Tracks.cpp revision 755b0a611f539dfa49e88aac592a938427c7e1b8 
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            IAudioFlinger::track_flags_t flags,
73            bool isOut,
74            bool useReadOnlyHeap)
75    :   RefBase(),
76        mThread(thread),
77        mClient(client),
78        mCblk(NULL),
79        // mBuffer
80        mState(IDLE),
81        mSampleRate(sampleRate),
82        mFormat(format),
83        mChannelMask(channelMask),
84        mChannelCount(popcount(channelMask)),
85        mFrameSize(audio_is_linear_pcm(format) ?
86                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
87        mFrameCount(frameCount),
88        mSessionId(sessionId),
89        mFlags(flags),
90        mIsOut(isOut),
91        mServerProxy(NULL),
92        mId(android_atomic_inc(&nextTrackId)),
93        mTerminated(false)
94{
95    // if the caller is us, trust the specified uid
96    if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
97        int newclientUid = IPCThreadState::self()->getCallingUid();
98        if (clientUid != -1 && clientUid != newclientUid) {
99            ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
100        }
101        clientUid = newclientUid;
102    }
103    // clientUid contains the uid of the app that is responsible for this track, so we can blame
104    // battery usage on it.
105    mUid = clientUid;
106
107    // client == 0 implies sharedBuffer == 0
108    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
109
110    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
111            sharedBuffer->size());
112
113    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
114    size_t size = sizeof(audio_track_cblk_t);
115    size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
116    if (sharedBuffer == 0 && !useReadOnlyHeap) {
117        size += bufferSize;
118    }
119
120    if (client != 0) {
121        mCblkMemory = client->heap()->allocate(size);
122        if (mCblkMemory == 0 ||
123                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
124            ALOGE("not enough memory for AudioTrack size=%u", size);
125            client->heap()->dump("AudioTrack");
126            mCblkMemory.clear();
127            return;
128        }
129    } else {
130        // this syntax avoids calling the audio_track_cblk_t constructor twice
131        mCblk = (audio_track_cblk_t *) new uint8_t[size];
132        // assume mCblk != NULL
133    }
134
135    // construct the shared structure in-place.
136    if (mCblk != NULL) {
137        new(mCblk) audio_track_cblk_t();
138        if (useReadOnlyHeap) {
139            const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
140            if (roHeap == 0 ||
141                    (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
142                    (mBuffer = mBufferMemory->pointer()) == NULL) {
143                ALOGE("not enough memory for read-only buffer size=%zu", bufferSize);
144                if (roHeap != 0) {
145                    roHeap->dump("buffer");
146                }
147                mCblkMemory.clear();
148                mBufferMemory.clear();
149                return;
150            }
151            memset(mBuffer, 0, bufferSize);
152        } else {
153            // clear all buffers
154            if (sharedBuffer == 0) {
155                mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
156                memset(mBuffer, 0, bufferSize);
157            } else {
158                mBuffer = sharedBuffer->pointer();
159#if 0
160                mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
161#endif
162            }
163        }
164
165#ifdef TEE_SINK
166        if (mTeeSinkTrackEnabled) {
167            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
168            if (Format_isValid(pipeFormat)) {
169                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
170                size_t numCounterOffers = 0;
171                const NBAIO_Format offers[1] = {pipeFormat};
172                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
173                ALOG_ASSERT(index == 0);
174                PipeReader *pipeReader = new PipeReader(*pipe);
175                numCounterOffers = 0;
176                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
177                ALOG_ASSERT(index == 0);
178                mTeeSink = pipe;
179                mTeeSource = pipeReader;
180            }
181        }
182#endif
183
184    }
185}
186
187AudioFlinger::ThreadBase::TrackBase::~TrackBase()
188{
189#ifdef TEE_SINK
190    dumpTee(-1, mTeeSource, mId);
191#endif
192    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
193    delete mServerProxy;
194    if (mCblk != NULL) {
195        if (mClient == 0) {
196            delete mCblk;
197        } else {
198            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
199        }
200    }
201    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
202    if (mClient != 0) {
203        // Client destructor must run with AudioFlinger mutex locked
204        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
205        // If the client's reference count drops to zero, the associated destructor
206        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
207        // relying on the automatic clear() at end of scope.
208        mClient.clear();
209    }
210}
211
212// AudioBufferProvider interface
213// getNextBuffer() = 0;
214// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
215void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
216{
217#ifdef TEE_SINK
218    if (mTeeSink != 0) {
219        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
220    }
221#endif
222
223    ServerProxy::Buffer buf;
224    buf.mFrameCount = buffer->frameCount;
225    buf.mRaw = buffer->raw;
226    buffer->frameCount = 0;
227    buffer->raw = NULL;
228    mServerProxy->releaseBuffer(&buf);
229}
230
231status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
232{
233    mSyncEvents.add(event);
234    return NO_ERROR;
235}
236
237// ----------------------------------------------------------------------------
238//      Playback
239// ----------------------------------------------------------------------------
240
241AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
242    : BnAudioTrack(),
243      mTrack(track)
244{
245}
246
247AudioFlinger::TrackHandle::~TrackHandle() {
248    // just stop the track on deletion, associated resources
249    // will be freed from the main thread once all pending buffers have
250    // been played. Unless it's not in the active track list, in which
251    // case we free everything now...
