Tracks.cpp revision e541269be94f3a1072932d51537905b120ef4733 
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(isOut ?
85                audio_channel_count_from_out_mask(channelMask) :
86                audio_channel_count_from_in_mask(channelMask)),
87        mFrameSize(audio_is_linear_pcm(format) ?
88                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
89        mFrameCount(frameCount),
90        mSessionId(sessionId),
91        mFlags(flags),
92        mIsOut(isOut),
93        mServerProxy(NULL),
94        mId(android_atomic_inc(&nextTrackId)),
95        mTerminated(false)
96{
97    // if the caller is us, trust the specified uid
98    if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
99        int newclientUid = IPCThreadState::self()->getCallingUid();
100        if (clientUid != -1 && clientUid != newclientUid) {
101            ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
102        }
103        clientUid = newclientUid;
104    }
105    // clientUid contains the uid of the app that is responsible for this track, so we can blame
106    // battery usage on it.
107    mUid = clientUid;
108
109    // client == 0 implies sharedBuffer == 0
110    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
111
112    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
113            sharedBuffer->size());
114
115    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
116    size_t size = sizeof(audio_track_cblk_t);
117    size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
118    if (sharedBuffer == 0 && !useReadOnlyHeap) {
119        size += bufferSize;
120    }
121
122    if (client != 0) {
123        mCblkMemory = client->heap()->allocate(size);
124        if (mCblkMemory == 0 ||
125                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
126            ALOGE("not enough memory for AudioTrack size=%u", size);
127            client->heap()->dump("AudioTrack");
128            mCblkMemory.clear();
129            return;
130        }
131    } else {
132        // this syntax avoids calling the audio_track_cblk_t constructor twice
133        mCblk = (audio_track_cblk_t *) new uint8_t[size];
134        // assume mCblk != NULL
135    }
136
137    // construct the shared structure in-place.
138    if (mCblk != NULL) {
139        new(mCblk) audio_track_cblk_t();
140        if (useReadOnlyHeap) {
141            const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
142            if (roHeap == 0 ||
143                    (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
144                    (mBuffer = mBufferMemory->pointer()) == NULL) {
145                ALOGE("not enough memory for read-only buffer size=%zu", bufferSize);
146                if (roHeap != 0) {
147                    roHeap->dump("buffer");
148                }
149                mCblkMemory.clear();
150                mBufferMemory.clear();
151                return;
152            }
153            memset(mBuffer, 0, bufferSize);
154        } else {
155            // clear all buffers
156            if (sharedBuffer == 0) {
157                mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
158                memset(mBuffer, 0, bufferSize);
159            } else {
160                mBuffer = sharedBuffer->pointer();
161#if 0
162                mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
163#endif
164            }
165        }
166
167#ifdef TEE_SINK
168        if (mTeeSinkTrackEnabled) {
169            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
170            if (Format_isValid(pipeFormat)) {
171                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
172                size_t numCounterOffers = 0;
173                const NBAIO_Format offers[1] = {pipeFormat};
174                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
175                ALOG_ASSERT(index == 0);
176                PipeReader *pipeReader = new PipeReader(*pipe);
177                numCounterOffers = 0;
178                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
179                ALOG_ASSERT(index == 0);
180                mTeeSink = pipe;
181                mTeeSource = pipeReader;
182            }
183        }
184#endif
185
186    }
187}
188
189AudioFlinger::ThreadBase::TrackBase::~TrackBase()
190{
191#ifdef TEE_SINK
192    dumpTee(-1, mTeeSource, mId);
193#endif
194    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
195    delete mServerProxy;
196    if (mCblk != NULL) {
197        if (mClient == 0) {
198            delete mCblk;
199        } else {
200            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
201        }
202    }
203    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
204    if (mClient != 0) {
205        // Client destructor must run with AudioFlinger client mutex locked
206        Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
207        // If the client's reference count drops to zero, the associated destructor
208        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
209        // relying on the automatic clear() at end of scope.
210        mClient.clear();
211    }
212}
213
214// AudioBufferProvider interface
215// getNextBuffer() = 0;
216// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
217void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
218{
219#ifdef TEE_SINK
220    if (mTeeSink != 0) {
221        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
222    }
223#endif
224
225    ServerProxy::Buffer buf;
226    buf.mFrameCount = buffer->frameCount;
227    buf.mRaw = buffer->raw;
228    buffer->frameCount = 0;
229    buffer->raw = NULL;
230    mServerProxy->releaseBuffer(&buf);
231}
232
233status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
234{
235    mSyncEvents.add(event);
236    return NO_ERROR;
237}
238
239// ----------------------------------------------------------------------------
240//      Playback
241// ----------------------------------------------------------------------------
242
243AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
244    : BnAudioTrack(),
245      mTrack(track)
246{
247}
248
249AudioFlinger::TrackHandle::~TrackHandle() {
250    // just stop the track on deletion, associated resources
251    // will be freed from the main thread once all pending buffers have
252    // been played. Unless it's not in the active track list, in which
253    // case we free everything now...
254    mTrack->destroy();
255}
256
257sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
258    return mTrack->getCblk();
259}
260
261status_t AudioFlinger::TrackHandle::start() {
262    return mTrack->start();
263}
264
265void AudioFlinger::TrackHandle::stop() {
266    mTrack->stop();
267}
268
269void AudioFlinger::TrackHandle::flush() {
270    mTrack->flush();
271}
272
273void AudioFlinger::TrackHandle::pause() {
274    mTrack->pause();
275}
276
277status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
278{
279    return mTrack->attachAuxEffect(EffectId);
280}
281
282status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
283                                                         sp<IMemory>* buffer) {
284    if (!mTrack->isTimedTrack())
285        return INVALID_OPERATION;
286
287    PlaybackThread::TimedTrack* tt =
288            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
289    return tt->allocateTimedBuffer(size, buffer);
290}
291
292status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
293                                                     int64_t pts) {
294    if (!mTrack->isTimedTrack())
295        return INVALID_OPERATION;
296
297    if (buffer == 0 || buffer->pointer() == NULL) {
298        ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
299        return BAD_VALUE;
300    }
301
302    PlaybackThread::TimedTrack* tt =
303            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
304    return tt->queueTimedBuffer(buffer, pts);
305}
306
307status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
308    const LinearTransform& xform, int target) {
309
310    if (!mTrack->isTimedTrack())
311        return INVALID_OPERATION;
312
313    PlaybackThread::TimedTrack* tt =
314            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
315    return tt->setMediaTimeTransform(
316        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
317}
318
319status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
320    return mTrack->setParameters(keyValuePairs);
321}
322
323status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
324{
325    return mTrack->getTimestamp(timestamp);
326}
327
328
329void AudioFlinger::TrackHandle::signal()
330{
331    return mTrack->signal();
332}
333
334status_t AudioFlinger::TrackHandle::onTransact(
335    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
336{
337    return BnAudioTrack::onTransact(code, data, reply, flags);
338}
339
340// ----------------------------------------------------------------------------
341
342// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
343AudioFlinger::PlaybackThread::Track::Track(
344            PlaybackThread *thread,
345            const sp<Client>& client,
346            audio_stream_type_t streamType,
347            uint32_t sampleRate,
348            audio_format_t format,
349            audio_channel_mask_t channelMask,
350            size_t frameCount,
351            const sp<IMemory>& sharedBuffer,
352            int sessionId,
353            int uid,
354            IAudioFlinger::track_flags_t flags)
355    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
356            sessionId, uid, flags, true /*isOut*/),
357    mFillingUpStatus(FS_INVALID),
358    // mRetryCount initialized later when needed
359    mSharedBuffer(sharedBuffer),
360    mStreamType(streamType),
361    mName(-1),  // see note below
362    mMainBuffer(thread->mixBuffer()),
363    mAuxBuffer(NULL),
364    mAuxEffectId(0), mHasVolumeController(false),
365    mPresentationCompleteFrames(0),
366    mFastIndex(-1),
367    mCachedVolume(1.0),
368    mIsInvalid(false),
369    mAudioTrackServerProxy(NULL),
370    mResumeToStopping(false),
371    mFlushHwPending(false)
372{
373    if (mCblk == NULL) {
374        return;
375    }
376
377    if (sharedBuffer == 0) {
378        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
379                mFrameSize);
380    } else {
381        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
382                mFrameSize);
383    }
384    mServerProxy = mAudioTrackServerProxy;
385
386    mName = thread->getTrackName_l(channelMask, sessionId);
387    if (mName < 0) {
388        ALOGE("no more track names available");
389        return;
390    }
391    // only allocate a fast track index if we were able to allocate a normal track name
392    if (flags & IAudioFlinger::TRACK_FAST) {
393        mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
394        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
395        int i = __builtin_ctz(thread->mFastTrackAvailMask);
396        ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
397        // FIXME This is too eager.  We allocate a fast track index before the
398        //       fast track becomes active.  Since fast tracks are a scarce resource,
399        //       this means we are potentially denying other more important fast tracks from
400        //       being created.  It would be better to allocate the index dynamically.
