Tracks.cpp revision e659ef420dae0caae84ab78f9df8952acb9ad3a0 
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 <sys/syscall.h>
25#include <utils/Log.h>
26
27#include <private/media/AudioTrackShared.h>
28
29#include <common_time/cc_helper.h>
30#include <common_time/local_clock.h>
31
32#include "AudioMixer.h"
33#include "AudioFlinger.h"
34#include "ServiceUtilities.h"
35
36#include <media/nbaio/Pipe.h>
37#include <media/nbaio/PipeReader.h>
38#include <audio_utils/minifloat.h>
39
40// ----------------------------------------------------------------------------
41
42// Note: the following macro is used for extremely verbose logging message.  In
43// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
44// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
45// are so verbose that we want to suppress them even when we have ALOG_ASSERT
46// turned on.  Do not uncomment the #def below unless you really know what you
47// are doing and want to see all of the extremely verbose messages.
48//#define VERY_VERY_VERBOSE_LOGGING
49#ifdef VERY_VERY_VERBOSE_LOGGING
50#define ALOGVV ALOGV
51#else
52#define ALOGVV(a...) do { } while(0)
53#endif
54
55namespace android {
56
57// ----------------------------------------------------------------------------
58//      TrackBase
59// ----------------------------------------------------------------------------
60
61static volatile int32_t nextTrackId = 55;
62
63// TrackBase constructor must be called with AudioFlinger::mLock held
64AudioFlinger::ThreadBase::TrackBase::TrackBase(
65            ThreadBase *thread,
66            const sp<Client>& client,
67            uint32_t sampleRate,
68            audio_format_t format,
69            audio_channel_mask_t channelMask,
70            size_t frameCount,
71            void *buffer,
72            int sessionId,
73            int clientUid,
74            IAudioFlinger::track_flags_t flags,
75            bool isOut,
76            alloc_type alloc,
77            track_type type)
78    :   RefBase(),
79        mThread(thread),
80        mClient(client),
81        mCblk(NULL),
82        // mBuffer
83        mState(IDLE),
84        mSampleRate(sampleRate),
85        mFormat(format),
86        mChannelMask(channelMask),
87        mChannelCount(isOut ?
88                audio_channel_count_from_out_mask(channelMask) :
89                audio_channel_count_from_in_mask(channelMask)),
90        mFrameSize(audio_is_linear_pcm(format) ?
91                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
92        mFrameCount(frameCount),
93        mSessionId(sessionId),
94        mFlags(flags),
95        mIsOut(isOut),
96        mServerProxy(NULL),
97        mId(android_atomic_inc(&nextTrackId)),
98        mTerminated(false),
99        mType(type),
100        mThreadIoHandle(thread->id())
101{
102    // if the caller is us, trust the specified uid
103    if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
104        int newclientUid = IPCThreadState::self()->getCallingUid();
105        if (clientUid != -1 && clientUid != newclientUid) {
106            ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
107        }
108        clientUid = newclientUid;
109    }
110    // clientUid contains the uid of the app that is responsible for this track, so we can blame
111    // battery usage on it.
112    mUid = clientUid;
113
114    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
115    size_t size = sizeof(audio_track_cblk_t);
116    size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize;
117    if (buffer == NULL && alloc == ALLOC_CBLK) {
118        size += bufferSize;
119    }
120
121    if (client != 0) {
122        mCblkMemory = client->heap()->allocate(size);
123        if (mCblkMemory == 0 ||
124                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
125            ALOGE("not enough memory for AudioTrack size=%u", size);
126            client->heap()->dump("AudioTrack");
127            mCblkMemory.clear();
128            return;
129        }
130    } else {
131        // this syntax avoids calling the audio_track_cblk_t constructor twice
132        mCblk = (audio_track_cblk_t *) new uint8_t[size];
133        // assume mCblk != NULL
134    }
135
136    // construct the shared structure in-place.
137    if (mCblk != NULL) {
138        new(mCblk) audio_track_cblk_t();
139        switch (alloc) {
140        case ALLOC_READONLY: {
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            } break;
155        case ALLOC_PIPE:
156            mBufferMemory = thread->pipeMemory();
157            // mBuffer is the virtual address as seen from current process (mediaserver),
158            // and should normally be coming from mBufferMemory->pointer().
159            // However in this case the TrackBase does not reference the buffer directly.
160            // It should references the buffer via the pipe.
161            // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
162            mBuffer = NULL;
163            break;
164        case ALLOC_CBLK:
165            // clear all buffers
166            if (buffer == NULL) {
167                mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
168                memset(mBuffer, 0, bufferSize);
169            } else {
170                mBuffer = buffer;
171#if 0
172                mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
173#endif
174            }
175            break;
176        case ALLOC_LOCAL:
177            mBuffer = calloc(1, bufferSize);
178            break;
179        case ALLOC_NONE:
180            mBuffer = buffer;
181            break;
182        }
183
184#ifdef TEE_SINK
185        if (mTeeSinkTrackEnabled) {
186            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat);
187            if (Format_isValid(pipeFormat)) {
188                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
189                size_t numCounterOffers = 0;
190                const NBAIO_Format offers[1] = {pipeFormat};
191                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
192                ALOG_ASSERT(index == 0);
193                PipeReader *pipeReader = new PipeReader(*pipe);
194                numCounterOffers = 0;
195                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
196                ALOG_ASSERT(index == 0);
197                mTeeSink = pipe;
198                mTeeSource = pipeReader;
199            }
200        }
201#endif
202
203    }
204}
205
206status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const
207{
208    status_t status;
209    if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) {
210        status = cblk() != NULL ? NO_ERROR : NO_MEMORY;
211    } else {
212        status = getCblk() != 0 ? NO_ERROR : NO_MEMORY;
213    }
214    return status;
215}
216
217AudioFlinger::ThreadBase::TrackBase::~TrackBase()
218{
219#ifdef TEE_SINK
220    dumpTee(-1, mTeeSource, mId);
221#endif
222    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
223    delete mServerProxy;
224    if (mCblk != NULL) {
225        if (mClient == 0) {
226            delete mCblk;
227        } else {
228            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
229        }
230    }
231    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
232    if (mClient != 0) {
233        // Client destructor must run with AudioFlinger client mutex locked
234        Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
235        // If the client's reference count drops to zero, the associated destructor
236        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
237        // relying on the automatic clear() at end of scope.
238        mClient.clear();
239    }
240    // flush the binder command buffer
241    IPCThreadState::self()->flushCommands();
242}
243
244// AudioBufferProvider interface
245// getNextBuffer() = 0;
246// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
247void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
248{
249#ifdef TEE_SINK
250    if (mTeeSink != 0) {
251        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
252    }
253#endif
254
255    ServerProxy::Buffer buf;
256    buf.mFrameCount = buffer->frameCount;
257    buf.mRaw = buffer->raw;
258    buffer->frameCount = 0;
259    buffer->raw = NULL;
260    mServerProxy->releaseBuffer(&buf);
261}
262
263status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
264{
265    mSyncEvents.add(event);
266    return NO_ERROR;
267}
268
269// ----------------------------------------------------------------------------
270//      Playback
271// ----------------------------------------------------------------------------
272
273AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
274    : BnAudioTrack(),
275      mTrack(track)
276{
277}
278
279AudioFlinger::TrackHandle::~TrackHandle() {
280    // just stop the track on deletion, associated resources
281    // will be freed from the main thread once all pending buffers have
282    // been played. Unless it's not in the active track list, in which
283    // case we free everything now...
