Tracks.cpp revision 6427cf119ca2c32bc39731768c91b408f1e666b6 
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        // FIXME Since we're using a raw pointer as the key, it is theoretically possible
902        //       for a brand new track to share the same address as a recently destroyed
903        //       track, and thus for us to get the frames released of the wrong track.
904        //       It is unlikely that we would be able to call getTimestamp() so quickly
905        //       right after creating a new track.  Nevertheless, the index here should
906        //       be changed to something that is unique.  Or use a completely different strategy.
907        ssize_t i = playbackThread->mLatchQ.mFramesReleased.indexOfKey(this);
908        uint32_t framesWritten = i >= 0 ?
909                playbackThread->mLatchQ.mFramesReleased[i] :
910                mAudioTrackServerProxy->framesReleased();
911        bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten;
912        if (framesWritten < unpresentedFrames) {
913            mPreviousValid = false;
914            return INVALID_OPERATION;
915        }
916        mPreviousFramesWritten = framesWritten;
917        uint32_t position = framesWritten - unpresentedFrames;
918        struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime;
919        if (checkPreviousTimestamp) {
920            if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec ||
921                    (time.tv_sec == mPreviousTimestamp.mTime.tv_sec &&
922                    time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) {
923                ALOGW("Time is going backwards");
924            }
925            // position can bobble slightly as an artifact; this hides the bobble
926            static const uint32_t MINIMUM_POSITION_DELTA = 8u;
927            if ((position <= mPreviousTimestamp.mPosition) ||
928                    (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) {
929                position = mPreviousTimestamp.mPosition;
930                time = mPreviousTimestamp.mTime;
931            }
932        }
933        timestamp.mPosition = position;
934        timestamp.mTime = time;
935        mPreviousTimestamp = timestamp;
936        mPreviousValid = true;
937        return NO_ERROR;
938    }
939
940    return playbackThread->getTimestamp_l(timestamp);
941}
942
943status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
944{
945    status_t status = DEAD_OBJECT;
946    sp<ThreadBase> thread = mThread.promote();
947    if (thread != 0) {
948        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
949        sp<AudioFlinger> af = mClient->audioFlinger();
950
951        Mutex::Autolock _l(af->mLock);
952
953        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
954
955        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
956            Mutex::Autolock _dl(playbackThread->mLock);
957            Mutex::Autolock _sl(srcThread->mLock);
958            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
959            if (chain == 0) {
960                return INVALID_OPERATION;
961            }
962
963            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
964            if (effect == 0) {
965                return INVALID_OPERATION;
966            }
967            srcThread->removeEffect_l(effect);
968            status = playbackThread->addEffect_l(effect);
969            if (status != NO_ERROR) {
970                srcThread->addEffect_l(effect);
971                return INVALID_OPERATION;
972            }
973            // removeEffect_l() has stopped the effect if it was active so it must be restarted
974            if (effect->state() == EffectModule::ACTIVE ||
975                    effect->state() == EffectModule::STOPPING) {
976                effect->start();
977            }
978
979            sp<EffectChain> dstChain = effect->chain().promote();
980            if (dstChain == 0) {
981                srcThread->addEffect_l(effect);
982                return INVALID_OPERATION;
983            }
984            AudioSystem::unregisterEffect(effect->id());
985            AudioSystem::registerEffect(&effect->desc(),
986                                        srcThread->id(),
987                                        dstChain->strategy(),
988                                        AUDIO_SESSION_OUTPUT_MIX,
989                                        effect->id());
990            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
991        }
992        status = playbackThread->attachAuxEffect(this, EffectId);
993    }
994    return status;
995}
996
997void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
998{
999    mAuxEffectId = EffectId;
1000    mAuxBuffer = buffer;
1001}
1002
1003bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
1004                                                         size_t audioHalFrames)
1005{
1006    // a track is considered presented when the total number of frames written to audio HAL
1007    // corresponds to the number of frames written when presentationComplete() is called for the
1008    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
1009    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
1010    // to detect when all frames have been played. In this case framesWritten isn't
1011    // useful because it doesn't always reflect whether there is data in the h/w
1012    // buffers, particularly if a track has been paused and resumed during draining
1013    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
1014                      mPresentationCompleteFrames, framesWritten);
1015    if (mPresentationCompleteFrames == 0) {
1016        mPresentationCompleteFrames = framesWritten + audioHalFrames;
1017        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
1018                  mPresentationCompleteFrames, audioHalFrames);
1019    }
1020
1021    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
1022        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
1023        mAudioTrackServerProxy->setStreamEndDone();
1024        return true;
1025    }
1026    return false;
1027}
1028
1029void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
1030{
1031    for (size_t i = 0; i < mSyncEvents.size(); i++) {
1032        if (mSyncEvents[i]->type() == type) {
1033            mSyncEvents[i]->trigger();
1034            mSyncEvents.removeAt(i);
1035            i--;
1036        }
1037    }
1038}
1039
1040// implement VolumeBufferProvider interface
1041
1042gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
1043{
1044    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