401        mFastIndex = i;
402        // Read the initial underruns because this field is never cleared by the fast mixer
403        mObservedUnderruns = thread->getFastTrackUnderruns(i);
404        thread->mFastTrackAvailMask &= ~(1 << i);
405    }
406}
407
408AudioFlinger::PlaybackThread::Track::~Track()
409{
410    ALOGV("PlaybackThread::Track destructor");
411
412    // The destructor would clear mSharedBuffer,
413    // but it will not push the decremented reference count,
414    // leaving the client's IMemory dangling indefinitely.
415    // This prevents that leak.
416    if (mSharedBuffer != 0) {
417        mSharedBuffer.clear();
418        // flush the binder command buffer
419        IPCThreadState::self()->flushCommands();
420    }
421}
422
423status_t AudioFlinger::PlaybackThread::Track::initCheck() const
424{
425    status_t status = TrackBase::initCheck();
426    if (status == NO_ERROR && mName < 0) {
427        status = NO_MEMORY;
428    }
429    return status;
430}
431
432void AudioFlinger::PlaybackThread::Track::destroy()
433{
434    // NOTE: destroyTrack_l() can remove a strong reference to this Track
435    // by removing it from mTracks vector, so there is a risk that this Tracks's
436    // destructor is called. As the destructor needs to lock mLock,
437    // we must acquire a strong reference on this Track before locking mLock
438    // here so that the destructor is called only when exiting this function.
439    // On the other hand, as long as Track::destroy() is only called by
440    // TrackHandle destructor, the TrackHandle still holds a strong ref on
441    // this Track with its member mTrack.
442    sp<Track> keep(this);
443    { // scope for mLock
444        sp<ThreadBase> thread = mThread.promote();
445        if (thread != 0) {
446            Mutex::Autolock _l(thread->mLock);
447            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
448            bool wasActive = playbackThread->destroyTrack_l(this);
449            if (!isOutputTrack() && !wasActive) {
450                AudioSystem::releaseOutput(thread->id());
451            }
452        }
453    }
454}
455
456/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
457{
458    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
459                  "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
460}
461
462void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
463{
464    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
465    if (isFastTrack()) {
466        sprintf(buffer, "    F %2d", mFastIndex);
467    } else if (mName >= AudioMixer::TRACK0) {
468        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
469    } else {
470        sprintf(buffer, "    none");
471    }
472    track_state state = mState;
473    char stateChar;
474    if (isTerminated()) {
475        stateChar = 'T';
476    } else {
477        switch (state) {
478        case IDLE:
479            stateChar = 'I';
480            break;
481        case STOPPING_1:
482            stateChar = 's';
483            break;
484        case STOPPING_2:
485            stateChar = '5';
486            break;
487        case STOPPED:
488            stateChar = 'S';
489            break;
490        case RESUMING:
491            stateChar = 'R';
492            break;
493        case ACTIVE:
494            stateChar = 'A';
495            break;
496        case PAUSING:
497            stateChar = 'p';
498            break;
499        case PAUSED:
500            stateChar = 'P';
501            break;
502        case FLUSHED:
503            stateChar = 'F';
504            break;
505        default:
506            stateChar = '?';
507            break;
508        }
509    }
510    char nowInUnderrun;
511    switch (mObservedUnderruns.mBitFields.mMostRecent) {
512    case UNDERRUN_FULL:
513        nowInUnderrun = ' ';
514        break;
515    case UNDERRUN_PARTIAL:
516        nowInUnderrun = '<';
517        break;
518    case UNDERRUN_EMPTY:
519        nowInUnderrun = '*';
520        break;
521    default:
522        nowInUnderrun = '?';
523        break;
524    }
525    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
526                                 "%08X %p %p 0x%03X %9u%c\n",
527            active ? "yes" : "no",
528            (mClient == 0) ? getpid_cached : mClient->pid(),
529            mStreamType,
530            mFormat,
531            mChannelMask,
532            mSessionId,
533            mFrameCount,
534            stateChar,
535            mFillingUpStatus,
536            mAudioTrackServerProxy->getSampleRate(),
537            20.0 * log10((vlr & 0xFFFF) / 4096.0),
538            20.0 * log10((vlr >> 16) / 4096.0),
539            mCblk->mServer,
540            mMainBuffer,
541            mAuxBuffer,
542            mCblk->mFlags,
543            mAudioTrackServerProxy->getUnderrunFrames(),
544            nowInUnderrun);
545}
546
547uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
548    return mAudioTrackServerProxy->getSampleRate();
549}
550
551// AudioBufferProvider interface
552status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
553        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
554{
555    ServerProxy::Buffer buf;
556    size_t desiredFrames = buffer->frameCount;
557    buf.mFrameCount = desiredFrames;
558    status_t status = mServerProxy->obtainBuffer(&buf);
559    buffer->frameCount = buf.mFrameCount;
560    buffer->raw = buf.mRaw;
561    if (buf.mFrameCount == 0) {
562        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
563    }
564    return status;
565}
566
567// releaseBuffer() is not overridden
568
569// ExtendedAudioBufferProvider interface
570
571// Note that framesReady() takes a mutex on the control block using tryLock().
572// This could result in priority inversion if framesReady() is called by the normal mixer,
573// as the normal mixer thread runs at lower
574// priority than the client's callback thread:  there is a short window within framesReady()
575// during which the normal mixer could be preempted, and the client callback would block.
