Tracks.cpp revision d054c32443a493513ab63529b0c8b1aca290278c 
1/*
2**
3** Copyright 2012, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <math.h>
24#include <utils/Log.h>
25
26#include <private/media/AudioTrackShared.h>
27
28#include <common_time/cc_helper.h>
29#include <common_time/local_clock.h>
30
31#include "AudioMixer.h"
32#include "AudioFlinger.h"
33#include "ServiceUtilities.h"
34
35#include <media/nbaio/Pipe.h>
36#include <media/nbaio/PipeReader.h>
37
38// ----------------------------------------------------------------------------
39
40// Note: the following macro is used for extremely verbose logging message.  In
41// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
42// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
43// are so verbose that we want to suppress them even when we have ALOG_ASSERT
44// turned on.  Do not uncomment the #def below unless you really know what you
45// are doing and want to see all of the extremely verbose messages.
46//#define VERY_VERY_VERBOSE_LOGGING
47#ifdef VERY_VERY_VERBOSE_LOGGING
48#define ALOGVV ALOGV
49#else
50#define ALOGVV(a...) do { } while(0)
51#endif
52
53namespace android {
54
55// ----------------------------------------------------------------------------
56//      TrackBase
57// ----------------------------------------------------------------------------
58
59static volatile int32_t nextTrackId = 55;
60
61// TrackBase constructor must be called with AudioFlinger::mLock held
62AudioFlinger::ThreadBase::TrackBase::TrackBase(
63            ThreadBase *thread,
64            const sp<Client>& client,
65            uint32_t sampleRate,
66            audio_format_t format,
67            audio_channel_mask_t channelMask,
68            size_t frameCount,
69            const sp<IMemory>& sharedBuffer,
70            int sessionId,
71            bool isOut)
72    :   RefBase(),
73        mThread(thread),
74        mClient(client),
75        mCblk(NULL),
76        // mBuffer
77        mState(IDLE),
78        mSampleRate(sampleRate),
79        mFormat(format),
80        mChannelMask(channelMask),
81        mChannelCount(popcount(channelMask)),
82        mFrameSize(audio_is_linear_pcm(format) ?
83                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
84        mFrameCount(frameCount),
85        mSessionId(sessionId),
86        mIsOut(isOut),
87        mServerProxy(NULL),
88        mId(android_atomic_inc(&nextTrackId)),
89        mTerminated(false)
90{
91    // client == 0 implies sharedBuffer == 0
92    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
93
94    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
95            sharedBuffer->size());
96
97    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
98    size_t size = sizeof(audio_track_cblk_t);
99    size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
100    if (sharedBuffer == 0) {
101        size += bufferSize;
102    }
103
104    if (client != 0) {
105        mCblkMemory = client->heap()->allocate(size);
106        if (mCblkMemory != 0) {
107            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
108            // can't assume mCblk != NULL
109        } else {
110            ALOGE("not enough memory for AudioTrack size=%u", size);
111            client->heap()->dump("AudioTrack");
112            return;
113        }
114    } else {
115        // this syntax avoids calling the audio_track_cblk_t constructor twice
116        mCblk = (audio_track_cblk_t *) new uint8_t[size];
117        // assume mCblk != NULL
118    }
119
120    // construct the shared structure in-place.
121    if (mCblk != NULL) {
122        new(mCblk) audio_track_cblk_t();
123        // clear all buffers
124        mCblk->frameCount_ = frameCount;
125        if (sharedBuffer == 0) {
126            mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
127            memset(mBuffer, 0, bufferSize);
128        } else {
129            mBuffer = sharedBuffer->pointer();
130#if 0
131            mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
132#endif
133        }
134
135#ifdef TEE_SINK
136        if (mTeeSinkTrackEnabled) {
137            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
138            if (pipeFormat != Format_Invalid) {
139                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
140                size_t numCounterOffers = 0;
141                const NBAIO_Format offers[1] = {pipeFormat};
142                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
143                ALOG_ASSERT(index == 0);
144                PipeReader *pipeReader = new PipeReader(*pipe);
145                numCounterOffers = 0;
146                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
147                ALOG_ASSERT(index == 0);
148                mTeeSink = pipe;
149                mTeeSource = pipeReader;
150            }
151        }
152#endif
153
154    }
155}
156
157AudioFlinger::ThreadBase::TrackBase::~TrackBase()
158{
159#ifdef TEE_SINK
160    dumpTee(-1, mTeeSource, mId);
161#endif
162    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
163    delete mServerProxy;
164    if (mCblk != NULL) {
165        if (mClient == 0) {
166            delete mCblk;
167        } else {
168            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
169        }
170    }
171    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
172    if (mClient != 0) {
173        // Client destructor must run with AudioFlinger mutex locked
174        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
175        // If the client's reference count drops to zero, the associated destructor
176        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
177        // relying on the automatic clear() at end of scope.
178        mClient.clear();
179    }
180}
181
182// AudioBufferProvider interface
183// getNextBuffer() = 0;
184// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
185void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
186{
187#ifdef TEE_SINK
188    if (mTeeSink != 0) {
189        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
190    }
191#endif
192
193    ServerProxy::Buffer buf;
194    buf.mFrameCount = buffer->frameCount;
195    buf.mRaw = buffer->raw;
196    buffer->frameCount = 0;
197    buffer->raw = NULL;
198    mServerProxy->releaseBuffer(&buf);
199}
200
201status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
202{
203    mSyncEvents.add(event);
204    return NO_ERROR;
205}
206
207// ----------------------------------------------------------------------------
208//      Playback
209// ----------------------------------------------------------------------------
210
211AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
212    : BnAudioTrack(),
213      mTrack(track)
214{
215}
216
217AudioFlinger::TrackHandle::~TrackHandle() {
218    // just stop the track on deletion, associated resources
219    // will be freed from the main thread once all pending buffers have
220    // been played. Unless it's not in the active track list, in which
221    // case we free everything now...