1045    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
1046    gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
1047    float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
1048    float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
1049    // track volumes come from shared memory, so can't be trusted and must be clamped
1050    if (vl > GAIN_FLOAT_UNITY) {
1051        vl = GAIN_FLOAT_UNITY;
1052    }
1053    if (vr > GAIN_FLOAT_UNITY) {
1054        vr = GAIN_FLOAT_UNITY;
1055    }
1056    // now apply the cached master volume and stream type volume;
1057    // this is trusted but lacks any synchronization or barrier so may be stale
1058    float v = mCachedVolume;
1059    vl *= v;
1060    vr *= v;
1061    // re-combine into packed minifloat
1062    vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
1063    // FIXME look at mute, pause, and stop flags
1064    return vlr;
1065}
1066
1067status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
1068{
1069    if (isTerminated() || mState == PAUSED ||
1070            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
1071                                      (mState == STOPPED)))) {
1072        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
1073              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
1074        event->cancel();
1075        return INVALID_OPERATION;
1076    }
1077    (void) TrackBase::setSyncEvent(event);
1078    return NO_ERROR;
1079}
1080
1081void AudioFlinger::PlaybackThread::Track::invalidate()
1082{
1083    // FIXME should use proxy, and needs work
1084    audio_track_cblk_t* cblk = mCblk;
1085    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1086    android_atomic_release_store(0x40000000, &cblk->mFutex);
1087    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1088    (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
1089    mIsInvalid = true;
1090}
1091
1092void AudioFlinger::PlaybackThread::Track::signal()
1093{
1094    sp<ThreadBase> thread = mThread.promote();
1095    if (thread != 0) {
1096        PlaybackThread *t = (PlaybackThread *)thread.get();
1097        Mutex::Autolock _l(t->mLock);
1098        t->broadcast_l();
1099    }
1100}
1101
1102//To be called with thread lock held
1103bool AudioFlinger::PlaybackThread::Track::isResumePending() {
1104
1105    if (mState == RESUMING)
1106        return true;
1107    /* Resume is pending if track was stopping before pause was called */
1108    if (mState == STOPPING_1 &&
1109        mResumeToStopping)
1110        return true;
1111
1112    return false;
1113}
1114
1115//To be called with thread lock held
1116void AudioFlinger::PlaybackThread::Track::resumeAck() {
1117
1118
1119    if (mState == RESUMING)
1120        mState = ACTIVE;
1121
1122    // Other possibility of  pending resume is stopping_1 state
1123    // Do not update the state from stopping as this prevents
1124    // drain being called.
1125    if (mState == STOPPING_1) {
1126        mResumeToStopping = false;
1127    }
1128}
1129// ----------------------------------------------------------------------------
1130
1131sp<AudioFlinger::PlaybackThread::TimedTrack>
1132AudioFlinger::PlaybackThread::TimedTrack::create(
1133            PlaybackThread *thread,
1134            const sp<Client>& client,
1135            audio_stream_type_t streamType,
1136            uint32_t sampleRate,
1137            audio_format_t format,
1138            audio_channel_mask_t channelMask,
1139            size_t frameCount,
1140            const sp<IMemory>& sharedBuffer,
1141            int sessionId,
1142            int uid)
1143{
1144    if (!client->reserveTimedTrack())
1145        return 0;
1146
1147    return new TimedTrack(
1148        thread, client, streamType, sampleRate, format, channelMask, frameCount,
1149        sharedBuffer, sessionId, uid);
1150}
1151
1152AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
1153            PlaybackThread *thread,
1154            const sp<Client>& client,
1155            audio_stream_type_t streamType,
1156            uint32_t sampleRate,
1157            audio_format_t format,
1158            audio_channel_mask_t channelMask,
1159            size_t frameCount,
1160            const sp<IMemory>& sharedBuffer,
1161            int sessionId,
1162            int uid)
1163    : Track(thread, client, streamType, sampleRate, format, channelMask,
1164            frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer,
1165                    sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED),
1166      mQueueHeadInFlight(false),
1167      mTrimQueueHeadOnRelease(false),
1168      mFramesPendingInQueue(0),
1169      mTimedSilenceBuffer(NULL),
1170      mTimedSilenceBufferSize(0),
1171      mTimedAudioOutputOnTime(false),
1172      mMediaTimeTransformValid(false)
1173{
1174    LocalClock lc;
1175    mLocalTimeFreq = lc.getLocalFreq();
1176
1177    mLocalTimeToSampleTransform.a_zero = 0;
1178    mLocalTimeToSampleTransform.b_zero = 0;
1179    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1180    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1181    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1182                            &mLocalTimeToSampleTransform.a_to_b_denom);
1183
1184    mMediaTimeToSampleTransform.a_zero = 0;
1185    mMediaTimeToSampleTransform.b_zero = 0;
1186    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1187    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1188    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1189                            &mMediaTimeToSampleTransform.a_to_b_denom);
1190}
1191
1192AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1193    mClient->releaseTimedTrack();
1194    delete [] mTimedSilenceBuffer;
1195}
1196
1197status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1198    size_t size, sp<IMemory>* buffer) {
1199
1200    Mutex::Autolock _l(mTimedBufferQueueLock);
1201
1202    trimTimedBufferQueue_l();
1203
1204    // lazily initialize the shared memory heap for timed buffers
1205    if (mTimedMemoryDealer == NULL) {
1206        const int kTimedBufferHeapSize = 512 << 10;
1207
1208        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1209                                              "AudioFlingerTimed");
1210        if (mTimedMemoryDealer == NULL) {
1211            return NO_MEMORY;
1212        }
1213    }
1214
1215    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1216    if (newBuffer == 0 || newBuffer->pointer() == NULL) {
1217        return NO_MEMORY;
1218    }
1219
1220    *buffer = newBuffer;
1221    return NO_ERROR;
1222}
1223
1224// caller must hold mTimedBufferQueueLock
