Tracks.cpp revision fe9570c7b937b49d3603ccb394aed732b79bc6be 
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            bool useReadOnlyHeap)
74    :   RefBase(),
75        mThread(thread),
76        mClient(client),
77        mCblk(NULL),
78        // mBuffer
79        mState(IDLE),
80        mSampleRate(sampleRate),
81        mFormat(format),
82        mChannelMask(channelMask),
83        mChannelCount(popcount(channelMask)),
84        mFrameSize(audio_is_linear_pcm(format) ?
85                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
86        mFrameCount(frameCount),
87        mSessionId(sessionId),
88        mIsOut(isOut),
89        mServerProxy(NULL),
90        mId(android_atomic_inc(&nextTrackId)),
91        mTerminated(false)
92{
93    // if the caller is us, trust the specified uid
94    if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
95        int newclientUid = IPCThreadState::self()->getCallingUid();
96        if (clientUid != -1 && clientUid != newclientUid) {
97            ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
98        }
99        clientUid = newclientUid;
100    }
101    // clientUid contains the uid of the app that is responsible for this track, so we can blame
102    // battery usage on it.
103    mUid = clientUid;
104
105    // client == 0 implies sharedBuffer == 0
106    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
107
108    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
109            sharedBuffer->size());
110
111    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
112    size_t size = sizeof(audio_track_cblk_t);
113    size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
114    if (sharedBuffer == 0 && !useReadOnlyHeap) {
115        size += bufferSize;
116    }
117
118    if (client != 0) {
119        mCblkMemory = client->heap()->allocate(size);
120        if (mCblkMemory == 0 ||
121                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
122            ALOGE("not enough memory for AudioTrack size=%u", size);
123            client->heap()->dump("AudioTrack");
124            mCblkMemory.clear();
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        if (useReadOnlyHeap) {
137            const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
138            if (roHeap == 0 ||
139                    (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
140                    (mBuffer = mBufferMemory->pointer()) == NULL) {
141                ALOGE("not enough memory for read-only buffer size=%zu", bufferSize);
142                if (roHeap != 0) {
143                    roHeap->dump("buffer");
144                }
145                mCblkMemory.clear();
146                mBufferMemory.clear();
147                return;
148            }
149            memset(mBuffer, 0, bufferSize);
150        } else {
151            // clear all buffers
152            if (sharedBuffer == 0) {
153                mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
154                memset(mBuffer, 0, bufferSize);
155            } else {
156                mBuffer = sharedBuffer->pointer();
157#if 0
158                mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
159#endif
160            }
161        }
162
163#ifdef TEE_SINK
164        if (mTeeSinkTrackEnabled) {
165            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
166            if (Format_isValid(pipeFormat)) {
167                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
168                size_t numCounterOffers = 0;
169                const NBAIO_Format offers[1] = {pipeFormat};
170                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
171                ALOG_ASSERT(index == 0);
172                PipeReader *pipeReader = new PipeReader(*pipe);
173                numCounterOffers = 0;
174                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
175                ALOG_ASSERT(index == 0);
176                mTeeSink = pipe;
177                mTeeSource = pipeReader;
178            }
179        }
180#endif
181
182    }
183}
184
185AudioFlinger::ThreadBase::TrackBase::~TrackBase()
186{
187#ifdef TEE_SINK
188    dumpTee(-1, mTeeSource, mId);
189#endif
190    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
191    delete mServerProxy;
192    if (mCblk != NULL) {
193        if (mClient == 0) {
194            delete mCblk;
195        } else {
196            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
197        }
198    }
199    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
200    if (mClient != 0) {
201        // Client destructor must run with AudioFlinger mutex locked
202        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
203        // If the client's reference count drops to zero, the associated destructor
204        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
205        // relying on the automatic clear() at end of scope.
206        mClient.clear();
207    }
208}
209
210// AudioBufferProvider interface
211// getNextBuffer() = 0;
212// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
213void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
214{
215#ifdef TEE_SINK
216    if (mTeeSink != 0) {
217        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
218    }
219#endif
220
221    ServerProxy::Buffer buf;
222    buf.mFrameCount = buffer->frameCount;
223    buf.mRaw = buffer->raw;
224    buffer->frameCount = 0;
225    buffer->raw = NULL;
226    mServerProxy->releaseBuffer(&buf);
227}
228
229status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
230{
231    mSyncEvents.add(event);
232    return NO_ERROR;
233}
234
235// ----------------------------------------------------------------------------
236//      Playback
237// ----------------------------------------------------------------------------
238
239AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
240    : BnAudioTrack(),
241      mTrack(track)
242{
243}
244
245AudioFlinger::TrackHandle::~TrackHandle() {
246    // just stop the track on deletion, associated resources
247    // will be freed from the main thread once all pending buffers have
248    // been played. Unless it's not in the active track list, in which
249    // case we free everything now...
250    mTrack->destroy();
251}
252
253sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
254    return mTrack->getCblk();
255}
256
257status_t AudioFlinger::TrackHandle::start() {
258    return mTrack->start();
259}
260
261void AudioFlinger::TrackHandle::stop() {
262    mTrack->stop();
263}
264
265void AudioFlinger::TrackHandle::flush() {
266    mTrack->flush();
267}
268
269void AudioFlinger::TrackHandle::pause() {
270    mTrack->pause();
271}
272
273status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
274{
275    return mTrack->attachAuxEffect(EffectId);
276}
277
278status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
279                                                         sp<IMemory>* buffer) {
280    if (!mTrack->isTimedTrack())
281        return INVALID_OPERATION;
282
283    PlaybackThread::TimedTrack* tt =
284            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
285    return tt->allocateTimedBuffer(size, buffer);
286}
287
288status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
289                                                     int64_t pts) {
290    if (!mTrack->isTimedTrack())
291        return INVALID_OPERATION;
292
293    if (buffer == 0 || buffer->pointer() == NULL) {
294        ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
295        return BAD_VALUE;
296    }
297
298    PlaybackThread::TimedTrack* tt =
299            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
300    return tt->queueTimedBuffer(buffer, pts);
301}
302
303status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
304    const LinearTransform& xform, int target) {
305
306    if (!mTrack->isTimedTrack())
307        return INVALID_OPERATION;
308
309    PlaybackThread::TimedTrack* tt =
310            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
311    return tt->setMediaTimeTransform(
312        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
313}
314
315status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
316    return mTrack->setParameters(keyValuePairs);
317}
318
319status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
320{
321    return mTrack->getTimestamp(timestamp);
322}
323
324
325void AudioFlinger::TrackHandle::signal()
326{
327    return mTrack->signal();
328}
329
330status_t AudioFlinger::TrackHandle::onTransact(
331    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
332{
333    return BnAudioTrack::onTransact(code, data, reply, flags);
334}
335
336// ----------------------------------------------------------------------------
337
338// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
339AudioFlinger::PlaybackThread::Track::Track(
340            PlaybackThread *thread,
341            const sp<Client>& client,
342            audio_stream_type_t streamType,
343            uint32_t sampleRate,
344            audio_format_t format,
345            audio_channel_mask_t channelMask,
346            size_t frameCount,
347            const sp<IMemory>& sharedBuffer,
348            int sessionId,
349            int uid,
350            IAudioFlinger::track_flags_t flags)
351    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
352            sessionId, uid, true /*isOut*/),
353    mFillingUpStatus(FS_INVALID),
354    // mRetryCount initialized later when needed
355    mSharedBuffer(sharedBuffer),
356    mStreamType(streamType),
357    mName(-1),  // see note below
358    mMainBuffer(thread->mixBuffer()),
359    mAuxBuffer(NULL),
360    mAuxEffectId(0), mHasVolumeController(false),
361    mPresentationCompleteFrames(0),
362    mFlags(flags),
363    mFastIndex(-1),
364    mCachedVolume(1.0),
365    mIsInvalid(false),
366    mAudioTrackServerProxy(NULL),
367    mResumeToStopping(false),
368    mFlushHwPending(false)
369{
370    if (mCblk == NULL) {
371        return;
372    }
373
374    if (sharedBuffer == 0) {
375        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
376                mFrameSize);
377    } else {
378        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
379                mFrameSize);
380    }
381    mServerProxy = mAudioTrackServerProxy;
382
383    mName = thread->getTrackName_l(channelMask, sessionId);
384    if (mName < 0) {
385        ALOGE("no more track names available");
386        return;
387    }
388    // only allocate a fast track index if we were able to allocate a normal track name
389    if (flags & IAudioFlinger::TRACK_FAST) {
390        mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
391        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
392        int i = __builtin_ctz(thread->mFastTrackAvailMask);
393        ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
394        // FIXME This is too eager.  We allocate a fast track index before the
395        //       fast track becomes active.  Since fast tracks are a scarce resource,
396        //       this means we are potentially denying other more important fast tracks from
397        //       being created.  It would be better to allocate the index dynamically.