222    mTrack->destroy();
223}
224
225sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
226    return mTrack->getCblk();
227}
228
229status_t AudioFlinger::TrackHandle::start() {
230    return mTrack->start();
231}
232
233void AudioFlinger::TrackHandle::stop() {
234    mTrack->stop();
235}
236
237void AudioFlinger::TrackHandle::flush() {
238    mTrack->flush();
239}
240
241void AudioFlinger::TrackHandle::pause() {
242    mTrack->pause();
243}
244
245status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
246{
247    return mTrack->attachAuxEffect(EffectId);
248}
249
250status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
251                                                         sp<IMemory>* buffer) {
252    if (!mTrack->isTimedTrack())
253        return INVALID_OPERATION;
254
255    PlaybackThread::TimedTrack* tt =
256            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
257    return tt->allocateTimedBuffer(size, buffer);
258}
259
260status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
261                                                     int64_t pts) {
262    if (!mTrack->isTimedTrack())
263        return INVALID_OPERATION;
264
265    PlaybackThread::TimedTrack* tt =
266            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
267    return tt->queueTimedBuffer(buffer, pts);
268}
269
270status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
271    const LinearTransform& xform, int target) {
272
273    if (!mTrack->isTimedTrack())
274        return INVALID_OPERATION;
275
276    PlaybackThread::TimedTrack* tt =
277            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
278    return tt->setMediaTimeTransform(
279        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
280}
281
282status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
283    return mTrack->setParameters(keyValuePairs);
284}
285
286status_t AudioFlinger::TrackHandle::onTransact(
287    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
288{
289    return BnAudioTrack::onTransact(code, data, reply, flags);
290}
291
292// ----------------------------------------------------------------------------
293
294// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
295AudioFlinger::PlaybackThread::Track::Track(
296            PlaybackThread *thread,
297            const sp<Client>& client,
298            audio_stream_type_t streamType,
299            uint32_t sampleRate,
300            audio_format_t format,
301            audio_channel_mask_t channelMask,
302            size_t frameCount,
303            const sp<IMemory>& sharedBuffer,
304            int sessionId,
305            IAudioFlinger::track_flags_t flags)
306    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
307            sessionId, true /*isOut*/),
308    mFillingUpStatus(FS_INVALID),
309    // mRetryCount initialized later when needed
310    mSharedBuffer(sharedBuffer),
311    mStreamType(streamType),
312    mName(-1),  // see note below
313    mMainBuffer(thread->mixBuffer()),
314    mAuxBuffer(NULL),
315    mAuxEffectId(0), mHasVolumeController(false),
316    mPresentationCompleteFrames(0),
317    mFlags(flags),
318    mFastIndex(-1),
319    mUnderrunCount(0),
320    mCachedVolume(1.0),
321    mIsInvalid(false),
322    mAudioTrackServerProxy(NULL),
323    mResumeToStopping(false)
324{
325    if (mCblk != NULL) {
326        if (sharedBuffer == 0) {
327            mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
328                    mFrameSize);
329        } else {
330            mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
331                    mFrameSize);
332        }
333        mServerProxy = mAudioTrackServerProxy;
334        // to avoid leaking a track name, do not allocate one unless there is an mCblk
335        mName = thread->getTrackName_l(channelMask, sessionId);
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            mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
343            ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
344            int i = __builtin_ctz(thread->mFastTrackAvailMask);
345            ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
346            // FIXME This is too eager.  We allocate a fast track index before the
347            //       fast track becomes active.  Since fast tracks are a scarce resource,
348            //       this means we are potentially denying other more important fast tracks from
349            //       being created.  It would be better to allocate the index dynamically.
350            mFastIndex = 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            Mutex::Autolock _l(thread->mLock);
380            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
381            bool wasActive = playbackThread->destroyTrack_l(this);
382            if (!isOutputTrack() && !wasActive) {
383                AudioSystem::releaseOutput(thread->id());
384            }
385        }
386    }
387}
388
389/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
390{
391    result.append("   Name Client Type Fmt Chn mask Session fCount S F SRate  "
392                  "L dB  R dB    Server Main buf  Aux Buf Flags Underruns\n");
393}
394
395void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
396{
397    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
398    if (isFastTrack()) {
399        sprintf(buffer, "   F %2d", mFastIndex);
400    } else {
401        sprintf(buffer, "   %4d", mName - AudioMixer::TRACK0);
402    }
403    track_state state = mState;
404    char stateChar;
405    if (isTerminated()) {
406        stateChar = 'T';
407    } else {
408        switch (state) {
409        case IDLE:
410            stateChar = 'I';
411            break;
412        case STOPPING_1:
413            stateChar = 's';
414            break;
415        case STOPPING_2:
416            stateChar = '5';
417            break;
418        case STOPPED:
419            stateChar = 'S';
420            break;
421        case RESUMING:
422            stateChar = 'R';
423            break;
424        case ACTIVE:
425            stateChar = 'A';
426            break;
427        case PAUSING:
428            stateChar = 'p';
429            break;
430        case PAUSED:
431            stateChar = 'P';
432            break;
433        case FLUSHED:
434            stateChar = 'F';
435            break;
436        default:
437            stateChar = '?';
438            break;
439        }
440    }
441    char nowInUnderrun;
442    switch (mObservedUnderruns.mBitFields.mMostRecent) {
443    case UNDERRUN_FULL:
444        nowInUnderrun = ' ';
445        break;
446    case UNDERRUN_PARTIAL:
447        nowInUnderrun = '<';
448        break;
449    case UNDERRUN_EMPTY:
450        nowInUnderrun = '*';