576// Another problem can occur if framesReady() is called by the fast mixer:
577// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
578// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
579size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
580    return mAudioTrackServerProxy->framesReady();
581}
582
583size_t AudioFlinger::PlaybackThread::Track::framesReleased() const
584{
585    return mAudioTrackServerProxy->framesReleased();
586}
587
588// Don't call for fast tracks; the framesReady() could result in priority inversion
589bool AudioFlinger::PlaybackThread::Track::isReady() const {
590    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
591        return true;
592    }
593
594    if (isStopping()) {
595        if (framesReady() > 0) {
596            mFillingUpStatus = FS_FILLED;
597        }
598        return true;
599    }
600
601    if (framesReady() >= mFrameCount ||
602            (mCblk->mFlags & CBLK_FORCEREADY)) {
603        mFillingUpStatus = FS_FILLED;
604        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
605        return true;
606    }
607    return false;
608}
609
610status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
611                                                    int triggerSession __unused)
612{
613    status_t status = NO_ERROR;
614    ALOGV("start(%d), calling pid %d session %d",
615            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
616
617    sp<ThreadBase> thread = mThread.promote();
618    if (thread != 0) {
619        if (isOffloaded()) {
620            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
621            Mutex::Autolock _lth(thread->mLock);
622            sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
623            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
624                    (ec != 0 && ec->isNonOffloadableEnabled())) {
625                invalidate();
626                return PERMISSION_DENIED;
627            }
628        }
629        Mutex::Autolock _lth(thread->mLock);
630        track_state state = mState;
631        // here the track could be either new, or restarted
632        // in both cases "unstop" the track
633
634        // initial state-stopping. next state-pausing.
635        // What if resume is called ?
636
637        if (state == PAUSED || state == PAUSING) {
638            if (mResumeToStopping) {
639                // happened we need to resume to STOPPING_1
640                mState = TrackBase::STOPPING_1;
641                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
642            } else {
643                mState = TrackBase::RESUMING;
644                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
645            }
646        } else {
647            mState = TrackBase::ACTIVE;
648            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
649        }
650
651        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
652        status = playbackThread->addTrack_l(this);
653        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
654            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
655            //  restore previous state if start was rejected by policy manager
656            if (status == PERMISSION_DENIED) {
657                mState = state;
658            }
659        }
660        // track was already in the active list, not a problem
661        if (status == ALREADY_EXISTS) {
662            status = NO_ERROR;
663        } else {
664            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
665            // It is usually unsafe to access the server proxy from a binder thread.
666            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
667            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
668            // and for fast tracks the track is not yet in the fast mixer thread's active set.
669            ServerProxy::Buffer buffer;
670            buffer.mFrameCount = 1;
671            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
672        }
673    } else {
674        status = BAD_VALUE;
675    }
676    return status;
677}
678
679void AudioFlinger::PlaybackThread::Track::stop()
680{
681    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
682    sp<ThreadBase> thread = mThread.promote();
683    if (thread != 0) {
684        Mutex::Autolock _l(thread->mLock);
685        track_state state = mState;
686        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
687            // If the track is not active (PAUSED and buffers full), flush buffers
688            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
689            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
690                reset();
691                mState = STOPPED;
692            } else if (!isFastTrack() && !isOffloaded()) {
693                mState = STOPPED;
694            } else {
695                // For fast tracks prepareTracks_l() will set state to STOPPING_2
696                // presentation is complete
697                // For an offloaded track this starts a drain and state will
698                // move to STOPPING_2 when drain completes and then STOPPED
699                mState = STOPPING_1;
700            }
701            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
702                    playbackThread);
703        }
704    }
705}
706
707void AudioFlinger::PlaybackThread::Track::pause()
708{
709    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
710    sp<ThreadBase> thread = mThread.promote();
711    if (thread != 0) {
712        Mutex::Autolock _l(thread->mLock);
713        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
714        switch (mState) {
715        case STOPPING_1:
716        case STOPPING_2:
717            if (!isOffloaded()) {
718                /* nothing to do if track is not offloaded */
719                break;
720            }
721
722            // Offloaded track was draining, we need to carry on draining when resumed
723            mResumeToStopping = true;
724            // fall through...
725        case ACTIVE:
726        case RESUMING:
727            mState = PAUSING;
728            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
729            playbackThread->broadcast_l();
730            break;
731
732        default:
733            break;
734        }
735    }
736}
737
738void AudioFlinger::PlaybackThread::Track::flush()
739{
740    ALOGV("flush(%d)", mName);
741    sp<ThreadBase> thread = mThread.promote();
742    if (thread != 0) {
743        Mutex::Autolock _l(thread->mLock);
744        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
745
746        if (isOffloaded()) {
747            // If offloaded we allow flush during any state except terminated
748            // and keep the track active to avoid problems if user is seeking
749            // rapidly and underlying hardware has a significant delay handling
750            // a pause
751            if (isTerminated()) {
752                return;
753            }
754
755            ALOGV("flush: offload flush");
756            reset();
757
758            if (mState == STOPPING_1 || mState == STOPPING_2) {
759                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
760                mState = ACTIVE;
761            }
762
763            if (mState == ACTIVE) {
764                ALOGV("flush called in active state, resetting buffer time out retry count");
765                mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
766            }
767
768            mFlushHwPending = true;
769            mResumeToStopping = false;
770        } else {
771            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
772                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
773                return;
774            }
775            // No point remaining in PAUSED state after a flush => go to
776            // FLUSHED state
777            mState = FLUSHED;
778            // do not reset the track if it is still in the process of being stopped or paused.
779            // this will be done by prepareTracks_l() when the track is stopped.
780            // prepareTracks_l() will see mState == FLUSHED, then
781            // remove from active track list, reset(), and trigger presentation complete
782            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
783                reset();
784            }
785        }
786        // Prevent flush being lost if the track is flushed and then resumed
787        // before mixer thread can run. This is important when offloading
788        // because the hardware buffer could hold a large amount of audio
789        playbackThread->broadcast_l();
790    }
791}
792
793// must be called with thread lock held
794void AudioFlinger::PlaybackThread::Track::flushAck()
795{
796    if (!isOffloaded())
797        return;
798
799    mFlushHwPending = false;
800}
801
802void AudioFlinger::PlaybackThread::Track::reset()
803{
804    // Do not reset twice to avoid discarding data written just after a flush and before
805    // the audioflinger thread detects the track is stopped.