252    mTrack->destroy();
253}
254
255sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
256    return mTrack->getCblk();
257}
258
259status_t AudioFlinger::TrackHandle::start() {
260    return mTrack->start();
261}
262
263void AudioFlinger::TrackHandle::stop() {
264    mTrack->stop();
265}
266
267void AudioFlinger::TrackHandle::flush() {
268    mTrack->flush();
269}
270
271void AudioFlinger::TrackHandle::pause() {
272    mTrack->pause();
273}
274
275status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
276{
277    return mTrack->attachAuxEffect(EffectId);
278}
279
280status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
281                                                         sp<IMemory>* buffer) {
282    if (!mTrack->isTimedTrack())
283        return INVALID_OPERATION;
284
285    PlaybackThread::TimedTrack* tt =
286            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
287    return tt->allocateTimedBuffer(size, buffer);
288}
289
290status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
291                                                     int64_t pts) {
292    if (!mTrack->isTimedTrack())
293        return INVALID_OPERATION;
294
295    if (buffer == 0 || buffer->pointer() == NULL) {
296        ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
297        return BAD_VALUE;
298    }
299
300    PlaybackThread::TimedTrack* tt =
301            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
302    return tt->queueTimedBuffer(buffer, pts);
303}
304
305status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
306    const LinearTransform& xform, int target) {
307
308    if (!mTrack->isTimedTrack())
309        return INVALID_OPERATION;
310
311    PlaybackThread::TimedTrack* tt =
312            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
313    return tt->setMediaTimeTransform(
314        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
315}
316
317status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
318    return mTrack->setParameters(keyValuePairs);
319}
320
321status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
322{
323    return mTrack->getTimestamp(timestamp);
324}
325
326
327void AudioFlinger::TrackHandle::signal()
328{
329    return mTrack->signal();
330}
331
332status_t AudioFlinger::TrackHandle::onTransact(
333    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
334{
335    return BnAudioTrack::onTransact(code, data, reply, flags);
336}
337
338// ----------------------------------------------------------------------------
339
340// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
341AudioFlinger::PlaybackThread::Track::Track(
342            PlaybackThread *thread,
343            const sp<Client>& client,
344            audio_stream_type_t streamType,
345            uint32_t sampleRate,
346            audio_format_t format,
347            audio_channel_mask_t channelMask,
348            size_t frameCount,
349            const sp<IMemory>& sharedBuffer,
350            int sessionId,
351            int uid,
352            IAudioFlinger::track_flags_t flags)
353    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
354            sessionId, uid, flags, true /*isOut*/),
355    mFillingUpStatus(FS_INVALID),
356    // mRetryCount initialized later when needed
357    mSharedBuffer(sharedBuffer),
358    mStreamType(streamType),
359    mName(-1),  // see note below
360    mMainBuffer(thread->mixBuffer()),
361    mAuxBuffer(NULL),
362    mAuxEffectId(0), mHasVolumeController(false),
363    mPresentationCompleteFrames(0),
364    mFastIndex(-1),
365    mCachedVolume(1.0),
366    mIsInvalid(false),
367    mAudioTrackServerProxy(NULL),
368    mResumeToStopping(false),
369    mFlushHwPending(false)
370{
371    if (mCblk == NULL) {
372        return;
373    }
374
375    if (sharedBuffer == 0) {
376        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
377                mFrameSize);
378    } else {
379        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
380                mFrameSize);
381    }
382    mServerProxy = mAudioTrackServerProxy;
383
384    mName = thread->getTrackName_l(channelMask, sessionId);
385    if (mName < 0) {
386        ALOGE("no more track names available");
387        return;
388    }
389    // only allocate a fast track index if we were able to allocate a normal track name
390    if (flags & IAudioFlinger::TRACK_FAST) {
391        mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
392        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
393        int i = __builtin_ctz(thread->mFastTrackAvailMask);
394        ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
395        // FIXME This is too eager.  We allocate a fast track index before the
396        //       fast track becomes active.  Since fast tracks are a scarce resource,
397        //       this means we are potentially denying other more important fast tracks from
398        //       being created.  It would be better to allocate the index dynamically.
399        mFastIndex = i;
400        // Read the initial underruns because this field is never cleared by the fast mixer
401        mObservedUnderruns = thread->getFastTrackUnderruns(i);
402        thread->mFastTrackAvailMask &= ~(1 << i);
403    }
404}
405
406AudioFlinger::PlaybackThread::Track::~Track()
407{
408    ALOGV("PlaybackThread::Track destructor");
409
410    // The destructor would clear mSharedBuffer,
411    // but it will not push the decremented reference count,
412    // leaving the client's IMemory dangling indefinitely.
413    // This prevents that leak.
414    if (mSharedBuffer != 0) {
415        mSharedBuffer.clear();
416        // flush the binder command buffer
417        IPCThreadState::self()->flushCommands();
418    }
419}
420
421status_t AudioFlinger::PlaybackThread::Track::initCheck() const
422{
423    status_t status = TrackBase::initCheck();
424    if (status == NO_ERROR && mName < 0) {
425        status = NO_MEMORY;
426    }
427    return status;
428}
429
430void AudioFlinger::PlaybackThread::Track::destroy()
431{
432    // NOTE: destroyTrack_l() can remove a strong reference to this Track
433    // by removing it from mTracks vector, so there is a risk that this Tracks's
434    // destructor is called. As the destructor needs to lock mLock,
435    // we must acquire a strong reference on this Track before locking mLock
436    // here so that the destructor is called only when exiting this function.
437    // On the other hand, as long as Track::destroy() is only called by
438    // TrackHandle destructor, the TrackHandle still holds a strong ref on
439    // this Track with its member mTrack.
440    sp<Track> keep(this);
441    { // scope for mLock
442        sp<ThreadBase> thread = mThread.promote();
443        if (thread != 0) {
444            Mutex::Autolock _l(thread->mLock);
445            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
446            bool wasActive = playbackThread->destroyTrack_l(this);
447            if (!isOutputTrack() && !wasActive) {
448                AudioSystem::releaseOutput(thread->id());
449            }
450        }
451    }
452}
453
454/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
455{
456    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
457                  "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
458}
459
460void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
461{
462    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
463    if (isFastTrack()) {
464        sprintf(buffer, "    F %2d", mFastIndex);
465    } else if (mName >= AudioMixer::TRACK0) {
466        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
467    } else {
468        sprintf(buffer, "    none");
469    }
470    track_state state = mState;
471    char stateChar;
472    if (isTerminated()) {
473        stateChar = 'T';
474    } else {
475        switch (state) {
476        case IDLE:
477            stateChar = 'I';
478            break;
479        case STOPPING_1:
480            stateChar = 's';
481            break;
482        case STOPPING_2:
483            stateChar = '5';
484            break;
485        case STOPPED:
486            stateChar = 'S';
487            break;
488        case RESUMING:
489            stateChar = 'R';
490            break;
491        case ACTIVE:
492            stateChar = 'A';
493            break;
494        case PAUSING:
495            stateChar = 'p';
496            break;
497        case PAUSED:
498            stateChar = 'P';
499            break;
500        case FLUSHED:
501            stateChar = 'F';
502            break;
503        default:
504            stateChar = '?';
505            break;
506        }
507    }
508    char nowInUnderrun;
509    switch (mObservedUnderruns.mBitFields.mMostRecent) {
510    case UNDERRUN_FULL:
511        nowInUnderrun = ' ';
512        break;
513    case UNDERRUN_PARTIAL:
514        nowInUnderrun = '<';
515        break;
516    case UNDERRUN_EMPTY:
517        nowInUnderrun = '*';
518        break;
519    default:
520        nowInUnderrun = '?';
521        break;
522    }
523    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
524                                 "%08X %p %p 0x%03X %9u%c\n",
525            active ? "yes" : "no",
526            (mClient == 0) ? getpid_cached : mClient->pid(),
527            mStreamType,
528            mFormat,
529            mChannelMask,
530            mSessionId,
531            mFrameCount,
532            stateChar,
533            mFillingUpStatus,
534            mAudioTrackServerProxy->getSampleRate(),
535            20.0 * log10((vlr & 0xFFFF) / 4096.0),
536            20.0 * log10((vlr >> 16) / 4096.0),
537            mCblk->mServer,
538            mMainBuffer,
539            mAuxBuffer,
540            mCblk->mFlags,
541            mAudioTrackServerProxy->getUnderrunFrames(),
542            nowInUnderrun);
543}
544
545uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
546    return mAudioTrackServerProxy->getSampleRate();
547}
548
549// AudioBufferProvider interface
550status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
551        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
552{
553    ServerProxy::Buffer buf;
554    size_t desiredFrames = buffer->frameCount;
555    buf.mFrameCount = desiredFrames;
556    status_t status = mServerProxy->obtainBuffer(&buf);
557    buffer->frameCount = buf.mFrameCount;
558    buffer->raw = buf.mRaw;
559    if (buf.mFrameCount == 0) {
560        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
561    }
562    return status;
563}
564
565// releaseBuffer() is not overridden
566
567// ExtendedAudioBufferProvider interface
568
569// Note that framesReady() takes a mutex on the control block using tryLock().