451        break;
452    default:
453        nowInUnderrun = '?';
454        break;
455    }
456    snprintf(&buffer[7], size-7, " %6u %4u %3u %08X %7u %6u %1c %1d %5u %5.2g %5.2g  "
457                                 "%08X %08X %08X 0x%03X %9u%c\n",
458            (mClient == 0) ? getpid_cached : mClient->pid(),
459            mStreamType,
460            mFormat,
461            mChannelMask,
462            mSessionId,
463            mFrameCount,
464            stateChar,
465            mFillingUpStatus,
466            mAudioTrackServerProxy->getSampleRate(),
467            20.0 * log10((vlr & 0xFFFF) / 4096.0),
468            20.0 * log10((vlr >> 16) / 4096.0),
469            mCblk->mServer,
470            (int)mMainBuffer,
471            (int)mAuxBuffer,
472            mCblk->mFlags,
473            mUnderrunCount,
474            nowInUnderrun);
475}
476
477uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
478    return mAudioTrackServerProxy->getSampleRate();
479}
480
481// AudioBufferProvider interface
482status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
483        AudioBufferProvider::Buffer* buffer, int64_t pts)
484{
485    ServerProxy::Buffer buf;
486    size_t desiredFrames = buffer->frameCount;
487    buf.mFrameCount = desiredFrames;
488    status_t status = mServerProxy->obtainBuffer(&buf);
489    buffer->frameCount = buf.mFrameCount;
490    buffer->raw = buf.mRaw;
491    if (buf.mFrameCount == 0) {
492        // only implemented so far for normal tracks, not fast tracks
493        mCblk->u.mStreaming.mUnderrunFrames += desiredFrames;
494        // FIXME also wake futex so that underrun is noticed more quickly
495        (void) android_atomic_or(CBLK_UNDERRUN, &mCblk->mFlags);
496    }
497    return status;
498}
499
500// Note that framesReady() takes a mutex on the control block using tryLock().
501// This could result in priority inversion if framesReady() is called by the normal mixer,
502// as the normal mixer thread runs at lower
503// priority than the client's callback thread:  there is a short window within framesReady()
504// during which the normal mixer could be preempted, and the client callback would block.
505// Another problem can occur if framesReady() is called by the fast mixer:
506// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
507// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
508size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
509    return mAudioTrackServerProxy->framesReady();
510}
511
512// Don't call for fast tracks; the framesReady() could result in priority inversion
513bool AudioFlinger::PlaybackThread::Track::isReady() const {
514    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
515        return true;
516    }
517
518    if (framesReady() >= mFrameCount ||
519            (mCblk->mFlags & CBLK_FORCEREADY)) {
520        mFillingUpStatus = FS_FILLED;
521        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
522        return true;
523    }
524    return false;
525}
526
527status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
528                                                    int triggerSession)
529{
530    status_t status = NO_ERROR;
531    ALOGV("start(%d), calling pid %d session %d",
532            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
533
534    sp<ThreadBase> thread = mThread.promote();
535    if (thread != 0) {
536        Mutex::Autolock _l(thread->mLock);
537        track_state state = mState;
538        // here the track could be either new, or restarted
539        // in both cases "unstop" the track
540
541        if (state == PAUSED) {
542            if (mResumeToStopping) {
543                // happened we need to resume to STOPPING_1
544                mState = TrackBase::STOPPING_1;
545                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
546            } else {
547                mState = TrackBase::RESUMING;
548                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
549            }
550        } else {
551            mState = TrackBase::ACTIVE;
552            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
553        }
554
555        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
556        status = playbackThread->addTrack_l(this);
557        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
558            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
559            //  restore previous state if start was rejected by policy manager
560            if (status == PERMISSION_DENIED) {
561                mState = state;
562            }
563        }
564        // track was already in the active list, not a problem
565        if (status == ALREADY_EXISTS) {
566            status = NO_ERROR;
567        }
568    } else {
569        status = BAD_VALUE;
570    }
571    return status;
572}
573
574void AudioFlinger::PlaybackThread::Track::stop()
575{
576    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
577    sp<ThreadBase> thread = mThread.promote();
578    if (thread != 0) {
579        Mutex::Autolock _l(thread->mLock);
580        track_state state = mState;
581        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
582            // If the track is not active (PAUSED and buffers full), flush buffers
583            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
584            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
585                reset();
586                mState = STOPPED;
587            } else if (!isFastTrack() && !isOffloaded()) {
588                mState = STOPPED;
589            } else {
590                // For fast tracks prepareTracks_l() will set state to STOPPING_2
591                // presentation is complete
592                // For an offloaded track this starts a drain and state will
593                // move to STOPPING_2 when drain completes and then STOPPED
594                mState = STOPPING_1;
595            }
596            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
597                    playbackThread);
598        }
599    }
600}
601
602void AudioFlinger::PlaybackThread::Track::pause()
603{
604    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
605    sp<ThreadBase> thread = mThread.promote();
606    if (thread != 0) {
607        Mutex::Autolock _l(thread->mLock);
608        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
609        switch (mState) {
610        case STOPPING_1:
611        case STOPPING_2:
612            if (!isOffloaded()) {
613                /* nothing to do if track is not offloaded */
614                break;
615            }
616
617            // Offloaded track was draining, we need to carry on draining when resumed
618            mResumeToStopping = true;
619            // fall through...