1225void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1226    int64_t mediaTimeNow;
1227    {
1228        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1229        if (!mMediaTimeTransformValid)
1230            return;
1231
1232        int64_t targetTimeNow;
1233        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1234            ? mCCHelper.getCommonTime(&targetTimeNow)
1235            : mCCHelper.getLocalTime(&targetTimeNow);
1236
1237        if (OK != res)
1238            return;
1239
1240        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1241                                                    &mediaTimeNow)) {
1242            return;
1243        }
1244    }
1245
1246    size_t trimEnd;
1247    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1248        int64_t bufEnd;
1249
1250        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1251            // We have a next buffer.  Just use its PTS as the PTS of the frame
1252            // following the last frame in this buffer.  If the stream is sparse
1253            // (ie, there are deliberate gaps left in the stream which should be
1254            // filled with silence by the TimedAudioTrack), then this can result
1255            // in one extra buffer being left un-trimmed when it could have
1256            // been.  In general, this is not typical, and we would rather
1257            // optimized away the TS calculation below for the more common case
1258            // where PTSes are contiguous.
1259            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1260        } else {
1261            // We have no next buffer.  Compute the PTS of the frame following
1262            // the last frame in this buffer by computing the duration of of
1263            // this frame in media time units and adding it to the PTS of the
1264            // buffer.
1265            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1266                               / mFrameSize;
1267
1268            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1269                                                                &bufEnd)) {
1270                ALOGE("Failed to convert frame count of %lld to media time"
1271                      " duration" " (scale factor %d/%u) in %s",
1272                      frameCount,
1273                      mMediaTimeToSampleTransform.a_to_b_numer,
1274                      mMediaTimeToSampleTransform.a_to_b_denom,
1275                      __PRETTY_FUNCTION__);
1276                break;
1277            }
1278            bufEnd += mTimedBufferQueue[trimEnd].pts();
1279        }
1280
1281        if (bufEnd > mediaTimeNow)
1282            break;
1283
1284        // Is the buffer we want to use in the middle of a mix operation right
1285        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1286        // from the mixer which should be coming back shortly.
1287        if (!trimEnd && mQueueHeadInFlight) {
1288            mTrimQueueHeadOnRelease = true;
1289        }
1290    }
1291
1292    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1293    if (trimStart < trimEnd) {
1294        // Update the bookkeeping for framesReady()
1295        for (size_t i = trimStart; i < trimEnd; ++i) {
1296            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1297        }
1298
1299        // Now actually remove the buffers from the queue.
1300        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1301    }
1302}
1303
1304void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1305        const char* logTag) {
1306    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1307                "%s called (reason \"%s\"), but timed buffer queue has no"
1308                " elements to trim.", __FUNCTION__, logTag);
1309
1310    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1311    mTimedBufferQueue.removeAt(0);
1312}
1313
1314void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1315        const TimedBuffer& buf,
1316        const char* logTag __unused) {
1317    uint32_t bufBytes        = buf.buffer()->size();
1318    uint32_t consumedAlready = buf.position();
1319
1320    ALOG_ASSERT(consumedAlready <= bufBytes,
1321                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1322                " only %u bytes long, but claims to have consumed %u"
1323                " bytes.  (update reason: \"%s\")",
1324                bufBytes, consumedAlready, logTag);
1325
1326    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1327    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1328                "Bad bookkeeping while updating frames pending.  Should have at"
1329                " least %u queued frames, but we think we have only %u.  (update"
1330                " reason: \"%s\")",
1331                bufFrames, mFramesPendingInQueue, logTag);
1332
1333    mFramesPendingInQueue -= bufFrames;
1334}
1335
1336status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1337    const sp<IMemory>& buffer, int64_t pts) {
1338
1339    {
1340        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1341        if (!mMediaTimeTransformValid)
1342            return INVALID_OPERATION;
1343    }
1344
1345    Mutex::Autolock _l(mTimedBufferQueueLock);
1346
1347    uint32_t bufFrames = buffer->size() / mFrameSize;
1348    mFramesPendingInQueue += bufFrames;
1349    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1350
1351    return NO_ERROR;
1352}
1353
1354status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1355    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1356
1357    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1358           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1359           target);
1360
1361    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1362          target == TimedAudioTrack::COMMON_TIME)) {
1363        return BAD_VALUE;
1364    }
1365
1366    Mutex::Autolock lock(mMediaTimeTransformLock);
1367    mMediaTimeTransform = xform;
1368    mMediaTimeTransformTarget = target;
1369    mMediaTimeTransformValid = true;
1370
1371    return NO_ERROR;
1372}
1373
1374#define min(a, b) ((a) < (b) ? (a) : (b))
1375
1376// implementation of getNextBuffer for tracks whose buffers have timestamps
1377status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1378    AudioBufferProvider::Buffer* buffer, int64_t pts)
1379{
1380    if (pts == AudioBufferProvider::kInvalidPTS) {
1381        buffer->raw = NULL;
1382        buffer->frameCount = 0;
1383        mTimedAudioOutputOnTime = false;
1384        return INVALID_OPERATION;
1385    }
1386
1387    Mutex::Autolock _l(mTimedBufferQueueLock);
1388
