                    buffer->frameCount, mFramesPendingInQueue);
1457
1458        mFramesPendingInQueue -= buffer->frameCount;
1459
1460        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1461            || mTrimQueueHeadOnRelease) {
1462            trimTimedBufferQueueHead_l("releaseBuffer");
1463            mTrimQueueHeadOnRelease = false;
1464        }
1465    } else {
1466        LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1467                  " buffers in the timed buffer queue");
1468    }
1469
1470done:
1471    buffer->raw = 0;
1472    buffer->frameCount = 0;
1473}
1474
1475size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1476    Mutex::Autolock _l(mTimedBufferQueueLock);
1477    return mFramesPendingInQueue;
1478}
1479
1480AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1481        : mPTS(0), mPosition(0) {}
1482
1483AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1484    const sp<IMemory>& buffer, int64_t pts)
1485        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1486
1487
1488// ----------------------------------------------------------------------------
1489
1490AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1491            PlaybackThread *playbackThread,
1492            DuplicatingThread *sourceThread,
1493            uint32_t sampleRate,
1494            audio_format_t format,
1495            audio_channel_mask_t channelMask,
1496            size_t frameCount,
1497            int uid)
1498    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1499                NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
1500    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1501{
1502
1503    if (mCblk != NULL) {
1504        mOutBuffer.frameCount = 0;
1505        playbackThread->mTracks.add(this);
1506        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1507                "mCblk->frameCount_ %u, mChannelMask 0x%08x",
1508                mCblk, mBuffer,
1509                mCblk->frameCount_, mChannelMask);
1510        // since client and server are in the same process,
1511        // the buffer has the same virtual address on both sides
1512        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1513        mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
1514        mClientProxy->setSendLevel(0.0);
1515        mClientProxy->setSampleRate(sampleRate);
1516        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1517                true /*clientInServer*/);
1518    } else {
1519        ALOGW("Error creating output track on thread %p", playbackThread);
1520    }
1521}
1522
1523AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1524{
1525    clearBufferQueue();
1526    delete mClientProxy;
1527    // superclass destructor will now delete the server proxy and shared memory both refer to
1528}
1529
1530status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1531                                                          int triggerSession)
1532{
1533    status_t status = Track::start(event, triggerSession);
1534    if (status != NO_ERROR) {
1535        return status;
1536    }
1537
1538    mActive = true;
1539    mRetryCount = 127;
1540    return status;
1541}
1542
1543void AudioFlinger::PlaybackThread::OutputTrack::stop()
1544{
1545    Track::stop();
1546    clearBufferQueue();
1547    mOutBuffer.frameCount = 0;
1548    mActive = false;
1549}
1550
1551bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1552{
1553    Buffer *pInBuffer;
1554    Buffer inBuffer;
1555    uint32_t channelCount = mChannelCount;
1556    bool outputBufferFull = false;
1557    inBuffer.frameCount = frames;
1558    inBuffer.i16 = data;
1559
1560    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1561
1562    if (!mActive && frames != 0) {
1563        start();
1564        sp<ThreadBase> thread = mThread.promote();
1565        if (thread != 0) {
1566            MixerThread *mixerThread = (MixerThread *)thread.get();
1567            if (mFrameCount > frames) {
1568                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1569                    uint32_t startFrames = (mFrameCount - frames);
1570                    pInBuffer = new Buffer;
1571                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1572                    pInBuffer->frameCount = startFrames;
1573                    pInBuffer->i16 = pInBuffer->mBuffer;
1574                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1575                    mBufferQueue.add(pInBuffer);
1576                } else {
1577                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1578                }
1579            }
1580        }
1581    }
1582
1583    while (waitTimeLeftMs) {
1584        // First write pending buffers, then new data
1585        if (mBufferQueue.size()) {
1586            pInBuffer = mBufferQueue.itemAt(0);
1587        } else {
1588            pInBuffer = &inBuffer;
1589        }
1590
1591        if (pInBuffer->frameCount == 0) {