284    mTrack->destroy();
285}
286
287sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
288    return mTrack->getCblk();
289}
290
291status_t AudioFlinger::TrackHandle::start() {
292    return mTrack->start();
293}
294
295void AudioFlinger::TrackHandle::stop() {
296    mTrack->stop();
297}
298
299void AudioFlinger::TrackHandle::flush() {
300    mTrack->flush();
301}
302
303void AudioFlinger::TrackHandle::pause() {
304    mTrack->pause();
305}
306
307status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
308{
309    return mTrack->attachAuxEffect(EffectId);
310}
311
312status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
313                                                         sp<IMemory>* buffer) {
314    if (!mTrack->isTimedTrack())
315        return INVALID_OPERATION;
316
317    PlaybackThread::TimedTrack* tt =
318            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
319    return tt->allocateTimedBuffer(size, buffer);
320}
321
322status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
323                                                     int64_t pts) {
324    if (!mTrack->isTimedTrack())
325        return INVALID_OPERATION;
326
327    if (buffer == 0 || buffer->pointer() == NULL) {
328        ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
329        return BAD_VALUE;
330    }
331
332    PlaybackThread::TimedTrack* tt =
333            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
334    return tt->queueTimedBuffer(buffer, pts);
335}
336
337status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
338    const LinearTransform& xform, int target) {
339
340    if (!mTrack->isTimedTrack())
341        return INVALID_OPERATION;
342
343    PlaybackThread::TimedTrack* tt =
344            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
345    return tt->setMediaTimeTransform(
346        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
347}
348
349status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
350    return mTrack->setParameters(keyValuePairs);
351}
352
353status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
354{
355    return mTrack->getTimestamp(timestamp);
356}
357
358
359void AudioFlinger::TrackHandle::signal()
360{
361    return mTrack->signal();
362}
363
364status_t AudioFlinger::TrackHandle::onTransact(
365    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
366{
367    return BnAudioTrack::onTransact(code, data, reply, flags);
368}
369
370// ----------------------------------------------------------------------------
371
372// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
373AudioFlinger::PlaybackThread::Track::Track(
374            PlaybackThread *thread,
375            const sp<Client>& client,
376            audio_stream_type_t streamType,
377            uint32_t sampleRate,
378            audio_format_t format,
379            audio_channel_mask_t channelMask,
380            size_t frameCount,
381            void *buffer,
382            const sp<IMemory>& sharedBuffer,
383            int sessionId,
384            int uid,
385            IAudioFlinger::track_flags_t flags,
386            track_type type)
387    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
388                  (sharedBuffer != 0) ? sharedBuffer->pointer() : buffer,
389                  sessionId, uid, flags, true /*isOut*/,
390                  (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
391                  type),
392    mFillingUpStatus(FS_INVALID),
393    // mRetryCount initialized later when needed
394    mSharedBuffer(sharedBuffer),
395    mStreamType(streamType),
396    mName(-1),  // see note below
397    mMainBuffer(thread->mixBuffer()),
398    mAuxBuffer(NULL),
399    mAuxEffectId(0), mHasVolumeController(false),
400    mPresentationCompleteFrames(0),
401    mFastIndex(-1),
402    mCachedVolume(1.0),
403    mIsInvalid(false),
404    mAudioTrackServerProxy(NULL),
405    mResumeToStopping(false),
406    mFlushHwPending(false),
407    mPreviousValid(false),
408    mPreviousFramesWritten(0)
409    // mPreviousTimestamp
410{
411    // client == 0 implies sharedBuffer == 0
412    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
413
414    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
415            sharedBuffer->size());
416
417    if (mCblk == NULL) {
418        return;
419    }
420
421    if (sharedBuffer == 0) {
422        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
423                mFrameSize, !isExternalTrack(), sampleRate);
424    } else {
425        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
426                mFrameSize);
427    }
428    mServerProxy = mAudioTrackServerProxy;
429
430    mName = thread->getTrackName_l(channelMask, format, sessionId);
431    if (mName < 0) {
432        ALOGE("no more track names available");
433        return;
434    }
435    // only allocate a fast track index if we were able to allocate a normal track name
436    if (flags & IAudioFlinger::TRACK_FAST) {
437        mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
438        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
439        int i = __builtin_ctz(thread->mFastTrackAvailMask);
440        ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
441        // FIXME This is too eager.  We allocate a fast track index before the
442        //       fast track becomes active.  Since fast tracks are a scarce resource,
443        //       this means we are potentially denying other more important fast tracks from
444        //       being created.  It would be better to allocate the index dynamically.
445        mFastIndex = i;
446        // Read the initial underruns because this field is never cleared by the fast mixer
447        mObservedUnderruns = thread->getFastTrackUnderruns(i);
448        thread->mFastTrackAvailMask &= ~(1 << i);
449    }
450}
451
452AudioFlinger::PlaybackThread::Track::~Track()
453{
454    ALOGV("PlaybackThread::Track destructor");
455
456    // The destructor would clear mSharedBuffer,
457    // but it will not push the decremented reference count,
458    // leaving the client's IMemory dangling indefinitely.
459    // This prevents that leak.
460    if (mSharedBuffer != 0) {
461        mSharedBuffer.clear();
462    }
463}
464
465status_t AudioFlinger::PlaybackThread::Track::initCheck() const
466{
467    status_t status = TrackBase::initCheck();
468    if (status == NO_ERROR && mName < 0) {
469        status = NO_MEMORY;
470    }
471    return status;
472}
473
474void AudioFlinger::PlaybackThread::Track::destroy()
475{
476    // NOTE: destroyTrack_l() can remove a strong reference to this Track
477    // by removing it from mTracks vector, so there is a risk that this Tracks's
478    // destructor is called. As the destructor needs to lock mLock,
479    // we must acquire a strong reference on this Track before locking mLock
480    // here so that the destructor is called only when exiting this function.
481    // On the other hand, as long as Track::destroy() is only called by
482    // TrackHandle destructor, the TrackHandle still holds a strong ref on
483    // this Track with its member mTrack.
484    sp<Track> keep(this);
485    { // scope for mLock
486        bool wasActive = false;
487        sp<ThreadBase> thread = mThread.promote();
488        if (thread != 0) {
489            Mutex::Autolock _l(thread->mLock);
490            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
491            wasActive = playbackThread->destroyTrack_l(this);
492        }
493        if (isExternalTrack() && !wasActive) {
494            AudioSystem::releaseOutput(mThreadIoHandle);
495        }
496    }
497}
498
499/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
500{
501    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
502                  "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
503}
504
505void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
506{
507    gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
508    if (isFastTrack()) {
509        sprintf(buffer, "    F %2d", mFastIndex);
510    } else if (mName >= AudioMixer::TRACK0) {
511        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
512    } else {
513        sprintf(buffer, "    none");
514    }
515    track_state state = mState;
516    char stateChar;
517    if (isTerminated()) {
518        stateChar = 'T';
519    } else {
520        switch (state) {
521        case IDLE:
522            stateChar = 'I';
523            break;
524        case STOPPING_1:
525            stateChar = 's';
526            break;
527        case STOPPING_2:
528            stateChar = '5';
529            break;
530        case STOPPED:
531            stateChar = 'S';
532            break;
533        case RESUMING:
534            stateChar = 'R';
535            break;
536        case ACTIVE:
537            stateChar = 'A';
538            break;
539        case PAUSING:
540            stateChar = 'p';
541            break;
542        case PAUSED:
543            stateChar = 'P';
544            break;
545        case FLUSHED:
546            stateChar = 'F';
547            break;
548        default:
549            stateChar = '?';
550            break;
551        }
552    }
553    char nowInUnderrun;
554    switch (mObservedUnderruns.mBitFields.mMostRecent) {
555    case UNDERRUN_FULL:
556        nowInUnderrun = ' ';
557        break;
558    case UNDERRUN_PARTIAL:
559        nowInUnderrun = '<';
560        break;
561    case UNDERRUN_EMPTY:
562        nowInUnderrun = '*';
563        break;
564    default:
565        nowInUnderrun = '?';
566        break;
567    }
568    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
569                                 "%08X %p %p 0x%03X %9u%c\n",
570            active ? "yes" : "no",
571            (mClient == 0) ? getpid_cached : mClient->pid(),
572            mStreamType,
573            mFormat,
574            mChannelMask,
575            mSessionId,
576            mFrameCount,
577            stateChar,
578            mFillingUpStatus,
579            mAudioTrackServerProxy->getSampleRate(),
580            20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
581            20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
582            mCblk->mServer,
583            mMainBuffer,
584            mAuxBuffer,
585            mCblk->mFlags,
586            mAudioTrackServerProxy->getUnderrunFrames(),
587            nowInUnderrun);
588}
589
590uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
591    return mAudioTrackServerProxy->getSampleRate();
592}
593
594// AudioBufferProvider interface
595status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
596        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
597{
598    ServerProxy::Buffer buf;
599    size_t desiredFrames = buffer->frameCount;
600    buf.mFrameCount = desiredFrames;
601    status_t status = mServerProxy->obtainBuffer(&buf);
602    buffer->frameCount = buf.mFrameCount;
603    buffer->raw = buf.mRaw;
604    if (buf.mFrameCount == 0) {
605        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
606    }
607    return status;
608}
609
610// releaseBuffer() is not overridden
611
612// ExtendedAudioBufferProvider interface
613
614// Note that framesReady() takes a mutex on the control block using tryLock().