1389    ALOG_ASSERT(!mQueueHeadInFlight,
1390                "getNextBuffer called without releaseBuffer!");
1391
1392    while (true) {
1393
1394        // if we have no timed buffers, then fail
1395        if (mTimedBufferQueue.isEmpty()) {
1396            buffer->raw = NULL;
1397            buffer->frameCount = 0;
1398            return NOT_ENOUGH_DATA;
1399        }
1400
1401        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1402
1403        // calculate the PTS of the head of the timed buffer queue expressed in
1404        // local time
1405        int64_t headLocalPTS;
1406        {
1407            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1408
1409            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1410
1411            if (mMediaTimeTransform.a_to_b_denom == 0) {
1412                // the transform represents a pause, so yield silence
1413                timedYieldSilence_l(buffer->frameCount, buffer);
1414                return NO_ERROR;
1415            }
1416
1417            int64_t transformedPTS;
1418            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1419                                                        &transformedPTS)) {
1420                // the transform failed.  this shouldn't happen, but if it does
1421                // then just drop this buffer
1422                ALOGW("timedGetNextBuffer transform failed");
1423                buffer->raw = NULL;
1424                buffer->frameCount = 0;
1425                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1426                return NO_ERROR;
1427            }
1428
1429            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1430                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1431                                                          &headLocalPTS)) {
1432                    buffer->raw = NULL;
1433                    buffer->frameCount = 0;
1434                    return INVALID_OPERATION;
1435                }
1436            } else {
1437                headLocalPTS = transformedPTS;
1438            }
1439        }
1440
1441        uint32_t sr = sampleRate();
1442
1443        // adjust the head buffer's PTS to reflect the portion of the head buffer
1444        // that has already been consumed
1445        int64_t effectivePTS = headLocalPTS +
1446                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1447
1448        // Calculate the delta in samples between the head of the input buffer
1449        // queue and the start of the next output buffer that will be written.
1450        // If the transformation fails because of over or underflow, it means
1451        // that the sample's position in the output stream is so far out of
1452        // whack that it should just be dropped.
1453        int64_t sampleDelta;
1454        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1455            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1456            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1457                                       " mix");
1458            continue;
1459        }
1460        if (!mLocalTimeToSampleTransform.doForwardTransform(
1461                (effectivePTS - pts) << 32, &sampleDelta)) {
1462            ALOGV("*** too late during sample rate transform: dropped buffer");
1463            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1464            continue;
1465        }
1466
1467        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1468               " sampleDelta=[%d.%08x]",
1469               head.pts(), head.position(), pts,
1470               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1471                   + (sampleDelta >> 32)),
1472               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1473
1474        // if the delta between the ideal placement for the next input sample and
1475        // the current output position is within this threshold, then we will
1476        // concatenate the next input samples to the previous output
1477        const int64_t kSampleContinuityThreshold =
1478                (static_cast<int64_t>(sr) << 32) / 250;
1479
1480        // if this is the first buffer of audio that we're emitting from this track
1481        // then it should be almost exactly on time.
1482        const int64_t kSampleStartupThreshold = 1LL << 32;
1483
1484        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1485           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1486            // the next input is close enough to being on time, so concatenate it
1487            // with the last output
1488            timedYieldSamples_l(buffer);
1489
1490            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1491                    head.position(), buffer->frameCount);
1492            return NO_ERROR;
1493        }
1494
1495        // Looks like our output is not on time.  Reset our on timed status.
1496        // Next time we mix samples from our input queue, then should be within
1497        // the StartupThreshold.
1498        mTimedAudioOutputOnTime = false;
1499        if (sampleDelta > 0) {
1500            // the gap between the current output position and the proper start of
1501            // the next input sample is too big, so fill it with silence
1502            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1503
1504            timedYieldSilence_l(framesUntilNextInput, buffer);
1505            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1506            return NO_ERROR;
1507        } else {
1508            // the next input sample is late
1509            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1510            size_t onTimeSamplePosition =
1511                    head.position() + lateFrames * mFrameSize;
1512
1513            if (onTimeSamplePosition > head.buffer()->size()) {
1514                // all the remaining samples in the head are too late, so
1515                // drop it and move on
1516                ALOGV("*** too late: dropped buffer");
1517                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1518                continue;
1519            } else {
1520                // skip over the late samples
1521                head.setPosition(onTimeSamplePosition);
1522
1523                // yield the available samples
1524                timedYieldSamples_l(buffer);
1525
1526                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1527                return NO_ERROR;
1528            }
1529        }
1530    }
1531}
1532
1533// Yield samples from the timed buffer queue head up to the given output
1534// buffer's capacity.