1592            break;
1593        }
1594
1595        if (mOutBuffer.frameCount == 0) {
1596            mOutBuffer.frameCount = pInBuffer->frameCount;
1597            nsecs_t startTime = systemTime();
1598            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1599            if (status != NO_ERROR) {
1600                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1601                        mThread.unsafe_get(), status);
1602                outputBufferFull = true;
1603                break;
1604            }
1605            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1606            if (waitTimeLeftMs >= waitTimeMs) {
1607                waitTimeLeftMs -= waitTimeMs;
1608            } else {
1609                waitTimeLeftMs = 0;
1610            }
1611        }
1612
1613        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1614                pInBuffer->frameCount;
1615        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1616        Proxy::Buffer buf;
1617        buf.mFrameCount = outFrames;
1618        buf.mRaw = NULL;
1619        mClientProxy->releaseBuffer(&buf);
1620        pInBuffer->frameCount -= outFrames;
1621        pInBuffer->i16 += outFrames * channelCount;
1622        mOutBuffer.frameCount -= outFrames;
1623        mOutBuffer.i16 += outFrames * channelCount;
1624
1625        if (pInBuffer->frameCount == 0) {
1626            if (mBufferQueue.size()) {
1627                mBufferQueue.removeAt(0);
1628                delete [] pInBuffer->mBuffer;
1629                delete pInBuffer;
1630                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1631                        mThread.unsafe_get(), mBufferQueue.size());
1632            } else {
1633                break;
1634            }
1635        }
1636    }
1637
1638    // If we could not write all frames, allocate a buffer and queue it for next time.
1639    if (inBuffer.frameCount) {
1640        sp<ThreadBase> thread = mThread.promote();
1641        if (thread != 0 && !thread->standby()) {
1642            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1643                pInBuffer = new Buffer;
1644                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1645                pInBuffer->frameCount = inBuffer.frameCount;
1646                pInBuffer->i16 = pInBuffer->mBuffer;
1647                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1648                        sizeof(int16_t));
1649                mBufferQueue.add(pInBuffer);
1650                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1651                        mThread.unsafe_get(), mBufferQueue.size());
1652            } else {
1653                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1654                        mThread.unsafe_get(), this);
1655            }
1656        }
1657    }
1658
1659    // Calling write() with a 0 length buffer, means that no more data will be written:
1660    // If no more buffers are pending, fill output track buffer to make sure it is started
1661    // by output mixer.
1662    if (frames == 0 && mBufferQueue.size() == 0) {
1663        // FIXME borken, replace by getting framesReady() from proxy
1664        size_t user = 0;    // was mCblk->user
1665        if (user < mFrameCount) {
1666            frames = mFrameCount - user;
1667            pInBuffer = new Buffer;
1668            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1669            pInBuffer->frameCount = frames;
1670            pInBuffer->i16 = pInBuffer->mBuffer;
1671            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1672            mBufferQueue.add(pInBuffer);
1673        } else if (mActive) {
1674            stop();
1675        }
1676    }
1677
1678    return outputBufferFull;
1679}
1680
1681status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1682        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1683{
1684    ClientProxy::Buffer buf;
1685    buf.mFrameCount = buffer->frameCount;
1686    struct timespec timeout;
1687    timeout.tv_sec = waitTimeMs / 1000;
1688    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1689    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1690    buffer->frameCount = buf.mFrameCount;
1691    buffer->raw = buf.mRaw;
1692    return status;
1693}
1694
1695void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1696{
1697    size_t size = mBufferQueue.size();
1698
1699    for (size_t i = 0; i < size; i++) {
1700        Buffer *pBuffer = mBufferQueue.itemAt(i);
1701        delete [] pBuffer->mBuffer;
1702        delete pBuffer;
1703    }
1704    mBufferQueue.clear();
1705}
1706
1707
1708// ----------------------------------------------------------------------------
1709//      Record
1710// ----------------------------------------------------------------------------
1711
1712AudioFlinger::RecordHandle::RecordHandle(
1713        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1714    : BnAudioRecord(),
