Tracks.cpp revision e186b51e0a9834b287d7a509e960eaf1b688db75 
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 <math.h>
23#include <cutils/compiler.h>
24#include <utils/Log.h>
25
26#include <private/media/AudioTrackShared.h>
27
28#include <common_time/cc_helper.h>
29#include <common_time/local_clock.h>
30
31#include "AudioMixer.h"
32#include "AudioFlinger.h"
33#include "ServiceUtilities.h"
34
35// ----------------------------------------------------------------------------
36
37// Note: the following macro is used for extremely verbose logging message.  In
38// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
39// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
40// are so verbose that we want to suppress them even when we have ALOG_ASSERT
41// turned on.  Do not uncomment the #def below unless you really know what you
42// are doing and want to see all of the extremely verbose messages.
43//#define VERY_VERY_VERBOSE_LOGGING
44#ifdef VERY_VERY_VERBOSE_LOGGING
45#define ALOGVV ALOGV
46#else
47#define ALOGVV(a...) do { } while(0)
48#endif
49
50namespace android {
51
52// ----------------------------------------------------------------------------
53//      TrackBase
54// ----------------------------------------------------------------------------
55
56// TrackBase constructor must be called with AudioFlinger::mLock held
57AudioFlinger::ThreadBase::TrackBase::TrackBase(
58            ThreadBase *thread,
59            const sp<Client>& client,
60            uint32_t sampleRate,
61            audio_format_t format,
62            audio_channel_mask_t channelMask,
63            size_t frameCount,
64            const sp<IMemory>& sharedBuffer,
65            int sessionId,
66            bool isOut)
67    :   RefBase(),
68        mThread(thread),
69        mClient(client),
70        mCblk(NULL),
71        // mBuffer
72        // mBufferEnd
73        mStepCount(0),
74        mState(IDLE),
75        mSampleRate(sampleRate),
76        mFormat(format),
77        mChannelMask(channelMask),
78        mChannelCount(popcount(channelMask)),
79        mFrameSize(audio_is_linear_pcm(format) ?
80                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
81        mFrameCount(frameCount),
82        mStepServerFailed(false),
83        mSessionId(sessionId),
84        mIsOut(isOut),
85        mServerProxy(NULL)
86{
87    // client == 0 implies sharedBuffer == 0
88    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
89
90    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
91            sharedBuffer->size());
92
93    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
94    size_t size = sizeof(audio_track_cblk_t);
95    size_t bufferSize = frameCount * mFrameSize;
96    if (sharedBuffer == 0) {
97        size += bufferSize;
98    }
99
100    if (client != 0) {
101        mCblkMemory = client->heap()->allocate(size);
102        if (mCblkMemory != 0) {
103            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
104            // can't assume mCblk != NULL
105        } else {
106            ALOGE("not enough memory for AudioTrack size=%u", size);
107            client->heap()->dump("AudioTrack");
108            return;
109        }
110    } else {
111        // this syntax avoids calling the audio_track_cblk_t constructor twice
112        mCblk = (audio_track_cblk_t *) new uint8_t[size];
113        // assume mCblk != NULL
114    }
115
116    // construct the shared structure in-place.
117    if (mCblk != NULL) {
118        new(mCblk) audio_track_cblk_t();
119        // clear all buffers
120        mCblk->frameCount_ = frameCount;
121// uncomment the following lines to quickly test 32-bit wraparound
122//      mCblk->user = 0xffff0000;
123//      mCblk->server = 0xffff0000;
124//      mCblk->userBase = 0xffff0000;
125//      mCblk->serverBase = 0xffff0000;
126        if (sharedBuffer == 0) {
127            mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
128            memset(mBuffer, 0, bufferSize);
129            // Force underrun condition to avoid false underrun callback until first data is
130            // written to buffer (other flags are cleared)
131            mCblk->flags = CBLK_UNDERRUN;
132        } else {
133            mBuffer = sharedBuffer->pointer();
134        }
135        mBufferEnd = (uint8_t *)mBuffer + bufferSize;
136        mServerProxy = new ServerProxy(mCblk, mBuffer, frameCount, mFrameSize, isOut);
137    }
138}
139
140AudioFlinger::ThreadBase::TrackBase::~TrackBase()
141{
142    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
143    delete mServerProxy;
144    if (mCblk != NULL) {
145        if (mClient == 0) {
146            delete mCblk;
147        } else {
148            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
149        }
150    }
151    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
152    if (mClient != 0) {
153        // Client destructor must run with AudioFlinger mutex locked
154        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
155        // If the client's reference count drops to zero, the associated destructor
156        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
157        // relying on the automatic clear() at end of scope.
158        mClient.clear();
159    }
160}
161
162// AudioBufferProvider interface
163// getNextBuffer() = 0;
164// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
165void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
166{
167    buffer->raw = NULL;
168    mStepCount = buffer->frameCount;
169    // FIXME See note at getNextBuffer()
170    (void) step();      // ignore return value of step()
171    buffer->frameCount = 0;
172}
173
174bool AudioFlinger::ThreadBase::TrackBase::step() {
175    bool result = mServerProxy->step(mStepCount);
176    if (!result) {
177        ALOGV("stepServer failed acquiring cblk mutex");
178        mStepServerFailed = true;
179    }
180    return result;
181}
182
183void AudioFlinger::ThreadBase::TrackBase::reset() {
184    audio_track_cblk_t* cblk = this->cblk();
185
186    cblk->user = 0;
187    cblk->server = 0;
188    cblk->userBase = 0;
189    cblk->serverBase = 0;
190    mStepServerFailed = false;
191    ALOGV("TrackBase::reset");
192}
193
194uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
195    return mServerProxy->getSampleRate();
196}
197
198void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
199    audio_track_cblk_t* cblk = this->cblk();
200    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
201    int8_t *bufferEnd = bufferStart + frames * mFrameSize;
202
203    // Check validity of returned pointer in case the track control block would have been corrupted.
204    ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
205            "TrackBase::getBuffer buffer out of range:\n"
206                "    start: %p, end %p , mBuffer %p mBufferEnd %p\n"
207                "    server %u, serverBase %u, user %u, userBase %u, frameSize %u",
208                bufferStart, bufferEnd, mBuffer, mBufferEnd,
209                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
210
211    return bufferStart;
212}
213
214status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
215{
216    mSyncEvents.add(event);
217    return NO_ERROR;
218}
219
220// ----------------------------------------------------------------------------
221//      Playback
222// ----------------------------------------------------------------------------
223
224AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
225    : BnAudioTrack(),
226      mTrack(track)
227{
228}
229
230AudioFlinger::TrackHandle::~TrackHandle() {
231    // just stop the track on deletion, associated resources
232    // will be freed from the main thread once all pending buffers have
233    // been played. Unless it's not in the active track list, in which
234    // case we free everything now...