1535//
1536// Caller must hold mTimedBufferQueueLock
1537void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1538    AudioBufferProvider::Buffer* buffer) {
1539
1540    const TimedBuffer& head = mTimedBufferQueue[0];
1541
1542    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1543                   head.position());
1544
1545    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1546                                 mFrameSize);
1547    size_t framesRequested = buffer->frameCount;
1548    buffer->frameCount = min(framesLeftInHead, framesRequested);
1549
1550    mQueueHeadInFlight = true;
1551    mTimedAudioOutputOnTime = true;
1552}
1553
1554// Yield samples of silence up to the given output buffer's capacity
1555//
1556// Caller must hold mTimedBufferQueueLock
1557void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1558    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1559
1560    // lazily allocate a buffer filled with silence
1561    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1562        delete [] mTimedSilenceBuffer;
1563        mTimedSilenceBufferSize = numFrames * mFrameSize;
1564        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1565        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1566    }
1567
1568    buffer->raw = mTimedSilenceBuffer;
1569    size_t framesRequested = buffer->frameCount;
1570    buffer->frameCount = min(numFrames, framesRequested);
1571
1572    mTimedAudioOutputOnTime = false;
1573}
1574
1575// AudioBufferProvider interface
1576void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1577    AudioBufferProvider::Buffer* buffer) {
1578
1579    Mutex::Autolock _l(mTimedBufferQueueLock);
1580
1581    // If the buffer which was just released is part of the buffer at the head
1582    // of the queue, be sure to update the amt of the buffer which has been
1583    // consumed.  If the buffer being returned is not part of the head of the
1584    // queue, its either because the buffer is part of the silence buffer, or
1585    // because the head of the timed queue was trimmed after the mixer called
1586    // getNextBuffer but before the mixer called releaseBuffer.
1587    if (buffer->raw == mTimedSilenceBuffer) {
1588        ALOG_ASSERT(!mQueueHeadInFlight,
1589                    "Queue head in flight during release of silence buffer!");
1590        goto done;
1591    }
1592
1593    ALOG_ASSERT(mQueueHeadInFlight,
1594                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1595                " head in flight.");
1596
1597    if (mTimedBufferQueue.size()) {
1598        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1599
1600        void* start = head.buffer()->pointer();
1601        void* end   = reinterpret_cast<void*>(
1602                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1603                        + head.buffer()->size());
1604
1605        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1606                    "released buffer not within the head of the timed buffer"
1607                    " queue; qHead = [%p, %p], released buffer = %p",
1608                    start, end, buffer->raw);
1609
1610        head.setPosition(head.position() +
1611                (buffer->frameCount * mFrameSize));
1612        mQueueHeadInFlight = false;
1613
1614        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1615                    "Bad bookkeeping during releaseBuffer!  Should have at"
1616                    " least %u queued frames, but we think we have only %u",
1617                    buffer->frameCount, mFramesPendingInQueue);
1618
1619        mFramesPendingInQueue -= buffer->frameCount;
1620
1621        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1622            || mTrimQueueHeadOnRelease) {
1623            trimTimedBufferQueueHead_l("releaseBuffer");
1624            mTrimQueueHeadOnRelease = false;
1625        }
1626    } else {
1627        LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1628                  " buffers in the timed buffer queue");
1629    }
1630
1631done:
1632    buffer->raw = 0;
1633    buffer->frameCount = 0;
1634}
1635
1636size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1637    Mutex::Autolock _l(mTimedBufferQueueLock);
1638    return mFramesPendingInQueue;
1639}
1640
1641AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1642        : mPTS(0), mPosition(0) {}
1643
1644AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1645    const sp<IMemory>& buffer, int64_t pts)
1646        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1647
1648
1649// ----------------------------------------------------------------------------
1650
1651AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1652            PlaybackThread *playbackThread,
1653            DuplicatingThread *sourceThread,
1654            uint32_t sampleRate,
1655            audio_format_t format,
1656            audio_channel_mask_t channelMask,
1657            size_t frameCount,
1658            int uid)
1659    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1660                NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT),
1661    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1662{
1663
1664    if (mCblk != NULL) {
1665        mOutBuffer.frameCount = 0;
1666        playbackThread->mTracks.add(this);
1667        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1668                "frameCount %u, mChannelMask 0x%08x",
1669                mCblk, mBuffer,
1670                frameCount, mChannelMask);
1671        // since client and server are in the same process,
1672        // the buffer has the same virtual address on both sides
1673        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1674                true /*clientInServer*/);