620        case ACTIVE:
621        case RESUMING:
622            mState = PAUSING;
623            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
624            playbackThread->signal_l();
625            break;
626
627        default:
628            break;
629        }
630    }
631}
632
633void AudioFlinger::PlaybackThread::Track::flush()
634{
635    ALOGV("flush(%d)", mName);
636    sp<ThreadBase> thread = mThread.promote();
637    if (thread != 0) {
638        Mutex::Autolock _l(thread->mLock);
639        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
640
641        if (isOffloaded()) {
642            // If offloaded we allow flush during any state except terminated
643            // and keep the track active to avoid problems if user is seeking
644            // rapidly and underlying hardware has a significant delay handling
645            // a pause
646            if (isTerminated()) {
647                return;
648            }
649
650            ALOGV("flush: offload flush");
651            reset();
652
653            if (mState == STOPPING_1 || mState == STOPPING_2) {
654                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
655                mState = ACTIVE;
656            }
657
658            if (mState == ACTIVE) {
659                ALOGV("flush called in active state, resetting buffer time out retry count");
660                mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
661            }
662
663            mResumeToStopping = false;
664        } else {
665            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
666                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
667                return;
668            }
669            // No point remaining in PAUSED state after a flush => go to
670            // FLUSHED state
671            mState = FLUSHED;
672            // do not reset the track if it is still in the process of being stopped or paused.
673            // this will be done by prepareTracks_l() when the track is stopped.
674            // prepareTracks_l() will see mState == FLUSHED, then
675            // remove from active track list, reset(), and trigger presentation complete
676            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
677                reset();
678            }
679        }
680        // Prevent flush being lost if the track is flushed and then resumed
681        // before mixer thread can run. This is important when offloading
682        // because the hardware buffer could hold a large amount of audio
683        playbackThread->flushOutput_l();
684        playbackThread->signal_l();
685    }
686}
687
688void AudioFlinger::PlaybackThread::Track::reset()
689{
690    // Do not reset twice to avoid discarding data written just after a flush and before
691    // the audioflinger thread detects the track is stopped.
692    if (!mResetDone) {
693        // Force underrun condition to avoid false underrun callback until first data is
694        // written to buffer
695        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
696        mFillingUpStatus = FS_FILLING;
697        mResetDone = true;
698        if (mState == FLUSHED) {
699            mState = IDLE;
700        }
701    }
702}
703
704status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
705{
706    sp<ThreadBase> thread = mThread.promote();
707    if (thread == 0) {
708        ALOGE("thread is dead");
709        return FAILED_TRANSACTION;
710    } else if ((thread->type() == ThreadBase::DIRECT) ||
711                    (thread->type() == ThreadBase::OFFLOAD)) {
712        return thread->setParameters(keyValuePairs);
713    } else {
714        return PERMISSION_DENIED;
715    }
716}
717
718status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
719{
720    status_t status = DEAD_OBJECT;
721    sp<ThreadBase> thread = mThread.promote();
722    if (thread != 0) {
723        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
724        sp<AudioFlinger> af = mClient->audioFlinger();
725
726        Mutex::Autolock _l(af->mLock);
727
728        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
729
730        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
731            Mutex::Autolock _dl(playbackThread->mLock);
732            Mutex::Autolock _sl(srcThread->mLock);
733            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
734            if (chain == 0) {
735                return INVALID_OPERATION;
736            }
737
738            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
739            if (effect == 0) {
740                return INVALID_OPERATION;
741            }
742            srcThread->removeEffect_l(effect);
743            playbackThread->addEffect_l(effect);
744            // removeEffect_l() has stopped the effect if it was active so it must be restarted
745            if (effect->state() == EffectModule::ACTIVE ||
746                    effect->state() == EffectModule::STOPPING) {
747                effect->start();
748            }
749
750            sp<EffectChain> dstChain = effect->chain().promote();
751            if (dstChain == 0) {
752                srcThread->addEffect_l(effect);
753                return INVALID_OPERATION;
754            }
755            AudioSystem::unregisterEffect(effect->id());
756            AudioSystem::registerEffect(&effect->desc(),
757                                        srcThread->id(),
758                                        dstChain->strategy(),
759                                        AUDIO_SESSION_OUTPUT_MIX,
760                                        effect->id());
761        }
762        status = playbackThread->attachAuxEffect(this, EffectId);
763    }
764    return status;
765}
766
767void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
768{
769    mAuxEffectId = EffectId;
770    mAuxBuffer = buffer;
771}
772
773bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
774                                                         size_t audioHalFrames)
775{
776    // a track is considered presented when the total number of frames written to audio HAL
777    // corresponds to the number of frames written when presentationComplete() is called for the
778    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
779    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
780    // to detect when all frames have been played. In this case framesWritten isn't
781    // useful because it doesn't always reflect whether there is data in the h/w
782    // buffers, particularly if a track has been paused and resumed during draining
783    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
784                      mPresentationCompleteFrames, framesWritten);
785    if (mPresentationCompleteFrames == 0) {
786        mPresentationCompleteFrames = framesWritten + audioHalFrames;
787        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
788                  mPresentationCompleteFrames, audioHalFrames);
789    }
790
791    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
792        ALOGV("presentationComplete() session %d complete: framesWritten %d",
793                  mSessionId, framesWritten);
794        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
795        mAudioTrackServerProxy->setStreamEndDone();
796        return true;
797    }
798    return false;
799}
800
801void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
802{
803    for (int i = 0; i < (int)mSyncEvents.size(); i++) {
804        if (mSyncEvents[i]->type() == type) {
805            mSyncEvents[i]->trigger();
806            mSyncEvents.removeAt(i);
807            i--;
808        }
809    }
810}
811
812// implement VolumeBufferProvider interface
813
814uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
815{
816    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
817    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
818    uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