615// This could result in priority inversion if framesReady() is called by the normal mixer,
616// as the normal mixer thread runs at lower
617// priority than the client's callback thread:  there is a short window within framesReady()
618// during which the normal mixer could be preempted, and the client callback would block.
619// Another problem can occur if framesReady() is called by the fast mixer:
620// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
621// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
622size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
623    return mAudioTrackServerProxy->framesReady();
624}
625
626size_t AudioFlinger::PlaybackThread::Track::framesReleased() const
627{
628    return mAudioTrackServerProxy->framesReleased();
629}
630
631// Don't call for fast tracks; the framesReady() could result in priority inversion
632bool AudioFlinger::PlaybackThread::Track::isReady() const {
633    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
634        return true;
635    }
636
637    if (isStopping()) {
638        if (framesReady() > 0) {
639            mFillingUpStatus = FS_FILLED;
640        }
641        return true;
642    }
643
644    if (framesReady() >= mFrameCount ||
645            (mCblk->mFlags & CBLK_FORCEREADY)) {
646        mFillingUpStatus = FS_FILLED;
647        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
648        return true;
649    }
650    return false;
651}
652
653status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
654                                                    int triggerSession __unused)
655{
656    status_t status = NO_ERROR;
657    ALOGV("start(%d), calling pid %d session %d",
658            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
659
660    sp<ThreadBase> thread = mThread.promote();
661    if (thread != 0) {
662        if (isOffloaded()) {
663            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
664            Mutex::Autolock _lth(thread->mLock);
665            sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
666            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
667                    (ec != 0 && ec->isNonOffloadableEnabled())) {
668                invalidate();
669                return PERMISSION_DENIED;
670            }
671        }
672        Mutex::Autolock _lth(thread->mLock);
673        track_state state = mState;
674        // here the track could be either new, or restarted
675        // in both cases "unstop" the track
676
677        // initial state-stopping. next state-pausing.
678        // What if resume is called ?
679
680        if (state == PAUSED || state == PAUSING) {
681            if (mResumeToStopping) {
682                // happened we need to resume to STOPPING_1
683                mState = TrackBase::STOPPING_1;
684                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
685            } else {
686                mState = TrackBase::RESUMING;
687                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
688            }
689        } else {
690            mState = TrackBase::ACTIVE;
691            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
692        }
693
694        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
695        status = playbackThread->addTrack_l(this);
696        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
697            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
698            //  restore previous state if start was rejected by policy manager
699            if (status == PERMISSION_DENIED) {
700                mState = state;
701            }
702        }
703        // track was already in the active list, not a problem
704        if (status == ALREADY_EXISTS) {
705            status = NO_ERROR;
706        } else {
707            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
708            // It is usually unsafe to access the server proxy from a binder thread.
709            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
710            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
711            // and for fast tracks the track is not yet in the fast mixer thread's active set.
712            ServerProxy::Buffer buffer;
713            buffer.mFrameCount = 1;
714            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
715        }
716    } else {
717        status = BAD_VALUE;
718    }
719    return status;
720}
721
722void AudioFlinger::PlaybackThread::Track::stop()
723{
724    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
725    sp<ThreadBase> thread = mThread.promote();
726    if (thread != 0) {
727        Mutex::Autolock _l(thread->mLock);
728        track_state state = mState;
729        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
730            // If the track is not active (PAUSED and buffers full), flush buffers
731            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
732            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
733                reset();
734                mState = STOPPED;
735            } else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
736                mState = STOPPED;
737            } else {
738                // For fast tracks prepareTracks_l() will set state to STOPPING_2
739                // presentation is complete
740                // For an offloaded track this starts a drain and state will
741                // move to STOPPING_2 when drain completes and then STOPPED
742                mState = STOPPING_1;
743            }
744            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
745                    playbackThread);
746        }
747    }
748}
749
750void AudioFlinger::PlaybackThread::Track::pause()
751{
752    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
753    sp<ThreadBase> thread = mThread.promote();
754    if (thread != 0) {
755        Mutex::Autolock _l(thread->mLock);
756        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
757        switch (mState) {
758        case STOPPING_1:
759        case STOPPING_2:
760            if (!isOffloaded()) {
761                /* nothing to do if track is not offloaded */
762                break;
763            }
764
765            // Offloaded track was draining, we need to carry on draining when resumed
766            mResumeToStopping = true;
767            // fall through...
768        case ACTIVE:
769        case RESUMING:
770            mState = PAUSING;
771            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
772            playbackThread->broadcast_l();
773            break;
774
775        default:
776            break;
777        }
778    }
779}
780
781void AudioFlinger::PlaybackThread::Track::flush()
782{
783    ALOGV("flush(%d)", mName);
784    sp<ThreadBase> thread = mThread.promote();
785    if (thread != 0) {
786        Mutex::Autolock _l(thread->mLock);
787        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
788
789        if (isOffloaded()) {
790            // If offloaded we allow flush during any state except terminated
791            // and keep the track active to avoid problems if user is seeking
792            // rapidly and underlying hardware has a significant delay handling
793            // a pause
794            if (isTerminated()) {
795                return;
796            }
797
798            ALOGV("flush: offload flush");
799            reset();
800
801            if (mState == STOPPING_1 || mState == STOPPING_2) {
802                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
803                mState = ACTIVE;
804            }
805
806            if (mState == ACTIVE) {
807                ALOGV("flush called in active state, resetting buffer time out retry count");
808                mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
809            }
810
811            mFlushHwPending = true;
812            mResumeToStopping = false;
813        } else {
814            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
815                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
816                return;
817            }
818            // No point remaining in PAUSED state after a flush => go to
819            // FLUSHED state
820            mState = FLUSHED;
821            // do not reset the track if it is still in the process of being stopped or paused.
822            // this will be done by prepareTracks_l() when the track is stopped.
823            // prepareTracks_l() will see mState == FLUSHED, then
824            // remove from active track list, reset(), and trigger presentation complete
825            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
826                reset();
827                if (thread->type() == ThreadBase::DIRECT) {
828                    DirectOutputThread *t = (DirectOutputThread *)playbackThread;
829                    t->flushHw_l();
830                }
831            }
832        }
833        // Prevent flush being lost if the track is flushed and then resumed
834        // before mixer thread can run. This is important when offloading
835        // because the hardware buffer could hold a large amount of audio
836        playbackThread->broadcast_l();
837    }
838}
839
840// must be called with thread lock held
841void AudioFlinger::PlaybackThread::Track::flushAck()
842{
843    if (!isOffloaded())
844        return;
845
846    mFlushHwPending = false;
847}
848
849void AudioFlinger::PlaybackThread::Track::reset()
850{
851    // Do not reset twice to avoid discarding data written just after a flush and before
852    // the audioflinger thread detects the track is stopped.