1675        mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
1676        mClientProxy->setSendLevel(0.0);
1677        mClientProxy->setSampleRate(sampleRate);
1678    } else {
1679        ALOGW("Error creating output track on thread %p", playbackThread);
1680    }
1681}
1682
1683AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1684{
1685    clearBufferQueue();
1686    delete mClientProxy;
1687    // superclass destructor will now delete the server proxy and shared memory both refer to
1688}
1689
1690status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1691                                                          int triggerSession)
1692{
1693    status_t status = Track::start(event, triggerSession);
1694    if (status != NO_ERROR) {
1695        return status;
1696    }
1697
1698    mActive = true;
1699    mRetryCount = 127;
1700    return status;
1701}
1702
1703void AudioFlinger::PlaybackThread::OutputTrack::stop()
1704{
1705    Track::stop();
1706    clearBufferQueue();
1707    mOutBuffer.frameCount = 0;
1708    mActive = false;
1709}
1710
1711bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1712{
1713    Buffer *pInBuffer;
1714    Buffer inBuffer;
1715    uint32_t channelCount = mChannelCount;
1716    bool outputBufferFull = false;
1717    inBuffer.frameCount = frames;
1718    inBuffer.i16 = data;
1719
1720    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1721
1722    if (!mActive && frames != 0) {
1723        start();
1724        sp<ThreadBase> thread = mThread.promote();
1725        if (thread != 0) {
1726            MixerThread *mixerThread = (MixerThread *)thread.get();
1727            if (mFrameCount > frames) {
1728                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1729                    uint32_t startFrames = (mFrameCount - frames);
1730                    pInBuffer = new Buffer;
1731                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1732                    pInBuffer->frameCount = startFrames;
1733                    pInBuffer->i16 = pInBuffer->mBuffer;
1734                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1735                    mBufferQueue.add(pInBuffer);
1736                } else {
1737                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1738                }
1739            }
1740        }
1741    }
1742
1743    while (waitTimeLeftMs) {
1744        // First write pending buffers, then new data
1745        if (mBufferQueue.size()) {
1746            pInBuffer = mBufferQueue.itemAt(0);
1747        } else {
1748            pInBuffer = &inBuffer;
1749        }
1750
1751        if (pInBuffer->frameCount == 0) {
1752            break;
1753        }
1754
1755        if (mOutBuffer.frameCount == 0) {
1756            mOutBuffer.frameCount = pInBuffer->frameCount;
1757            nsecs_t startTime = systemTime();
1758            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1759            if (status != NO_ERROR) {
1760                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1761                        mThread.unsafe_get(), status);
1762                outputBufferFull = true;
1763                break;
1764            }
1765            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1766            if (waitTimeLeftMs >= waitTimeMs) {
1767                waitTimeLeftMs -= waitTimeMs;
1768            } else {
1769                waitTimeLeftMs = 0;
1770            }
1771        }
1772
1773        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1774                pInBuffer->frameCount;
1775        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1776        Proxy::Buffer buf;
1777        buf.mFrameCount = outFrames;
1778        buf.mRaw = NULL;
1779        mClientProxy->releaseBuffer(&buf);
1780        pInBuffer->frameCount -= outFrames;
1781        pInBuffer->i16 += outFrames * channelCount;
1782        mOutBuffer.frameCount -= outFrames;
1783        mOutBuffer.i16 += outFrames * channelCount;
1784
1785        if (pInBuffer->frameCount == 0) {
1786            if (mBufferQueue.size()) {
1787                mBufferQueue.removeAt(0);
1788                delete [] pInBuffer->mBuffer;
1789                delete pInBuffer;
1790                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1791                        mThread.unsafe_get(), mBufferQueue.size());
1792            } else {
1793                break;
1794            }
1795        }
1796    }
1797
1798    // If we could not write all frames, allocate a buffer and queue it for next time.
1799    if (inBuffer.frameCount) {
1800        sp<ThreadBase> thread = mThread.promote();
1801        if (thread != 0 && !thread->standby()) {
1802            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1803                pInBuffer = new Buffer;
1804                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1805                pInBuffer->frameCount = inBuffer.frameCount;
1806                pInBuffer->i16 = pInBuffer->mBuffer;
1807                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1808                        sizeof(int16_t));
1809                mBufferQueue.add(pInBuffer);
1810                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1811                        mThread.unsafe_get(), mBufferQueue.size());
1812            } else {
1813                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1814                        mThread.unsafe_get(), this);
1815            }
1816        }
1817    }
1818
1819    // Calling write() with a 0 length buffer, means that no more data will be written:
1820    // If no more buffers are pending, fill output track buffer to make sure it is started
1821    // by output mixer.