398        mFastIndex = i;
399        // Read the initial underruns because this field is never cleared by the fast mixer
400        mObservedUnderruns = thread->getFastTrackUnderruns(i);
401        thread->mFastTrackAvailMask &= ~(1 << i);
402    }
403}
404
405AudioFlinger::PlaybackThread::Track::~Track()
406{
407    ALOGV("PlaybackThread::Track destructor");
408
409    // The destructor would clear mSharedBuffer,
410    // but it will not push the decremented reference count,
411    // leaving the client's IMemory dangling indefinitely.
412    // This prevents that leak.
413    if (mSharedBuffer != 0) {
414        mSharedBuffer.clear();
415        // flush the binder command buffer
416        IPCThreadState::self()->flushCommands();
417    }
418}
419
420status_t AudioFlinger::PlaybackThread::Track::initCheck() const
421{
422    status_t status = TrackBase::initCheck();
423    if (status == NO_ERROR && mName < 0) {
424        status = NO_MEMORY;
425    }
426    return status;
427}
428
429void AudioFlinger::PlaybackThread::Track::destroy()
430{
431    // NOTE: destroyTrack_l() can remove a strong reference to this Track
432    // by removing it from mTracks vector, so there is a risk that this Tracks's
433    // destructor is called. As the destructor needs to lock mLock,
434    // we must acquire a strong reference on this Track before locking mLock
435    // here so that the destructor is called only when exiting this function.
436    // On the other hand, as long as Track::destroy() is only called by
437    // TrackHandle destructor, the TrackHandle still holds a strong ref on
438    // this Track with its member mTrack.
439    sp<Track> keep(this);
440    { // scope for mLock
441        sp<ThreadBase> thread = mThread.promote();
442        if (thread != 0) {
443            Mutex::Autolock _l(thread->mLock);
444            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
445            bool wasActive = playbackThread->destroyTrack_l(this);
446            if (!isOutputTrack() && !wasActive) {
447                AudioSystem::releaseOutput(thread->id());
448            }
449        }
450    }
451}
452
453/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
454{
455    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
456                  "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
457}
458
459void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
460{
461    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
462    if (isFastTrack()) {
463        sprintf(buffer, "    F %2d", mFastIndex);
464    } else if (mName >= AudioMixer::TRACK0) {
465        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
466    } else {
467        sprintf(buffer, "    none");
468    }
469    track_state state = mState;
470    char stateChar;
471    if (isTerminated()) {
472        stateChar = 'T';
473    } else {
474        switch (state) {
475        case IDLE:
476            stateChar = 'I';
477            break;
478        case STOPPING_1:
479            stateChar = 's';
480            break;
481        case STOPPING_2:
482            stateChar = '5';
483            break;
484        case STOPPED:
485            stateChar = 'S';
486            break;
487        case RESUMING:
488            stateChar = 'R';
489            break;
490        case ACTIVE:
491            stateChar = 'A';
492            break;
493        case PAUSING:
494            stateChar = 'p';
495            break;
496        case PAUSED:
497            stateChar = 'P';
498            break;
499        case FLUSHED:
500            stateChar = 'F';
501            break;
502        default:
503            stateChar = '?';
504            break;
505        }
506    }
507    char nowInUnderrun;
508    switch (mObservedUnderruns.mBitFields.mMostRecent) {
509    case UNDERRUN_FULL:
510        nowInUnderrun = ' ';
511        break;
512    case UNDERRUN_PARTIAL:
513        nowInUnderrun = '<';
514        break;
515    case UNDERRUN_EMPTY:
516        nowInUnderrun = '*';
517        break;
518    default:
519        nowInUnderrun = '?';
520        break;
521    }
522    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
523                                 "%08X %p %p 0x%03X %9u%c\n",
524            active ? "yes" : "no",
525            (mClient == 0) ? getpid_cached : mClient->pid(),
526            mStreamType,
527            mFormat,
528            mChannelMask,
529            mSessionId,
530            mFrameCount,
531            stateChar,
532            mFillingUpStatus,
533            mAudioTrackServerProxy->getSampleRate(),
534            20.0 * log10((vlr & 0xFFFF) / 4096.0),
535            20.0 * log10((vlr >> 16) / 4096.0),
536            mCblk->mServer,
537            mMainBuffer,
538            mAuxBuffer,
539            mCblk->mFlags,
540            mAudioTrackServerProxy->getUnderrunFrames(),
541            nowInUnderrun);
542}
543
544uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
545    return mAudioTrackServerProxy->getSampleRate();
546}
547
548// AudioBufferProvider interface
549status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
550        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
551{
552    ServerProxy::Buffer buf;
553    size_t desiredFrames = buffer->frameCount;
554    buf.mFrameCount = desiredFrames;
555    status_t status = mServerProxy->obtainBuffer(&buf);
556    buffer->frameCount = buf.mFrameCount;
557    buffer->raw = buf.mRaw;
558    if (buf.mFrameCount == 0) {
559        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
560    }
561    return status;
562}
563
564// releaseBuffer() is not overridden
565
566// ExtendedAudioBufferProvider interface
567
568// Note that framesReady() takes a mutex on the control block using tryLock().
569// This could result in priority inversion if framesReady() is called by the normal mixer,
570// as the normal mixer thread runs at lower
571// priority than the client's callback thread:  there is a short window within framesReady()
572// during which the normal mixer could be preempted, and the client callback would block.