853    if (!mResetDone) {
854        // Force underrun condition to avoid false underrun callback until first data is
855        // written to buffer
856        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
857        mFillingUpStatus = FS_FILLING;
858        mResetDone = true;
859        if (mState == FLUSHED) {
860            mState = IDLE;
861        }
862    }
863}
864
865status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
866{
867    sp<ThreadBase> thread = mThread.promote();
868    if (thread == 0) {
869        ALOGE("thread is dead");
870        return FAILED_TRANSACTION;
871    } else if ((thread->type() == ThreadBase::DIRECT) ||
872                    (thread->type() == ThreadBase::OFFLOAD)) {
873        return thread->setParameters(keyValuePairs);
874    } else {
875        return PERMISSION_DENIED;
876    }
877}
878
879status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
880{
881    // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
882    if (isFastTrack()) {
883        // FIXME no lock held to set mPreviousValid = false
884        return INVALID_OPERATION;
885    }
886    sp<ThreadBase> thread = mThread.promote();
887    if (thread == 0) {
888        // FIXME no lock held to set mPreviousValid = false
889        return INVALID_OPERATION;
890    }
891    Mutex::Autolock _l(thread->mLock);
892    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
893    if (!isOffloaded() && !isDirect()) {
894        if (!playbackThread->mLatchQValid) {
895            mPreviousValid = false;
896            return INVALID_OPERATION;
897        }
898        uint32_t unpresentedFrames =
899                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
900                playbackThread->mSampleRate;
901        uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
902        bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten;
903        if (framesWritten < unpresentedFrames) {
904            mPreviousValid = false;
905            return INVALID_OPERATION;
906        }
907        mPreviousFramesWritten = framesWritten;
908        uint32_t position = framesWritten - unpresentedFrames;
909        struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime;
910        if (checkPreviousTimestamp) {
911            if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec ||
912                    (time.tv_sec == mPreviousTimestamp.mTime.tv_sec &&
913                    time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) {
914                ALOGW("Time is going backwards");
915            }
916            // position can bobble slightly as an artifact; this hides the bobble
917            static const uint32_t MINIMUM_POSITION_DELTA = 8u;
918            if ((position <= mPreviousTimestamp.mPosition) ||
919                    (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) {
920                position = mPreviousTimestamp.mPosition;
921                time = mPreviousTimestamp.mTime;
922            }
923        }
924        timestamp.mPosition = position;
925        timestamp.mTime = time;
926        mPreviousTimestamp = timestamp;
927        mPreviousValid = true;
928        return NO_ERROR;
929    }
930
931    return playbackThread->getTimestamp_l(timestamp);
932}
933
934status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
935{
936    status_t status = DEAD_OBJECT;
937    sp<ThreadBase> thread = mThread.promote();
938    if (thread != 0) {
939        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
940        sp<AudioFlinger> af = mClient->audioFlinger();
941
942        Mutex::Autolock _l(af->mLock);
943
944        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
945
946        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
947            Mutex::Autolock _dl(playbackThread->mLock);
948            Mutex::Autolock _sl(srcThread->mLock);
949            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
950            if (chain == 0) {
951                return INVALID_OPERATION;
952            }
953
954            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
955            if (effect == 0) {
956                return INVALID_OPERATION;
957            }
958            srcThread->removeEffect_l(effect);
959            status = playbackThread->addEffect_l(effect);
960            if (status != NO_ERROR) {
961                srcThread->addEffect_l(effect);
962                return INVALID_OPERATION;
963            }
964            // removeEffect_l() has stopped the effect if it was active so it must be restarted
965            if (effect->state() == EffectModule::ACTIVE ||
966                    effect->state() == EffectModule::STOPPING) {
967                effect->start();
968            }
969
970            sp<EffectChain> dstChain = effect->chain().promote();
971            if (dstChain == 0) {
972                srcThread->addEffect_l(effect);
973                return INVALID_OPERATION;
974            }
975            AudioSystem::unregisterEffect(effect->id());
976            AudioSystem::registerEffect(&effect->desc(),
977                                        srcThread->id(),
978                                        dstChain->strategy(),
979                                        AUDIO_SESSION_OUTPUT_MIX,
980                                        effect->id());
981            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
982        }
983        status = playbackThread->attachAuxEffect(this, EffectId);
984    }
985    return status;
986}
987
988void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
989{
990    mAuxEffectId = EffectId;
991    mAuxBuffer = buffer;
992}
993
994bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
995                                                         size_t audioHalFrames)
996{
997    // a track is considered presented when the total number of frames written to audio HAL
998    // corresponds to the number of frames written when presentationComplete() is called for the
999    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
1000    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
1001    // to detect when all frames have been played. In this case framesWritten isn't
1002    // useful because it doesn't always reflect whether there is data in the h/w
1003    // buffers, particularly if a track has been paused and resumed during draining
1004    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
1005                      mPresentationCompleteFrames, framesWritten);
1006    if (mPresentationCompleteFrames == 0) {
1007        mPresentationCompleteFrames = framesWritten + audioHalFrames;
1008        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
1009                  mPresentationCompleteFrames, audioHalFrames);
1010    }
1011
1012    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
1013        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
1014        mAudioTrackServerProxy->setStreamEndDone();
1015        return true;
1016    }
1017    return false;
1018}
1019
1020void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
1021{
1022    for (size_t i = 0; i < mSyncEvents.size(); i++) {
1023        if (mSyncEvents[i]->type() == type) {
1024            mSyncEvents[i]->trigger();
1025            mSyncEvents.removeAt(i);
1026            i--;
1027        }
1028    }
1029}
1030
1031// implement VolumeBufferProvider interface
1032
1033gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
1034{
1035    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
1036    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
1037    gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
1038    float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
1039    float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
1040    // track volumes come from shared memory, so can't be trusted and must be clamped
1041    if (vl > GAIN_FLOAT_UNITY) {
1042        vl = GAIN_FLOAT_UNITY;
1043    }
1044    if (vr > GAIN_FLOAT_UNITY) {
1045        vr = GAIN_FLOAT_UNITY;
1046    }
1047    // now apply the cached master volume and stream type volume;
1048    // this is trusted but lacks any synchronization or barrier so may be stale
1049    float v = mCachedVolume;
1050    vl *= v;
1051    vr *= v;
1052    // re-combine into packed minifloat
1053    vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
1054    // FIXME look at mute, pause, and stop flags
1055    return vlr;
1056}
1057
1058status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
1059{
1060    if (isTerminated() || mState == PAUSED ||
1061            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
1062                                      (mState == STOPPED)))) {
1063        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
1064              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
1065        event->cancel();
1066        return INVALID_OPERATION;
1067    }
1068    (void) TrackBase::setSyncEvent(event);
1069    return NO_ERROR;
1070}
1071
1072void AudioFlinger::PlaybackThread::Track::invalidate()
1073{
1074    // FIXME should use proxy, and needs work
1075    audio_track_cblk_t* cblk = mCblk;
1076    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1077    android_atomic_release_store(0x40000000, &cblk->mFutex);
1078    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1079    (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
1080    mIsInvalid = true;
1081}
1082
1083void AudioFlinger::PlaybackThread::Track::signal()
1084{
1085    sp<ThreadBase> thread = mThread.promote();
1086    if (thread != 0) {
1087        PlaybackThread *t = (PlaybackThread *)thread.get();
1088        Mutex::Autolock _l(t->mLock);
1089        t->broadcast_l();
1090    }
1091}
1092
1093//To be called with thread lock held
1094bool AudioFlinger::PlaybackThread::Track::isResumePending() {
1095
1096    if (mState == RESUMING)
1097        return true;
1098    /* Resume is pending if track was stopping before pause was called */
1099    if (mState == STOPPING_1 &&
1100        mResumeToStopping)
1101        return true;
1102
1103    return false;
1104}
1105
1106//To be called with thread lock held
1107void AudioFlinger::PlaybackThread::Track::resumeAck() {
1108
1109
1110    if (mState == RESUMING)
1111        mState = ACTIVE;
1112
1113    // Other possibility of  pending resume is stopping_1 state
1114    // Do not update the state from stopping as this prevents
1115    // drain being called.