235    mTrack->destroy();
236}
237
238sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
239    return mTrack->getCblk();
240}
241
242status_t AudioFlinger::TrackHandle::start() {
243    return mTrack->start();
244}
245
246void AudioFlinger::TrackHandle::stop() {
247    mTrack->stop();
248}
249
250void AudioFlinger::TrackHandle::flush() {
251    mTrack->flush();
252}
253
254void AudioFlinger::TrackHandle::pause() {
255    mTrack->pause();
256}
257
258status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
259{
260    return mTrack->attachAuxEffect(EffectId);
261}
262
263status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
264                                                         sp<IMemory>* buffer) {
265    if (!mTrack->isTimedTrack())
266        return INVALID_OPERATION;
267
268    PlaybackThread::TimedTrack* tt =
269            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
270    return tt->allocateTimedBuffer(size, buffer);
271}
272
273status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
274                                                     int64_t pts) {
275    if (!mTrack->isTimedTrack())
276        return INVALID_OPERATION;
277
278    PlaybackThread::TimedTrack* tt =
279            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
280    return tt->queueTimedBuffer(buffer, pts);
281}
282
283status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
284    const LinearTransform& xform, int target) {
285
286    if (!mTrack->isTimedTrack())
287        return INVALID_OPERATION;
288
289    PlaybackThread::TimedTrack* tt =
290            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
291    return tt->setMediaTimeTransform(
292        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
293}
294
295status_t AudioFlinger::TrackHandle::onTransact(
296    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
297{
298    return BnAudioTrack::onTransact(code, data, reply, flags);
299}
300
301// ----------------------------------------------------------------------------
302
303// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
304AudioFlinger::PlaybackThread::Track::Track(
305            PlaybackThread *thread,
306            const sp<Client>& client,
307            audio_stream_type_t streamType,
308            uint32_t sampleRate,
309            audio_format_t format,
310            audio_channel_mask_t channelMask,
311            size_t frameCount,
312            const sp<IMemory>& sharedBuffer,
313            int sessionId,
314            IAudioFlinger::track_flags_t flags)
315    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
316            sessionId, true /*isOut*/),
317    mFillingUpStatus(FS_INVALID),
318    // mRetryCount initialized later when needed
319    mSharedBuffer(sharedBuffer),
320    mStreamType(streamType),
321    mName(-1),  // see note below
322    mMainBuffer(thread->mixBuffer()),
323    mAuxBuffer(NULL),
324    mAuxEffectId(0), mHasVolumeController(false),
325    mPresentationCompleteFrames(0),
326    mFlags(flags),
327    mFastIndex(-1),
328    mUnderrunCount(0),
329    mCachedVolume(1.0),
330    mIsInvalid(false)
331{
332    if (mCblk != NULL) {
333        // to avoid leaking a track name, do not allocate one unless there is an mCblk
334        mName = thread->getTrackName_l(channelMask, sessionId);
335        mCblk->mName = mName;
336        if (mName < 0) {
337            ALOGE("no more track names available");
338            return;
339        }
340        // only allocate a fast track index if we were able to allocate a normal track name
341        if (flags & IAudioFlinger::TRACK_FAST) {
342            ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
343            int i = __builtin_ctz(thread->mFastTrackAvailMask);
344            ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
345            // FIXME This is too eager.  We allocate a fast track index before the
346            //       fast track becomes active.  Since fast tracks are a scarce resource,
347            //       this means we are potentially denying other more important fast tracks from
348            //       being created.  It would be better to allocate the index dynamically.
349            mFastIndex = i;
350            mCblk->mName = i;
351            // Read the initial underruns because this field is never cleared by the fast mixer
352            mObservedUnderruns = thread->getFastTrackUnderruns(i);
353            thread->mFastTrackAvailMask &= ~(1 << i);
354        }
355    }
356    ALOGV("Track constructor name %d, calling pid %d", mName,
357            IPCThreadState::self()->getCallingPid());
358}
359
360AudioFlinger::PlaybackThread::Track::~Track()
361{
362    ALOGV("PlaybackThread::Track destructor");
363}
364
365void AudioFlinger::PlaybackThread::Track::destroy()
366{
367    // NOTE: destroyTrack_l() can remove a strong reference to this Track
368    // by removing it from mTracks vector, so there is a risk that this Tracks's
369    // destructor is called. As the destructor needs to lock mLock,
370    // we must acquire a strong reference on this Track before locking mLock
371    // here so that the destructor is called only when exiting this function.
372    // On the other hand, as long as Track::destroy() is only called by
373    // TrackHandle destructor, the TrackHandle still holds a strong ref on
374    // this Track with its member mTrack.
375    sp<Track> keep(this);
376    { // scope for mLock
377        sp<ThreadBase> thread = mThread.promote();
378        if (thread != 0) {
379            if (!isOutputTrack()) {
380                if (mState == ACTIVE || mState == RESUMING) {
381                    AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
382
383#ifdef ADD_BATTERY_DATA
384                    // to track the speaker usage
385                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
386#endif
387                }
388                AudioSystem::releaseOutput(thread->id());
389            }
390            Mutex::Autolock _l(thread->mLock);
391            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
392            playbackThread->destroyTrack_l(this);
393        }
394    }
395}
396
397/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
398{
399    result.append("   Name Client Type Fmt Chn mask   Session StpCnt fCount S F SRate  "
400                  "L dB  R dB    Server      User     Main buf    Aux Buf  Flags Underruns\n");
401}
402
403void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
404{
405    uint32_t vlr = mServerProxy->getVolumeLR();
406    if (isFastTrack()) {
407        sprintf(buffer, "   F %2d", mFastIndex);
408    } else {
409        sprintf(buffer, "   %4d", mName - AudioMixer::TRACK0);
410    }
411    track_state state = mState;
412    char stateChar;
413    switch (state) {
414    case IDLE:
415        stateChar = 'I';
416        break;
417    case TERMINATED:
418        stateChar = 'T';
419        break;
420    case STOPPING_1:
421        stateChar = 's';
422        break;
423    case STOPPING_2:
424        stateChar = '5';
425        break;
426    case STOPPED:
427        stateChar = 'S';
428        break;
429    case RESUMING:
430        stateChar = 'R';
431        break;
432    case ACTIVE:
433        stateChar = 'A';
434        break;
435    case PAUSING:
436        stateChar = 'p';
437        break;
438    case PAUSED:
439        stateChar = 'P';
440        break;
441    case FLUSHED:
442        stateChar = 'F';
443        break;
444    default:
445        stateChar = '?';
446        break;
447    }
448    char nowInUnderrun;
449    switch (mObservedUnderruns.mBitFields.mMostRecent) {
450    case UNDERRUN_FULL:
451        nowInUnderrun = ' ';
452        break;
453    case UNDERRUN_PARTIAL:
454        nowInUnderrun = '<';
455        break;
456    case UNDERRUN_EMPTY:
457        nowInUnderrun = '*';
458        break;
459    default:
460        nowInUnderrun = '?';
461        break;
462    }
463    snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %5u %5.2g %5.2g  "
464            "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
465            (mClient == 0) ? getpid_cached : mClient->pid(),
466            mStreamType,
467            mFormat,
468            mChannelMask,
469            mSessionId,
470            mStepCount,
471            mFrameCount,
472            stateChar,
473            mFillingUpStatus,
474            mServerProxy->getSampleRate(),
475            20.0 * log10((vlr & 0xFFFF) / 4096.0),
476            20.0 * log10((vlr >> 16) / 4096.0),
477            mCblk->server,
478            mCblk->user,
479            (int)mMainBuffer,
480            (int)mAuxBuffer,
481            mCblk->flags,
482            mUnderrunCount,
483            nowInUnderrun);
484}
485
486// AudioBufferProvider interface
487status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
488        AudioBufferProvider::Buffer* buffer, int64_t pts)
489{
490    audio_track_cblk_t* cblk = this->cblk();
491    uint32_t framesReady;
492    uint32_t framesReq = buffer->frameCount;
493
494    // Check if last stepServer failed, try to step now
495    if (mStepServerFailed) {
496        // FIXME When called by fast mixer, this takes a mutex with tryLock().