573// Another problem can occur if framesReady() is called by the fast mixer:
574// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
575// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
576size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
577    return mAudioTrackServerProxy->framesReady();
578}
579
580size_t AudioFlinger::PlaybackThread::Track::framesReleased() const
581{
582    return mAudioTrackServerProxy->framesReleased();
583}
584
585// Don't call for fast tracks; the framesReady() could result in priority inversion
586bool AudioFlinger::PlaybackThread::Track::isReady() const {
587    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
588        return true;
589    }
590
591    if (isStopping()) {
592        if (framesReady() > 0) {
593            mFillingUpStatus = FS_FILLED;
594        }
595        return true;
596    }
597
598    if (framesReady() >= mFrameCount ||
599            (mCblk->mFlags & CBLK_FORCEREADY)) {
600        mFillingUpStatus = FS_FILLED;
601        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
602        return true;
603    }
604    return false;
605}
606
607status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
608                                                    int triggerSession __unused)
609{
610    status_t status = NO_ERROR;
611    ALOGV("start(%d), calling pid %d session %d",
612            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
613
614    sp<ThreadBase> thread = mThread.promote();
615    if (thread != 0) {
616        if (isOffloaded()) {
617            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
618            Mutex::Autolock _lth(thread->mLock);
619            sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
620            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
621                    (ec != 0 && ec->isNonOffloadableEnabled())) {
622                invalidate();
623                return PERMISSION_DENIED;
624            }
625        }
626        Mutex::Autolock _lth(thread->mLock);
627        track_state state = mState;
628        // here the track could be either new, or restarted
629        // in both cases "unstop" the track
630
631        // initial state-stopping. next state-pausing.
632        // What if resume is called ?
633
634        if (state == PAUSED || state == PAUSING) {
635            if (mResumeToStopping) {
636                // happened we need to resume to STOPPING_1
637                mState = TrackBase::STOPPING_1;
638                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
639            } else {
640                mState = TrackBase::RESUMING;
641                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
642            }
643        } else {
644            mState = TrackBase::ACTIVE;
645            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
646        }
647
648        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
649        status = playbackThread->addTrack_l(this);
650        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
651            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
652            //  restore previous state if start was rejected by policy manager
653            if (status == PERMISSION_DENIED) {
654                mState = state;
655            }
656        }
657        // track was already in the active list, not a problem
658        if (status == ALREADY_EXISTS) {
659            status = NO_ERROR;
660        } else {
661            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
662            // It is usually unsafe to access the server proxy from a binder thread.
663            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
664            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
665            // and for fast tracks the track is not yet in the fast mixer thread's active set.
666            ServerProxy::Buffer buffer;
667            buffer.mFrameCount = 1;
668            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
669        }
670    } else {
671        status = BAD_VALUE;
672    }
673    return status;
674}
675
676void AudioFlinger::PlaybackThread::Track::stop()
677{
678    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
679    sp<ThreadBase> thread = mThread.promote();
680    if (thread != 0) {
681        Mutex::Autolock _l(thread->mLock);
682        track_state state = mState;
683        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
684            // If the track is not active (PAUSED and buffers full), flush buffers
685            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
686            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
687                reset();
688                mState = STOPPED;
689            } else if (!isFastTrack() && !isOffloaded()) {
690                mState = STOPPED;
691            } else {
692                // For fast tracks prepareTracks_l() will set state to STOPPING_2
693                // presentation is complete
694                // For an offloaded track this starts a drain and state will
695                // move to STOPPING_2 when drain completes and then STOPPED
696                mState = STOPPING_1;
697            }
698            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
699                    playbackThread);
700        }
701    }
702}
703
704void AudioFlinger::PlaybackThread::Track::pause()
705{
706    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
707    sp<ThreadBase> thread = mThread.promote();
708    if (thread != 0) {
709        Mutex::Autolock _l(thread->mLock);
710        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
711        switch (mState) {
712        case STOPPING_1:
713        case STOPPING_2:
714            if (!isOffloaded()) {
715                /* nothing to do if track is not offloaded */
716                break;
717            }
718
719            // Offloaded track was draining, we need to carry on draining when resumed
720            mResumeToStopping = true;
721            // fall through...
722        case ACTIVE:
723        case RESUMING:
724            mState = PAUSING;
725            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
726            playbackThread->broadcast_l();
727            break;
728
729        default:
730            break;
731        }
732    }
733}
734
735void AudioFlinger::PlaybackThread::Track::flush()
736{
737    ALOGV("flush(%d)", mName);
738    sp<ThreadBase> thread = mThread.promote();
739    if (thread != 0) {
740        Mutex::Autolock _l(thread->mLock);
741        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
742
743        if (isOffloaded()) {
744            // If offloaded we allow flush during any state except terminated
745            // and keep the track active to avoid problems if user is seeking
746            // rapidly and underlying hardware has a significant delay handling
747            // a pause
748            if (isTerminated()) {
749                return;
750            }
751
752            ALOGV("flush: offload flush");
753            reset();
754
755            if (mState == STOPPING_1 || mState == STOPPING_2) {
756                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
757                mState = ACTIVE;
758            }
759
760            if (mState == ACTIVE) {
761                ALOGV("flush called in active state, resetting buffer time out retry count");
762                mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
763            }
764
765            mFlushHwPending = true;
766            mResumeToStopping = false;
767        } else {
768            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
769                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
770                return;
771            }
772            // No point remaining in PAUSED state after a flush => go to
773            // FLUSHED state
774            mState = FLUSHED;
775            // do not reset the track if it is still in the process of being stopped or paused.
776            // this will be done by prepareTracks_l() when the track is stopped.
777            // prepareTracks_l() will see mState == FLUSHED, then
778            // remove from active track list, reset(), and trigger presentation complete
779            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
780                reset();
781            }
782        }
783        // Prevent flush being lost if the track is flushed and then resumed
784        // before mixer thread can run. This is important when offloading
785        // because the hardware buffer could hold a large amount of audio
786        playbackThread->broadcast_l();
787    }
788}
789
790// must be called with thread lock held
791void AudioFlinger::PlaybackThread::Track::flushAck()
792{
793    if (!isOffloaded())
794        return;
795
796    mFlushHwPending = false;
797}
798
799void AudioFlinger::PlaybackThread::Track::reset()
800{
801    // Do not reset twice to avoid discarding data written just after a flush and before
802    // the audioflinger thread detects the track is stopped.