819    uint32_t vl = vlr & 0xFFFF;
820    uint32_t vr = vlr >> 16;
821    // track volumes come from shared memory, so can't be trusted and must be clamped
822    if (vl > MAX_GAIN_INT) {
823        vl = MAX_GAIN_INT;
824    }
825    if (vr > MAX_GAIN_INT) {
826        vr = MAX_GAIN_INT;
827    }
828    // now apply the cached master volume and stream type volume;
829    // this is trusted but lacks any synchronization or barrier so may be stale
830    float v = mCachedVolume;
831    vl *= v;
832    vr *= v;
833    // re-combine into U4.16
834    vlr = (vr << 16) | (vl & 0xFFFF);
835    // FIXME look at mute, pause, and stop flags
836    return vlr;
837}
838
839status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
840{
841    if (isTerminated() || mState == PAUSED ||
842            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
843                                      (mState == STOPPED)))) {
844        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
845              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
846        event->cancel();
847        return INVALID_OPERATION;
848    }
849    (void) TrackBase::setSyncEvent(event);
850    return NO_ERROR;
851}
852
853void AudioFlinger::PlaybackThread::Track::invalidate()
854{
855    // FIXME should use proxy, and needs work
856    audio_track_cblk_t* cblk = mCblk;
857    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
858    android_atomic_release_store(0x40000000, &cblk->mFutex);
859    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
860    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
861    mIsInvalid = true;
862}
863
864// ----------------------------------------------------------------------------
865
866sp<AudioFlinger::PlaybackThread::TimedTrack>
867AudioFlinger::PlaybackThread::TimedTrack::create(
868            PlaybackThread *thread,
869            const sp<Client>& client,
870            audio_stream_type_t streamType,
871            uint32_t sampleRate,
872            audio_format_t format,
873            audio_channel_mask_t channelMask,
874            size_t frameCount,
875            const sp<IMemory>& sharedBuffer,
876            int sessionId) {
877    if (!client->reserveTimedTrack())
878        return 0;
879
880    return new TimedTrack(
881        thread, client, streamType, sampleRate, format, channelMask, frameCount,
882        sharedBuffer, sessionId);
883}
884
885AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
886            PlaybackThread *thread,
887            const sp<Client>& client,
888            audio_stream_type_t streamType,
889            uint32_t sampleRate,
890            audio_format_t format,
891            audio_channel_mask_t channelMask,
892            size_t frameCount,
893            const sp<IMemory>& sharedBuffer,
894            int sessionId)
895    : Track(thread, client, streamType, sampleRate, format, channelMask,
896            frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
897      mQueueHeadInFlight(false),
898      mTrimQueueHeadOnRelease(false),
899      mFramesPendingInQueue(0),
900      mTimedSilenceBuffer(NULL),
901      mTimedSilenceBufferSize(0),
902      mTimedAudioOutputOnTime(false),
903      mMediaTimeTransformValid(false)
904{
905    LocalClock lc;
906    mLocalTimeFreq = lc.getLocalFreq();
907
908    mLocalTimeToSampleTransform.a_zero = 0;
909    mLocalTimeToSampleTransform.b_zero = 0;
910    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
911    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
912    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
913                            &mLocalTimeToSampleTransform.a_to_b_denom);
914
915    mMediaTimeToSampleTransform.a_zero = 0;
916    mMediaTimeToSampleTransform.b_zero = 0;
917    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
918    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
919    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
920                            &mMediaTimeToSampleTransform.a_to_b_denom);
921}
922
923AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
924    mClient->releaseTimedTrack();
925    delete [] mTimedSilenceBuffer;
926}
927
928status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
929    size_t size, sp<IMemory>* buffer) {
930
931    Mutex::Autolock _l(mTimedBufferQueueLock);
932
933    trimTimedBufferQueue_l();
934
935    // lazily initialize the shared memory heap for timed buffers
936    if (mTimedMemoryDealer == NULL) {
937        const int kTimedBufferHeapSize = 512 << 10;
938
939        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
940                                              "AudioFlingerTimed");
941        if (mTimedMemoryDealer == NULL)
942            return NO_MEMORY;
943    }
944
945    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
946    if (newBuffer == NULL) {
947        newBuffer = mTimedMemoryDealer->allocate(size);
948        if (newBuffer == NULL)
949            return NO_MEMORY;
950    }
951
952    *buffer = newBuffer;
953    return NO_ERROR;
954}
955
956// caller must hold mTimedBufferQueueLock
957void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
958    int64_t mediaTimeNow;
959    {
960        Mutex::Autolock mttLock(mMediaTimeTransformLock);
961        if (!mMediaTimeTransformValid)
962            return;
963
964        int64_t targetTimeNow;
965        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
966            ? mCCHelper.getCommonTime(&targetTimeNow)
967            : mCCHelper.getLocalTime(&targetTimeNow);
968
969        if (OK != res)
970            return;
971
972        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
973                                                    &mediaTimeNow)) {
974            return;
975        }
976    }
977
978    size_t trimEnd;
979    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
980        int64_t bufEnd;
981
982        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
983            // We have a next buffer.  Just use its PTS as the PTS of the frame
984            // following the last frame in this buffer.  If the stream is sparse
985            // (ie, there are deliberate gaps left in the stream which should be
986            // filled with silence by the TimedAudioTrack), then this can result
987            // in one extra buffer being left un-trimmed when it could have
988            // been.  In general, this is not typical, and we would rather
989            // optimized away the TS calculation below for the more common case
990            // where PTSes are contiguous.
991            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
992        } else {
993            // We have no next buffer.  Compute the PTS of the frame following
994            // the last frame in this buffer by computing the duration of of
995            // this frame in media time units and adding it to the PTS of the
996            // buffer.
997            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
998                               / mFrameSize;
999
1000            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1001                                                                &bufEnd)) {
1002                ALOGE("Failed to convert frame count of %lld to media time"
1003                      " duration" " (scale factor %d/%u) in %s",
1004                      frameCount,
1005                      mMediaTimeToSampleTransform.a_to_b_numer,
1006                      mMediaTimeToSampleTransform.a_to_b_denom,
1007                      __PRETTY_FUNCTION__);
1008                break;
1009            }
1010            bufEnd += mTimedBufferQueue[trimEnd].pts();
1011        }
1012
1013        if (bufEnd > mediaTimeNow)
1014            break;
1015
1016        // Is the buffer we want to use in the middle of a mix operation right
1017        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1018        // from the mixer which should be coming back shortly.