1116    if (mState == STOPPING_1) {
1117        mResumeToStopping = false;
1118    }
1119}
1120// ----------------------------------------------------------------------------
1121
1122sp<AudioFlinger::PlaybackThread::TimedTrack>
1123AudioFlinger::PlaybackThread::TimedTrack::create(
1124            PlaybackThread *thread,
1125            const sp<Client>& client,
1126            audio_stream_type_t streamType,
1127            uint32_t sampleRate,
1128            audio_format_t format,
1129            audio_channel_mask_t channelMask,
1130            size_t frameCount,
1131            const sp<IMemory>& sharedBuffer,
1132            int sessionId,
1133            int uid)
1134{
1135    if (!client->reserveTimedTrack())
1136        return 0;
1137
1138    return new TimedTrack(
1139        thread, client, streamType, sampleRate, format, channelMask, frameCount,
1140        sharedBuffer, sessionId, uid);
1141}
1142
1143AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
1144            PlaybackThread *thread,
1145            const sp<Client>& client,
1146            audio_stream_type_t streamType,
1147            uint32_t sampleRate,
1148            audio_format_t format,
1149            audio_channel_mask_t channelMask,
1150            size_t frameCount,
1151            const sp<IMemory>& sharedBuffer,
1152            int sessionId,
1153            int uid)
1154    : Track(thread, client, streamType, sampleRate, format, channelMask,
1155            frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer,
1156                    sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED),
1157      mQueueHeadInFlight(false),
1158      mTrimQueueHeadOnRelease(false),
1159      mFramesPendingInQueue(0),
1160      mTimedSilenceBuffer(NULL),
1161      mTimedSilenceBufferSize(0),
1162      mTimedAudioOutputOnTime(false),
1163      mMediaTimeTransformValid(false)
1164{
1165    LocalClock lc;
1166    mLocalTimeFreq = lc.getLocalFreq();
1167
1168    mLocalTimeToSampleTransform.a_zero = 0;
1169    mLocalTimeToSampleTransform.b_zero = 0;
1170    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1171    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1172    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1173                            &mLocalTimeToSampleTransform.a_to_b_denom);
1174
1175    mMediaTimeToSampleTransform.a_zero = 0;
1176    mMediaTimeToSampleTransform.b_zero = 0;
1177    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1178    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1179    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1180                            &mMediaTimeToSampleTransform.a_to_b_denom);
1181}
1182
1183AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1184    mClient->releaseTimedTrack();
1185    delete [] mTimedSilenceBuffer;
1186}
1187
1188status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1189    size_t size, sp<IMemory>* buffer) {
1190
1191    Mutex::Autolock _l(mTimedBufferQueueLock);
1192
1193    trimTimedBufferQueue_l();
1194
1195    // lazily initialize the shared memory heap for timed buffers
1196    if (mTimedMemoryDealer == NULL) {
1197        const int kTimedBufferHeapSize = 512 << 10;
1198
1199        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1200                                              "AudioFlingerTimed");
1201        if (mTimedMemoryDealer == NULL) {
1202            return NO_MEMORY;
1203        }
1204    }
1205
1206    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1207    if (newBuffer == 0 || newBuffer->pointer() == NULL) {
1208        return NO_MEMORY;
1209    }
1210
1211    *buffer = newBuffer;
1212    return NO_ERROR;
1213}
1214
1215// caller must hold mTimedBufferQueueLock
1216void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1217    int64_t mediaTimeNow;
1218    {
1219        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1220        if (!mMediaTimeTransformValid)
1221            return;
1222
1223        int64_t targetTimeNow;
1224        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1225            ? mCCHelper.getCommonTime(&targetTimeNow)
1226            : mCCHelper.getLocalTime(&targetTimeNow);
1227
1228        if (OK != res)
1229            return;
1230
1231        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1232                                                    &mediaTimeNow)) {
1233            return;
1234        }
1235    }
1236
1237    size_t trimEnd;
1238    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1239        int64_t bufEnd;
1240
1241        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1242            // We have a next buffer.  Just use its PTS as the PTS of the frame
1243            // following the last frame in this buffer.  If the stream is sparse
1244            // (ie, there are deliberate gaps left in the stream which should be
1245            // filled with silence by the TimedAudioTrack), then this can result
1246            // in one extra buffer being left un-trimmed when it could have
1247            // been.  In general, this is not typical, and we would rather
1248            // optimized away the TS calculation below for the more common case
1249            // where PTSes are contiguous.
1250            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1251        } else {
1252            // We have no next buffer.  Compute the PTS of the frame following
1253            // the last frame in this buffer by computing the duration of of
1254            // this frame in media time units and adding it to the PTS of the
1255            // buffer.
1256            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1257                               / mFrameSize;
1258
1259            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1260                                                                &bufEnd)) {
1261                ALOGE("Failed to convert frame count of %lld to media time"
1262                      " duration" " (scale factor %d/%u) in %s",
1263                      frameCount,
1264                      mMediaTimeToSampleTransform.a_to_b_numer,
1265                      mMediaTimeToSampleTransform.a_to_b_denom,
1266                      __PRETTY_FUNCTION__);
1267                break;
1268            }
1269            bufEnd += mTimedBufferQueue[trimEnd].pts();
1270        }
1271
1272        if (bufEnd > mediaTimeNow)
1273            break;
1274
1275        // Is the buffer we want to use in the middle of a mix operation right
1276        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1277        // from the mixer which should be coming back shortly.
1278        if (!trimEnd && mQueueHeadInFlight) {
1279            mTrimQueueHeadOnRelease = true;
1280        }
1281    }
1282
1283    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1284    if (trimStart < trimEnd) {
1285        // Update the bookkeeping for framesReady()
1286        for (size_t i = trimStart; i < trimEnd; ++i) {
1287            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1288        }
1289
1290        // Now actually remove the buffers from the queue.
1291        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1292    }
1293}
1294
1295void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1296        const char* logTag) {
1297    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1298                "%s called (reason \"%s\"), but timed buffer queue has no"
1299                " elements to trim.", __FUNCTION__, logTag);
1300
1301    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1302    mTimedBufferQueue.removeAt(0);
1303}
1304
1305void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1306        const TimedBuffer& buf,
1307        const char* logTag __unused) {
1308    uint32_t bufBytes        = buf.buffer()->size();
1309    uint32_t consumedAlready = buf.position();
1310
1311    ALOG_ASSERT(consumedAlready <= bufBytes,
1312                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1313                " only %u bytes long, but claims to have consumed %u"
1314                " bytes.  (update reason: \"%s\")",
1315                bufBytes, consumedAlready, logTag);
1316
1317    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1318    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1319                "Bad bookkeeping while updating frames pending.  Should have at"
1320                " least %u queued frames, but we think we have only %u.  (update"
1321                " reason: \"%s\")",
1322                bufFrames, mFramesPendingInQueue, logTag);
1323
1324    mFramesPendingInQueue -= bufFrames;
1325}
1326
1327status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1328    const sp<IMemory>& buffer, int64_t pts) {
1329
1330    {
1331        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1332        if (!mMediaTimeTransformValid)
1333            return INVALID_OPERATION;
1334    }
1335
1336    Mutex::Autolock _l(mTimedBufferQueueLock);
1337
1338    uint32_t bufFrames = buffer->size() / mFrameSize;
1339    mFramesPendingInQueue += bufFrames;
1340    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1341
1342    return NO_ERROR;
1343}
1344
1345status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1346    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1347
1348    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1349           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1350           target);
1351
1352    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1353          target == TimedAudioTrack::COMMON_TIME)) {
1354        return BAD_VALUE;
1355    }
1356
1357    Mutex::Autolock lock(mMediaTimeTransformLock);
1358    mMediaTimeTransform = xform;
1359    mMediaTimeTransformTarget = target;
1360    mMediaTimeTransformValid = true;
1361
1362    return NO_ERROR;
1363}
1364
1365#define min(a, b) ((a) < (b) ? (a) : (b))
1366
1367// implementation of getNextBuffer for tracks whose buffers have timestamps
1368status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1369    AudioBufferProvider::Buffer* buffer, int64_t pts)
1370{
1371    if (pts == AudioBufferProvider::kInvalidPTS) {
1372        buffer->raw = NULL;
1373        buffer->frameCount = 0;
1374        mTimedAudioOutputOnTime = false;
1375        return INVALID_OPERATION;
1376    }
1377
1378    Mutex::Autolock _l(mTimedBufferQueueLock);
1379
1380    ALOG_ASSERT(!mQueueHeadInFlight,
1381                "getNextBuffer called without releaseBuffer!");
1382
1383    while (true) {
1384
1385        // if we have no timed buffers, then fail
1386        if (mTimedBufferQueue.isEmpty()) {
1387            buffer->raw = NULL;
1388            buffer->frameCount = 0;
1389            return NOT_ENOUGH_DATA;
1390        }
1391
1392        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1393
1394        // calculate the PTS of the head of the timed buffer queue expressed in
1395        // local time
1396        int64_t headLocalPTS;
1397        {
1398            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1399
1400            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1401
1402            if (mMediaTimeTransform.a_to_b_denom == 0) {
1403                // the transform represents a pause, so yield silence
1404                timedYieldSilence_l(buffer->frameCount, buffer);
1405                return NO_ERROR;
1406            }
1407
1408            int64_t transformedPTS;
1409            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1410                                                        &transformedPTS)) {
1411                // the transform failed.  this shouldn't happen, but if it does
1412                // then just drop this buffer
1413                ALOGW("timedGetNextBuffer transform failed");
1414                buffer->raw = NULL;
1415                buffer->frameCount = 0;
1416                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1417                return NO_ERROR;
1418            }
1419
1420            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1421                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1422                                                          &headLocalPTS)) {
1423                    buffer->raw = NULL;
1424                    buffer->frameCount = 0;
1425                    return INVALID_OPERATION;
1426                }
1427            } else {
1428                headLocalPTS = transformedPTS;
1429            }
1430        }
1431
1432        uint32_t sr = sampleRate();
1433
1434        // adjust the head buffer's PTS to reflect the portion of the head buffer
1435        // that has already been consumed
1436        int64_t effectivePTS = headLocalPTS +
1437                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1438
1439        // Calculate the delta in samples between the head of the input buffer
1440        // queue and the start of the next output buffer that will be written.