1822    if (frames == 0 && mBufferQueue.size() == 0) {
1823        // FIXME borken, replace by getting framesReady() from proxy
1824        size_t user = 0;    // was mCblk->user
1825        if (user < mFrameCount) {
1826            frames = mFrameCount - user;
1827            pInBuffer = new Buffer;
1828            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1829            pInBuffer->frameCount = frames;
1830            pInBuffer->i16 = pInBuffer->mBuffer;
1831            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1832            mBufferQueue.add(pInBuffer);
1833        } else if (mActive) {
1834            stop();
1835        }
1836    }
1837
1838    return outputBufferFull;
1839}
1840
1841status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1842        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1843{
1844    ClientProxy::Buffer buf;
1845    buf.mFrameCount = buffer->frameCount;
1846    struct timespec timeout;
1847    timeout.tv_sec = waitTimeMs / 1000;
1848    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1849    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1850    buffer->frameCount = buf.mFrameCount;
1851    buffer->raw = buf.mRaw;
1852    return status;
1853}
1854
1855void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1856{
1857    size_t size = mBufferQueue.size();
1858
1859    for (size_t i = 0; i < size; i++) {
1860        Buffer *pBuffer = mBufferQueue.itemAt(i);
1861        delete [] pBuffer->mBuffer;
1862        delete pBuffer;
1863    }
1864    mBufferQueue.clear();
1865}
1866
1867
1868AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread,
1869                                                     uint32_t sampleRate,
1870                                                     audio_channel_mask_t channelMask,
1871                                                     audio_format_t format,
1872                                                     size_t frameCount,
1873                                                     void *buffer,
1874                                                     IAudioFlinger::track_flags_t flags)
1875    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1876              buffer, 0, 0, getuid(), flags, TYPE_PATCH),
1877              mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true))
1878{
1879    uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) /
1880                                                                    playbackThread->sampleRate();
1881    mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
1882    mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
1883
1884    ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec",
1885                                      this, sampleRate,
1886                                      (int)mPeerTimeout.tv_sec,
1887                                      (int)(mPeerTimeout.tv_nsec / 1000000));
1888}
1889
1890AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack()
1891{
1892}
1893
1894// AudioBufferProvider interface
1895status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer(
1896        AudioBufferProvider::Buffer* buffer, int64_t pts)
1897{
1898    ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy");
1899    Proxy::Buffer buf;
1900    buf.mFrameCount = buffer->frameCount;
1901    status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
1902    ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status);
1903    buffer->frameCount = buf.mFrameCount;
1904    if (buf.mFrameCount == 0) {
1905        return WOULD_BLOCK;
1906    }
1907    status = Track::getNextBuffer(buffer, pts);
1908    return status;
1909}
1910
1911void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer)
1912{
1913    ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy");
1914    Proxy::Buffer buf;
1915    buf.mFrameCount = buffer->frameCount;
1916    buf.mRaw = buffer->raw;
1917    mPeerProxy->releaseBuffer(&buf);
1918    TrackBase::releaseBuffer(buffer);
1919}
1920
1921status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
1922                                                                const struct timespec *timeOut)
1923{
1924    return mProxy->obtainBuffer(buffer, timeOut);
1925}
1926
1927void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
1928{
1929    mProxy->releaseBuffer(buffer);
1930    if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) {
1931        ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting");
1932        start();
1933    }
1934    android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
1935}
1936
1937// ----------------------------------------------------------------------------
1938//      Record
1939// ----------------------------------------------------------------------------
1940
1941AudioFlinger::RecordHandle::RecordHandle(
1942        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1943    : BnAudioRecord(),
1944    mRecordTrack(recordTrack)
1945{
1946}
1947
1948AudioFlinger::RecordHandle::~RecordHandle() {
1949    stop_nonvirtual();
1950    mRecordTrack->destroy();
1951}
1952
1953status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1954        int triggerSession) {
1955    ALOGV("RecordHandle::start()");
1956    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1957}
1958
1959void AudioFlinger::RecordHandle::stop() {
1960    stop_nonvirtual();
1961}
1962
1963void AudioFlinger::RecordHandle::stop_nonvirtual() {
1964    ALOGV("RecordHandle::stop()");
1965    mRecordTrack->stop();
1966}
1967
1968status_t AudioFlinger::RecordHandle::onTransact(
1969    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1970{
1971    return BnAudioRecord::onTransact(code, data, reply, flags);
1972}
1973
1974// ----------------------------------------------------------------------------
1975
1976// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
1977AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1978            RecordThread *thread,
1979            const sp<Client>& client,
1980            uint32_t sampleRate,
1981            audio_format_t format,
1982            audio_channel_mask_t channelMask,
1983            size_t frameCount,
1984            void *buffer,
1985            int sessionId,
1986            int uid,
1987            IAudioFlinger::track_flags_t flags,
1988            track_type type)
1989    :   TrackBase(thread, client, sampleRate, format,
1990                  channelMask, frameCount, buffer, sessionId, uid,
1991                  flags, false /*isOut*/,
1992                  (type == TYPE_DEFAULT) ?