1019        if (!trimEnd && mQueueHeadInFlight) {
1020            mTrimQueueHeadOnRelease = true;
1021        }
1022    }
1023
1024    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1025    if (trimStart < trimEnd) {
1026        // Update the bookkeeping for framesReady()
1027        for (size_t i = trimStart; i < trimEnd; ++i) {
1028            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1029        }
1030
1031        // Now actually remove the buffers from the queue.
1032        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1033    }
1034}
1035
1036void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1037        const char* logTag) {
1038    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1039                "%s called (reason \"%s\"), but timed buffer queue has no"
1040                " elements to trim.", __FUNCTION__, logTag);
1041
1042    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1043    mTimedBufferQueue.removeAt(0);
1044}
1045
1046void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1047        const TimedBuffer& buf,
1048        const char* logTag) {
1049    uint32_t bufBytes        = buf.buffer()->size();
1050    uint32_t consumedAlready = buf.position();
1051
1052    ALOG_ASSERT(consumedAlready <= bufBytes,
1053                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1054                " only %u bytes long, but claims to have consumed %u"
1055                " bytes.  (update reason: \"%s\")",
1056                bufBytes, consumedAlready, logTag);
1057
1058    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1059    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1060                "Bad bookkeeping while updating frames pending.  Should have at"
1061                " least %u queued frames, but we think we have only %u.  (update"
1062                " reason: \"%s\")",
1063                bufFrames, mFramesPendingInQueue, logTag);
1064
1065    mFramesPendingInQueue -= bufFrames;
1066}
1067
1068status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1069    const sp<IMemory>& buffer, int64_t pts) {
1070
1071    {
1072        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1073        if (!mMediaTimeTransformValid)
1074            return INVALID_OPERATION;
1075    }
1076
1077    Mutex::Autolock _l(mTimedBufferQueueLock);
1078
1079    uint32_t bufFrames = buffer->size() / mFrameSize;
1080    mFramesPendingInQueue += bufFrames;
1081    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1082
1083    return NO_ERROR;
1084}
1085
1086status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1087    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1088
1089    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1090           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1091           target);
1092
1093    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1094          target == TimedAudioTrack::COMMON_TIME)) {
1095        return BAD_VALUE;
1096    }
1097
1098    Mutex::Autolock lock(mMediaTimeTransformLock);
1099    mMediaTimeTransform = xform;
1100    mMediaTimeTransformTarget = target;
1101    mMediaTimeTransformValid = true;
1102
1103    return NO_ERROR;
1104}
1105
1106#define min(a, b) ((a) < (b) ? (a) : (b))
1107
1108// implementation of getNextBuffer for tracks whose buffers have timestamps
1109status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1110    AudioBufferProvider::Buffer* buffer, int64_t pts)
1111{
1112    if (pts == AudioBufferProvider::kInvalidPTS) {
1113        buffer->raw = NULL;
1114        buffer->frameCount = 0;
1115        mTimedAudioOutputOnTime = false;
1116        return INVALID_OPERATION;
1117    }
1118
1119    Mutex::Autolock _l(mTimedBufferQueueLock);
1120
1121    ALOG_ASSERT(!mQueueHeadInFlight,
1122                "getNextBuffer called without releaseBuffer!");
1123
1124    while (true) {
1125
1126        // if we have no timed buffers, then fail
1127        if (mTimedBufferQueue.isEmpty()) {
1128            buffer->raw = NULL;
1129            buffer->frameCount = 0;
1130            return NOT_ENOUGH_DATA;
1131        }
1132
1133        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1134
1135        // calculate the PTS of the head of the timed buffer queue expressed in
1136        // local time
1137        int64_t headLocalPTS;
1138        {
1139            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1140
1141            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1142
1143            if (mMediaTimeTransform.a_to_b_denom == 0) {
1144                // the transform represents a pause, so yield silence
1145                timedYieldSilence_l(buffer->frameCount, buffer);
1146                return NO_ERROR;
1147            }
1148
1149            int64_t transformedPTS;
1150            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1151                                                        &transformedPTS)) {
1152                // the transform failed.  this shouldn't happen, but if it does
1153                // then just drop this buffer
1154                ALOGW("timedGetNextBuffer transform failed");
1155                buffer->raw = NULL;
1156                buffer->frameCount = 0;
1157                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1158                return NO_ERROR;
1159            }
1160
1161            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1162                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1163                                                          &headLocalPTS)) {
1164                    buffer->raw = NULL;
1165                    buffer->frameCount = 0;
1166                    return INVALID_OPERATION;
1167                }
1168            } else {
1169                headLocalPTS = transformedPTS;
1170            }
1171        }
1172
1173        uint32_t sr = sampleRate();
1174
1175        // adjust the head buffer's PTS to reflect the portion of the head buffer
1176        // that has already been consumed
1177        int64_t effectivePTS = headLocalPTS +
1178                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1179
1180        // Calculate the delta in samples between the head of the input buffer
1181        // queue and the start of the next output buffer that will be written.
1182        // If the transformation fails because of over or underflow, it means
1183        // that the sample's position in the output stream is so far out of
1184        // whack that it should just be dropped.
1185        int64_t sampleDelta;
1186        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1187            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1188            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1189                                       " mix");
1190            continue;
1191        }
1192        if (!mLocalTimeToSampleTransform.doForwardTransform(
1193                (effectivePTS - pts) << 32, &sampleDelta)) {
1194            ALOGV("*** too late during sample rate transform: dropped buffer");
1195            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1196            continue;
1197        }
1198
1199        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1200               " sampleDelta=[%d.%08x]",
1201               head.pts(), head.position(), pts,
1202               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1203                   + (sampleDelta >> 32)),
1204               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1205
1206        // if the delta between the ideal placement for the next input sample and
1207        // the current output position is within this threshold, then we will
1208        // concatenate the next input samples to the previous output
1209        const int64_t kSampleContinuityThreshold =
1210                (static_cast<int64_t>(sr) << 32) / 250;
1211
1212        // if this is the first buffer of audio that we're emitting from this track
1213        // then it should be almost exactly on time.