1441        // If the transformation fails because of over or underflow, it means
1442        // that the sample's position in the output stream is so far out of
1443        // whack that it should just be dropped.
1444        int64_t sampleDelta;
1445        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1446            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1447            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1448                                       " mix");
1449            continue;
1450        }
1451        if (!mLocalTimeToSampleTransform.doForwardTransform(
1452                (effectivePTS - pts) << 32, &sampleDelta)) {
1453            ALOGV("*** too late during sample rate transform: dropped buffer");
1454            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1455            continue;
1456        }
1457
1458        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1459               " sampleDelta=[%d.%08x]",
1460               head.pts(), head.position(), pts,
1461               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1462                   + (sampleDelta >> 32)),
1463               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1464
1465        // if the delta between the ideal placement for the next input sample and
1466        // the current output position is within this threshold, then we will
1467        // concatenate the next input samples to the previous output
1468        const int64_t kSampleContinuityThreshold =
1469                (static_cast<int64_t>(sr) << 32) / 250;
1470
1471        // if this is the first buffer of audio that we're emitting from this track
1472        // then it should be almost exactly on time.
1473        const int64_t kSampleStartupThreshold = 1LL << 32;
1474
1475        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1476           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1477            // the next input is close enough to being on time, so concatenate it
1478            // with the last output
1479            timedYieldSamples_l(buffer);
1480
1481            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1482                    head.position(), buffer->frameCount);
1483            return NO_ERROR;
1484        }
1485
1486        // Looks like our output is not on time.  Reset our on timed status.
1487        // Next time we mix samples from our input queue, then should be within
1488        // the StartupThreshold.
1489        mTimedAudioOutputOnTime = false;
1490        if (sampleDelta > 0) {
1491            // the gap between the current output position and the proper start of
1492            // the next input sample is too big, so fill it with silence
1493            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1494
1495            timedYieldSilence_l(framesUntilNextInput, buffer);
1496            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1497            return NO_ERROR;
1498        } else {
1499            // the next input sample is late
1500            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1501            size_t onTimeSamplePosition =
1502                    head.position() + lateFrames * mFrameSize;
1503
1504            if (onTimeSamplePosition > head.buffer()->size()) {
1505                // all the remaining samples in the head are too late, so
1506                // drop it and move on
1507                ALOGV("*** too late: dropped buffer");
1508                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1509                continue;
1510            } else {
1511                // skip over the late samples
1512                head.setPosition(onTimeSamplePosition);
1513
1514                // yield the available samples
1515                timedYieldSamples_l(buffer);
1516
1517                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1518                return NO_ERROR;
1519            }
1520        }
1521    }
1522}
1523
1524// Yield samples from the timed buffer queue head up to the given output
1525// buffer's capacity.
1526//
1527// Caller must hold mTimedBufferQueueLock
1528void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1529    AudioBufferProvider::Buffer* buffer) {
1530
1531    const TimedBuffer& head = mTimedBufferQueue[0];
1532
1533    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1534                   head.position());
1535
1536    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1537                                 mFrameSize);
1538    size_t framesRequested = buffer->frameCount;
1539    buffer->frameCount = min(framesLeftInHead, framesRequested);
1540
1541    mQueueHeadInFlight = true;
1542    mTimedAudioOutputOnTime = true;
1543}
1544
1545// Yield samples of silence up to the given output buffer's capacity
1546//
1547// Caller must hold mTimedBufferQueueLock
1548void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1549    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1550
1551    // lazily allocate a buffer filled with silence
1552    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1553        delete [] mTimedSilenceBuffer;
1554        mTimedSilenceBufferSize = numFrames * mFrameSize;
1555        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1556        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1557    }
1558
1559    buffer->raw = mTimedSilenceBuffer;
1560    size_t framesRequested = buffer->frameCount;
1561    buffer->frameCount = min(numFrames, framesRequested);
1562
1563    mTimedAudioOutputOnTime = false;
1564}
1565
1566// AudioBufferProvider interface
1567void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1568    AudioBufferProvider::Buffer* buffer) {
1569
1570    Mutex::Autolock _l(mTimedBufferQueueLock);
1571
1572    // If the buffer which was just released is part of the buffer at the head
1573    // of the queue, be sure to update the amt of the buffer which has been
1574    // consumed.  If the buffer being returned is not part of the head of the
1575    // queue, its either because the buffer is part of the silence buffer, or
1576    // because the head of the timed queue was trimmed after the mixer called
1577    // getNextBuffer but before the mixer called releaseBuffer.
1578    if (buffer->raw == mTimedSilenceBuffer) {
1579        ALOG_ASSERT(!mQueueHeadInFlight,
1580                    "Queue head in flight during release of silence buffer!");
1581        goto done;
1582    }
1583
1584    ALOG_ASSERT(mQueueHeadInFlight,
1585                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1586                " head in flight.");
1587
1588    if (mTimedBufferQueue.size()) {
1589        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1590
1591        void* start = head.buffer()->pointer();
1592        void* end   = reinterpret_cast<void*>(
1593                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1594                        + head.buffer()->size());
1595
1596        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1597                    "released buffer not within the head of the timed buffer"
1598                    " queue; qHead = [%p, %p], released buffer = %p",
1599                    start, end, buffer->raw);
1600
1601        head.setPosition(head.position() +
1602                (buffer->frameCount * mFrameSize));
1603        mQueueHeadInFlight = false;
1604
1605        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1606                    "Bad bookkeeping during releaseBuffer!  Should have at"
1607                    " least %u queued frames, but we think we have only %u",
1608                    buffer->frameCount, mFramesPendingInQueue);
1609
1610        mFramesPendingInQueue -= buffer->frameCount;
1611
1612        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1613            || mTrimQueueHeadOnRelease) {
1614            trimTimedBufferQueueHead_l("releaseBuffer");
1615            mTrimQueueHeadOnRelease = false;
1616        }
1617    } else {
1618        LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1619                  " buffers in the timed buffer queue");
1620    }
1621
1622done:
1623    buffer->raw = 0;
1624    buffer->frameCount = 0;
1625}
1626
1627size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1628    Mutex::Autolock _l(mTimedBufferQueueLock);
1629    return mFramesPendingInQueue;
1630}
1631
1632AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1633        : mPTS(0), mPosition(0) {}
1634
1635AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1636    const sp<IMemory>& buffer, int64_t pts)
1637        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1638
1639
1640// ----------------------------------------------------------------------------
1641
1642AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1643            PlaybackThread *playbackThread,
1644            DuplicatingThread *sourceThread,
1645            uint32_t sampleRate,
1646            audio_format_t format,
1647            audio_channel_mask_t channelMask,
1648            size_t frameCount,
1649            int uid)
1650    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1651                NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT),
1652    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1653{
1654
1655    if (mCblk != NULL) {
1656        mOutBuffer.frameCount = 0;
1657        playbackThread->mTracks.add(this);
1658        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1659                "frameCount %u, mChannelMask 0x%08x",
1660                mCblk, mBuffer,
1661                frameCount, mChannelMask);
1662        // since client and server are in the same process,
1663        // the buffer has the same virtual address on both sides
1664        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1665                true /*clientInServer*/);
1666        mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
1667        mClientProxy->setSendLevel(0.0);
1668        mClientProxy->setSampleRate(sampleRate);
1669    } else {
1670        ALOGW("Error creating output track on thread %p", playbackThread);
1671    }
1672}
1673
1674AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1675{
1676    clearBufferQueue();
1677    delete mClientProxy;
1678    // superclass destructor will now delete the server proxy and shared memory both refer to
1679}
1680
1681status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1682                                                          int triggerSession)
1683{
1684    status_t status = Track::start(event, triggerSession);
1685    if (status != NO_ERROR) {
1686        return status;
1687    }
1688
1689    mActive = true;
1690    mRetryCount = 127;
1691    return status;
1692}
1693
1694void AudioFlinger::PlaybackThread::OutputTrack::stop()
1695{
1696    Track::stop();
1697    clearBufferQueue();
1698    mOutBuffer.frameCount = 0;
1699    mActive = false;
1700}
1701
1702bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1703{
1704    Buffer *pInBuffer;
1705    Buffer inBuffer;