570// This could result in priority inversion if framesReady() is called by the normal mixer,
571// as the normal mixer thread runs at lower
572// priority than the client's callback thread:  there is a short window within framesReady()
573// during which the normal mixer could be preempted, and the client callback would block.
574// Another problem can occur if framesReady() is called by the fast mixer:
575// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
576// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
577size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
578    return mAudioTrackServerProxy->framesReady();
579}
580
581size_t AudioFlinger::PlaybackThread::Track::framesReleased() const
582{
583    return mAudioTrackServerProxy->framesReleased();
584}
585
586// Don't call for fast tracks; the framesReady() could result in priority inversion
587bool AudioFlinger::PlaybackThread::Track::isReady() const {
588    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
589        return true;
590    }
591
592    if (isStopping()) {
593        if (framesReady() > 0) {
594            mFillingUpStatus = FS_FILLED;
595        }
596        return true;
597    }
598
599    if (framesReady() >= mFrameCount ||
600            (mCblk->mFlags & CBLK_FORCEREADY)) {
601        mFillingUpStatus = FS_FILLED;
602        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
603        return true;
604    }
605    return false;
606}
607
608status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
609                                                    int triggerSession __unused)
610{
611    status_t status = NO_ERROR;
612    ALOGV("start(%d), calling pid %d session %d",
613            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
614
615    sp<ThreadBase> thread = mThread.promote();
616    if (thread != 0) {
617        if (isOffloaded()) {
618            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
619            Mutex::Autolock _lth(thread->mLock);
620            sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
621            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
622                    (ec != 0 && ec->isNonOffloadableEnabled())) {
623                invalidate();
624                return PERMISSION_DENIED;
625            }
626        }
627        Mutex::Autolock _lth(thread->mLock);
628        track_state state = mState;
629        // here the track could be either new, or restarted
630        // in both cases "unstop" the track
631
632        // initial state-stopping. next state-pausing.
633        // What if resume is called ?
634
635        if (state == PAUSED || state == PAUSING) {
636            if (mResumeToStopping) {
637                // happened we need to resume to STOPPING_1
638                mState = TrackBase::STOPPING_1;
639                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
640            } else {
641                mState = TrackBase::RESUMING;
642                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
643            }
644        } else {
645            mState = TrackBase::ACTIVE;
646            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
647        }
648
649        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
650        status = playbackThread->addTrack_l(this);
651        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
652            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
653            //  restore previous state if start was rejected by policy manager
654            if (status == PERMISSION_DENIED) {
655                mState = state;
656            }
657        }
658        // track was already in the active list, not a problem
659        if (status == ALREADY_EXISTS) {
660            status = NO_ERROR;
661        } else {
662            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
663            // It is usually unsafe to access the server proxy from a binder thread.
664            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
665            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
666            // and for fast tracks the track is not yet in the fast mixer thread's active set.
667            ServerProxy::Buffer buffer;
668            buffer.mFrameCount = 1;
669            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
670        }
671    } else {
672        status = BAD_VALUE;
673    }
674    return status;
675}
676
677void AudioFlinger::PlaybackThread::Track::stop()
678{
679    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
680    sp<ThreadBase> thread = mThread.promote();
681    if (thread != 0) {
682        Mutex::Autolock _l(thread->mLock);
683        track_state state = mState;
684        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
685            // If the track is not active (PAUSED and buffers full), flush buffers
686            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
687            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
688                reset();
689                mState = STOPPED;
690            } else if (!isFastTrack() && !isOffloaded()) {
691                mState = STOPPED;
692            } else {
693                // For fast tracks prepareTracks_l() will set state to STOPPING_2
694                // presentation is complete
695                // For an offloaded track this starts a drain and state will
696                // move to STOPPING_2 when drain completes and then STOPPED
697                mState = STOPPING_1;
698            }
699            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
700                    playbackThread);
701        }
702    }
703}
704
705void AudioFlinger::PlaybackThread::Track::pause()
706{
707    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
708    sp<ThreadBase> thread = mThread.promote();
709    if (thread != 0) {
710        Mutex::Autolock _l(thread->mLock);
711        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
712        switch (mState) {
713        case STOPPING_1:
714        case STOPPING_2:
715            if (!isOffloaded()) {
716                /* nothing to do if track is not offloaded */
717                break;
718            }
719
720            // Offloaded track was draining, we need to carry on draining when resumed
721            mResumeToStopping = true;
722            // fall through...
723        case ACTIVE:
724        case RESUMING:
725            mState = PAUSING;
726            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
727            playbackThread->broadcast_l();
728            break;
729
730        default:
731            break;
732        }
733    }
734}
735
736void AudioFlinger::PlaybackThread::Track::flush()
737{
738    ALOGV("flush(%d)", mName);
739    sp<ThreadBase> thread = mThread.promote();
740    if (thread != 0) {
741        Mutex::Autolock _l(thread->mLock);
742        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
743
744        if (isOffloaded()) {
745            // If offloaded we allow flush during any state except terminated
746            // and keep the track active to avoid problems if user is seeking
747            // rapidly and underlying hardware has a significant delay handling
748            // a pause
749            if (isTerminated()) {
750                return;
751            }
752
753            ALOGV("flush: offload flush");
754            reset();
755
756            if (mState == STOPPING_1 || mState == STOPPING_2) {
757                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
758                mState = ACTIVE;
759            }
760
761            if (mState == ACTIVE) {
762                ALOGV("flush called in active state, resetting buffer time out retry count");
763                mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
764            }
765
766            mFlushHwPending = true;
767            mResumeToStopping = false;
768        } else {
769            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
770                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
771                return;
772            }
773            // No point remaining in PAUSED state after a flush => go to
774            // FLUSHED state
775            mState = FLUSHED;
776            // do not reset the track if it is still in the process of being stopped or paused.
777            // this will be done by prepareTracks_l() when the track is stopped.
778            // prepareTracks_l() will see mState == FLUSHED, then
779            // remove from active track list, reset(), and trigger presentation complete
780            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
781                reset();
782            }
783        }
784        // Prevent flush being lost if the track is flushed and then resumed
785        // before mixer thread can run. This is important when offloading
786        // because the hardware buffer could hold a large amount of audio
787        playbackThread->broadcast_l();
788    }
789}
790
791// must be called with thread lock held
792void AudioFlinger::PlaybackThread::Track::flushAck()
793{
794    if (!isOffloaded())
795        return;
796
797    mFlushHwPending = false;
798}
799
800void AudioFlinger::PlaybackThread::Track::reset()
801{
802    // Do not reset twice to avoid discarding data written just after a flush and before
803    // the audioflinger thread detects the track is stopped.