497        //       Since the fast mixer is higher priority than client callback thread,
498        //       it does not result in priority inversion for client.
499        //       But a non-blocking solution would be preferable to avoid
500        //       fast mixer being unable to tryLock(), and
501        //       to avoid the extra context switches if the client wakes up,
502        //       discovers the mutex is locked, then has to wait for fast mixer to unlock.
503        if (!step())  goto getNextBuffer_exit;
504        ALOGV("stepServer recovered");
505        mStepServerFailed = false;
506    }
507
508    // FIXME Same as above
509    framesReady = mServerProxy->framesReady();
510
511    if (CC_LIKELY(framesReady)) {
512        uint32_t s = cblk->server;
513        uint32_t bufferEnd = cblk->serverBase + mFrameCount;
514
515        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
516        if (framesReq > framesReady) {
517            framesReq = framesReady;
518        }
519        if (framesReq > bufferEnd - s) {
520            framesReq = bufferEnd - s;
521        }
522
523        buffer->raw = getBuffer(s, framesReq);
524        buffer->frameCount = framesReq;
525        return NO_ERROR;
526    }
527
528getNextBuffer_exit:
529    buffer->raw = NULL;
530    buffer->frameCount = 0;
531    ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
532    return NOT_ENOUGH_DATA;
533}
534
535// Note that framesReady() takes a mutex on the control block using tryLock().
536// This could result in priority inversion if framesReady() is called by the normal mixer,
537// as the normal mixer thread runs at lower
538// priority than the client's callback thread:  there is a short window within framesReady()
539// during which the normal mixer could be preempted, and the client callback would block.
540// Another problem can occur if framesReady() is called by the fast mixer:
541// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
542// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
543size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
544    return mServerProxy->framesReady();
545}
546
547// Don't call for fast tracks; the framesReady() could result in priority inversion
548bool AudioFlinger::PlaybackThread::Track::isReady() const {
549    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
550        return true;
551    }
552
553    if (framesReady() >= mFrameCount ||
554            (mCblk->flags & CBLK_FORCEREADY)) {
555        mFillingUpStatus = FS_FILLED;
556        android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
557        return true;
558    }
559    return false;
560}
561
562status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
563                                                    int triggerSession)
564{
565    status_t status = NO_ERROR;
566    ALOGV("start(%d), calling pid %d session %d",
567            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
568
569    sp<ThreadBase> thread = mThread.promote();
570    if (thread != 0) {
571        Mutex::Autolock _l(thread->mLock);
572        thread->mNBLogWriter->logf("start mName=%d", mName);
573        track_state state = mState;
574        // here the track could be either new, or restarted
575        // in both cases "unstop" the track
576        if (mState == PAUSED) {
577            mState = TrackBase::RESUMING;
578            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
579        } else {
580            mState = TrackBase::ACTIVE;
581            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
582        }
583
584        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
585            thread->mLock.unlock();
586            status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
587            thread->mLock.lock();
588
589#ifdef ADD_BATTERY_DATA
590            // to track the speaker usage
591            if (status == NO_ERROR) {
592                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
593            }
594#endif
595        }
596        if (status == NO_ERROR) {
597            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
598            playbackThread->addTrack_l(this);
599        } else {
600            mState = state;
601            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
602        }
603    } else {
604        status = BAD_VALUE;
605    }
606    return status;
607}
608
609void AudioFlinger::PlaybackThread::Track::stop()
610{
611    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
612    sp<ThreadBase> thread = mThread.promote();
613    if (thread != 0) {
614        Mutex::Autolock _l(thread->mLock);
615        thread->mNBLogWriter->logf("stop mName=%d", mName);
616        track_state state = mState;
617        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
618            // If the track is not active (PAUSED and buffers full), flush buffers
619            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
620            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
621                reset();
622                mState = STOPPED;
623            } else if (!isFastTrack()) {
624                mState = STOPPED;
625            } else {
626                // prepareTracks_l() will set state to STOPPING_2 after next underrun,
627                // and then to STOPPED and reset() when presentation is complete
628                mState = STOPPING_1;
629            }
630            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
631                    playbackThread);
632        }
633        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
634            thread->mLock.unlock();
635            AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
636            thread->mLock.lock();
637
638#ifdef ADD_BATTERY_DATA
639            // to track the speaker usage
640            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
641#endif
642        }
643    }
644}
645
646void AudioFlinger::PlaybackThread::Track::pause()
647{
648    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
649    sp<ThreadBase> thread = mThread.promote();
650    if (thread != 0) {
651        Mutex::Autolock _l(thread->mLock);
652        thread->mNBLogWriter->logf("pause mName=%d", mName);
653        if (mState == ACTIVE || mState == RESUMING) {
654            mState = PAUSING;
655            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
656            if (!isOutputTrack()) {
657                thread->mLock.unlock();
658                AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
659                thread->mLock.lock();
660
661#ifdef ADD_BATTERY_DATA
662                // to track the speaker usage
663                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
664#endif
665            }
666        }
667    }
668}
669
670void AudioFlinger::PlaybackThread::Track::flush()
671{
672    ALOGV("flush(%d)", mName);
673    sp<ThreadBase> thread = mThread.promote();
674    if (thread != 0) {
675        Mutex::Autolock _l(thread->mLock);
676        thread->mNBLogWriter->logf("flush mName=%d", mName);
677        if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED &&
678                mState != PAUSING && mState != IDLE && mState != FLUSHED) {
679            return;
680        }
681        // No point remaining in PAUSED state after a flush => go to
682        // FLUSHED state
683        mState = FLUSHED;
684        // do not reset the track if it is still in the process of being stopped or paused.