806    if (!mResetDone) {
807        // Force underrun condition to avoid false underrun callback until first data is
808        // written to buffer
809        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
810        mFillingUpStatus = FS_FILLING;
811        mResetDone = true;
812        if (mState == FLUSHED) {
813            mState = IDLE;
814        }
815    }
816}
817
818status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
819{
820    sp<ThreadBase> thread = mThread.promote();
821    if (thread == 0) {
822        ALOGE("thread is dead");
823        return FAILED_TRANSACTION;
824    } else if ((thread->type() == ThreadBase::DIRECT) ||
825                    (thread->type() == ThreadBase::OFFLOAD)) {
826        return thread->setParameters(keyValuePairs);
827    } else {
828        return PERMISSION_DENIED;
829    }
830}
831
832status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
833{
834    // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
835    if (isFastTrack()) {
836        return INVALID_OPERATION;
837    }
838    sp<ThreadBase> thread = mThread.promote();
839    if (thread == 0) {
840        return INVALID_OPERATION;
841    }
842    Mutex::Autolock _l(thread->mLock);
843    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
844    if (!isOffloaded()) {
845        if (!playbackThread->mLatchQValid) {
846            return INVALID_OPERATION;
847        }
848        uint32_t unpresentedFrames =
849                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
850                playbackThread->mSampleRate;
851        uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
852        if (framesWritten < unpresentedFrames) {
853            return INVALID_OPERATION;
854        }
855        timestamp.mPosition = framesWritten - unpresentedFrames;
856        timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
857        return NO_ERROR;
858    }
859
860    return playbackThread->getTimestamp_l(timestamp);
861}
862
863status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
864{
865    status_t status = DEAD_OBJECT;
866    sp<ThreadBase> thread = mThread.promote();
867    if (thread != 0) {
868        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
869        sp<AudioFlinger> af = mClient->audioFlinger();
870
871        Mutex::Autolock _l(af->mLock);
872
873        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
874
875        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
876            Mutex::Autolock _dl(playbackThread->mLock);
877            Mutex::Autolock _sl(srcThread->mLock);
878            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
879            if (chain == 0) {
880                return INVALID_OPERATION;
881            }
882
883            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
884            if (effect == 0) {
885                return INVALID_OPERATION;
886            }
887            srcThread->removeEffect_l(effect);
888            status = playbackThread->addEffect_l(effect);
889            if (status != NO_ERROR) {
890                srcThread->addEffect_l(effect);
891                return INVALID_OPERATION;
892            }
893            // removeEffect_l() has stopped the effect if it was active so it must be restarted
894            if (effect->state() == EffectModule::ACTIVE ||
895                    effect->state() == EffectModule::STOPPING) {
896                effect->start();
897            }
898
899            sp<EffectChain> dstChain = effect->chain().promote();
900            if (dstChain == 0) {
901                srcThread->addEffect_l(effect);
902                return INVALID_OPERATION;
903            }
904            AudioSystem::unregisterEffect(effect->id());
905            AudioSystem::registerEffect(&effect->desc(),
906                                        srcThread->id(),
907                                        dstChain->strategy(),
908                                        AUDIO_SESSION_OUTPUT_MIX,
909                                        effect->id());
910            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
911        }
912        status = playbackThread->attachAuxEffect(this, EffectId);
913    }
914    return status;
915}
916
917void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
918{
919    mAuxEffectId = EffectId;
920    mAuxBuffer = buffer;
921}
922
923bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
924                                                         size_t audioHalFrames)
925{
926    // a track is considered presented when the total number of frames written to audio HAL
927    // corresponds to the number of frames written when presentationComplete() is called for the
928    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
929    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
930    // to detect when all frames have been played. In this case framesWritten isn't
931    // useful because it doesn't always reflect whether there is data in the h/w
932    // buffers, particularly if a track has been paused and resumed during draining
933    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
934                      mPresentationCompleteFrames, framesWritten);
935    if (mPresentationCompleteFrames == 0) {
936        mPresentationCompleteFrames = framesWritten + audioHalFrames;
937        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
938                  mPresentationCompleteFrames, audioHalFrames);
939    }
940
941    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
942        ALOGV("presentationComplete() session %d complete: framesWritten %d",
943                  mSessionId, framesWritten);
944        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
945        mAudioTrackServerProxy->setStreamEndDone();
946        return true;
947    }
948    return false;
949}
950
951void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
952{
953    for (size_t i = 0; i < mSyncEvents.size(); i++) {
954        if (mSyncEvents[i]->type() == type) {
955            mSyncEvents[i]->trigger();
956            mSyncEvents.removeAt(i);
957            i--;
958        }
959    }
960}
961
962// implement VolumeBufferProvider interface
963
964uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
965{
966    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
967    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
968    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
969    uint32_t vl = vlr & 0xFFFF;
970    uint32_t vr = vlr >> 16;
971    // track volumes come from shared memory, so can't be trusted and must be clamped
972    if (vl > MAX_GAIN_INT) {
973        vl = MAX_GAIN_INT;
974    }
975    if (vr > MAX_GAIN_INT) {
976        vr = MAX_GAIN_INT;
977    }
978    // now apply the cached master volume and stream type volume;
979    // this is trusted but lacks any synchronization or barrier so may be stale
980    float v = mCachedVolume;
981    vl *= v;
982    vr *= v;
983    // re-combine into U4.16
984    vlr = (vr << 16) | (vl & 0xFFFF);
985    // FIXME look at mute, pause, and stop flags
986    return vlr;
987}
988
989status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
990{
991    if (isTerminated() || mState == PAUSED ||
992            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
993                                      (mState == STOPPED)))) {
994        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
995              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
996        event->cancel();
997        return INVALID_OPERATION;
998    }
999    (void) TrackBase::setSyncEvent(event);
1000    return NO_ERROR;
1001}
1002
1003void AudioFlinger::PlaybackThread::Track::invalidate()
1004{
1005    // FIXME should use proxy, and needs work
1006    audio_track_cblk_t* cblk = mCblk;
1007    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1008    android_atomic_release_store(0x40000000, &cblk->mFutex);
1009    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1010    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1011    mIsInvalid = true;
1012}
1013
1014void AudioFlinger::PlaybackThread::Track::signal()
1015{
1016    sp<ThreadBase> thread = mThread.promote();
1017    if (thread != 0) {
1018        PlaybackThread *t = (PlaybackThread *)thread.get();
1019        Mutex::Autolock _l(t->mLock);
1020        t->broadcast_l();
1021    }
1022}
1023
1024//To be called with thread lock held
1025bool AudioFlinger::PlaybackThread::Track::isResumePending() {
1026
1027    if (mState == RESUMING)
1028        return true;
1029    /* Resume is pending if track was stopping before pause was called */
1030    if (mState == STOPPING_1 &&
1031        mResumeToStopping)
1032        return true;
1033
1034    return false;
1035}
1036
1037//To be called with thread lock held
1038void AudioFlinger::PlaybackThread::Track::resumeAck() {
1039
1040
1041    if (mState == RESUMING)
1042        mState = ACTIVE;
1043
1044    // Other possibility of  pending resume is stopping_1 state
1045    // Do not update the state from stopping as this prevents
1046    // drain being called.