803    if (!mResetDone) {
804        // Force underrun condition to avoid false underrun callback until first data is
805        // written to buffer
806        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
807        mFillingUpStatus = FS_FILLING;
808        mResetDone = true;
809        if (mState == FLUSHED) {
810            mState = IDLE;
811        }
812    }
813}
814
815status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
816{
817    sp<ThreadBase> thread = mThread.promote();
818    if (thread == 0) {
819        ALOGE("thread is dead");
820        return FAILED_TRANSACTION;
821    } else if ((thread->type() == ThreadBase::DIRECT) ||
822                    (thread->type() == ThreadBase::OFFLOAD)) {
823        return thread->setParameters(keyValuePairs);
824    } else {
825        return PERMISSION_DENIED;
826    }
827}
828
829status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
830{
831    // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
832    if (isFastTrack()) {
833        return INVALID_OPERATION;
834    }
835    sp<ThreadBase> thread = mThread.promote();
836    if (thread == 0) {
837        return INVALID_OPERATION;
838    }
839    Mutex::Autolock _l(thread->mLock);
840    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
841    if (!isOffloaded()) {
842        if (!playbackThread->mLatchQValid) {
843            return INVALID_OPERATION;
844        }
845        uint32_t unpresentedFrames =
846                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
847                playbackThread->mSampleRate;
848        uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
849        if (framesWritten < unpresentedFrames) {
850            return INVALID_OPERATION;
851        }
852        timestamp.mPosition = framesWritten - unpresentedFrames;
853        timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
854        return NO_ERROR;
855    }
856
857    return playbackThread->getTimestamp_l(timestamp);
858}
859
860status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
861{
862    status_t status = DEAD_OBJECT;
863    sp<ThreadBase> thread = mThread.promote();
864    if (thread != 0) {
865        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
866        sp<AudioFlinger> af = mClient->audioFlinger();
867
868        Mutex::Autolock _l(af->mLock);
869
870        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
871
872        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
873            Mutex::Autolock _dl(playbackThread->mLock);
874            Mutex::Autolock _sl(srcThread->mLock);
875            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
876            if (chain == 0) {
877                return INVALID_OPERATION;
878            }
879
880            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
881            if (effect == 0) {
882                return INVALID_OPERATION;
883            }
884            srcThread->removeEffect_l(effect);
885            status = playbackThread->addEffect_l(effect);
886            if (status != NO_ERROR) {
887                srcThread->addEffect_l(effect);
888                return INVALID_OPERATION;
889            }
890            // removeEffect_l() has stopped the effect if it was active so it must be restarted
891            if (effect->state() == EffectModule::ACTIVE ||
892                    effect->state() == EffectModule::STOPPING) {
893                effect->start();
894            }
895
896            sp<EffectChain> dstChain = effect->chain().promote();
897            if (dstChain == 0) {
898                srcThread->addEffect_l(effect);
899                return INVALID_OPERATION;
900            }
901            AudioSystem::unregisterEffect(effect->id());
902            AudioSystem::registerEffect(&effect->desc(),
903                                        srcThread->id(),
904                                        dstChain->strategy(),
905                                        AUDIO_SESSION_OUTPUT_MIX,
906                                        effect->id());
907            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
908        }
909        status = playbackThread->attachAuxEffect(this, EffectId);
910    }
911    return status;
912}
913
914void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
915{
916    mAuxEffectId = EffectId;
917    mAuxBuffer = buffer;
918}
919
920bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
921                                                         size_t audioHalFrames)
922{
923    // a track is considered presented when the total number of frames written to audio HAL
924    // corresponds to the number of frames written when presentationComplete() is called for the
925    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
926    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
927    // to detect when all frames have been played. In this case framesWritten isn't
928    // useful because it doesn't always reflect whether there is data in the h/w
929    // buffers, particularly if a track has been paused and resumed during draining
930    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
931                      mPresentationCompleteFrames, framesWritten);
932    if (mPresentationCompleteFrames == 0) {
933        mPresentationCompleteFrames = framesWritten + audioHalFrames;
934        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
935                  mPresentationCompleteFrames, audioHalFrames);
936    }
937
938    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
939        ALOGV("presentationComplete() session %d complete: framesWritten %d",
940                  mSessionId, framesWritten);
941        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
942        mAudioTrackServerProxy->setStreamEndDone();
943        return true;
944    }
945    return false;
946}
947
948void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
949{
950    for (size_t i = 0; i < mSyncEvents.size(); i++) {
951        if (mSyncEvents[i]->type() == type) {
952            mSyncEvents[i]->trigger();
953            mSyncEvents.removeAt(i);
954            i--;
955        }
956    }
957}
958
959// implement VolumeBufferProvider interface
960
961uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
962{
963    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
964    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
965    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
966    uint32_t vl = vlr & 0xFFFF;
967    uint32_t vr = vlr >> 16;
968    // track volumes come from shared memory, so can't be trusted and must be clamped
969    if (vl > MAX_GAIN_INT) {
970        vl = MAX_GAIN_INT;
971    }
972    if (vr > MAX_GAIN_INT) {
973        vr = MAX_GAIN_INT;
974    }
975    // now apply the cached master volume and stream type volume;
976    // this is trusted but lacks any synchronization or barrier so may be stale
977    float v = mCachedVolume;
978    vl *= v;
979    vr *= v;
980    // re-combine into U4.16
981    vlr = (vr << 16) | (vl & 0xFFFF);
982    // FIXME look at mute, pause, and stop flags
983    return vlr;
984}
985
986status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
987{
988    if (isTerminated() || mState == PAUSED ||
989            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
990                                      (mState == STOPPED)))) {
991        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
992              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
993        event->cancel();
994        return INVALID_OPERATION;
995    }
996    (void) TrackBase::setSyncEvent(event);
997    return NO_ERROR;
998}
999
1000void AudioFlinger::PlaybackThread::Track::invalidate()
1001{
1002    // FIXME should use proxy, and needs work
1003    audio_track_cblk_t* cblk = mCblk;
1004    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1005    android_atomic_release_store(0x40000000, &cblk->mFutex);
1006    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1007    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1008    mIsInvalid = true;
1009}
1010
1011void AudioFlinger::PlaybackThread::Track::signal()
1012{
1013    sp<ThreadBase> thread = mThread.promote();
1014    if (thread != 0) {
1015        PlaybackThread *t = (PlaybackThread *)thread.get();
1016        Mutex::Autolock _l(t->mLock);
1017        t->broadcast_l();
1018    }
1019}
1020
1021//To be called with thread lock held
1022bool AudioFlinger::PlaybackThread::Track::isResumePending() {
1023
1024    if (mState == RESUMING)
1025        return true;
1026    /* Resume is pending if track was stopping before pause was called */
1027    if (mState == STOPPING_1 &&
1028        mResumeToStopping)
1029        return true;
1030
1031    return false;
1032}
1033
1034//To be called with thread lock held
1035void AudioFlinger::PlaybackThread::Track::resumeAck() {
1036
1037
1038    if (mState == RESUMING)
1039        mState = ACTIVE;
1040
1041    // Other possibility of  pending resume is stopping_1 state
1042    // Do not update the state from stopping as this prevents
1043    // drain being called.