685        // this will be done by prepareTracks_l() when the track is stopped.
686        // prepareTracks_l() will see mState == FLUSHED, then
687        // remove from active track list, reset(), and trigger presentation complete
688        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
689        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
690            reset();
691        }
692    }
693}
694
695void AudioFlinger::PlaybackThread::Track::reset()
696{
697    // Do not reset twice to avoid discarding data written just after a flush and before
698    // the audioflinger thread detects the track is stopped.
699    if (!mResetDone) {
700        TrackBase::reset();
701        // Force underrun condition to avoid false underrun callback until first data is
702        // written to buffer
703        android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
704        android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
705        mFillingUpStatus = FS_FILLING;
706        mResetDone = true;
707        if (mState == FLUSHED) {
708            mState = IDLE;
709        }
710    }
711}
712
713status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
714{
715    status_t status = DEAD_OBJECT;
716    sp<ThreadBase> thread = mThread.promote();
717    if (thread != 0) {
718        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
719        sp<AudioFlinger> af = mClient->audioFlinger();
720
721        Mutex::Autolock _l(af->mLock);
722
723        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
724
725        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
726            Mutex::Autolock _dl(playbackThread->mLock);
727            Mutex::Autolock _sl(srcThread->mLock);
728            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
729            if (chain == 0) {
730                return INVALID_OPERATION;
731            }
732
733            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
734            if (effect == 0) {
735                return INVALID_OPERATION;
736            }
737            srcThread->removeEffect_l(effect);
738            playbackThread->addEffect_l(effect);
739            // removeEffect_l() has stopped the effect if it was active so it must be restarted
740            if (effect->state() == EffectModule::ACTIVE ||
741                    effect->state() == EffectModule::STOPPING) {
742                effect->start();
743            }
744
745            sp<EffectChain> dstChain = effect->chain().promote();
746            if (dstChain == 0) {
747                srcThread->addEffect_l(effect);
748                return INVALID_OPERATION;
749            }
750            AudioSystem::unregisterEffect(effect->id());
751            AudioSystem::registerEffect(&effect->desc(),
752                                        srcThread->id(),
753                                        dstChain->strategy(),
754                                        AUDIO_SESSION_OUTPUT_MIX,
755                                        effect->id());
756        }
757        status = playbackThread->attachAuxEffect(this, EffectId);
758    }
759    return status;
760}
761
762void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
763{
764    mAuxEffectId = EffectId;
765    mAuxBuffer = buffer;
766}
767
768bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
769                                                         size_t audioHalFrames)
770{
771    // a track is considered presented when the total number of frames written to audio HAL
772    // corresponds to the number of frames written when presentationComplete() is called for the
773    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
774    if (mPresentationCompleteFrames == 0) {
775        mPresentationCompleteFrames = framesWritten + audioHalFrames;
776        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
777                  mPresentationCompleteFrames, audioHalFrames);
778    }
779    if (framesWritten >= mPresentationCompleteFrames) {
780        ALOGV("presentationComplete() session %d complete: framesWritten %d",
781                  mSessionId, framesWritten);
782        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
783        return true;
784    }
785    return false;
786}
787
788void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
789{
790    for (int i = 0; i < (int)mSyncEvents.size(); i++) {
791        if (mSyncEvents[i]->type() == type) {
792            mSyncEvents[i]->trigger();
793            mSyncEvents.removeAt(i);
794            i--;
795        }
796    }
797}
798
799// implement VolumeBufferProvider interface
800
801uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
802{
803    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
804    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
805    uint32_t vlr = mServerProxy->getVolumeLR();
806    uint32_t vl = vlr & 0xFFFF;
807    uint32_t vr = vlr >> 16;
808    // track volumes come from shared memory, so can't be trusted and must be clamped
809    if (vl > MAX_GAIN_INT) {
810        vl = MAX_GAIN_INT;
811    }
812    if (vr > MAX_GAIN_INT) {
813        vr = MAX_GAIN_INT;
814    }
815    // now apply the cached master volume and stream type volume;
816    // this is trusted but lacks any synchronization or barrier so may be stale
817    float v = mCachedVolume;
818    vl *= v;
819    vr *= v;
820    // re-combine into U4.16
821    vlr = (vr << 16) | (vl & 0xFFFF);
822    // FIXME look at mute, pause, and stop flags
823    return vlr;
824}
825
826status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
827{
828    if (mState == TERMINATED || mState == PAUSED ||
829            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
830                                      (mState == STOPPED)))) {
831        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
832              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
833        event->cancel();
834        return INVALID_OPERATION;
835    }
836    (void) TrackBase::setSyncEvent(event);
837    return NO_ERROR;
838}
839
840void AudioFlinger::PlaybackThread::Track::invalidate()
841{
842    // FIXME should use proxy
843    android_atomic_or(CBLK_INVALID, &mCblk->flags);
844    mCblk->cv.signal();
845    mIsInvalid = true;
846}
847
848// ----------------------------------------------------------------------------
849
850sp<AudioFlinger::PlaybackThread::TimedTrack>
851AudioFlinger::PlaybackThread::TimedTrack::create(
852            PlaybackThread *thread,
853            const sp<Client>& client,
854            audio_stream_type_t streamType,
855            uint32_t sampleRate,
856            audio_format_t format,
857            audio_channel_mask_t channelMask,
858            size_t frameCount,
859            const sp<IMemory>& sharedBuffer,
860            int sessionId) {
861    if (!client->reserveTimedTrack())
862        return 0;
863
864    return new TimedTrack(
865        thread, client, streamType, sampleRate, format, channelMask, frameCount,
866        sharedBuffer, sessionId);
867}
868
869AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
870            PlaybackThread *thread,
871            const sp<Client>& client,
872            audio_stream_type_t streamType,
873            uint32_t sampleRate,
874            audio_format_t format,
875            audio_channel_mask_t channelMask,
876            size_t frameCount,
877            const sp<IMemory>& sharedBuffer,
878            int sessionId)
879    : Track(thread, client, streamType, sampleRate, format, channelMask,
880            frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
881      mQueueHeadInFlight(false),
882      mTrimQueueHeadOnRelease(false),
883      mFramesPendingInQueue(0),
884      mTimedSilenceBuffer(NULL),
885      mTimedSilenceBufferSize(0),
886      mTimedAudioOutputOnTime(false),
887      mMediaTimeTransformValid(false)
888{
889    LocalClock lc;
890    mLocalTimeFreq = lc.getLocalFreq();
891
892    mLocalTimeToSampleTransform.a_zero = 0;
893    mLocalTimeToSampleTransform.b_zero = 0;
894    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
895    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
896    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
897                            &mLocalTimeToSampleTransform.a_to_b_denom);
898
899    mMediaTimeToSampleTransform.a_zero = 0;
900    mMediaTimeToSampleTransform.b_zero = 0;
901    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
902    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
903    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
904                            &mMediaTimeToSampleTransform.a_to_b_denom);
905}
906
907AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
908    mClient->releaseTimedTrack();
909    delete [] mTimedSilenceBuffer;
910}
911
912status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
913    size_t size, sp<IMemory>* buffer) {
914
915    Mutex::Autolock _l(mTimedBufferQueueLock);
916
917    trimTimedBufferQueue_l();
918
919    // lazily initialize the shared memory heap for timed buffers
920    if (mTimedMemoryDealer == NULL) {
921        const int kTimedBufferHeapSize = 512 << 10;
922
923        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
924                                              "AudioFlingerTimed");
925        if (mTimedMemoryDealer == NULL)
926            return NO_MEMORY;
927    }
928
929    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