1047    if (mState == STOPPING_1) {
1048        mResumeToStopping = false;
1049    }
1050}
1051// ----------------------------------------------------------------------------
1052
1053sp<AudioFlinger::PlaybackThread::TimedTrack>
1054AudioFlinger::PlaybackThread::TimedTrack::create(
1055            PlaybackThread *thread,
1056            const sp<Client>& client,
1057            audio_stream_type_t streamType,
1058            uint32_t sampleRate,
1059            audio_format_t format,
1060            audio_channel_mask_t channelMask,
1061            size_t frameCount,
1062            const sp<IMemory>& sharedBuffer,
1063            int sessionId,
1064            int uid)
1065{
1066    if (!client->reserveTimedTrack())
1067        return 0;
1068
1069    return new TimedTrack(
1070        thread, client, streamType, sampleRate, format, channelMask, frameCount,
1071        sharedBuffer, sessionId, uid);
1072}
1073
1074AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
1075            PlaybackThread *thread,
1076            const sp<Client>& client,
1077            audio_stream_type_t streamType,
1078            uint32_t sampleRate,
1079            audio_format_t format,
1080            audio_channel_mask_t channelMask,
1081            size_t frameCount,
1082            const sp<IMemory>& sharedBuffer,
1083            int sessionId,
1084            int uid)
1085    : Track(thread, client, streamType, sampleRate, format, channelMask,
1086            frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
1087      mQueueHeadInFlight(false),
1088      mTrimQueueHeadOnRelease(false),
1089      mFramesPendingInQueue(0),
1090      mTimedSilenceBuffer(NULL),
1091      mTimedSilenceBufferSize(0),
1092      mTimedAudioOutputOnTime(false),
1093      mMediaTimeTransformValid(false)
1094{
1095    LocalClock lc;
1096    mLocalTimeFreq = lc.getLocalFreq();
1097
1098    mLocalTimeToSampleTransform.a_zero = 0;
1099    mLocalTimeToSampleTransform.b_zero = 0;
1100    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1101    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1102    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1103                            &mLocalTimeToSampleTransform.a_to_b_denom);
1104
1105    mMediaTimeToSampleTransform.a_zero = 0;
1106    mMediaTimeToSampleTransform.b_zero = 0;
1107    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1108    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1109    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1110                            &mMediaTimeToSampleTransform.a_to_b_denom);
1111}
1112
1113AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1114    mClient->releaseTimedTrack();
1115    delete [] mTimedSilenceBuffer;
1116}
1117
1118status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1119    size_t size, sp<IMemory>* buffer) {
1120
1121    Mutex::Autolock _l(mTimedBufferQueueLock);
1122
1123    trimTimedBufferQueue_l();
1124
1125    // lazily initialize the shared memory heap for timed buffers
1126    if (mTimedMemoryDealer == NULL) {
1127        const int kTimedBufferHeapSize = 512 << 10;
1128
1129        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1130                                              "AudioFlingerTimed");
1131        if (mTimedMemoryDealer == NULL) {
1132            return NO_MEMORY;
1133        }
1134    }
1135
1136    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1137    if (newBuffer == 0 || newBuffer->pointer() == NULL) {
1138        return NO_MEMORY;
1139    }
1140
1141    *buffer = newBuffer;
1142    return NO_ERROR;
1143}
1144
1145// caller must hold mTimedBufferQueueLock
1146void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1147    int64_t mediaTimeNow;
1148    {
1149        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1150        if (!mMediaTimeTransformValid)
1151            return;
1152
1153        int64_t targetTimeNow;
1154        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1155            ? mCCHelper.getCommonTime(&targetTimeNow)
1156            : mCCHelper.getLocalTime(&targetTimeNow);
1157
1158        if (OK != res)
1159            return;
1160
1161        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1162                                                    &mediaTimeNow)) {
1163            return;
1164        }
1165    }
1166
1167    size_t trimEnd;
1168    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1169        int64_t bufEnd;
1170
1171        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1172            // We have a next buffer.  Just use its PTS as the PTS of the frame
1173            // following the last frame in this buffer.  If the stream is sparse
1174            // (ie, there are deliberate gaps left in the stream which should be
1175            // filled with silence by the TimedAudioTrack), then this can result
1176            // in one extra buffer being left un-trimmed when it could have
1177            // been.  In general, this is not typical, and we would rather
1178            // optimized away the TS calculation below for the more common case
1179            // where PTSes are contiguous.
1180            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1181        } else {
1182            // We have no next buffer.  Compute the PTS of the frame following
1183            // the last frame in this buffer by computing the duration of of
1184            // this frame in media time units and adding it to the PTS of the
1185            // buffer.
1186            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1187                               / mFrameSize;
1188
1189            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1190                                                                &bufEnd)) {
1191                ALOGE("Failed to convert frame count of %lld to media time"
1192                      " duration" " (scale factor %d/%u) in %s",
1193                      frameCount,
1194                      mMediaTimeToSampleTransform.a_to_b_numer,
1195                      mMediaTimeToSampleTransform.a_to_b_denom,
1196                      __PRETTY_FUNCTION__);
1197                break;
1198            }
1199            bufEnd += mTimedBufferQueue[trimEnd].pts();
1200        }
1201
1202        if (bufEnd > mediaTimeNow)
1203            break;
1204
1205        // Is the buffer we want to use in the middle of a mix operation right
1206        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1207        // from the mixer which should be coming back shortly.
1208        if (!trimEnd && mQueueHeadInFlight) {
1209            mTrimQueueHeadOnRelease = true;
1210        }
1211    }
1212
1213    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1214    if (trimStart < trimEnd) {
1215        // Update the bookkeeping for framesReady()
1216        for (size_t i = trimStart; i < trimEnd; ++i) {
1217            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1218        }
1219
1220        // Now actually remove the buffers from the queue.