804    if (!mResetDone) {
805        // Force underrun condition to avoid false underrun callback until first data is
806        // written to buffer
807        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
808        mFillingUpStatus = FS_FILLING;
809        mResetDone = true;
810        if (mState == FLUSHED) {
811            mState = IDLE;
812        }
813    }
814}
815
816status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
817{
818    sp<ThreadBase> thread = mThread.promote();
819    if (thread == 0) {
820        ALOGE("thread is dead");
821        return FAILED_TRANSACTION;
822    } else if ((thread->type() == ThreadBase::DIRECT) ||
823                    (thread->type() == ThreadBase::OFFLOAD)) {
824        return thread->setParameters(keyValuePairs);
825    } else {
826        return PERMISSION_DENIED;
827    }
828}
829
830status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
831{
832    // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
833    if (isFastTrack()) {
834        return INVALID_OPERATION;
835    }
836    sp<ThreadBase> thread = mThread.promote();
837    if (thread == 0) {
838        return INVALID_OPERATION;
839    }
840    Mutex::Autolock _l(thread->mLock);
841    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
842    if (!isOffloaded()) {
843        if (!playbackThread->mLatchQValid) {
844            return INVALID_OPERATION;
845        }
846        uint32_t unpresentedFrames =
847                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
848                playbackThread->mSampleRate;
849        uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
850        if (framesWritten < unpresentedFrames) {
851            return INVALID_OPERATION;
852        }
853        timestamp.mPosition = framesWritten - unpresentedFrames;
854        timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
855        return NO_ERROR;
856    }
857
858    return playbackThread->getTimestamp_l(timestamp);
859}
860
861status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
862{
863    status_t status = DEAD_OBJECT;
864    sp<ThreadBase> thread = mThread.promote();
865    if (thread != 0) {
866        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
867        sp<AudioFlinger> af = mClient->audioFlinger();
868
869        Mutex::Autolock _l(af->mLock);
870
871        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
872
873        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
874            Mutex::Autolock _dl(playbackThread->mLock);
875            Mutex::Autolock _sl(srcThread->mLock);
876            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
877            if (chain == 0) {
878                return INVALID_OPERATION;
879            }
880
881            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
882            if (effect == 0) {
883                return INVALID_OPERATION;
884            }
885            srcThread->removeEffect_l(effect);
886            status = playbackThread->addEffect_l(effect);
887            if (status != NO_ERROR) {
888                srcThread->addEffect_l(effect);
889                return INVALID_OPERATION;
890            }
891            // removeEffect_l() has stopped the effect if it was active so it must be restarted
892            if (effect->state() == EffectModule::ACTIVE ||
893                    effect->state() == EffectModule::STOPPING) {
894                effect->start();
895            }
896
897            sp<EffectChain> dstChain = effect->chain().promote();
898            if (dstChain == 0) {
899                srcThread->addEffect_l(effect);
900                return INVALID_OPERATION;
901            }
902            AudioSystem::unregisterEffect(effect->id());
903            AudioSystem::registerEffect(&effect->desc(),
904                                        srcThread->id(),
905                                        dstChain->strategy(),
906                                        AUDIO_SESSION_OUTPUT_MIX,
907                                        effect->id());
908            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
909        }
910        status = playbackThread->attachAuxEffect(this, EffectId);
911    }
912    return status;
913}
914
915void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
916{
917    mAuxEffectId = EffectId;
918    mAuxBuffer = buffer;
919}
920
921bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
922                                                         size_t audioHalFrames)
923{
924    // a track is considered presented when the total number of frames written to audio HAL
925    // corresponds to the number of frames written when presentationComplete() is called for the
926    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
927    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
928    // to detect when all frames have been played. In this case framesWritten isn't
929    // useful because it doesn't always reflect whether there is data in the h/w
930    // buffers, particularly if a track has been paused and resumed during draining
931    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
932                      mPresentationCompleteFrames, framesWritten);
933    if (mPresentationCompleteFrames == 0) {
934        mPresentationCompleteFrames = framesWritten + audioHalFrames;
935        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
936                  mPresentationCompleteFrames, audioHalFrames);
937    }
938
939    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
940        ALOGV("presentationComplete() session %d complete: framesWritten %d",
941                  mSessionId, framesWritten);
942        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
943        mAudioTrackServerProxy->setStreamEndDone();
944        return true;
945    }
946    return false;
947}
948
949void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
950{
951    for (size_t i = 0; i < mSyncEvents.size(); i++) {
952        if (mSyncEvents[i]->type() == type) {
953            mSyncEvents[i]->trigger();
954            mSyncEvents.removeAt(i);
955            i--;
956        }
957    }
958}
959
960// implement VolumeBufferProvider interface
961
962uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
963{
964    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
965    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
966    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
967    uint32_t vl = vlr & 0xFFFF;
968    uint32_t vr = vlr >> 16;
969    // track volumes come from shared memory, so can't be trusted and must be clamped
970    if (vl > MAX_GAIN_INT) {
971        vl = MAX_GAIN_INT;
972    }
973    if (vr > MAX_GAIN_INT) {
974        vr = MAX_GAIN_INT;
975    }
976    // now apply the cached master volume and stream type volume;
977    // this is trusted but lacks any synchronization or barrier so may be stale
978    float v = mCachedVolume;
979    vl *= v;
980    vr *= v;
981    // re-combine into U4.16
982    vlr = (vr << 16) | (vl & 0xFFFF);
983    // FIXME look at mute, pause, and stop flags
984    return vlr;
985}
986
987status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
988{
989    if (isTerminated() || mState == PAUSED ||
990            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
991                                      (mState == STOPPED)))) {
992        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
993              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
994        event->cancel();
995        return INVALID_OPERATION;
996    }
997    (void) TrackBase::setSyncEvent(event);
998    return NO_ERROR;
999}
1000
1001void AudioFlinger::PlaybackThread::Track::invalidate()
1002{
1003    // FIXME should use proxy, and needs work
1004    audio_track_cblk_t* cblk = mCblk;
1005    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1006    android_atomic_release_store(0x40000000, &cblk->mFutex);
1007    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1008    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1009    mIsInvalid = true;
1010}
1011
1012void AudioFlinger::PlaybackThread::Track::signal()
1013{
1014    sp<ThreadBase> thread = mThread.promote();
1015    if (thread != 0) {
1016        PlaybackThread *t = (PlaybackThread *)thread.get();
1017        Mutex::Autolock _l(t->mLock);
1018        t->broadcast_l();
1019    }
1020}
1021
1022//To be called with thread lock held
1023bool AudioFlinger::PlaybackThread::Track::isResumePending() {
1024
1025    if (mState == RESUMING)
1026        return true;
1027    /* Resume is pending if track was stopping before pause was called */
1028    if (mState == STOPPING_1 &&
1029        mResumeToStopping)
1030        return true;
1031
1032    return false;
1033}
1034
1035//To be called with thread lock held
1036void AudioFlinger::PlaybackThread::Track::resumeAck() {
1037
1038
1039    if (mState == RESUMING)
1040        mState = ACTIVE;
1041
1042    // Other possibility of  pending resume is stopping_1 state
1043    // Do not update the state from stopping as this prevents
1044    // drain being called.