930    if (newBuffer == NULL) {
931        newBuffer = mTimedMemoryDealer->allocate(size);
932        if (newBuffer == NULL)
933            return NO_MEMORY;
934    }
935
936    *buffer = newBuffer;
937    return NO_ERROR;
938}
939
940// caller must hold mTimedBufferQueueLock
941void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
942    int64_t mediaTimeNow;
943    {
944        Mutex::Autolock mttLock(mMediaTimeTransformLock);
945        if (!mMediaTimeTransformValid)
946            return;
947
948        int64_t targetTimeNow;
949        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
950            ? mCCHelper.getCommonTime(&targetTimeNow)
951            : mCCHelper.getLocalTime(&targetTimeNow);
952
953        if (OK != res)
954            return;
955
956        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
957                                                    &mediaTimeNow)) {
958            return;
959        }
960    }
961
962    size_t trimEnd;
963    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
964        int64_t bufEnd;
965
966        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
967            // We have a next buffer.  Just use its PTS as the PTS of the frame
968            // following the last frame in this buffer.  If the stream is sparse
969            // (ie, there are deliberate gaps left in the stream which should be
970            // filled with silence by the TimedAudioTrack), then this can result
971            // in one extra buffer being left un-trimmed when it could have
972            // been.  In general, this is not typical, and we would rather
973            // optimized away the TS calculation below for the more common case
974            // where PTSes are contiguous.
975            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
976        } else {
977            // We have no next buffer.  Compute the PTS of the frame following
978            // the last frame in this buffer by computing the duration of of
979            // this frame in media time units and adding it to the PTS of the
980            // buffer.
981            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
982                               / mFrameSize;
983
984            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
985                                                                &bufEnd)) {
986                ALOGE("Failed to convert frame count of %lld to media time"
987                      " duration" " (scale factor %d/%u) in %s",
988                      frameCount,
989                      mMediaTimeToSampleTransform.a_to_b_numer,
990                      mMediaTimeToSampleTransform.a_to_b_denom,
991                      __PRETTY_FUNCTION__);
992                break;
993            }
994            bufEnd += mTimedBufferQueue[trimEnd].pts();
995        }
996
997        if (bufEnd > mediaTimeNow)
998            break;
999
1000        // Is the buffer we want to use in the middle of a mix operation right
1001        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1002        // from the mixer which should be coming back shortly.
1003        if (!trimEnd && mQueueHeadInFlight) {
1004            mTrimQueueHeadOnRelease = true;
1005        }
1006    }
1007
1008    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1009    if (trimStart < trimEnd) {
1010        // Update the bookkeeping for framesReady()
1011        for (size_t i = trimStart; i < trimEnd; ++i) {
1012            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1013        }
1014
1015        // Now actually remove the buffers from the queue.
1016        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1017    }
1018}
1019
1020void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1021        const char* logTag) {
1022    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1023                "%s called (reason \"%s\"), but timed buffer queue has no"
1024                " elements to trim.", __FUNCTION__, logTag);
1025
1026    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1027    mTimedBufferQueue.removeAt(0);
1028}
1029
1030void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1031        const TimedBuffer& buf,
1032        const char* logTag) {
1033    uint32_t bufBytes        = buf.buffer()->size();
1034    uint32_t consumedAlready = buf.position();
1035
1036    ALOG_ASSERT(consumedAlready <= bufBytes,
1037                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1038                " only %u bytes long, but claims to have consumed %u"
1039                " bytes.  (update reason: \"%s\")",
1040                bufBytes, consumedAlready, logTag);
1041
1042    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1043    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1044                "Bad bookkeeping while updating frames pending.  Should have at"
1045                " least %u queued frames, but we think we have only %u.  (update"
1046                " reason: \"%s\")",
1047                bufFrames, mFramesPendingInQueue, logTag);
1048
1049    mFramesPendingInQueue -= bufFrames;
1050}
1051
1052status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1053    const sp<IMemory>& buffer, int64_t pts) {
1054
1055    {
1056        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1057        if (!mMediaTimeTransformValid)
1058            return INVALID_OPERATION;
1059    }
1060
1061    Mutex::Autolock _l(mTimedBufferQueueLock);
1062
1063    uint32_t bufFrames = buffer->size() / mFrameSize;
1064    mFramesPendingInQueue += bufFrames;
1065    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1066
1067    return NO_ERROR;
1068}
1069
1070status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1071    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1072
1073    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1074           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1075           target);
1076
1077    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1078          target == TimedAudioTrack::COMMON_TIME)) {
1079        return BAD_VALUE;
1080    }
1081
1082    Mutex::Autolock lock(mMediaTimeTransformLock);
1083    mMediaTimeTransform = xform;
1084    mMediaTimeTransformTarget = target;
1085    mMediaTimeTransformValid = true;
1086
1087    return NO_ERROR;
1088}
1089
1090#define min(a, b) ((a) < (b) ? (a) : (b))
1091
1092// implementation of getNextBuffer for tracks whose buffers have timestamps
1093status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1094    AudioBufferProvider::Buffer* buffer, int64_t pts)
1095{
1096    if (pts == AudioBufferProvider::kInvalidPTS) {
1097        buffer->raw = NULL;
1098        buffer->frameCount = 0;
1099        mTimedAudioOutputOnTime = false;
1100        return INVALID_OPERATION;
1101    }
1102
1103    Mutex::Autolock _l(mTimedBufferQueueLock);
1104
1105    ALOG_ASSERT(!mQueueHeadInFlight,
1106                "getNextBuffer called without releaseBuffer!");
1107
1108    while (true) {
1109
1110        // if we have no timed buffers, then fail
1111        if (mTimedBufferQueue.isEmpty()) {
1112            buffer->raw = NULL;
1113            buffer->frameCount = 0;
1114            return NOT_ENOUGH_DATA;
1115        }
1116
1117        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1118
1119        // calculate the PTS of the head of the timed buffer queue expressed in
1120        // local time
1121        int64_t headLocalPTS;
1122        {
1123            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1124
1125            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1126
1127            if (mMediaTimeTransform.a_to_b_denom == 0) {
1128                // the transform represents a pause, so yield silence
1129                timedYieldSilence_l(buffer->frameCount, buffer);
1130                return NO_ERROR;
1131            }
1132
1133            int64_t transformedPTS;
1134            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1135                                                        &transformedPTS)) {
1136                // the transform failed.  this shouldn't happen, but if it does
1137                // then just drop this buffer
1138                ALOGW("timedGetNextBuffer transform failed");
1139                buffer->raw = NULL;
1140                buffer->frameCount = 0;
1141                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1142                return NO_ERROR;
1143            }
1144
1145            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1146                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1147                                                          &headLocalPTS)) {
1148                    buffer->raw = NULL;
1149                    buffer->frameCount = 0;
1150                    return INVALID_OPERATION;
1151                }
1152            } else {
1153                headLocalPTS = transformedPTS;
1154            }
1155        }
1156
1157        // adjust the head buffer's PTS to reflect the portion of the head buffer
1158        // that has already been consumed
1159        int64_t effectivePTS = headLocalPTS +
1160                ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
1161
1162        // Calculate the delta in samples between the head of the input buffer
1163        // queue and the start of the next output buffer that will be written.