1715    mRecordTrack(recordTrack)
1716{
1717}
1718
1719AudioFlinger::RecordHandle::~RecordHandle() {
1720    stop_nonvirtual();
1721    mRecordTrack->destroy();
1722}
1723
1724sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
1725    return mRecordTrack->getCblk();
1726}
1727
1728status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1729        int triggerSession) {
1730    ALOGV("RecordHandle::start()");
1731    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1732}
1733
1734void AudioFlinger::RecordHandle::stop() {
1735    stop_nonvirtual();
1736}
1737
1738void AudioFlinger::RecordHandle::stop_nonvirtual() {
1739    ALOGV("RecordHandle::stop()");
1740    mRecordTrack->stop();
1741}
1742
1743status_t AudioFlinger::RecordHandle::onTransact(
1744    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1745{
1746    return BnAudioRecord::onTransact(code, data, reply, flags);
1747}
1748
1749// ----------------------------------------------------------------------------
1750
1751// RecordTrack constructor must be called with AudioFlinger::mLock held
1752AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1753            RecordThread *thread,
1754            const sp<Client>& client,
1755            uint32_t sampleRate,
1756            audio_format_t format,
1757            audio_channel_mask_t channelMask,
1758            size_t frameCount,
1759            int sessionId,
1760            int uid)
1761    :   TrackBase(thread, client, sampleRate, format,
1762                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/),
1763        mOverflow(false)
1764{
1765    ALOGV("RecordTrack constructor");
1766    if (mCblk != NULL) {
1767        mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
1768                mFrameSize);
1769        mServerProxy = mAudioRecordServerProxy;
1770    }
1771}
1772
1773AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1774{
1775    ALOGV("%s", __func__);
1776}
1777
1778// AudioBufferProvider interface
1779status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1780        int64_t pts)
1781{
1782    ServerProxy::Buffer buf;
1783    buf.mFrameCount = buffer->frameCount;
1784    status_t status = mServerProxy->obtainBuffer(&buf);
1785    buffer->frameCount = buf.mFrameCount;
1786    buffer->raw = buf.mRaw;
1787    if (buf.mFrameCount == 0) {
1788        // FIXME also wake futex so that overrun is noticed more quickly
1789        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1790    }
1791    return status;
1792}
1793
1794status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1795                                                        int triggerSession)
1796{
1797    sp<ThreadBase> thread = mThread.promote();
1798    if (thread != 0) {
1799        RecordThread *recordThread = (RecordThread *)thread.get();
1800        return recordThread->start(this, event, triggerSession);
1801    } else {
1802        return BAD_VALUE;
1803    }
1804}
1805
1806void AudioFlinger::RecordThread::RecordTrack::stop()
1807{
1808    sp<ThreadBase> thread = mThread.promote();
1809    if (thread != 0) {
1810        RecordThread *recordThread = (RecordThread *)thread.get();
1811        if (recordThread->stop(this)) {
1812            AudioSystem::stopInput(recordThread->id());
1813        }
1814    }
1815}
1816
1817void AudioFlinger::RecordThread::RecordTrack::destroy()
1818{
1819    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1820    sp<RecordTrack> keep(this);
1821    {
1822        sp<ThreadBase> thread = mThread.promote();
1823        if (thread != 0) {
1824            if (mState == ACTIVE || mState == RESUMING) {
1825                AudioSystem::stopInput(thread->id());
1826            }
1827            AudioSystem::releaseInput(thread->id());
1828            Mutex::Autolock _l(thread->mLock);
1829            RecordThread *recordThread = (RecordThread *) thread.get();
1830            recordThread->destroyTrack_l(this);
1831        }
1832    }
1833}
1834
1835void AudioFlinger::RecordThread::RecordTrack::invalidate()
1836{
1837    // FIXME should use proxy, and needs work
1838    audio_track_cblk_t* cblk = mCblk;
1839    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1840    android_atomic_release_store(0x40000000, &cblk->mFutex);
1841    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1842    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1843}
1844
1845
1846/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1847{
1848    result.append("Client Fmt Chn mask Session S   Server fCount\n");
1849}
1850
1851void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
1852{
1853    snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6zu\n",
1854            (mClient == 0) ? getpid_cached : mClient->pid(),
1855            mFormat,
1856            mChannelMask,
1857            mSessionId,
1858            mState,
1859            mCblk->mServer,
1860            mFrameCount);
1861}
1862
1863}; // namespace android
1864