1221        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1222    }
1223}
1224
1225void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1226        const char* logTag) {
1227    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1228                "%s called (reason \"%s\"), but timed buffer queue has no"
1229                " elements to trim.", __FUNCTION__, logTag);
1230
1231    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1232    mTimedBufferQueue.removeAt(0);
1233}
1234
1235void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1236        const TimedBuffer& buf,
1237        const char* logTag __unused) {
1238    uint32_t bufBytes        = buf.buffer()->size();
1239    uint32_t consumedAlready = buf.position();
1240
1241    ALOG_ASSERT(consumedAlready <= bufBytes,
1242                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1243                " only %u bytes long, but claims to have consumed %u"
1244                " bytes.  (update reason: \"%s\")",
1245                bufBytes, consumedAlready, logTag);
1246
1247    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1248    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1249                "Bad bookkeeping while updating frames pending.  Should have at"
1250                " least %u queued frames, but we think we have only %u.  (update"
1251                " reason: \"%s\")",
1252                bufFrames, mFramesPendingInQueue, logTag);
1253
1254    mFramesPendingInQueue -= bufFrames;
1255}
1256
1257status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1258    const sp<IMemory>& buffer, int64_t pts) {
1259
1260    {
1261        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1262        if (!mMediaTimeTransformValid)
1263            return INVALID_OPERATION;
1264    }
1265
1266    Mutex::Autolock _l(mTimedBufferQueueLock);
1267
1268    uint32_t bufFrames = buffer->size() / mFrameSize;
1269    mFramesPendingInQueue += bufFrames;
1270    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1271
1272    return NO_ERROR;
1273}
1274
1275status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1276    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1277
1278    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1279           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1280           target);
1281
1282    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1283          target == TimedAudioTrack::COMMON_TIME)) {
1284        return BAD_VALUE;
1285    }
1286
1287    Mutex::Autolock lock(mMediaTimeTransformLock);
1288    mMediaTimeTransform = xform;
1289    mMediaTimeTransformTarget = target;
1290    mMediaTimeTransformValid = true;
1291
1292    return NO_ERROR;
1293}
1294
1295#define min(a, b) ((a) < (b) ? (a) : (b))
1296
1297// implementation of getNextBuffer for tracks whose buffers have timestamps
1298status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1299    AudioBufferProvider::Buffer* buffer, int64_t pts)
1300{
1301    if (pts == AudioBufferProvider::kInvalidPTS) {
1302        buffer->raw = NULL;
1303        buffer->frameCount = 0;
1304        mTimedAudioOutputOnTime = false;
1305        return INVALID_OPERATION;
1306    }
1307
1308    Mutex::Autolock _l(mTimedBufferQueueLock);
1309
1310    ALOG_ASSERT(!mQueueHeadInFlight,
1311                "getNextBuffer called without releaseBuffer!");
1312
1313    while (true) {
1314
1315        // if we have no timed buffers, then fail
1316        if (mTimedBufferQueue.isEmpty()) {
1317            buffer->raw = NULL;
1318            buffer->frameCount = 0;
1319            return NOT_ENOUGH_DATA;
1320        }
1321
1322        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1323
1324        // calculate the PTS of the head of the timed buffer queue expressed in
1325        // local time
1326        int64_t headLocalPTS;
1327        {
1328            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1329
1330            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1331
1332            if (mMediaTimeTransform.a_to_b_denom == 0) {
1333                // the transform represents a pause, so yield silence
1334                timedYieldSilence_l(buffer->frameCount, buffer);
1335                return NO_ERROR;
1336            }
1337
1338            int64_t transformedPTS;
1339            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1340                                                        &transformedPTS)) {
1341                // the transform failed.  this shouldn't happen, but if it does
1342                // then just drop this buffer
1343                ALOGW("timedGetNextBuffer transform failed");
1344                buffer->raw = NULL;
1345                buffer->frameCount = 0;
1346                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1347                return NO_ERROR;
1348            }
1349
1350            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1351                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1352                                                          &headLocalPTS)) {
1353                    buffer->raw = NULL;
1354                    buffer->frameCount = 0;
1355                    return INVALID_OPERATION;
1356                }
1357            } else {
1358                headLocalPTS = transformedPTS;
1359            }
1360        }
1361
1362        uint32_t sr = sampleRate();
1363
1364        // adjust the head buffer's PTS to reflect the portion of the head buffer
1365        // that has already been consumed
1366        int64_t effectivePTS = headLocalPTS +
1367                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1368
1369        // Calculate the delta in samples between the head of the input buffer
1370        // queue and the start of the next output buffer that will be written.
1371        // If the transformation fails because of over or underflow, it means
1372        // that the sample's position in the output stream is so far out of
1373        // whack that it should just be dropped.
1374        int64_t sampleDelta;
1375        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1376            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1377            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1378                                       " mix");
1379            continue;
1380        }
1381        if (!mLocalTimeToSampleTransform.doForwardTransform(
1382                (effectivePTS - pts) << 32, &sampleDelta)) {
1383            ALOGV("*** too late during sample rate transform: dropped buffer");
1384            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1385            continue;
1386        }
1387
1388        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1389               " sampleDelta=[%d.%08x]",
1390               head.pts(), head.position(), pts,
1391               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1392                   + (sampleDelta >> 32)),
1393               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1394
1395        // if the delta between the ideal placement for the next input sample and
1396        // the current output position is within this threshold, then we will
1397        // concatenate the next input samples to the previous output
1398        const int64_t kSampleContinuityThreshold =
1399                (static_cast<int64_t>(sr) << 32) / 250;
1400
1401        // if this is the first buffer of audio that we're emitting from this track
1402        // then it should be almost exactly on time.
1403        const int64_t kSampleStartupThreshold = 1LL << 32;
1404
1405        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1406           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1407            // the next input is close enough to being on time, so concatenate it
1408            // with the last output
1409            timedYieldSamples_l(buffer);
1410
1411            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1412                    head.position(), buffer->frameCount);
1413            return NO_ERROR;
1414        }
1415
1416        // Looks like our output is not on time.  Reset our on timed status.
1417        // Next time we mix samples from our input queue, then should be within
1418        // the StartupThreshold.
1419        mTimedAudioOutputOnTime = false;
1420        if (sampleDelta > 0) {
1421            // the gap between the current output position and the proper start of
1422            // the next input sample is too big, so fill it with silence
1423            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1424
1425            timedYieldSilence_l(framesUntilNextInput, buffer);
1426            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1427            return NO_ERROR;
1428        } else {
1429            // the next input sample is late
1430            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1431            size_t onTimeSamplePosition =
1432                    head.position() + lateFrames * mFrameSize;
1433
1434            if (onTimeSamplePosition > head.buffer()->size()) {
1435                // all the remaining samples in the head are too late, so
1436                // drop it and move on
1437                ALOGV("*** too late: dropped buffer");
1438                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1439                continue;
1440            } else {
1441                // skip over the late samples
1442                head.setPosition(onTimeSamplePosition);
1443
1444                // yield the available samples
1445                timedYieldSamples_l(buffer);
1446
1447                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1448                return NO_ERROR;
1449            }
1450        }
1451    }
1452}
1453
1454// Yield samples from the timed buffer queue head up to the given output
1455// buffer's capacity.
1456//
1457// Caller must hold mTimedBufferQueueLock
1458void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1459    AudioBufferProvider::Buffer* buffer) {
1460
1461    const TimedBuffer& head = mTimedBufferQueue[0];
1462
1463    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1464                   head.position());
1465
1466    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1467                                 mFrameSize);
1468    size_t framesRequested = buffer->frameCount;
1469    buffer->frameCount = min(framesLeftInHead, framesRequested);
1470
1471    mQueueHeadInFlight = true;
1472    mTimedAudioOutputOnTime = true;
1473}
1474
1475// Yield samples of silence up to the given output buffer's capacity
1476//
1477// Caller must hold mTimedBufferQueueLock
1478void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1479    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1480
1481    // lazily allocate a buffer filled with silence
1482    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1483        delete [] mTimedSilenceBuffer;
1484        mTimedSilenceBufferSize = numFrames * mFrameSize;
1485        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1486        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1487    }
1488
1489    buffer->raw = mTimedSilenceBuffer;
1490    size_t framesRequested = buffer->frameCount;
1491    buffer->frameCount = min(numFrames, framesRequested);
1492
1493    mTimedAudioOutputOnTime = false;
1494}
1495
1496// AudioBufferProvider interface
1497void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1498    AudioBufferProvider::Buffer* buffer) {
1499
1500    Mutex::Autolock _l(mTimedBufferQueueLock);
1501
1502    // If the buffer which was just released is part of the buffer at the head
1503    // of the queue, be sure to update the amt of the buffer which has been
1504    // consumed.  If the buffer being returned is not part of the head of the
1505    // queue, its either because the buffer is part of the silence buffer, or
1506    // because the head of the timed queue was trimmed after the mixer called
1507    // getNextBuffer but before the mixer called releaseBuffer.