1164        // If the transformation fails because of over or underflow, it means
1165        // that the sample's position in the output stream is so far out of
1166        // whack that it should just be dropped.
1167        int64_t sampleDelta;
1168        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1169            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1170            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1171                                       " mix");
1172            continue;
1173        }
1174        if (!mLocalTimeToSampleTransform.doForwardTransform(
1175                (effectivePTS - pts) << 32, &sampleDelta)) {
1176            ALOGV("*** too late during sample rate transform: dropped buffer");
1177            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1178            continue;
1179        }
1180
1181        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1182               " sampleDelta=[%d.%08x]",
1183               head.pts(), head.position(), pts,
1184               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1185                   + (sampleDelta >> 32)),
1186               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1187
1188        // if the delta between the ideal placement for the next input sample and
1189        // the current output position is within this threshold, then we will
1190        // concatenate the next input samples to the previous output
1191        const int64_t kSampleContinuityThreshold =
1192                (static_cast<int64_t>(sampleRate()) << 32) / 250;
1193
1194        // if this is the first buffer of audio that we're emitting from this track
1195        // then it should be almost exactly on time.
1196        const int64_t kSampleStartupThreshold = 1LL << 32;
1197
1198        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1199           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1200            // the next input is close enough to being on time, so concatenate it
1201            // with the last output
1202            timedYieldSamples_l(buffer);
1203
1204            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1205                    head.position(), buffer->frameCount);
1206            return NO_ERROR;
1207        }
1208
1209        // Looks like our output is not on time.  Reset our on timed status.
1210        // Next time we mix samples from our input queue, then should be within
1211        // the StartupThreshold.
1212        mTimedAudioOutputOnTime = false;
1213        if (sampleDelta > 0) {
1214            // the gap between the current output position and the proper start of
1215            // the next input sample is too big, so fill it with silence
1216            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1217
1218            timedYieldSilence_l(framesUntilNextInput, buffer);
1219            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1220            return NO_ERROR;
1221        } else {
1222            // the next input sample is late
1223            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1224            size_t onTimeSamplePosition =
1225                    head.position() + lateFrames * mFrameSize;
1226
1227            if (onTimeSamplePosition > head.buffer()->size()) {
1228                // all the remaining samples in the head are too late, so
1229                // drop it and move on
1230                ALOGV("*** too late: dropped buffer");
1231                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1232                continue;
1233            } else {
1234                // skip over the late samples
1235                head.setPosition(onTimeSamplePosition);
1236
1237                // yield the available samples
1238                timedYieldSamples_l(buffer);
1239
1240                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1241                return NO_ERROR;
1242            }
1243        }
1244    }
1245}
1246
1247// Yield samples from the timed buffer queue head up to the given output
1248// buffer's capacity.
1249//
1250// Caller must hold mTimedBufferQueueLock
1251void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1252    AudioBufferProvider::Buffer* buffer) {
1253
1254    const TimedBuffer& head = mTimedBufferQueue[0];
1255
1256    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1257                   head.position());
1258
1259    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1260                                 mFrameSize);
1261    size_t framesRequested = buffer->frameCount;
1262    buffer->frameCount = min(framesLeftInHead, framesRequested);
1263
1264    mQueueHeadInFlight = true;
1265    mTimedAudioOutputOnTime = true;
1266}
1267
1268// Yield samples of silence up to the given output buffer's capacity
1269//
1270// Caller must hold mTimedBufferQueueLock
1271void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1272    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1273
1274    // lazily allocate a buffer filled with silence
1275    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1276        delete [] mTimedSilenceBuffer;
1277        mTimedSilenceBufferSize = numFrames * mFrameSize;
1278        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1279        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1280    }
1281
1282    buffer->raw = mTimedSilenceBuffer;
1283    size_t framesRequested = buffer->frameCount;
1284    buffer->frameCount = min(numFrames, framesRequested);
1285
1286    mTimedAudioOutputOnTime = false;
1287}
1288
1289// AudioBufferProvider interface
1290void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1291    AudioBufferProvider::Buffer* buffer) {
1292
1293    Mutex::Autolock _l(mTimedBufferQueueLock);
1294
1295    // If the buffer which was just released is part of the buffer at the head
1296    // of the queue, be sure to update the amt of the buffer which has been
1297    // consumed.  If the buffer being returned is not part of the head of the
1298    // queue, its either because the buffer is part of the silence buffer, or
1299    // because the head of the timed queue was trimmed after the mixer called
1300    // getNextBuffer but before the mixer called releaseBuffer.