1706    uint32_t channelCount = mChannelCount;
1707    bool outputBufferFull = false;
1708    inBuffer.frameCount = frames;
1709    inBuffer.i16 = data;
1710
1711    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1712
1713    if (!mActive && frames != 0) {
1714        start();
1715        sp<ThreadBase> thread = mThread.promote();
1716        if (thread != 0) {
1717            MixerThread *mixerThread = (MixerThread *)thread.get();
1718            if (mFrameCount > frames) {
1719                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1720                    uint32_t startFrames = (mFrameCount - frames);
1721                    pInBuffer = new Buffer;
1722                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1723                    pInBuffer->frameCount = startFrames;
1724                    pInBuffer->i16 = pInBuffer->mBuffer;
1725                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1726                    mBufferQueue.add(pInBuffer);
1727                } else {
1728                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1729                }
1730            }
1731        }
1732    }
1733
1734    while (waitTimeLeftMs) {
1735        // First write pending buffers, then new data
1736        if (mBufferQueue.size()) {
1737            pInBuffer = mBufferQueue.itemAt(0);
1738        } else {
1739            pInBuffer = &inBuffer;
1740        }
1741
1742        if (pInBuffer->frameCount == 0) {
1743            break;
1744        }
1745
1746        if (mOutBuffer.frameCount == 0) {
1747            mOutBuffer.frameCount = pInBuffer->frameCount;
1748            nsecs_t startTime = systemTime();
1749            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1750            if (status != NO_ERROR) {
1751                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1752                        mThread.unsafe_get(), status);
1753                outputBufferFull = true;
1754                break;
1755            }
1756            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1757            if (waitTimeLeftMs >= waitTimeMs) {
1758                waitTimeLeftMs -= waitTimeMs;
1759            } else {
1760                waitTimeLeftMs = 0;
1761            }
1762        }
1763
1764        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1765                pInBuffer->frameCount;
1766        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1767        Proxy::Buffer buf;
1768        buf.mFrameCount = outFrames;
1769        buf.mRaw = NULL;
1770        mClientProxy->releaseBuffer(&buf);
1771        pInBuffer->frameCount -= outFrames;
1772        pInBuffer->i16 += outFrames * channelCount;
1773        mOutBuffer.frameCount -= outFrames;
1774        mOutBuffer.i16 += outFrames * channelCount;
1775
1776        if (pInBuffer->frameCount == 0) {
1777            if (mBufferQueue.size()) {
1778                mBufferQueue.removeAt(0);
1779                delete [] pInBuffer->mBuffer;
1780                delete pInBuffer;
1781                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1782                        mThread.unsafe_get(), mBufferQueue.size());
1783            } else {
1784                break;
1785            }
1786        }
1787    }
1788
1789    // If we could not write all frames, allocate a buffer and queue it for next time.
1790    if (inBuffer.frameCount) {
1791        sp<ThreadBase> thread = mThread.promote();
1792        if (thread != 0 && !thread->standby()) {
1793            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1794                pInBuffer = new Buffer;
1795                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1796                pInBuffer->frameCount = inBuffer.frameCount;
1797                pInBuffer->i16 = pInBuffer->mBuffer;
1798                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1799                        sizeof(int16_t));
1800                mBufferQueue.add(pInBuffer);
1801                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1802                        mThread.unsafe_get(), mBufferQueue.size());
1803            } else {
1804                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1805                        mThread.unsafe_get(), this);
1806            }
1807        }
1808    }
1809
1810    // Calling write() with a 0 length buffer, means that no more data will be written:
1811    // If no more buffers are pending, fill output track buffer to make sure it is started
1812    // by output mixer.
1813    if (frames == 0 && mBufferQueue.size() == 0) {
1814        // FIXME borken, replace by getting framesReady() from proxy
1815        size_t user = 0;    // was mCblk->user
1816        if (user < mFrameCount) {
1817            frames = mFrameCount - user;
1818            pInBuffer = new Buffer;
1819            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1820            pInBuffer->frameCount = frames;
1821            pInBuffer->i16 = pInBuffer->mBuffer;
1822            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1823            mBufferQueue.add(pInBuffer);
1824        } else if (mActive) {
1825            stop();
1826        }
1827    }
1828
1829    return outputBufferFull;
1830}
1831
1832status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1833        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1834{
1835    ClientProxy::Buffer buf;
1836    buf.mFrameCount = buffer->frameCount;
1837    struct timespec timeout;
1838    timeout.tv_sec = waitTimeMs / 1000;
1839    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1840    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1841    buffer->frameCount = buf.mFrameCount;
1842    buffer->raw = buf.mRaw;
1843    return status;
1844}
1845
1846void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1847{
1848    size_t size = mBufferQueue.size();
1849
1850    for (size_t i = 0; i < size; i++) {
1851        Buffer *pBuffer = mBufferQueue.itemAt(i);
1852        delete [] pBuffer->mBuffer;
1853        delete pBuffer;
1854    }
1855    mBufferQueue.clear();
1856}
1857
1858
1859AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread,
1860                                                     uint32_t sampleRate,
1861                                                     audio_channel_mask_t channelMask,
1862                                                     audio_format_t format,
1863                                                     size_t frameCount,
1864                                                     void *buffer,
1865                                                     IAudioFlinger::track_flags_t flags)
1866    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1867              buffer, 0, 0, getuid(), flags, TYPE_PATCH),
1868              mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true))
1869{
1870    uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) /
1871                                                                    playbackThread->sampleRate();
1872    mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
1873    mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
1874
1875    ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec",
1876                                      this, sampleRate,
1877                                      (int)mPeerTimeout.tv_sec,
1878                                      (int)(mPeerTimeout.tv_nsec / 1000000));
1879}
1880
1881AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack()
1882{
1883}
1884
1885// AudioBufferProvider interface
1886status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer(
1887        AudioBufferProvider::Buffer* buffer, int64_t pts)
1888{
1889    ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy");
1890    Proxy::Buffer buf;
1891    buf.mFrameCount = buffer->frameCount;
1892    status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
1893    ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status);
1894    buffer->frameCount = buf.mFrameCount;
1895    if (buf.mFrameCount == 0) {
1896        return WOULD_BLOCK;
1897    }
1898    status = Track::getNextBuffer(buffer, pts);
1899    return status;
1900}
1901
1902void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer)
1903{
1904    ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy");
1905    Proxy::Buffer buf;
1906    buf.mFrameCount = buffer->frameCount;
1907    buf.mRaw = buffer->raw;
1908    mPeerProxy->releaseBuffer(&buf);
1909    TrackBase::releaseBuffer(buffer);
1910}
1911
1912status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
1913                                                                const struct timespec *timeOut)
1914{
1915    return mProxy->obtainBuffer(buffer, timeOut);
1916}
1917
1918void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
1919{
1920    mProxy->releaseBuffer(buffer);
1921    if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) {
1922        ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting");
1923        start();
1924    }
1925    android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
1926}
1927
1928// ----------------------------------------------------------------------------
1929//      Record
1930// ----------------------------------------------------------------------------
1931
1932AudioFlinger::RecordHandle::RecordHandle(
1933        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1934    : BnAudioRecord(),
1935    mRecordTrack(recordTrack)
1936{
1937}
1938
1939AudioFlinger::RecordHandle::~RecordHandle() {
1940    stop_nonvirtual();
1941    mRecordTrack->destroy();
1942}
1943
1944status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1945        int triggerSession) {
1946    ALOGV("RecordHandle::start()");
1947    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1948}
1949
1950void AudioFlinger::RecordHandle::stop() {
1951    stop_nonvirtual();
1952}
1953
1954void AudioFlinger::RecordHandle::stop_nonvirtual() {
1955    ALOGV("RecordHandle::stop()");
1956    mRecordTrack->stop();
1957}
1958
1959status_t AudioFlinger::RecordHandle::onTransact(
1960    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1961{
1962    return BnAudioRecord::onTransact(code, data, reply, flags);
1963}
1964
1965// ----------------------------------------------------------------------------
1966
1967// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
1968AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1969            RecordThread *thread,
1970            const sp<Client>& client,
1971            uint32_t sampleRate,
1972            audio_format_t format,
1973            audio_channel_mask_t channelMask,
1974            size_t frameCount,
1975            void *buffer,
1976            int sessionId,
1977            int uid,
1978            IAudioFlinger::track_flags_t flags,
1979            track_type type)
1980    :   TrackBase(thread, client, sampleRate, format,
1981                  channelMask, frameCount, buffer, sessionId, uid,
1982                  flags, false /*isOut*/,
1983                  (type == TYPE_DEFAULT) ?