1508    if (buffer->raw == mTimedSilenceBuffer) {
1509        ALOG_ASSERT(!mQueueHeadInFlight,
1510                    "Queue head in flight during release of silence buffer!");
1511        goto done;
1512    }
1513
1514    ALOG_ASSERT(mQueueHeadInFlight,
1515                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1516                " head in flight.");
1517
1518    if (mTimedBufferQueue.size()) {
1519        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1520
1521        void* start = head.buffer()->pointer();
1522        void* end   = reinterpret_cast<void*>(
1523                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1524                        + head.buffer()->size());
1525
1526        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1527                    "released buffer not within the head of the timed buffer"
1528                    " queue; qHead = [%p, %p], released buffer = %p",
1529                    start, end, buffer->raw);
1530
1531        head.setPosition(head.position() +
1532                (buffer->frameCount * mFrameSize));
1533        mQueueHeadInFlight = false;
1534
1535        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1536                    "Bad bookkeeping during releaseBuffer!  Should have at"
1537                    " least %u queued frames, but we think we have only %u",
1538                    buffer->frameCount, mFramesPendingInQueue);
1539
1540        mFramesPendingInQueue -= buffer->frameCount;
1541
1542        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1543            || mTrimQueueHeadOnRelease) {
1544            trimTimedBufferQueueHead_l("releaseBuffer");
1545            mTrimQueueHeadOnRelease = false;
1546        }
1547    } else {
1548        LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1549                  " buffers in the timed buffer queue");
1550    }
1551
1552done:
1553    buffer->raw = 0;
1554    buffer->frameCount = 0;
1555}
1556
1557size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1558    Mutex::Autolock _l(mTimedBufferQueueLock);
1559    return mFramesPendingInQueue;
1560}
1561
1562AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1563        : mPTS(0), mPosition(0) {}
1564
1565AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1566    const sp<IMemory>& buffer, int64_t pts)
1567        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1568
1569
1570// ----------------------------------------------------------------------------
1571
1572AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1573            PlaybackThread *playbackThread,
1574            DuplicatingThread *sourceThread,
1575            uint32_t sampleRate,
1576            audio_format_t format,
1577            audio_channel_mask_t channelMask,
1578            size_t frameCount,
1579            int uid)
1580    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1581                NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
1582    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1583{
1584
1585    if (mCblk != NULL) {
1586        mOutBuffer.frameCount = 0;
1587        playbackThread->mTracks.add(this);
1588        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1589                "frameCount %u, mChannelMask 0x%08x",
1590                mCblk, mBuffer,
1591                frameCount, mChannelMask);
1592        // since client and server are in the same process,
1593        // the buffer has the same virtual address on both sides
1594        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1595        mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
1596        mClientProxy->setSendLevel(0.0);
1597        mClientProxy->setSampleRate(sampleRate);
1598        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1599                true /*clientInServer*/);
1600    } else {
1601        ALOGW("Error creating output track on thread %p", playbackThread);
1602    }
1603}
1604
1605AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1606{
1607    clearBufferQueue();
1608    delete mClientProxy;
1609    // superclass destructor will now delete the server proxy and shared memory both refer to
1610}
1611
1612status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1613                                                          int triggerSession)
1614{
1615    status_t status = Track::start(event, triggerSession);
1616    if (status != NO_ERROR) {
1617        return status;
1618    }
1619
1620    mActive = true;
1621    mRetryCount = 127;
1622    return status;
1623}
1624
1625void AudioFlinger::PlaybackThread::OutputTrack::stop()
1626{
1627    Track::stop();
1628    clearBufferQueue();
1629    mOutBuffer.frameCount = 0;
1630    mActive = false;
1631}
1632
1633bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1634{
1635    Buffer *pInBuffer;
1636    Buffer inBuffer;
1637    uint32_t channelCount = mChannelCount;
1638    bool outputBufferFull = false;
1639    inBuffer.frameCount = frames;
1640    inBuffer.i16 = data;
1641
1642    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1643
1644    if (!mActive && frames != 0) {
1645        start();
1646        sp<ThreadBase> thread = mThread.promote();
1647        if (thread != 0) {
1648            MixerThread *mixerThread = (MixerThread *)thread.get();
1649            if (mFrameCount > frames) {
1650                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1651                    uint32_t startFrames = (mFrameCount - frames);
1652                    pInBuffer = new Buffer;
1653                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1654                    pInBuffer->frameCount = startFrames;
1655                    pInBuffer->i16 = pInBuffer->mBuffer;
1656                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1657                    mBufferQueue.add(pInBuffer);
1658                } else {
1659                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1660                }
1661            }
1662        }
1663    }
1664
1665    while (waitTimeLeftMs) {
1666        // First write pending buffers, then new data
1667        if (mBufferQueue.size()) {
1668            pInBuffer = mBufferQueue.itemAt(0);
1669        } else {
1670            pInBuffer = &inBuffer;
1671        }
1672
1673        if (pInBuffer->frameCount == 0) {
1674            break;
1675        }
1676
1677        if (mOutBuffer.frameCount == 0) {
1678            mOutBuffer.frameCount = pInBuffer->frameCount;
1679            nsecs_t startTime = systemTime();
1680            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1681            if (status != NO_ERROR) {
1682                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1683                        mThread.unsafe_get(), status);
1684                outputBufferFull = true;
1685                break;
1686            }
1687            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1688            if (waitTimeLeftMs >= waitTimeMs) {
1689                waitTimeLeftMs -= waitTimeMs;
1690            } else {
1691                waitTimeLeftMs = 0;
1692            }
1693        }
1694
1695        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1696                pInBuffer->frameCount;
1697        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1698        Proxy::Buffer buf;
1699        buf.mFrameCount = outFrames;
1700        buf.mRaw = NULL;
1701        mClientProxy->releaseBuffer(&buf);
1702        pInBuffer->frameCount -= outFrames;
1703        pInBuffer->i16 += outFrames * channelCount;
1704        mOutBuffer.frameCount -= outFrames;
1705        mOutBuffer.i16 += outFrames * channelCount;
1706
1707        if (pInBuffer->frameCount == 0) {
1708            if (mBufferQueue.size()) {
1709                mBufferQueue.removeAt(0);
1710                delete [] pInBuffer->mBuffer;
1711                delete pInBuffer;
1712                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1713                        mThread.unsafe_get(), mBufferQueue.size());
1714            } else {
1715                break;
1716            }
1717        }
1718    }
1719
1720    // If we could not write all frames, allocate a buffer and queue it for next time.
1721    if (inBuffer.frameCount) {
1722        sp<ThreadBase> thread = mThread.promote();
1723        if (thread != 0 && !thread->standby()) {
1724            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1725                pInBuffer = new Buffer;
1726                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1727                pInBuffer->frameCount = inBuffer.frameCount;
1728                pInBuffer->i16 = pInBuffer->mBuffer;
1729                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1730                        sizeof(int16_t));
1731                mBufferQueue.add(pInBuffer);
1732                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1733                        mThread.unsafe_get(), mBufferQueue.size());
1734            } else {
1735                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1736                        mThread.unsafe_get(), this);
1737            }
1738        }
1739    }
1740
1741    // Calling write() with a 0 length buffer, means that no more data will be written:
1742    // If no more buffers are pending, fill output track buffer to make sure it is started
1743    // by output mixer.