1301    if (buffer->raw == mTimedSilenceBuffer) {
1302        ALOG_ASSERT(!mQueueHeadInFlight,
1303                    "Queue head in flight during release of silence buffer!");
1304        goto done;
1305    }
1306
1307    ALOG_ASSERT(mQueueHeadInFlight,
1308                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1309                " head in flight.");
1310
1311    if (mTimedBufferQueue.size()) {
1312        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1313
1314        void* start = head.buffer()->pointer();
1315        void* end   = reinterpret_cast<void*>(
1316                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1317                        + head.buffer()->size());
1318
1319        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1320                    "released buffer not within the head of the timed buffer"
1321                    " queue; qHead = [%p, %p], released buffer = %p",
1322                    start, end, buffer->raw);
1323
1324        head.setPosition(head.position() +
1325                (buffer->frameCount * mFrameSize));
1326        mQueueHeadInFlight = false;
1327
1328        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1329                    "Bad bookkeeping during releaseBuffer!  Should have at"
1330                    " least %u queued frames, but we think we have only %u",
1331                    buffer->frameCount, mFramesPendingInQueue);
1332
1333        mFramesPendingInQueue -= buffer->frameCount;
1334
1335        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1336            || mTrimQueueHeadOnRelease) {
1337            trimTimedBufferQueueHead_l("releaseBuffer");
1338            mTrimQueueHeadOnRelease = false;
1339        }
1340    } else {
1341        LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1342                  " buffers in the timed buffer queue");
1343    }
1344
1345done:
1346    buffer->raw = 0;
1347    buffer->frameCount = 0;
1348}
1349
1350size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1351    Mutex::Autolock _l(mTimedBufferQueueLock);
1352    return mFramesPendingInQueue;
1353}
1354
1355AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1356        : mPTS(0), mPosition(0) {}
1357
1358AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1359    const sp<IMemory>& buffer, int64_t pts)
1360        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1361
1362
1363// ----------------------------------------------------------------------------
1364
1365AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1366            PlaybackThread *playbackThread,
1367            DuplicatingThread *sourceThread,
1368            uint32_t sampleRate,
1369            audio_format_t format,
1370            audio_channel_mask_t channelMask,
1371            size_t frameCount)
1372    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1373                NULL, 0, IAudioFlinger::TRACK_DEFAULT),
1374    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1375{
1376
1377    if (mCblk != NULL) {
1378        mOutBuffer.frameCount = 0;
1379        playbackThread->mTracks.add(this);
1380        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1381                "mCblk->frameCount_ %u, mChannelMask 0x%08x mBufferEnd %p",
1382                mCblk, mBuffer,
1383                mCblk->frameCount_, mChannelMask, mBufferEnd);
1384        // since client and server are in the same process,
1385        // the buffer has the same virtual address on both sides
1386        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1387    } else {
1388        ALOGW("Error creating output track on thread %p", playbackThread);
1389    }
1390}
1391
1392AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1393{
1394    clearBufferQueue();
1395    delete mClientProxy;
1396    // superclass destructor will now delete the server proxy and shared memory both refer to
1397}
1398
1399status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1400                                                          int triggerSession)
1401{
1402    status_t status = Track::start(event, triggerSession);
1403    if (status != NO_ERROR) {
1404        return status;
1405    }
1406
1407    mActive = true;
1408    mRetryCount = 127;
1409    return status;
1410}
1411
1412void AudioFlinger::PlaybackThread::OutputTrack::stop()
1413{
1414    Track::stop();
1415    clearBufferQueue();
1416    mOutBuffer.frameCount = 0;
1417    mActive = false;
1418}
1419
1420bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1421{
1422    Buffer *pInBuffer;
1423    Buffer inBuffer;
1424    uint32_t channelCount = mChannelCount;
1425    bool outputBufferFull = false;
1426    inBuffer.frameCount = frames;
1427    inBuffer.i16 = data;
1428
1429    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1430
1431    if (!mActive && frames != 0) {
1432        start();
1433        sp<ThreadBase> thread = mThread.promote();
1434        if (thread != 0) {
1435            MixerThread *mixerThread = (MixerThread *)thread.get();
1436            if (mFrameCount > frames) {
1437                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1438                    uint32_t startFrames = (mFrameCount - frames);
1439                    pInBuffer = new Buffer;
1440                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1441                    pInBuffer->frameCount = startFrames;
1442                    pInBuffer->i16 = pInBuffer->mBuffer;
1443                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1444                    mBufferQueue.add(pInBuffer);
1445                } else {
1446                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
1447                }
1448            }
1449        }
1450    }
1451
1452    while (waitTimeLeftMs) {
1453        // First write pending buffers, then new data
1454        if (mBufferQueue.size()) {
1455            pInBuffer = mBufferQueue.itemAt(0);
1456        } else {
1457            pInBuffer = &inBuffer;
1458        }
1459
1460        if (pInBuffer->frameCount == 0) {
1461            break;
1462        }
1463
1464        if (mOutBuffer.frameCount == 0) {
1465            mOutBuffer.frameCount = pInBuffer->frameCount;
1466            nsecs_t startTime = systemTime();
1467            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
1468                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
1469                        mThread.unsafe_get());
1470                outputBufferFull = true;
1471                break;
1472            }
1473            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1474            if (waitTimeLeftMs >= waitTimeMs) {
1475                waitTimeLeftMs -= waitTimeMs;
1476            } else {
1477                waitTimeLeftMs = 0;
1478            }
1479        }
1480
1481        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1482                pInBuffer->frameCount;
1483        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1484        mClientProxy->stepUser(outFrames);
1485        pInBuffer->frameCount -= outFrames;
1486        pInBuffer->i16 += outFrames * channelCount;
1487        mOutBuffer.frameCount -= outFrames;
1488        mOutBuffer.i16 += outFrames * channelCount;
1489
1490        if (pInBuffer->frameCount == 0) {
1491            if (mBufferQueue.size()) {
1492                mBufferQueue.removeAt(0);
1493                delete [] pInBuffer->mBuffer;
1494                delete pInBuffer;
1495                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1496                        mThread.unsafe_get(), mBufferQueue.size());
1497            } else {
1498                break;
1499            }
1500        }
1501    }
1502
1503    // If we could not write all frames, allocate a buffer and queue it for next time.
1504    if (inBuffer.frameCount) {
1505        sp<ThreadBase> thread = mThread.promote();
1506        if (thread != 0 && !thread->standby()) {
1507            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1508                pInBuffer = new Buffer;
1509                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1510                pInBuffer->frameCount = inBuffer.frameCount;
1511                pInBuffer->i16 = pInBuffer->mBuffer;
1512                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1513                        sizeof(int16_t));
1514                mBufferQueue.add(pInBuffer);
1515                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1516                        mThread.unsafe_get(), mBufferQueue.size());
1517            } else {
1518                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1519                        mThread.unsafe_get(), this);
1520            }
1521        }
1522    }
1523
1524    // Calling write() with a 0 length buffer, means that no more data will be written:
1525    // If no more buffers are pending, fill output track buffer to make sure it is started
1526    // by output mixer.