1993                          ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
1994                          ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
1995                  type),
1996        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
1997        // See real initialization of mRsmpInFront at RecordThread::start()
1998        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
1999{
2000    if (mCblk == NULL) {
2001        return;
2002    }
2003
2004    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
2005                                              mFrameSize, !isExternalTrack());
2006
2007    uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
2008    // FIXME I don't understand either of the channel count checks
2009    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
2010            channelCount <= FCC_2) {
2011        // sink SR
2012        mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
2013                thread->mChannelCount, sampleRate);
2014        // source SR
2015        mResampler->setSampleRate(thread->mSampleRate);
2016        mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
2017        mResamplerBufferProvider = new ResamplerBufferProvider(this);
2018    }
2019
2020    if (flags & IAudioFlinger::TRACK_FAST) {
2021        ALOG_ASSERT(thread->mFastTrackAvail);
2022        thread->mFastTrackAvail = false;
2023    }
2024}
2025
2026AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
2027{
2028    ALOGV("%s", __func__);
2029    delete mResampler;
2030    delete[] mRsmpOutBuffer;
2031    delete mResamplerBufferProvider;
2032}
2033
2034// AudioBufferProvider interface
2035status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
2036        int64_t pts __unused)
2037{
2038    ServerProxy::Buffer buf;
2039    buf.mFrameCount = buffer->frameCount;
2040    status_t status = mServerProxy->obtainBuffer(&buf);
2041    buffer->frameCount = buf.mFrameCount;
2042    buffer->raw = buf.mRaw;
2043    if (buf.mFrameCount == 0) {
2044        // FIXME also wake futex so that overrun is noticed more quickly
2045        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
2046    }
2047    return status;
2048}
2049
2050status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
2051                                                        int triggerSession)
2052{
2053    sp<ThreadBase> thread = mThread.promote();
2054    if (thread != 0) {
2055        RecordThread *recordThread = (RecordThread *)thread.get();
2056        return recordThread->start(this, event, triggerSession);
2057    } else {
2058        return BAD_VALUE;
2059    }
2060}
2061
2062void AudioFlinger::RecordThread::RecordTrack::stop()
2063{
2064    sp<ThreadBase> thread = mThread.promote();
2065    if (thread != 0) {
2066        RecordThread *recordThread = (RecordThread *)thread.get();
2067        if (recordThread->stop(this) && isExternalTrack()) {
2068            AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId);
2069        }
2070    }
2071}
2072
2073void AudioFlinger::RecordThread::RecordTrack::destroy()
2074{
2075    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
2076    sp<RecordTrack> keep(this);
2077    {
2078        if (isExternalTrack()) {
2079            if (mState == ACTIVE || mState == RESUMING) {
2080                AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId);
2081            }
2082            AudioSystem::releaseInput(mThreadIoHandle, (audio_session_t)mSessionId);
2083        }
2084        sp<ThreadBase> thread = mThread.promote();
2085        if (thread != 0) {
2086            Mutex::Autolock _l(thread->mLock);
2087            RecordThread *recordThread = (RecordThread *) thread.get();
2088            recordThread->destroyTrack_l(this);
2089        }
2090    }
2091}
2092
2093void AudioFlinger::RecordThread::RecordTrack::invalidate()
2094{
2095    // FIXME should use proxy, and needs work
2096    audio_track_cblk_t* cblk = mCblk;
2097    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
2098    android_atomic_release_store(0x40000000, &cblk->mFutex);
2099    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
2100    (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
2101}
2102
2103
2104/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
2105{
2106    result.append("    Active Client Fmt Chn mask Session S   Server fCount SRate\n");
2107}
2108
2109void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
2110{
2111    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n",
2112            active ? "yes" : "no",
2113            (mClient == 0) ? getpid_cached : mClient->pid(),
2114            mFormat,
2115            mChannelMask,
2116            mSessionId,
2117            mState,
2118            mCblk->mServer,
2119            mFrameCount,
2120            mSampleRate);
2121
2122}
2123
2124void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
2125{
2126    if (event == mSyncStartEvent) {
2127        ssize_t framesToDrop = 0;
2128        sp<ThreadBase> threadBase = mThread.promote();
2129        if (threadBase != 0) {
2130            // TODO: use actual buffer filling status instead of 2 buffers when info is available
2131            // from audio HAL
2132            framesToDrop = threadBase->mFrameCount * 2;
2133        }
2134        mFramesToDrop = framesToDrop;
2135    }
2136}
2137
2138void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
2139{
2140    if (mSyncStartEvent != 0) {
2141        mSyncStartEvent->cancel();
2142        mSyncStartEvent.clear();
2143    }
2144    mFramesToDrop = 0;
2145}
2146
2147
2148AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
2149                                                     uint32_t sampleRate,
2150                                                     audio_channel_mask_t channelMask,
2151                                                     audio_format_t format,
2152                                                     size_t frameCount,
2153                                                     void *buffer,
2154                                                     IAudioFlinger::track_flags_t flags)
2155    :   RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount,
2156                buffer, 0, getuid(), flags, TYPE_PATCH),
2157                mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true))
2158{
2159    uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) /
2160                                                                recordThread->sampleRate();
2161    mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
2162    mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
2163
2164    ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec",
2165                                      this, sampleRate,
2166                                      (int)mPeerTimeout.tv_sec,
2167                                      (int)(mPeerTimeout.tv_nsec / 1000000));
2168}
2169
2170AudioFlinger::RecordThread::PatchRecord::~PatchRecord()
2171{
2172}
2173
2174// AudioBufferProvider interface
2175status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
2176                                                  AudioBufferProvider::Buffer* buffer, int64_t pts)
2177{
2178    ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy");
2179    Proxy::Buffer buf;
2180    buf.mFrameCount = buffer->frameCount;
2181    status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
2182    ALOGV_IF(status != NO_ERROR,
2183             "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status);
2184    buffer->frameCount = buf.mFrameCount;
2185    if (buf.mFrameCount == 0) {
2186        return WOULD_BLOCK;
2187    }
2188    status = RecordTrack::getNextBuffer(buffer, pts);
2189    return status;
2190}
2191
2192void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer)
2193{
2194    ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy");
2195    Proxy::Buffer buf;
2196    buf.mFrameCount = buffer->frameCount;
2197    buf.mRaw = buffer->raw;
2198    mPeerProxy->releaseBuffer(&buf);
2199    TrackBase::releaseBuffer(buffer);
2200}
2201
2202status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer,
2203                                                               const struct timespec *timeOut)
2204{
2205    return mProxy->obtainBuffer(buffer, timeOut);
2206}
2207
2208void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer)
2209{
2210    mProxy->releaseBuffer(buffer);
2211}
2212
2213}; // namespace android
2214