1214        const int64_t kSampleStartupThreshold = 1LL << 32;
1215
1216        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1217           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1218            // the next input is close enough to being on time, so concatenate it
1219            // with the last output
1220            timedYieldSamples_l(buffer);
1221
1222            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1223                    head.position(), buffer->frameCount);
1224            return NO_ERROR;
1225        }
1226
1227        // Looks like our output is not on time.  Reset our on timed status.
1228        // Next time we mix samples from our input queue, then should be within
1229        // the StartupThreshold.
1230        mTimedAudioOutputOnTime = false;
1231        if (sampleDelta > 0) {
1232            // the gap between the current output position and the proper start of
1233            // the next input sample is too big, so fill it with silence
1234            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1235
1236            timedYieldSilence_l(framesUntilNextInput, buffer);
1237            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1238            return NO_ERROR;
1239        } else {
1240            // the next input sample is late
1241            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1242            size_t onTimeSamplePosition =
1243                    head.position() + lateFrames * mFrameSize;
1244
1245            if (onTimeSamplePosition > head.buffer()->size()) {
1246                // all the remaining samples in the head are too late, so
1247                // drop it and move on
1248                ALOGV("*** too late: dropped buffer");
1249                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1250                continue;
1251            } else {
1252                // skip over the late samples
1253                head.setPosition(onTimeSamplePosition);
1254
1255                // yield the available samples
1256                timedYieldSamples_l(buffer);
1257
1258                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1259                return NO_ERROR;
1260            }
1261        }
1262    }
1263}
1264
1265// Yield samples from the timed buffer queue head up to the given output
1266// buffer's capacity.
1267//
1268// Caller must hold mTimedBufferQueueLock
1269void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1270    AudioBufferProvider::Buffer* buffer) {
1271
1272    const TimedBuffer& head = mTimedBufferQueue[0];
1273
1274    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1275                   head.position());
1276
1277    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1278                                 mFrameSize);
1279    size_t framesRequested = buffer->frameCount;
1280    buffer->frameCount = min(framesLeftInHead, framesRequested);
1281
1282    mQueueHeadInFlight = true;
1283    mTimedAudioOutputOnTime = true;
1284}
1285
1286// Yield samples of silence up to the given output buffer's capacity
1287//
1288// Caller must hold mTimedBufferQueueLock
1289void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1290    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1291
1292    // lazily allocate a buffer filled with silence
1293    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1294        delete [] mTimedSilenceBuffer;
1295        mTimedSilenceBufferSize = numFrames * mFrameSize;
1296        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1297        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1298    }
1299
1300    buffer->raw = mTimedSilenceBuffer;
1301    size_t framesRequested = buffer->frameCount;
1302    buffer->frameCount = min(numFrames, framesRequested);
1303
1304    mTimedAudioOutputOnTime = false;
1305}
1306
1307// AudioBufferProvider interface
1308void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1309    AudioBufferProvider::Buffer* buffer) {
1310
1311    Mutex::Autolock _l(mTimedBufferQueueLock);
1312
1313    // If the buffer which was just released is part of the buffer at the head
1314    // of the queue, be sure to update the amt of the buffer which has been
1315    // consumed.  If the buffer being returned is not part of the head of the
1316    // queue, its either because the buffer is part of the silence buffer, or
1317    // because the head of the timed queue was trimmed after the mixer called
1318    // getNextBuffer but before the mixer called releaseBuffer.
1319    if (buffer->raw == mTimedSilenceBuffer) {
1320        ALOG_ASSERT(!mQueueHeadInFlight,
1321                    "Queue head in flight during release of silence buffer!");
1322        goto done;
1323    }
1324
1325    ALOG_ASSERT(mQueueHeadInFlight,
1326                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1327                " head in flight.");
1328
1329    if (mTimedBufferQueue.size()) {
1330        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1331
1332        void* start = head.buffer()->pointer();
1333        void* end   = reinterpret_cast<void*>(
1334                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1335                        + head.buffer()->size());
1336
1337        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1338                    "released buffer not within the head of the timed buffer"
1339                    " queue; qHead = [%p, %p], released buffer = %p",
1340                    start, end, buffer->raw);
1341
1342        head.setPosition(head.position() +
1343                (buffer->frameCount * mFrameSize));
1344        mQueueHeadInFlight = false;
1345
1346        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1347                    "Bad bookkeeping during releaseBuffer!  Should have at"
1348                    " least %u queued frames, but we think we have only %u",
1349                    buffer->frameCount, mFramesPendingInQueue);
1350
1351        mFramesPendingInQueue -= buffer->frameCount;
1352
1353        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1354            || mTrimQueueHeadOnRelease) {
1355            trimTimedBufferQueueHead_l("releaseBuffer");
1356            mTrimQueueHeadOnRelease = false;
1357        }
1358    } else {
1359        LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1360                  " buffers in the timed buffer queue");
1361    }
1362
1363done:
1364    buffer->raw = 0;
1365    buffer->frameCount = 0;
1366}
1367
1368size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1369    Mutex::Autolock _l(mTimedBufferQueueLock);
1370    return mFramesPendingInQueue;
1371}
1372
1373AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1374        : mPTS(0), mPosition(0) {}
1375
1376AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1377    const sp<IMemory>& buffer, int64_t pts)
1378        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1379
1380
1381// ----------------------------------------------------------------------------
1382
1383AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1384            PlaybackThread *playbackThread,