1045    if (mState == STOPPING_1) {
1046        mResumeToStopping = false;
1047    }
1048}
1049// ----------------------------------------------------------------------------
1050
1051sp<AudioFlinger::PlaybackThread::TimedTrack>
1052AudioFlinger::PlaybackThread::TimedTrack::create(
1053            PlaybackThread *thread,
1054            const sp<Client>& client,
1055            audio_stream_type_t streamType,
1056            uint32_t sampleRate,
1057            audio_format_t format,
1058            audio_channel_mask_t channelMask,
1059            size_t frameCount,
1060            const sp<IMemory>& sharedBuffer,
1061            int sessionId,
1062            int uid)
1063{
1064    if (!client->reserveTimedTrack())
1065        return 0;
1066
1067    return new TimedTrack(
1068        thread, client, streamType, sampleRate, format, channelMask, frameCount,
1069        sharedBuffer, sessionId, uid);
1070}
1071
1072AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
1073            PlaybackThread *thread,
1074            const sp<Client>& client,
1075            audio_stream_type_t streamType,
1076            uint32_t sampleRate,
1077            audio_format_t format,
1078            audio_channel_mask_t channelMask,
1079            size_t frameCount,
1080            const sp<IMemory>& sharedBuffer,
1081            int sessionId,
1082            int uid)
1083    : Track(thread, client, streamType, sampleRate, format, channelMask,
1084            frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
1085      mQueueHeadInFlight(false),
1086      mTrimQueueHeadOnRelease(false),
1087      mFramesPendingInQueue(0),
1088      mTimedSilenceBuffer(NULL),
1089      mTimedSilenceBufferSize(0),
1090      mTimedAudioOutputOnTime(false),
1091      mMediaTimeTransformValid(false)
1092{
1093    LocalClock lc;
1094    mLocalTimeFreq = lc.getLocalFreq();
1095
1096    mLocalTimeToSampleTransform.a_zero = 0;
1097    mLocalTimeToSampleTransform.b_zero = 0;
1098    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1099    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1100    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1101                            &mLocalTimeToSampleTransform.a_to_b_denom);
1102
1103    mMediaTimeToSampleTransform.a_zero = 0;
1104    mMediaTimeToSampleTransform.b_zero = 0;
1105    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1106    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1107    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1108                            &mMediaTimeToSampleTransform.a_to_b_denom);
1109}
1110
1111AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1112    mClient->releaseTimedTrack();
1113    delete [] mTimedSilenceBuffer;
1114}
1115
1116status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1117    size_t size, sp<IMemory>* buffer) {
1118
1119    Mutex::Autolock _l(mTimedBufferQueueLock);
1120
1121    trimTimedBufferQueue_l();
1122
1123    // lazily initialize the shared memory heap for timed buffers
1124    if (mTimedMemoryDealer == NULL) {
1125        const int kTimedBufferHeapSize = 512 << 10;
1126
1127        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1128                                              "AudioFlingerTimed");
1129        if (mTimedMemoryDealer == NULL) {
1130            return NO_MEMORY;
1131        }
1132    }
1133
1134    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1135    if (newBuffer == 0 || newBuffer->pointer() == NULL) {
1136        return NO_MEMORY;
1137    }
1138
1139    *buffer = newBuffer;
1140    return NO_ERROR;
1141}
1142
1143// caller must hold mTimedBufferQueueLock
1144void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1145    int64_t mediaTimeNow;
1146    {
1147        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1148        if (!mMediaTimeTransformValid)
1149            return;
1150
1151        int64_t targetTimeNow;
1152        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1153            ? mCCHelper.getCommonTime(&targetTimeNow)
1154            : mCCHelper.getLocalTime(&targetTimeNow);
1155
1156        if (OK != res)
1157            return;
1158
1159        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1160                                                    &mediaTimeNow)) {
1161            return;
1162        }
1163    }
1164
1165    size_t trimEnd;
1166    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1167        int64_t bufEnd;
1168
1169        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1170            // We have a next buffer.  Just use its PTS as the PTS of the frame
1171            // following the last frame in this buffer.  If the stream is sparse
1172            // (ie, there are deliberate gaps left in the stream which should be
1173            // filled with silence by the TimedAudioTrack), then this can result
1174            // in one extra buffer being left un-trimmed when it could have
1175            // been.  In general, this is not typical, and we would rather
1176            // optimized away the TS calculation below for the more common case
1177            // where PTSes are contiguous.
1178            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1179        } else {
1180            // We have no next buffer.  Compute the PTS of the frame following
1181            // the last frame in this buffer by computing the duration of of
1182            // this frame in media time units and adding it to the PTS of the
1183            // buffer.
1184            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1185                               / mFrameSize;
1186
1187            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1188                                                                &bufEnd)) {
1189                ALOGE("Failed to convert frame count of %lld to media time"
1190                      " duration" " (scale factor %d/%u) in %s",
1191                      frameCount,
1192                      mMediaTimeToSampleTransform.a_to_b_numer,
1193                      mMediaTimeToSampleTransform.a_to_b_denom,
1194                      __PRETTY_FUNCTION__);
1195                break;
1196            }
1197            bufEnd += mTimedBufferQueue[trimEnd].pts();
1198        }
1199
1200        if (bufEnd > mediaTimeNow)
1201            break;
1202
1203        // Is the buffer we want to use in the middle of a mix operation right
1204        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1205        // from the mixer which should be coming back shortly.
1206        if (!trimEnd && mQueueHeadInFlight) {
1207            mTrimQueueHeadOnRelease = true;
1208        }
1209    }
1210
1211    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1212    if (trimStart < trimEnd) {
1213        // Update the bookkeeping for framesReady()
1214        for (size_t i = trimStart; i < trimEnd; ++i) {
1215            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1216        }
1217
1218        // Now actually remove the buffers from the queue.