1044    if (mState == STOPPING_1) {
1045        mResumeToStopping = false;
1046    }
1047}
1048// ----------------------------------------------------------------------------
1049
1050sp<AudioFlinger::PlaybackThread::TimedTrack>
1051AudioFlinger::PlaybackThread::TimedTrack::create(
1052            PlaybackThread *thread,
1053            const sp<Client>& client,
1054            audio_stream_type_t streamType,
1055            uint32_t sampleRate,
1056            audio_format_t format,
1057            audio_channel_mask_t channelMask,
1058            size_t frameCount,
1059            const sp<IMemory>& sharedBuffer,
1060            int sessionId,
1061            int uid)
1062{
1063    if (!client->reserveTimedTrack())
1064        return 0;
1065
1066    return new TimedTrack(
1067        thread, client, streamType, sampleRate, format, channelMask, frameCount,
1068        sharedBuffer, sessionId, uid);
1069}
1070
1071AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
1072            PlaybackThread *thread,
1073            const sp<Client>& client,
1074            audio_stream_type_t streamType,
1075            uint32_t sampleRate,
1076            audio_format_t format,
1077            audio_channel_mask_t channelMask,
1078            size_t frameCount,
1079            const sp<IMemory>& sharedBuffer,
1080            int sessionId,
1081            int uid)
1082    : Track(thread, client, streamType, sampleRate, format, channelMask,
1083            frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
1084      mQueueHeadInFlight(false),
1085      mTrimQueueHeadOnRelease(false),
1086      mFramesPendingInQueue(0),
1087      mTimedSilenceBuffer(NULL),
1088      mTimedSilenceBufferSize(0),
1089      mTimedAudioOutputOnTime(false),
1090      mMediaTimeTransformValid(false)
1091{
1092    LocalClock lc;
1093    mLocalTimeFreq = lc.getLocalFreq();
1094
1095    mLocalTimeToSampleTransform.a_zero = 0;
1096    mLocalTimeToSampleTransform.b_zero = 0;
1097    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1098    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1099    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1100                            &mLocalTimeToSampleTransform.a_to_b_denom);
1101
1102    mMediaTimeToSampleTransform.a_zero = 0;
1103    mMediaTimeToSampleTransform.b_zero = 0;
1104    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1105    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1106    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1107                            &mMediaTimeToSampleTransform.a_to_b_denom);
1108}
1109
1110AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1111    mClient->releaseTimedTrack();
1112    delete [] mTimedSilenceBuffer;
1113}
1114
1115status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1116    size_t size, sp<IMemory>* buffer) {
1117
1118    Mutex::Autolock _l(mTimedBufferQueueLock);
1119
1120    trimTimedBufferQueue_l();
1121
1122    // lazily initialize the shared memory heap for timed buffers
1123    if (mTimedMemoryDealer == NULL) {
1124        const int kTimedBufferHeapSize = 512 << 10;
1125
1126        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1127                                              "AudioFlingerTimed");
1128        if (mTimedMemoryDealer == NULL) {
1129            return NO_MEMORY;
1130        }
1131    }
1132
1133    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1134    if (newBuffer == 0 || newBuffer->pointer() == NULL) {
1135        return NO_MEMORY;
1136    }
1137
1138    *buffer = newBuffer;
1139    return NO_ERROR;
1140}
1141
1142// caller must hold mTimedBufferQueueLock
1143void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1144    int64_t mediaTimeNow;
1145    {
1146        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1147        if (!mMediaTimeTransformValid)
1148            return;
1149
1150        int64_t targetTimeNow;
1151        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1152            ? mCCHelper.getCommonTime(&targetTimeNow)
1153            : mCCHelper.getLocalTime(&targetTimeNow);
1154
1155        if (OK != res)
1156            return;
1157
1158        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1159                                                    &mediaTimeNow)) {
1160            return;
1161        }
1162    }
1163
1164    size_t trimEnd;
1165    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1166        int64_t bufEnd;
1167
1168        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1169            // We have a next buffer.  Just use its PTS as the PTS of the frame
1170            // following the last frame in this buffer.  If the stream is sparse
1171            // (ie, there are deliberate gaps left in the stream which should be
1172            // filled with silence by the TimedAudioTrack), then this can result
1173            // in one extra buffer being left un-trimmed when it could have
1174            // been.  In general, this is not typical, and we would rather
1175            // optimized away the TS calculation below for the more common case
1176            // where PTSes are contiguous.
1177            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1178        } else {
1179            // We have no next buffer.  Compute the PTS of the frame following
1180            // the last frame in this buffer by computing the duration of of
1181            // this frame in media time units and adding it to the PTS of the
1182            // buffer.
1183            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1184                               / mFrameSize;
1185
1186            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1187                                                                &bufEnd)) {
1188                ALOGE("Failed to convert frame count of %lld to media time"
1189                      " duration" " (scale factor %d/%u) in %s",
1190                      frameCount,
1191                      mMediaTimeToSampleTransform.a_to_b_numer,
1192                      mMediaTimeToSampleTransform.a_to_b_denom,
1193                      __PRETTY_FUNCTION__);
1194                break;
1195            }
1196            bufEnd += mTimedBufferQueue[trimEnd].pts();
1197        }
1198
1199        if (bufEnd > mediaTimeNow)
1200            break;
1201
1202        // Is the buffer we want to use in the middle of a mix operation right
1203        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1204        // from the mixer which should be coming back shortly.
1205        if (!trimEnd && mQueueHeadInFlight) {
1206            mTrimQueueHeadOnRelease = true;
1207        }
1208    }
1209
1210    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1211    if (trimStart < trimEnd) {
1212        // Update the bookkeeping for framesReady()
1213        for (size_t i = trimStart; i < trimEnd; ++i) {
1214            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1215        }
1216
1217        // Now actually remove the buffers from the queue.