1744    if (frames == 0 && mBufferQueue.size() == 0) {
1745        // FIXME borken, replace by getting framesReady() from proxy
1746        size_t user = 0;    // was mCblk->user
1747        if (user < mFrameCount) {
1748            frames = mFrameCount - user;
1749            pInBuffer = new Buffer;
1750            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1751            pInBuffer->frameCount = frames;
1752            pInBuffer->i16 = pInBuffer->mBuffer;
1753            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1754            mBufferQueue.add(pInBuffer);
1755        } else if (mActive) {
1756            stop();
1757        }
1758    }
1759
1760    return outputBufferFull;
1761}
1762
1763status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1764        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1765{
1766    ClientProxy::Buffer buf;
1767    buf.mFrameCount = buffer->frameCount;
1768    struct timespec timeout;
1769    timeout.tv_sec = waitTimeMs / 1000;
1770    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1771    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1772    buffer->frameCount = buf.mFrameCount;
1773    buffer->raw = buf.mRaw;
1774    return status;
1775}
1776
1777void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1778{
1779    size_t size = mBufferQueue.size();
1780
1781    for (size_t i = 0; i < size; i++) {
1782        Buffer *pBuffer = mBufferQueue.itemAt(i);
1783        delete [] pBuffer->mBuffer;
1784        delete pBuffer;
1785    }
1786    mBufferQueue.clear();
1787}
1788
1789
1790// ----------------------------------------------------------------------------
1791//      Record
1792// ----------------------------------------------------------------------------
1793
1794AudioFlinger::RecordHandle::RecordHandle(
1795        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1796    : BnAudioRecord(),
1797    mRecordTrack(recordTrack)
1798{
1799}
1800
1801AudioFlinger::RecordHandle::~RecordHandle() {
1802    stop_nonvirtual();
1803    mRecordTrack->destroy();
1804}
1805
1806status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1807        int triggerSession) {
1808    ALOGV("RecordHandle::start()");
1809    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1810}
1811
1812void AudioFlinger::RecordHandle::stop() {
1813    stop_nonvirtual();
1814}
1815
1816void AudioFlinger::RecordHandle::stop_nonvirtual() {
1817    ALOGV("RecordHandle::stop()");
1818    mRecordTrack->stop();
1819}
1820
1821status_t AudioFlinger::RecordHandle::onTransact(
1822    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1823{
1824    return BnAudioRecord::onTransact(code, data, reply, flags);
1825}
1826
1827// ----------------------------------------------------------------------------
1828
1829// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
1830AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1831            RecordThread *thread,
1832            const sp<Client>& client,
1833            uint32_t sampleRate,
1834            audio_format_t format,
1835            audio_channel_mask_t channelMask,
1836            size_t frameCount,
1837            int sessionId,
1838            int uid,
1839            IAudioFlinger::track_flags_t flags)
1840    :   TrackBase(thread, client, sampleRate, format,
1841                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid,
1842                  flags, false /*isOut*/,
1843                  (flags & IAudioFlinger::TRACK_FAST) != 0 /*useReadOnlyHeap*/),
1844        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
1845        // See real initialization of mRsmpInFront at RecordThread::start()
1846        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
1847{
1848    if (mCblk == NULL) {
1849        return;
1850    }
1851
1852    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
1853
1854    uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
1855    // FIXME I don't understand either of the channel count checks
1856    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
1857            channelCount <= FCC_2) {
1858        // sink SR
1859        mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
1860        // source SR
1861        mResampler->setSampleRate(thread->mSampleRate);
1862        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
1863        mResamplerBufferProvider = new ResamplerBufferProvider(this);
1864    }
1865}
1866
1867AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1868{
1869    ALOGV("%s", __func__);
1870    delete mResampler;
1871    delete[] mRsmpOutBuffer;
1872    delete mResamplerBufferProvider;
1873}
1874
1875// AudioBufferProvider interface
1876status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1877        int64_t pts __unused)
1878{
1879    ServerProxy::Buffer buf;
1880    buf.mFrameCount = buffer->frameCount;
1881    status_t status = mServerProxy->obtainBuffer(&buf);
1882    buffer->frameCount = buf.mFrameCount;
1883    buffer->raw = buf.mRaw;
1884    if (buf.mFrameCount == 0) {
1885        // FIXME also wake futex so that overrun is noticed more quickly
1886        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1887    }
1888    return status;
1889}
1890
1891status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1892                                                        int triggerSession)
1893{
1894    sp<ThreadBase> thread = mThread.promote();
1895    if (thread != 0) {
1896        RecordThread *recordThread = (RecordThread *)thread.get();
1897        return recordThread->start(this, event, triggerSession);
1898    } else {
1899        return BAD_VALUE;
1900    }
1901}
1902
1903void AudioFlinger::RecordThread::RecordTrack::stop()
1904{
1905    sp<ThreadBase> thread = mThread.promote();
1906    if (thread != 0) {
1907        RecordThread *recordThread = (RecordThread *)thread.get();
1908        if (recordThread->stop(this)) {
1909            AudioSystem::stopInput(recordThread->id());
1910        }
1911    }
1912}
1913
1914void AudioFlinger::RecordThread::RecordTrack::destroy()
1915{
1916    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1917    sp<RecordTrack> keep(this);
1918    {
1919        sp<ThreadBase> thread = mThread.promote();
1920        if (thread != 0) {
1921            if (mState == ACTIVE || mState == RESUMING) {
1922                AudioSystem::stopInput(thread->id());
1923            }
1924            AudioSystem::releaseInput(thread->id());
1925            Mutex::Autolock _l(thread->mLock);
1926            RecordThread *recordThread = (RecordThread *) thread.get();
1927            recordThread->destroyTrack_l(this);
1928        }
1929    }
1930}
1931
1932void AudioFlinger::RecordThread::RecordTrack::invalidate()
1933{
1934    // FIXME should use proxy, and needs work
1935    audio_track_cblk_t* cblk = mCblk;
1936    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1937    android_atomic_release_store(0x40000000, &cblk->mFutex);
1938    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1939    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1940}
1941
1942
1943/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1944{
1945    result.append("    Active Client Fmt Chn mask Session S   Server fCount Resampling\n");
1946}
1947
1948void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
1949{
1950    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n",
1951            active ? "yes" : "no",
1952            (mClient == 0) ? getpid_cached : mClient->pid(),
1953            mFormat,
1954            mChannelMask,
1955            mSessionId,
1956            mState,
1957            mCblk->mServer,
1958            mFrameCount,
1959            mResampler != NULL);
1960
1961}
1962
1963void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
1964{
1965    if (event == mSyncStartEvent) {
1966        ssize_t framesToDrop = 0;
1967        sp<ThreadBase> threadBase = mThread.promote();
1968        if (threadBase != 0) {
1969            // TODO: use actual buffer filling status instead of 2 buffers when info is available
1970            // from audio HAL
1971            framesToDrop = threadBase->mFrameCount * 2;
1972        }
1973        mFramesToDrop = framesToDrop;
1974    }
1975}
1976
1977void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
1978{
1979    if (mSyncStartEvent != 0) {
1980        mSyncStartEvent->cancel();
1981        mSyncStartEvent.clear();
1982    }
1983    mFramesToDrop = 0;
1984}
1985
1986}; // namespace android
1987