1984                          ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
1985                          ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
1986                  type),
1987        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
1988        // See real initialization of mRsmpInFront at RecordThread::start()
1989        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
1990{
1991    if (mCblk == NULL) {
1992        return;
1993    }
1994
1995    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
1996                                              mFrameSize, !isExternalTrack());
1997
1998    uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
1999    // FIXME I don't understand either of the channel count checks
2000    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
2001            channelCount <= FCC_2) {
2002        // sink SR
2003        mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
2004                thread->mChannelCount, sampleRate);
2005        // source SR
2006        mResampler->setSampleRate(thread->mSampleRate);
2007        mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
2008        mResamplerBufferProvider = new ResamplerBufferProvider(this);
2009    }
2010
2011    if (flags & IAudioFlinger::TRACK_FAST) {
2012        ALOG_ASSERT(thread->mFastTrackAvail);
2013        thread->mFastTrackAvail = false;
2014    }
2015}
2016
2017AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
2018{
2019    ALOGV("%s", __func__);
2020    delete mResampler;
2021    delete[] mRsmpOutBuffer;
2022    delete mResamplerBufferProvider;
2023}
2024
2025// AudioBufferProvider interface
2026status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
2027        int64_t pts __unused)
2028{
2029    ServerProxy::Buffer buf;
2030    buf.mFrameCount = buffer->frameCount;
2031    status_t status = mServerProxy->obtainBuffer(&buf);
2032    buffer->frameCount = buf.mFrameCount;
2033    buffer->raw = buf.mRaw;
2034    if (buf.mFrameCount == 0) {
2035        // FIXME also wake futex so that overrun is noticed more quickly
2036        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
2037    }
2038    return status;
2039}
2040
2041status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
2042                                                        int triggerSession)
2043{
2044    sp<ThreadBase> thread = mThread.promote();
2045    if (thread != 0) {
2046        RecordThread *recordThread = (RecordThread *)thread.get();
2047        return recordThread->start(this, event, triggerSession);
2048    } else {
2049        return BAD_VALUE;
2050    }
2051}
2052
2053void AudioFlinger::RecordThread::RecordTrack::stop()
2054{
2055    sp<ThreadBase> thread = mThread.promote();
2056    if (thread != 0) {
2057        RecordThread *recordThread = (RecordThread *)thread.get();
2058        if (recordThread->stop(this) && isExternalTrack()) {
2059            AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId);
2060        }
2061    }
2062}
2063
2064void AudioFlinger::RecordThread::RecordTrack::destroy()
2065{
2066    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
2067    sp<RecordTrack> keep(this);
2068    {
2069        if (isExternalTrack()) {
2070            if (mState == ACTIVE || mState == RESUMING) {
2071                AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId);
2072            }
2073            AudioSystem::releaseInput(mThreadIoHandle, (audio_session_t)mSessionId);
2074        }
2075        sp<ThreadBase> thread = mThread.promote();
2076        if (thread != 0) {
2077            Mutex::Autolock _l(thread->mLock);
2078            RecordThread *recordThread = (RecordThread *) thread.get();
2079            recordThread->destroyTrack_l(this);
2080        }
2081    }
2082}
2083
2084void AudioFlinger::RecordThread::RecordTrack::invalidate()
2085{
2086    // FIXME should use proxy, and needs work
2087    audio_track_cblk_t* cblk = mCblk;
2088    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
2089    android_atomic_release_store(0x40000000, &cblk->mFutex);
2090    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
2091    (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
2092}
2093
2094
2095/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
2096{
2097    result.append("    Active Client Fmt Chn mask Session S   Server fCount SRate\n");
2098}
2099
2100void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
2101{
2102    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n",
2103            active ? "yes" : "no",
2104            (mClient == 0) ? getpid_cached : mClient->pid(),
2105            mFormat,
2106            mChannelMask,
2107            mSessionId,
2108            mState,
2109            mCblk->mServer,
2110            mFrameCount,
2111            mSampleRate);
2112
2113}
2114
2115void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
2116{
2117    if (event == mSyncStartEvent) {
2118        ssize_t framesToDrop = 0;
2119        sp<ThreadBase> threadBase = mThread.promote();
2120        if (threadBase != 0) {
2121            // TODO: use actual buffer filling status instead of 2 buffers when info is available
2122            // from audio HAL
2123            framesToDrop = threadBase->mFrameCount * 2;
2124        }
2125        mFramesToDrop = framesToDrop;
2126    }
2127}
2128
2129void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
2130{
2131    if (mSyncStartEvent != 0) {
2132        mSyncStartEvent->cancel();
2133        mSyncStartEvent.clear();
2134    }
2135    mFramesToDrop = 0;
2136}
2137
2138
2139AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
2140                                                     uint32_t sampleRate,
2141                                                     audio_channel_mask_t channelMask,
2142                                                     audio_format_t format,
2143                                                     size_t frameCount,
2144                                                     void *buffer,
2145                                                     IAudioFlinger::track_flags_t flags)
2146    :   RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount,
2147                buffer, 0, getuid(), flags, TYPE_PATCH),
2148                mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true))
2149{
2150    uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) /
2151                                                                recordThread->sampleRate();
2152    mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
2153    mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
2154
2155    ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec",
2156                                      this, sampleRate,
2157                                      (int)mPeerTimeout.tv_sec,
2158                                      (int)(mPeerTimeout.tv_nsec / 1000000));
2159}
2160
2161AudioFlinger::RecordThread::PatchRecord::~PatchRecord()
2162{
2163}
2164
2165// AudioBufferProvider interface
2166status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
2167                                                  AudioBufferProvider::Buffer* buffer, int64_t pts)
2168{
2169    ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy");
2170    Proxy::Buffer buf;
2171    buf.mFrameCount = buffer->frameCount;
2172    status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
2173    ALOGV_IF(status != NO_ERROR,
2174             "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status);
2175    buffer->frameCount = buf.mFrameCount;
2176    if (buf.mFrameCount == 0) {
2177        return WOULD_BLOCK;
2178    }
2179    status = RecordTrack::getNextBuffer(buffer, pts);
2180    return status;
2181}
2182
2183void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer)
2184{
2185    ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy");
2186    Proxy::Buffer buf;
2187    buf.mFrameCount = buffer->frameCount;
2188    buf.mRaw = buffer->raw;
2189    mPeerProxy->releaseBuffer(&buf);
2190    TrackBase::releaseBuffer(buffer);
2191}
2192
2193status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer,
2194                                                               const struct timespec *timeOut)
2195{
2196    return mProxy->obtainBuffer(buffer, timeOut);
2197}
2198
2199void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer)
2200{
2201    mProxy->releaseBuffer(buffer);
2202}
2203
2204}; // namespace android
2205