1218        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1219    }
1220}
1221
1222void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1223        const char* logTag) {
1224    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1225                "%s called (reason \"%s\"), but timed buffer queue has no"
1226                " elements to trim.", __FUNCTION__, logTag);
1227
1228    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1229    mTimedBufferQueue.removeAt(0);
1230}
1231
1232void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1233        const TimedBuffer& buf,
1234        const char* logTag __unused) {
1235    uint32_t bufBytes        = buf.buffer()->size();
1236    uint32_t consumedAlready = buf.position();
1237
1238    ALOG_ASSERT(consumedAlready <= bufBytes,
1239                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1240                " only %u bytes long, but claims to have consumed %u"
1241                " bytes.  (update reason: \"%s\")",
1242                bufBytes, consumedAlready, logTag);
1243
1244    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1245    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1246                "Bad bookkeeping while updating frames pending.  Should have at"
1247                " least %u queued frames, but we think we have only %u.  (update"
1248                " reason: \"%s\")",
1249                bufFrames, mFramesPendingInQueue, logTag);
1250
1251    mFramesPendingInQueue -= bufFrames;
1252}
1253
1254status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1255    const sp<IMemory>& buffer, int64_t pts) {
1256
1257    {
1258        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1259        if (!mMediaTimeTransformValid)
1260            return INVALID_OPERATION;
1261    }
1262
1263    Mutex::Autolock _l(mTimedBufferQueueLock);
1264
1265    uint32_t bufFrames = buffer->size() / mFrameSize;
1266    mFramesPendingInQueue += bufFrames;
1267    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1268
1269    return NO_ERROR;
1270}
1271
1272status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1273    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1274
1275    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1276           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1277           target);
1278
1279    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1280          target == TimedAudioTrack::COMMON_TIME)) {
1281        return BAD_VALUE;
1282    }
1283
1284    Mutex::Autolock lock(mMediaTimeTransformLock);
1285    mMediaTimeTransform = xform;
1286    mMediaTimeTransformTarget = target;
1287    mMediaTimeTransformValid = true;
1288
1289    return NO_ERROR;
1290}
1291
1292#define min(a, b) ((a) < (b) ? (a) : (b))
1293
1294// implementation of getNextBuffer for tracks whose buffers have timestamps
1295status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1296    AudioBufferProvider::Buffer* buffer, int64_t pts)
1297{
1298    if (pts == AudioBufferProvider::kInvalidPTS) {
1299        buffer->raw = NULL;
1300        buffer->frameCount = 0;
1301        mTimedAudioOutputOnTime = false;
1302        return INVALID_OPERATION;
1303    }
1304
1305    Mutex::Autolock _l(mTimedBufferQueueLock);
1306
1307    ALOG_ASSERT(!mQueueHeadInFlight,
1308                "getNextBuffer called without releaseBuffer!");
1309
1310    while (true) {
1311
1312        // if we have no timed buffers, then fail
1313        if (mTimedBufferQueue.isEmpty()) {
1314            buffer->raw = NULL;
1315            buffer->frameCount = 0;
1316            return NOT_ENOUGH_DATA;
1317        }
1318
1319        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1320
1321        // calculate the PTS of the head of the timed buffer queue expressed in
1322        // local time
1323        int64_t headLocalPTS;
1324        {
1325            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1326
1327            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1328
1329            if (mMediaTimeTransform.a_to_b_denom == 0) {
1330                // the transform represents a pause, so yield silence
1331                timedYieldSilence_l(buffer->frameCount, buffer);
1332                return NO_ERROR;
1333            }
1334
1335            int64_t transformedPTS;
1336            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1337                                                        &transformedPTS)) {
1338                // the transform failed.  this shouldn't happen, but if it does
1339                // then just drop this buffer
1340                ALOGW("timedGetNextBuffer transform failed");
1341                buffer->raw = NULL;
1342                buffer->frameCount = 0;
1343                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1344                return NO_ERROR;
1345            }
1346
1347            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1348                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1349                                                          &headLocalPTS)) {
1350                    buffer->raw = NULL;
1351                    buffer->frameCount = 0;
1352                    return INVALID_OPERATION;
1353                }
1354            } else {
1355                headLocalPTS = transformedPTS;
1356            }
1357        }
1358
1359        uint32_t sr = sampleRate();
1360
1361        // adjust the head buffer's PTS to reflect the portion of the head buffer
1362        // that has already been consumed
1363        int64_t effectivePTS = headLocalPTS +
1364                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1365
1366        // Calculate the delta in samples between the head of the input buffer
1367        // queue and the start of the next output buffer that will be written.
1368        // If the transformation fails because of over or underflow, it means
1369        // that the sample's position in the output stream is so far out of
1370        // whack that it should just be dropped.
1371        int64_t sampleDelta;
1372        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1373            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1374            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1375                                       " mix");
1376            continue;
1377        }
1378        if (!mLocalTimeToSampleTransform.doForwardTransform(
1379                (effectivePTS - pts) << 32, &sampleDelta)) {
1380            ALOGV("*** too late during sample rate transform: dropped buffer");
1381            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1382            continue;
1383        }
1384
1385        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1386               " sampleDelta=[%d.%08x]",
1387               head.pts(), head.position(), pts,
1388               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1389                   + (sampleDelta >> 32)),
1390               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1391
1392        // if the delta between the ideal placement for the next input sample and
1393        // the current output position is within this threshold, then we will
1394        // concatenate the next input samples to the previous output
1395        const int64_t kSampleContinuityThreshold =
1396                (static_cast<int64_t>(sr) << 32) / 250;
1397
1398        // if this is the first buffer of audio that we're emitting from this track
1399        // then it should be almost exactly on time.
1400        const int64_t kSampleStartupThreshold = 1LL << 32;
1401
1402        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1403           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1404            // the next input is close enough to being on time, so concatenate it
1405            // with the last output
1406            timedYieldSamples_l(buffer);
1407
1408            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1409                    head.position(), buffer->frameCount);
1410            return NO_ERROR;
1411        }
1412
1413        // Looks like our output is not on time.  Reset our on timed status.
1414        // Next time we mix samples from our input queue, then should be within
1415        // the StartupThreshold.
1416        mTimedAudioOutputOnTime = false;
1417        if (sampleDelta > 0) {
1418            // the gap between the current output position and the proper start of
1419            // the next input sample is too big, so fill it with silence
1420            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1421
1422            timedYieldSilence_l(framesUntilNextInput, buffer);
1423            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1424            return NO_ERROR;
1425        } else {
1426            // the next input sample is late
1427            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1428            size_t onTimeSamplePosition =
1429                    head.position() + lateFrames * mFrameSize;
1430
1431            if (onTimeSamplePosition > head.buffer()->size()) {
1432                // all the remaining samples in the head are too late, so
1433                // drop it and move on
1434                ALOGV("*** too late: dropped buffer");
1435                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1436                continue;
1437            } else {
1438                // skip over the late samples
1439                head.setPosition(onTimeSamplePosition);
1440
1441                // yield the available samples
1442                timedYieldSamples_l(buffer);
1443
1444                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1445                return NO_ERROR;
1446            }
1447        }
1448    }
1449}
1450
1451// Yield samples from the timed buffer queue head up to the given output
1452// buffer's capacity.
1453//
1454// Caller must hold mTimedBufferQueueLock
1455void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1456    AudioBufferProvider::Buffer* buffer) {
1457
1458    const TimedBuffer& head = mTimedBufferQueue[0];
1459
1460    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1461                   head.position());
1462
1463    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1464                                 mFrameSize);
1465    size_t framesRequested = buffer->frameCount;
1466    buffer->frameCount = min(framesLeftInHead, framesRequested);
1467
1468    mQueueHeadInFlight = true;
1469    mTimedAudioOutputOnTime = true;
1470}
1471
1472// Yield samples of silence up to the given output buffer's capacity
1473//
1474// Caller must hold mTimedBufferQueueLock
1475void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1476    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1477
1478    // lazily allocate a buffer filled with silence
1479    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1480        delete [] mTimedSilenceBuffer;
1481        mTimedSilenceBufferSize = numFrames * mFrameSize;
1482        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1483        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1484    }
1485
1486    buffer->raw = mTimedSilenceBuffer;
1487    size_t framesRequested = buffer->frameCount;
1488    buffer->frameCount = min(numFrames, framesRequested);
1489
1490    mTimedAudioOutputOnTime = false;
1491}
1492
1493// AudioBufferProvider interface
1494void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1495    AudioBufferProvider::Buffer* buffer) {
1496
1497    Mutex::Autolock _l(mTimedBufferQueueLock);
1498
1499    // If the buffer which was just released is part of the buffer at the head
1500    // of the queue, be sure to update the amt of the buffer which has been
1501    // consumed.  If the buffer being returned is not part of the head of the
1502    // queue, its either because the buffer is part of the silence buffer, or
1503    // because the head of the timed queue was trimmed after the mixer called
1504    // getNextBuffer but before the mixer called releaseBuffer.