1527    if (frames == 0 && mBufferQueue.size() == 0) {
1528        if (mCblk->user < mFrameCount) {
1529            frames = mFrameCount - mCblk->user;
1530            pInBuffer = new Buffer;
1531            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1532            pInBuffer->frameCount = frames;
1533            pInBuffer->i16 = pInBuffer->mBuffer;
1534            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1535            mBufferQueue.add(pInBuffer);
1536        } else if (mActive) {
1537            stop();
1538        }
1539    }
1540
1541    return outputBufferFull;
1542}
1543
1544status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1545        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1546{
1547    audio_track_cblk_t* cblk = mCblk;
1548    uint32_t framesReq = buffer->frameCount;
1549
1550    ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
1551    buffer->frameCount  = 0;
1552
1553    size_t framesAvail;
1554    {
1555        Mutex::Autolock _l(cblk->lock);
1556
1557        // read the server count again
1558        while (!(framesAvail = mClientProxy->framesAvailable_l())) {
1559            if (CC_UNLIKELY(!mActive)) {
1560                ALOGV("Not active and NO_MORE_BUFFERS");
1561                return NO_MORE_BUFFERS;
1562            }
1563            status_t result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
1564            if (result != NO_ERROR) {
1565                return NO_MORE_BUFFERS;
1566            }
1567        }
1568    }
1569
1570    if (framesReq > framesAvail) {
1571        framesReq = framesAvail;
1572    }
1573
1574    uint32_t u = cblk->user;
1575    uint32_t bufferEnd = cblk->userBase + mFrameCount;
1576
1577    if (framesReq > bufferEnd - u) {
1578        framesReq = bufferEnd - u;
1579    }
1580
1581    buffer->frameCount  = framesReq;
1582    buffer->raw         = mClientProxy->buffer(u);
1583    return NO_ERROR;
1584}
1585
1586
1587void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1588{
1589    size_t size = mBufferQueue.size();
1590
1591    for (size_t i = 0; i < size; i++) {
1592        Buffer *pBuffer = mBufferQueue.itemAt(i);
1593        delete [] pBuffer->mBuffer;
1594        delete pBuffer;
1595    }
1596    mBufferQueue.clear();
1597}
1598
1599
1600// ----------------------------------------------------------------------------
1601//      Record
1602// ----------------------------------------------------------------------------
1603
1604AudioFlinger::RecordHandle::RecordHandle(
1605        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1606    : BnAudioRecord(),
1607    mRecordTrack(recordTrack)
1608{
1609}
1610
1611AudioFlinger::RecordHandle::~RecordHandle() {
1612    stop_nonvirtual();
1613    mRecordTrack->destroy();
1614}
1615
1616sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
1617    return mRecordTrack->getCblk();
1618}
1619
1620status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1621        int triggerSession) {
1622    ALOGV("RecordHandle::start()");
1623    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1624}
1625
1626void AudioFlinger::RecordHandle::stop() {
1627    stop_nonvirtual();
1628}
1629
1630void AudioFlinger::RecordHandle::stop_nonvirtual() {
1631    ALOGV("RecordHandle::stop()");
1632    mRecordTrack->stop();
1633}
1634
1635status_t AudioFlinger::RecordHandle::onTransact(
1636    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1637{
1638    return BnAudioRecord::onTransact(code, data, reply, flags);
1639}
1640
1641// ----------------------------------------------------------------------------
1642
1643// RecordTrack constructor must be called with AudioFlinger::mLock held
1644AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1645            RecordThread *thread,
1646            const sp<Client>& client,
1647            uint32_t sampleRate,
1648            audio_format_t format,
1649            audio_channel_mask_t channelMask,
1650            size_t frameCount,
1651            int sessionId)
1652    :   TrackBase(thread, client, sampleRate, format,
1653                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
1654        mOverflow(false)
1655{
1656    ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
1657}
1658
1659AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1660{
1661    ALOGV("%s", __func__);
1662}
1663
1664// AudioBufferProvider interface
1665status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1666        int64_t pts)
1667{
1668    audio_track_cblk_t* cblk = this->cblk();
1669    uint32_t framesAvail;
1670    uint32_t framesReq = buffer->frameCount;
1671
1672    // Check if last stepServer failed, try to step now
1673    if (mStepServerFailed) {
1674        if (!step()) {
1675            goto getNextBuffer_exit;
1676        }
1677        ALOGV("stepServer recovered");
1678        mStepServerFailed = false;
1679    }
1680
1681    // FIXME lock is not actually held, so overrun is possible
1682    framesAvail = mServerProxy->framesAvailableIn_l();
1683
1684    if (CC_LIKELY(framesAvail)) {
1685        uint32_t s = cblk->server;
1686        uint32_t bufferEnd = cblk->serverBase + mFrameCount;
1687
1688        if (framesReq > framesAvail) {
1689            framesReq = framesAvail;
1690        }
1691        if (framesReq > bufferEnd - s) {
1692            framesReq = bufferEnd - s;
1693        }
1694
1695        buffer->raw = getBuffer(s, framesReq);
1696        buffer->frameCount = framesReq;
1697        return NO_ERROR;
1698    }
1699
1700getNextBuffer_exit:
1701    buffer->raw = NULL;
1702    buffer->frameCount = 0;
1703    return NOT_ENOUGH_DATA;
1704}
1705
1706status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1707                                                        int triggerSession)
1708{
1709    sp<ThreadBase> thread = mThread.promote();
1710    if (thread != 0) {
1711        RecordThread *recordThread = (RecordThread *)thread.get();
1712        return recordThread->start(this, event, triggerSession);
1713    } else {
1714        return BAD_VALUE;
1715    }
1716}
1717
1718void AudioFlinger::RecordThread::RecordTrack::stop()
1719{
1720    sp<ThreadBase> thread = mThread.promote();
1721    if (thread != 0) {
1722        RecordThread *recordThread = (RecordThread *)thread.get();
1723        recordThread->mLock.lock();
1724        bool doStop = recordThread->stop_l(this);
1725        if (doStop) {
1726            TrackBase::reset();
1727            // Force overrun condition to avoid false overrun callback until first data is
1728            // read from buffer
1729            android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
1730        }
1731        recordThread->mLock.unlock();
1732        if (doStop) {
1733            AudioSystem::stopInput(recordThread->id());
1734        }
1735    }
1736}
1737
1738void AudioFlinger::RecordThread::RecordTrack::destroy()
1739{
1740    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1741    sp<RecordTrack> keep(this);
1742    {
1743        sp<ThreadBase> thread = mThread.promote();
1744        if (thread != 0) {
1745            if (mState == ACTIVE || mState == RESUMING) {
1746                AudioSystem::stopInput(thread->id());
1747            }
1748            AudioSystem::releaseInput(thread->id());
1749            Mutex::Autolock _l(thread->mLock);
1750            RecordThread *recordThread = (RecordThread *) thread.get();
1751            recordThread->destroyTrack_l(this);
1752        }
1753    }
1754}
1755
1756
1757/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1758{
1759    result.append("   Clien Fmt Chn mask   Session Step S Serv     User   FrameCount\n");
1760}
1761
1762void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
1763{
1764    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %08x %08x %05d\n",
1765            (mClient == 0) ? getpid_cached : mClient->pid(),
1766            mFormat,
1767            mChannelMask,
1768            mSessionId,
1769            mStepCount,
1770            mState,
1771            mCblk->server,
1772            mCblk->user,
1773            mFrameCount);
1774}
1775
1776}; // namespace android
1777