1219        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1220    }
1221}
1222
1223void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1224        const char* logTag) {
1225    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1226                "%s called (reason \"%s\"), but timed buffer queue has no"
1227                " elements to trim.", __FUNCTION__, logTag);
1228
1229    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1230    mTimedBufferQueue.removeAt(0);
1231}
1232
1233void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1234        const TimedBuffer& buf,
1235        const char* logTag __unused) {
1236    uint32_t bufBytes        = buf.buffer()->size();
1237    uint32_t consumedAlready = buf.position();
1238
1239    ALOG_ASSERT(consumedAlready <= bufBytes,
1240                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1241                " only %u bytes long, but claims to have consumed %u"
1242                " bytes.  (update reason: \"%s\")",
1243                bufBytes, consumedAlready, logTag);
1244
1245    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1246    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1247                "Bad bookkeeping while updating frames pending.  Should have at"
1248                " least %u queued frames, but we think we have only %u.  (update"
1249                " reason: \"%s\")",
1250                bufFrames, mFramesPendingInQueue, logTag);
1251
1252    mFramesPendingInQueue -= bufFrames;
1253}
1254
1255status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1256    const sp<IMemory>& buffer, int64_t pts) {
1257
1258    {
1259        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1260        if (!mMediaTimeTransformValid)
1261            return INVALID_OPERATION;
1262    }
1263
1264    Mutex::Autolock _l(mTimedBufferQueueLock);
1265
1266    uint32_t bufFrames = buffer->size() / mFrameSize;
1267    mFramesPendingInQueue += bufFrames;
1268    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1269
1270    return NO_ERROR;
1271}
1272
1273status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1274    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1275
1276    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1277           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1278           target);
1279
1280    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1281          target == TimedAudioTrack::COMMON_TIME)) {
1282        return BAD_VALUE;
1283    }
1284
1285    Mutex::Autolock lock(mMediaTimeTransformLock);
1286    mMediaTimeTransform = xform;
1287    mMediaTimeTransformTarget = target;
1288    mMediaTimeTransformValid = true;
1289
1290    return NO_ERROR;
1291}
1292
1293#define min(a, b) ((a) < (b) ? (a) : (b))
1294
1295// implementation of getNextBuffer for tracks whose buffers have timestamps
1296status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1297    AudioBufferProvider::Buffer* buffer, int64_t pts)
1298{
1299    if (pts == AudioBufferProvider::kInvalidPTS) {
1300        buffer->raw = NULL;
1301        buffer->frameCount = 0;
1302        mTimedAudioOutputOnTime = false;
1303        return INVALID_OPERATION;
1304    }
1305
1306    Mutex::Autolock _l(mTimedBufferQueueLock);
1307
1308    ALOG_ASSERT(!mQueueHeadInFlight,
1309                "getNextBuffer called without releaseBuffer!");
1310
1311    while (true) {
1312
1313        // if we have no timed buffers, then fail
1314        if (mTimedBufferQueue.isEmpty()) {
1315            buffer->raw = NULL;
1316            buffer->frameCount = 0;
1317            return NOT_ENOUGH_DATA;
1318        }
1319
1320        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1321
1322        // calculate the PTS of the head of the timed buffer queue expressed in
1323        // local time
1324        int64_t headLocalPTS;
1325        {
1326            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1327
1328            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1329
1330            if (mMediaTimeTransform.a_to_b_denom == 0) {
1331                // the transform represents a pause, so yield silence
1332                timedYieldSilence_l(buffer->frameCount, buffer);
1333                return NO_ERROR;
1334            }
1335
1336            int64_t transformedPTS;
1337            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1338                                                        &transformedPTS)) {
1339                // the transform failed.  this shouldn't happen, but if it does
1340                // then just drop this buffer
1341                ALOGW("timedGetNextBuffer transform failed");
1342                buffer->raw = NULL;
1343                buffer->frameCount = 0;
1344                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1345                return NO_ERROR;
1346            }
1347
1348            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1349                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1350                                                          &headLocalPTS)) {
1351                    buffer->raw = NULL;
1352                    buffer->frameCount = 0;
1353                    return INVALID_OPERATION;
1354                }
1355            } else {
1356                headLocalPTS = transformedPTS;
1357            }
1358        }
1359
1360        uint32_t sr = sampleRate();
1361
1362        // adjust the head buffer's PTS to reflect the portion of the head buffer
1363        // that has already been consumed
1364        int64_t effectivePTS = headLocalPTS +
1365                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1366
1367        // Calculate the delta in samples between the head of the input buffer
1368        // queue and the start of the next output buffer that will be written.
1369        // If the transformation fails because of over or underflow, it means
1370        // that the sample's position in the output stream is so far out of
1371        // whack that it should just be dropped.
1372        int64_t sampleDelta;
1373        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1374            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1375            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1376                                       " mix");
1377            continue;
1378        }
1379        if (!mLocalTimeToSampleTransform.doForwardTransform(
1380                (effectivePTS - pts) << 32, &sampleDelta)) {
1381            ALOGV("*** too late during sample rate transform: dropped buffer");
1382            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1383            continue;
1384        }
1385
1386        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1387               " sampleDelta=[%d.%08x]",
1388               head.pts(), head.position(), pts,
1389               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1390                   + (sampleDelta >> 32)),
1391               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1392
1393        // if the delta between the ideal placement for the next input sample and
1394        // the current output position is within this threshold, then we will
1395        // concatenate the next input samples to the previous output
1396        const int64_t kSampleContinuityThreshold =
1397                (static_cast<int64_t>(sr) << 32) / 250;
1398
1399        // if this is the first buffer of audio that we're emitting from this track
1400        // then it should be almost exactly on time.
1401        const int64_t kSampleStartupThreshold = 1LL << 32;
1402
1403        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1404           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1405            // the next input is close enough to being on time, so concatenate it
1406            // with the last output
1407            timedYieldSamples_l(buffer);
1408
1409            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1410                    head.position(), buffer->frameCount);
1411            return NO_ERROR;
1412        }
1413
1414        // Looks like our output is not on time.  Reset our on timed status.
1415        // Next time we mix samples from our input queue, then should be within
1416        // the StartupThreshold.
1417        mTimedAudioOutputOnTime = false;
1418        if (sampleDelta > 0) {
1419            // the gap between the current output position and the proper start of
1420            // the next input sample is too big, so fill it with silence
1421            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1422
1423            timedYieldSilence_l(framesUntilNextInput, buffer);
1424            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1425            return NO_ERROR;
1426        } else {
1427            // the next input sample is late
1428            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1429            size_t onTimeSamplePosition =
1430                    head.position() + lateFrames * mFrameSize;
1431
1432            if (onTimeSamplePosition > head.buffer()->size()) {
1433                // all the remaining samples in the head are too late, so
1434                // drop it and move on
1435                ALOGV("*** too late: dropped buffer");
1436                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1437                continue;
1438            } else {
1439                // skip over the late samples
1440                head.setPosition(onTimeSamplePosition);
1441
1442                // yield the available samples
1443                timedYieldSamples_l(buffer);
1444
1445                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1446                return NO_ERROR;
1447            }
1448        }
1449    }
1450}
1451
1452// Yield samples from the timed buffer queue head up to the given output
1453// buffer's capacity.
1454//
1455// Caller must hold mTimedBufferQueueLock
1456void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1457    AudioBufferProvider::Buffer* buffer) {
1458
1459    const TimedBuffer& head = mTimedBufferQueue[0];
1460
1461    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1462                   head.position());
1463
1464    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1465                                 mFrameSize);
1466    size_t framesRequested = buffer->frameCount;
1467    buffer->frameCount = min(framesLeftInHead, framesRequested);
1468
1469    mQueueHeadInFlight = true;
1470    mTimedAudioOutputOnTime = true;
1471}
1472
1473// Yield samples of silence up to the given output buffer's capacity
1474//
1475// Caller must hold mTimedBufferQueueLock
1476void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1477    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1478
1479    // lazily allocate a buffer filled with silence
1480    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1481        delete [] mTimedSilenceBuffer;
1482        mTimedSilenceBufferSize = numFrames * mFrameSize;
1483        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1484        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1485    }
1486
1487    buffer->raw = mTimedSilenceBuffer;
1488    size_t framesRequested = buffer->frameCount;
1489    buffer->frameCount = min(numFrames, framesRequested);
1490
1491    mTimedAudioOutputOnTime = false;
1492}
1493
1494// AudioBufferProvider interface
1495void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1496    AudioBufferProvider::Buffer* buffer) {
1497
1498    Mutex::Autolock _l(mTimedBufferQueueLock);
1499
1500    // If the buffer which was just released is part of the buffer at the head
1501    // of the queue, be sure to update the amt of the buffer which has been
1502    // consumed.  If the buffer being returned is not part of the head of the
1503    // queue, its either because the buffer is part of the silence buffer, or
1504    // because the head of the timed queue was trimmed after the mixer called
1505    // getNextBuffer but before the mixer called releaseBuffer.