1505    if (buffer->raw == mTimedSilenceBuffer) {
1506        ALOG_ASSERT(!mQueueHeadInFlight,
1507                    "Queue head in flight during release of silence buffer!");
1508        goto done;
1509    }
1510
1511    ALOG_ASSERT(mQueueHeadInFlight,
1512                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1513                " head in flight.");
1514
1515    if (mTimedBufferQueue.size()) {
1516        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1517
1518        void* start = head.buffer()->pointer();
1519        void* end   = reinterpret_cast<void*>(
1520                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1521                        + head.buffer()->size());
1522
1523        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1524                    "released buffer not within the head of the timed buffer"
1525                    " queue; qHead = [%p, %p], released buffer = %p",
1526                    start, end, buffer->raw);
1527
1528        head.setPosition(head.position() +
1529                (buffer->frameCount * mFrameSize));
1530        mQueueHeadInFlight = false;
1531
1532        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1533                    "Bad bookkeeping during releaseBuffer!  Should have at"
1534                    " least %u queued frames, but we think we have only %u",
1535                    buffer->frameCount, mFramesPendingInQueue);
1536
1537        mFramesPendingInQueue -= buffer->frameCount;
1538
1539        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1540            || mTrimQueueHeadOnRelease) {
1541            trimTimedBufferQueueHead_l("releaseBuffer");
1542            mTrimQueueHeadOnRelease = false;
1543        }
1544    } else {
1545        LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1546                  " buffers in the timed buffer queue");
1547    }
1548
1549done:
1550    buffer->raw = 0;
1551    buffer->frameCount = 0;
1552}
1553
1554size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1555    Mutex::Autolock _l(mTimedBufferQueueLock);
1556    return mFramesPendingInQueue;
1557}
1558
1559AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1560        : mPTS(0), mPosition(0) {}
1561
1562AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1563    const sp<IMemory>& buffer, int64_t pts)
1564        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1565
1566
1567// ----------------------------------------------------------------------------
1568
1569AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1570            PlaybackThread *playbackThread,
1571            DuplicatingThread *sourceThread,
1572            uint32_t sampleRate,
1573            audio_format_t format,
1574            audio_channel_mask_t channelMask,
1575            size_t frameCount,
1576            int uid)
1577    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1578                NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
1579    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1580{
1581
1582    if (mCblk != NULL) {
1583        mOutBuffer.frameCount = 0;
1584        playbackThread->mTracks.add(this);
1585        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1586                "frameCount %u, mChannelMask 0x%08x",
1587                mCblk, mBuffer,
1588                frameCount, mChannelMask);
1589        // since client and server are in the same process,
1590        // the buffer has the same virtual address on both sides
1591        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1592        mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
1593        mClientProxy->setSendLevel(0.0);
1594        mClientProxy->setSampleRate(sampleRate);
1595        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1596                true /*clientInServer*/);
1597    } else {
1598        ALOGW("Error creating output track on thread %p", playbackThread);
1599    }
1600}
1601
1602AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1603{
1604    clearBufferQueue();
1605    delete mClientProxy;
1606    // superclass destructor will now delete the server proxy and shared memory both refer to
1607}
1608
1609status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1610                                                          int triggerSession)
1611{
1612    status_t status = Track::start(event, triggerSession);
1613    if (status != NO_ERROR) {
1614        return status;
1615    }
1616
1617    mActive = true;
1618    mRetryCount = 127;
1619    return status;
1620}
1621
1622void AudioFlinger::PlaybackThread::OutputTrack::stop()
1623{
1624    Track::stop();
1625    clearBufferQueue();
1626    mOutBuffer.frameCount = 0;
1627    mActive = false;
1628}
1629
1630bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1631{
1632    Buffer *pInBuffer;
1633    Buffer inBuffer;
1634    uint32_t channelCount = mChannelCount;
1635    bool outputBufferFull = false;
1636    inBuffer.frameCount = frames;
1637    inBuffer.i16 = data;
1638
1639    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1640
1641    if (!mActive && frames != 0) {
1642        start();
1643        sp<ThreadBase> thread = mThread.promote();
1644        if (thread != 0) {
1645            MixerThread *mixerThread = (MixerThread *)thread.get();
1646            if (mFrameCount > frames) {
1647                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1648                    uint32_t startFrames = (mFrameCount - frames);
1649                    pInBuffer = new Buffer;
1650                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1651                    pInBuffer->frameCount = startFrames;
1652                    pInBuffer->i16 = pInBuffer->mBuffer;
1653                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1654                    mBufferQueue.add(pInBuffer);
1655                } else {
1656                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1657                }
1658            }
1659        }
1660    }
1661
1662    while (waitTimeLeftMs) {
1663        // First write pending buffers, then new data
1664        if (mBufferQueue.size()) {
1665            pInBuffer = mBufferQueue.itemAt(0);
1666        } else {
1667            pInBuffer = &inBuffer;
1668        }
1669
1670        if (pInBuffer->frameCount == 0) {
1671            break;
1672        }
1673
1674        if (mOutBuffer.frameCount == 0) {
1675            mOutBuffer.frameCount = pInBuffer->frameCount;
1676            nsecs_t startTime = systemTime();
1677            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1678            if (status != NO_ERROR) {
1679                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1680                        mThread.unsafe_get(), status);
1681                outputBufferFull = true;
1682                break;
1683            }
1684            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1685            if (waitTimeLeftMs >= waitTimeMs) {
1686                waitTimeLeftMs -= waitTimeMs;
1687            } else {
1688                waitTimeLeftMs = 0;
1689            }
1690        }
1691
1692        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1693                pInBuffer->frameCount;
1694        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1695        Proxy::Buffer buf;
1696        buf.mFrameCount = outFrames;
1697        buf.mRaw = NULL;
1698        mClientProxy->releaseBuffer(&buf);
1699        pInBuffer->frameCount -= outFrames;
1700        pInBuffer->i16 += outFrames * channelCount;
1701        mOutBuffer.frameCount -= outFrames;
1702        mOutBuffer.i16 += outFrames * channelCount;
1703
1704        if (pInBuffer->frameCount == 0) {
1705            if (mBufferQueue.size()) {
1706                mBufferQueue.removeAt(0);
1707                delete [] pInBuffer->mBuffer;
1708                delete pInBuffer;
1709                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1710                        mThread.unsafe_get(), mBufferQueue.size());
1711            } else {
1712                break;
1713            }
1714        }
1715    }
1716
1717    // If we could not write all frames, allocate a buffer and queue it for next time.
1718    if (inBuffer.frameCount) {
1719        sp<ThreadBase> thread = mThread.promote();
1720        if (thread != 0 && !thread->standby()) {
1721            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1722                pInBuffer = new Buffer;
1723                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1724                pInBuffer->frameCount = inBuffer.frameCount;
1725                pInBuffer->i16 = pInBuffer->mBuffer;
1726                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1727                        sizeof(int16_t));
1728                mBufferQueue.add(pInBuffer);
1729                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1730                        mThread.unsafe_get(), mBufferQueue.size());
1731            } else {
1732                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1733                        mThread.unsafe_get(), this);
1734            }
1735        }
1736    }
1737
1738    // Calling write() with a 0 length buffer, means that no more data will be written:
1739    // If no more buffers are pending, fill output track buffer to make sure it is started
1740    // by output mixer.