1506    if (buffer->raw == mTimedSilenceBuffer) {
1507        ALOG_ASSERT(!mQueueHeadInFlight,
1508                    "Queue head in flight during release of silence buffer!");
1509        goto done;
1510    }
1511
1512    ALOG_ASSERT(mQueueHeadInFlight,
1513                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1514                " head in flight.");
1515
1516    if (mTimedBufferQueue.size()) {
1517        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1518
1519        void* start = head.buffer()->pointer();
1520        void* end   = reinterpret_cast<void*>(
1521                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1522                        + head.buffer()->size());
1523
1524        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1525                    "released buffer not within the head of the timed buffer"
1526                    " queue; qHead = [%p, %p], released buffer = %p",
1527                    start, end, buffer->raw);
1528
1529        head.setPosition(head.position() +
1530                (buffer->frameCount * mFrameSize));
1531        mQueueHeadInFlight = false;
1532
1533        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1534                    "Bad bookkeeping during releaseBuffer!  Should have at"
1535                    " least %u queued frames, but we think we have only %u",
1536                    buffer->frameCount, mFramesPendingInQueue);
1537
1538        mFramesPendingInQueue -= buffer->frameCount;
1539
1540        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1541            || mTrimQueueHeadOnRelease) {
1542            trimTimedBufferQueueHead_l("releaseBuffer");
1543            mTrimQueueHeadOnRelease = false;
1544        }
1545    } else {
1546        LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1547                  " buffers in the timed buffer queue");
1548    }
1549
1550done:
1551    buffer->raw = 0;
1552    buffer->frameCount = 0;
1553}
1554
1555size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1556    Mutex::Autolock _l(mTimedBufferQueueLock);
1557    return mFramesPendingInQueue;
1558}
1559
1560AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1561        : mPTS(0), mPosition(0) {}
1562
1563AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1564    const sp<IMemory>& buffer, int64_t pts)
1565        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1566
1567
1568// ----------------------------------------------------------------------------
1569
1570AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1571            PlaybackThread *playbackThread,
1572            DuplicatingThread *sourceThread,
1573            uint32_t sampleRate,
1574            audio_format_t format,
1575            audio_channel_mask_t channelMask,
1576            size_t frameCount,
1577            int uid)
1578    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1579                NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
1580    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1581{
1582
1583    if (mCblk != NULL) {
1584        mOutBuffer.frameCount = 0;
1585        playbackThread->mTracks.add(this);
1586        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1587                "frameCount %u, mChannelMask 0x%08x",
1588                mCblk, mBuffer,
1589                frameCount, mChannelMask);
1590        // since client and server are in the same process,
1591        // the buffer has the same virtual address on both sides
1592        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1593        mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
1594        mClientProxy->setSendLevel(0.0);
1595        mClientProxy->setSampleRate(sampleRate);
1596        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1597                true /*clientInServer*/);
1598    } else {
1599        ALOGW("Error creating output track on thread %p", playbackThread);
1600    }
1601}
1602
1603AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1604{
1605    clearBufferQueue();
1606    delete mClientProxy;
1607    // superclass destructor will now delete the server proxy and shared memory both refer to
1608}
1609
1610status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1611                                                          int triggerSession)
1612{
1613    status_t status = Track::start(event, triggerSession);
1614    if (status != NO_ERROR) {
1615        return status;
1616    }
1617
1618    mActive = true;
1619    mRetryCount = 127;
1620    return status;
1621}
1622
1623void AudioFlinger::PlaybackThread::OutputTrack::stop()
1624{
1625    Track::stop();
1626    clearBufferQueue();
1627    mOutBuffer.frameCount = 0;
1628    mActive = false;
1629}
1630
1631bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1632{
1633    Buffer *pInBuffer;
1634    Buffer inBuffer;
1635    uint32_t channelCount = mChannelCount;
1636    bool outputBufferFull = false;
1637    inBuffer.frameCount = frames;
1638    inBuffer.i16 = data;
1639
1640    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1641
1642    if (!mActive && frames != 0) {
1643        start();
1644        sp<ThreadBase> thread = mThread.promote();
1645        if (thread != 0) {
1646            MixerThread *mixerThread = (MixerThread *)thread.get();
1647            if (mFrameCount > frames) {
1648                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1649                    uint32_t startFrames = (mFrameCount - frames);
1650                    pInBuffer = new Buffer;
1651                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1652                    pInBuffer->frameCount = startFrames;
1653                    pInBuffer->i16 = pInBuffer->mBuffer;
1654                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1655                    mBufferQueue.add(pInBuffer);
1656                } else {
1657                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1658                }
1659            }
1660        }
1661    }
1662
1663    while (waitTimeLeftMs) {
1664        // First write pending buffers, then new data
1665        if (mBufferQueue.size()) {
1666            pInBuffer = mBufferQueue.itemAt(0);
1667        } else {
1668            pInBuffer = &inBuffer;
1669        }
1670
1671        if (pInBuffer->frameCount == 0) {
1672            break;
1673        }
1674
1675        if (mOutBuffer.frameCount == 0) {
1676            mOutBuffer.frameCount = pInBuffer->frameCount;
1677            nsecs_t startTime = systemTime();
1678            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1679            if (status != NO_ERROR) {
1680                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1681                        mThread.unsafe_get(), status);
1682                outputBufferFull = true;
1683                break;
1684            }
1685            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1686            if (waitTimeLeftMs >= waitTimeMs) {
1687                waitTimeLeftMs -= waitTimeMs;
1688            } else {
1689                waitTimeLeftMs = 0;
1690            }
1691        }
1692
1693        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1694                pInBuffer->frameCount;
1695        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1696        Proxy::Buffer buf;
1697        buf.mFrameCount = outFrames;
1698        buf.mRaw = NULL;
1699        mClientProxy->releaseBuffer(&buf);
1700        pInBuffer->frameCount -= outFrames;
1701        pInBuffer->i16 += outFrames * channelCount;
1702        mOutBuffer.frameCount -= outFrames;
1703        mOutBuffer.i16 += outFrames * channelCount;
1704
1705        if (pInBuffer->frameCount == 0) {
1706            if (mBufferQueue.size()) {
1707                mBufferQueue.removeAt(0);
1708                delete [] pInBuffer->mBuffer;
1709                delete pInBuffer;
1710                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1711                        mThread.unsafe_get(), mBufferQueue.size());
1712            } else {
1713                break;
1714            }
1715        }
1716    }
1717
1718    // If we could not write all frames, allocate a buffer and queue it for next time.
1719    if (inBuffer.frameCount) {
1720        sp<ThreadBase> thread = mThread.promote();
1721        if (thread != 0 && !thread->standby()) {
1722            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1723                pInBuffer = new Buffer;
1724                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1725                pInBuffer->frameCount = inBuffer.frameCount;
1726                pInBuffer->i16 = pInBuffer->mBuffer;
1727                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1728                        sizeof(int16_t));
1729                mBufferQueue.add(pInBuffer);
1730                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1731                        mThread.unsafe_get(), mBufferQueue.size());
1732            } else {
1733                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1734                        mThread.unsafe_get(), this);
1735            }
1736        }
1737    }
1738
1739    // Calling write() with a 0 length buffer, means that no more data will be written:
1740    // If no more buffers are pending, fill output track buffer to make sure it is started
1741    // by output mixer.