1741    if (frames == 0 && mBufferQueue.size() == 0) {
1742        // FIXME borken, replace by getting framesReady() from proxy
1743        size_t user = 0;    // was mCblk->user
1744        if (user < mFrameCount) {
1745            frames = mFrameCount - user;
1746            pInBuffer = new Buffer;
1747            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1748            pInBuffer->frameCount = frames;
1749            pInBuffer->i16 = pInBuffer->mBuffer;
1750            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1751            mBufferQueue.add(pInBuffer);
1752        } else if (mActive) {
1753            stop();
1754        }
1755    }
1756
1757    return outputBufferFull;
1758}
1759
1760status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1761        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1762{
1763    ClientProxy::Buffer buf;
1764    buf.mFrameCount = buffer->frameCount;
1765    struct timespec timeout;
1766    timeout.tv_sec = waitTimeMs / 1000;
1767    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1768    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1769    buffer->frameCount = buf.mFrameCount;
1770    buffer->raw = buf.mRaw;
1771    return status;
1772}
1773
1774void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1775{
1776    size_t size = mBufferQueue.size();
1777
1778    for (size_t i = 0; i < size; i++) {
1779        Buffer *pBuffer = mBufferQueue.itemAt(i);
1780        delete [] pBuffer->mBuffer;
1781        delete pBuffer;
1782    }
1783    mBufferQueue.clear();
1784}
1785
1786
1787// ----------------------------------------------------------------------------
1788//      Record
1789// ----------------------------------------------------------------------------
1790
1791AudioFlinger::RecordHandle::RecordHandle(
1792        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1793    : BnAudioRecord(),
1794    mRecordTrack(recordTrack)
1795{
1796}
1797
1798AudioFlinger::RecordHandle::~RecordHandle() {
1799    stop_nonvirtual();
1800    mRecordTrack->destroy();
1801}
1802
1803status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1804        int triggerSession) {
1805    ALOGV("RecordHandle::start()");
1806    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1807}
1808
1809void AudioFlinger::RecordHandle::stop() {
1810    stop_nonvirtual();
1811}
1812
1813void AudioFlinger::RecordHandle::stop_nonvirtual() {
1814    ALOGV("RecordHandle::stop()");
1815    mRecordTrack->stop();
1816}
1817
1818status_t AudioFlinger::RecordHandle::onTransact(
1819    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1820{
1821    return BnAudioRecord::onTransact(code, data, reply, flags);
1822}
1823
1824// ----------------------------------------------------------------------------
1825
1826// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
1827AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1828            RecordThread *thread,
1829            const sp<Client>& client,
1830            uint32_t sampleRate,
1831            audio_format_t format,
1832            audio_channel_mask_t channelMask,
1833            size_t frameCount,
1834            int sessionId,
1835            int uid,
1836            bool isFast)
1837    :   TrackBase(thread, client, sampleRate, format,
1838                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/,
1839                  isFast /*useReadOnlyHeap*/),
1840        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
1841        // See real initialization of mRsmpInFront at RecordThread::start()
1842        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
1843{
1844    if (mCblk == NULL) {
1845        return;
1846    }
1847
1848    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
1849
1850    uint32_t channelCount = popcount(channelMask);
1851    // FIXME I don't understand either of the channel count checks
1852    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
1853            channelCount <= FCC_2) {
1854        // sink SR
1855        mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
1856        // source SR
1857        mResampler->setSampleRate(thread->mSampleRate);
1858        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
1859        mResamplerBufferProvider = new ResamplerBufferProvider(this);
1860    }
1861}
1862
1863AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1864{
1865    ALOGV("%s", __func__);
1866    delete mResampler;
1867    delete[] mRsmpOutBuffer;
1868    delete mResamplerBufferProvider;
1869}
1870
1871// AudioBufferProvider interface
1872status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1873        int64_t pts __unused)
1874{
1875    ServerProxy::Buffer buf;
1876    buf.mFrameCount = buffer->frameCount;
1877    status_t status = mServerProxy->obtainBuffer(&buf);
1878    buffer->frameCount = buf.mFrameCount;
1879    buffer->raw = buf.mRaw;
1880    if (buf.mFrameCount == 0) {
1881        // FIXME also wake futex so that overrun is noticed more quickly
1882        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1883    }
1884    return status;
1885}
1886
1887status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1888                                                        int triggerSession)
1889{
1890    sp<ThreadBase> thread = mThread.promote();
1891    if (thread != 0) {
1892        RecordThread *recordThread = (RecordThread *)thread.get();
1893        return recordThread->start(this, event, triggerSession);
1894    } else {
1895        return BAD_VALUE;
1896    }
1897}
1898
1899void AudioFlinger::RecordThread::RecordTrack::stop()
1900{
1901    sp<ThreadBase> thread = mThread.promote();
1902    if (thread != 0) {
1903        RecordThread *recordThread = (RecordThread *)thread.get();
1904        if (recordThread->stop(this)) {
1905            AudioSystem::stopInput(recordThread->id());
1906        }
1907    }
1908}
1909
1910void AudioFlinger::RecordThread::RecordTrack::destroy()
1911{
1912    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1913    sp<RecordTrack> keep(this);
1914    {
1915        sp<ThreadBase> thread = mThread.promote();
1916        if (thread != 0) {
1917            if (mState == ACTIVE || mState == RESUMING) {
1918                AudioSystem::stopInput(thread->id());
1919            }
1920            AudioSystem::releaseInput(thread->id());
1921            Mutex::Autolock _l(thread->mLock);
1922            RecordThread *recordThread = (RecordThread *) thread.get();
1923            recordThread->destroyTrack_l(this);
1924        }
1925    }
1926}
1927
1928void AudioFlinger::RecordThread::RecordTrack::invalidate()
1929{
1930    // FIXME should use proxy, and needs work
1931    audio_track_cblk_t* cblk = mCblk;
1932    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1933    android_atomic_release_store(0x40000000, &cblk->mFutex);
1934    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1935    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1936}
1937
1938
1939/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1940{
1941    result.append("    Active Client Fmt Chn mask Session S   Server fCount Resampling\n");
1942}
1943
1944void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
1945{
1946    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n",
1947            active ? "yes" : "no",
1948            (mClient == 0) ? getpid_cached : mClient->pid(),
1949            mFormat,
1950            mChannelMask,
1951            mSessionId,
1952            mState,
1953            mCblk->mServer,
1954            mFrameCount,
1955            mResampler != NULL);
1956
1957}
1958
1959void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
1960{
1961    if (event == mSyncStartEvent) {
1962        ssize_t framesToDrop = 0;
1963        sp<ThreadBase> threadBase = mThread.promote();
1964        if (threadBase != 0) {
1965            // TODO: use actual buffer filling status instead of 2 buffers when info is available
1966            // from audio HAL
1967            framesToDrop = threadBase->mFrameCount * 2;
1968        }
1969        mFramesToDrop = framesToDrop;
1970    }
1971}
1972
1973void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
1974{
1975    if (mSyncStartEvent != 0) {
1976        mSyncStartEvent->cancel();
1977        mSyncStartEvent.clear();
1978    }
1979    mFramesToDrop = 0;
1980}
1981
1982}; // namespace android
1983