1742    if (frames == 0 && mBufferQueue.size() == 0) {
1743        // FIXME borken, replace by getting framesReady() from proxy
1744        size_t user = 0;    // was mCblk->user
1745        if (user < mFrameCount) {
1746            frames = mFrameCount - user;
1747            pInBuffer = new Buffer;
1748            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1749            pInBuffer->frameCount = frames;
1750            pInBuffer->i16 = pInBuffer->mBuffer;
1751            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1752            mBufferQueue.add(pInBuffer);
1753        } else if (mActive) {
1754            stop();
1755        }
1756    }
1757
1758    return outputBufferFull;
1759}
1760
1761status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1762        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1763{
1764    ClientProxy::Buffer buf;
1765    buf.mFrameCount = buffer->frameCount;
1766    struct timespec timeout;
1767    timeout.tv_sec = waitTimeMs / 1000;
1768    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1769    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1770    buffer->frameCount = buf.mFrameCount;
1771    buffer->raw = buf.mRaw;
1772    return status;
1773}
1774
1775void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1776{
1777    size_t size = mBufferQueue.size();
1778
1779    for (size_t i = 0; i < size; i++) {
1780        Buffer *pBuffer = mBufferQueue.itemAt(i);
1781        delete [] pBuffer->mBuffer;
1782        delete pBuffer;
1783    }
1784    mBufferQueue.clear();
1785}
1786
1787
1788// ----------------------------------------------------------------------------
1789//      Record
1790// ----------------------------------------------------------------------------
1791
1792AudioFlinger::RecordHandle::RecordHandle(
1793        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1794    : BnAudioRecord(),
1795    mRecordTrack(recordTrack)
1796{
1797}
1798
1799AudioFlinger::RecordHandle::~RecordHandle() {
1800    stop_nonvirtual();
1801    mRecordTrack->destroy();
1802}
1803
1804status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1805        int triggerSession) {
1806    ALOGV("RecordHandle::start()");
1807    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1808}
1809
1810void AudioFlinger::RecordHandle::stop() {
1811    stop_nonvirtual();
1812}
1813
1814void AudioFlinger::RecordHandle::stop_nonvirtual() {
1815    ALOGV("RecordHandle::stop()");
1816    mRecordTrack->stop();
1817}
1818
1819status_t AudioFlinger::RecordHandle::onTransact(
1820    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1821{
1822    return BnAudioRecord::onTransact(code, data, reply, flags);
1823}
1824
1825// ----------------------------------------------------------------------------
1826
1827// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
1828AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1829            RecordThread *thread,
1830            const sp<Client>& client,
1831            uint32_t sampleRate,
1832            audio_format_t format,
1833            audio_channel_mask_t channelMask,
1834            size_t frameCount,
1835            int sessionId,
1836            int uid,
1837            IAudioFlinger::track_flags_t flags)
1838    :   TrackBase(thread, client, sampleRate, format,
1839                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid,
1840                  flags, false /*isOut*/,
1841                  (flags & IAudioFlinger::TRACK_FAST) != 0 /*useReadOnlyHeap*/),
1842        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
1843        // See real initialization of mRsmpInFront at RecordThread::start()
1844        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
1845{
1846    if (mCblk == NULL) {
1847        return;
1848    }
1849
1850    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
1851
1852    uint32_t channelCount = popcount(channelMask);
1853    // FIXME I don't understand either of the channel count checks
1854    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
1855            channelCount <= FCC_2) {
1856        // sink SR
1857        mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
1858        // source SR
1859        mResampler->setSampleRate(thread->mSampleRate);
1860        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
1861        mResamplerBufferProvider = new ResamplerBufferProvider(this);
1862    }
1863}
1864
1865AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1866{
1867    ALOGV("%s", __func__);
1868    delete mResampler;
1869    delete[] mRsmpOutBuffer;
1870    delete mResamplerBufferProvider;
1871}
1872
1873// AudioBufferProvider interface
1874status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1875        int64_t pts __unused)
1876{
1877    ServerProxy::Buffer buf;
1878    buf.mFrameCount = buffer->frameCount;
1879    status_t status = mServerProxy->obtainBuffer(&buf);
1880    buffer->frameCount = buf.mFrameCount;
1881    buffer->raw = buf.mRaw;
1882    if (buf.mFrameCount == 0) {
1883        // FIXME also wake futex so that overrun is noticed more quickly
1884        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1885    }
1886    return status;
1887}
1888
1889status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1890                                                        int triggerSession)
1891{
1892    sp<ThreadBase> thread = mThread.promote();
1893    if (thread != 0) {
1894        RecordThread *recordThread = (RecordThread *)thread.get();
1895        return recordThread->start(this, event, triggerSession);
1896    } else {
1897        return BAD_VALUE;
1898    }
1899}
1900
1901void AudioFlinger::RecordThread::RecordTrack::stop()
1902{
1903    sp<ThreadBase> thread = mThread.promote();
1904    if (thread != 0) {
1905        RecordThread *recordThread = (RecordThread *)thread.get();
1906        if (recordThread->stop(this)) {
1907            AudioSystem::stopInput(recordThread->id());
1908        }
1909    }
1910}
1911
1912void AudioFlinger::RecordThread::RecordTrack::destroy()
1913{
1914    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1915    sp<RecordTrack> keep(this);
1916    {
1917        sp<ThreadBase> thread = mThread.promote();
1918        if (thread != 0) {
1919            if (mState == ACTIVE || mState == RESUMING) {
1920                AudioSystem::stopInput(thread->id());
1921            }
1922            AudioSystem::releaseInput(thread->id());
1923            Mutex::Autolock _l(thread->mLock);
1924            RecordThread *recordThread = (RecordThread *) thread.get();
1925            recordThread->destroyTrack_l(this);
1926        }
1927    }
1928}
1929
1930void AudioFlinger::RecordThread::RecordTrack::invalidate()
1931{
1932    // FIXME should use proxy, and needs work
1933    audio_track_cblk_t* cblk = mCblk;
1934    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1935    android_atomic_release_store(0x40000000, &cblk->mFutex);
1936    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1937    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1938}
1939
1940
1941/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1942{
1943    result.append("    Active Client Fmt Chn mask Session S   Server fCount Resampling\n");
1944}
1945
1946void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
1947{
1948    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n",
1949            active ? "yes" : "no",
1950            (mClient == 0) ? getpid_cached : mClient->pid(),
1951            mFormat,
1952            mChannelMask,
1953            mSessionId,
1954            mState,
1955            mCblk->mServer,
1956            mFrameCount,
1957            mResampler != NULL);
1958
1959}
1960
1961void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
1962{
1963    if (event == mSyncStartEvent) {
1964        ssize_t framesToDrop = 0;
1965        sp<ThreadBase> threadBase = mThread.promote();
1966        if (threadBase != 0) {
1967            // TODO: use actual buffer filling status instead of 2 buffers when info is available
1968            // from audio HAL
1969            framesToDrop = threadBase->mFrameCount * 2;
1970        }
1971        mFramesToDrop = framesToDrop;
1972    }
1973}
1974
1975void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
1976{
1977    if (mSyncStartEvent != 0) {
1978        mSyncStartEvent->cancel();
1979        mSyncStartEvent.clear();
1980    }
1981    mFramesToDrop = 0;
1982}
1983
1984}; // namespace android
1985