1385            DuplicatingThread *sourceThread,
1386            uint32_t sampleRate,
1387            audio_format_t format,
1388            audio_channel_mask_t channelMask,
1389            size_t frameCount)
1390    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1391                NULL, 0, IAudioFlinger::TRACK_DEFAULT),
1392    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1393{
1394
1395    if (mCblk != NULL) {
1396        mOutBuffer.frameCount = 0;
1397        playbackThread->mTracks.add(this);
1398        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1399                "mCblk->frameCount_ %u, mChannelMask 0x%08x",
1400                mCblk, mBuffer,
1401                mCblk->frameCount_, mChannelMask);
1402        // since client and server are in the same process,
1403        // the buffer has the same virtual address on both sides
1404        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1405        mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
1406        mClientProxy->setSendLevel(0.0);
1407        mClientProxy->setSampleRate(sampleRate);
1408        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1409                true /*clientInServer*/);
1410    } else {
1411        ALOGW("Error creating output track on thread %p", playbackThread);
1412    }
1413}
1414
1415AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1416{
1417    clearBufferQueue();
1418    delete mClientProxy;
1419    // superclass destructor will now delete the server proxy and shared memory both refer to
1420}
1421
1422status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1423                                                          int triggerSession)
1424{
1425    status_t status = Track::start(event, triggerSession);
1426    if (status != NO_ERROR) {
1427        return status;
1428    }
1429
1430    mActive = true;
1431    mRetryCount = 127;
1432    return status;
1433}
1434
1435void AudioFlinger::PlaybackThread::OutputTrack::stop()
1436{
1437    Track::stop();
1438    clearBufferQueue();
1439    mOutBuffer.frameCount = 0;
1440    mActive = false;
1441}
1442
1443bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1444{
1445    Buffer *pInBuffer;
1446    Buffer inBuffer;
1447    uint32_t channelCount = mChannelCount;
1448    bool outputBufferFull = false;
1449    inBuffer.frameCount = frames;
1450    inBuffer.i16 = data;
1451
1452    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1453
1454    if (!mActive && frames != 0) {
1455        start();
1456        sp<ThreadBase> thread = mThread.promote();
1457        if (thread != 0) {
1458            MixerThread *mixerThread = (MixerThread *)thread.get();
1459            if (mFrameCount > frames) {
1460                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1461                    uint32_t startFrames = (mFrameCount - frames);
1462                    pInBuffer = new Buffer;
1463                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1464                    pInBuffer->frameCount = startFrames;
1465                    pInBuffer->i16 = pInBuffer->mBuffer;
1466                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1467                    mBufferQueue.add(pInBuffer);
1468                } else {
1469                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1470                }
1471            }
1472        }
1473    }
1474
1475    while (waitTimeLeftMs) {
1476        // First write pending buffers, then new data
1477        if (mBufferQueue.size()) {
1478            pInBuffer = mBufferQueue.itemAt(0);
1479        } else {
1480            pInBuffer = &inBuffer;
1481        }
1482
1483        if (pInBuffer->frameCount == 0) {
1484            break;
1485        }
1486
1487        if (mOutBuffer.frameCount == 0) {
1488            mOutBuffer.frameCount = pInBuffer->frameCount;
1489            nsecs_t startTime = systemTime();
1490            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1491            if (status != NO_ERROR) {
1492                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1493                        mThread.unsafe_get(), status);
1494                outputBufferFull = true;
1495                break;
1496            }
1497            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1498            if (waitTimeLeftMs >= waitTimeMs) {
1499                waitTimeLeftMs -= waitTimeMs;
1500            } else {
1501                waitTimeLeftMs = 0;
1502            }
1503        }
1504
1505        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1506                pInBuffer->frameCount;
1507        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1508        Proxy::Buffer buf;
1509        buf.mFrameCount = outFrames;
1510        buf.mRaw = NULL;
1511        mClientProxy->releaseBuffer(&buf);
1512        pInBuffer->frameCount -= outFrames;
1513        pInBuffer->i16 += outFrames * channelCount;
1514        mOutBuffer.frameCount -= outFrames;
1515        mOutBuffer.i16 += outFrames * channelCount;
1516
1517        if (pInBuffer->frameCount == 0) {
1518            if (mBufferQueue.size()) {
1519                mBufferQueue.removeAt(0);
1520                delete [] pInBuffer->mBuffer;
1521                delete pInBuffer;
1522                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1523                        mThread.unsafe_get(), mBufferQueue.size());
1524            } else {
1525                break;
1526            }
1527        }
1528    }
1529
1530    // If we could not write all frames, allocate a buffer and queue it for next time.
1531    if (inBuffer.frameCount) {
1532        sp<ThreadBase> thread = mThread.promote();
1533        if (thread != 0 && !thread->standby()) {
1534            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1535                pInBuffer = new Buffer;
1536                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1537                pInBuffer->frameCount = inBuffer.frameCount;
1538                pInBuffer->i16 = pInBuffer->mBuffer;
1539                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1540                        sizeof(int16_t));
1541                mBufferQueue.add(pInBuffer);
1542                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1543                        mThread.unsafe_get(), mBufferQueue.size());
1544            } else {
1545                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1546                        mThread.unsafe_get(), this);
1547            }
1548        }
1549    }
1550
1551    // Calling write() with a 0 length buffer, means that no more data will be written:
1552    // If no more buffers are pending, fill output track buffer to make sure it is started
1553    // by output mixer.
1554    if (frames == 0 && mBufferQueue.size() == 0) {
1555        // FIXME borken, replace by getting framesReady() from proxy
1556        size_t user = 0;    // was mCblk->user
1557        if (user < mFrameCount) {
1558            frames = mFrameCount - user;
1559            pInBuffer = new Buffer;
1560            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1561            pInBuffer->frameCount = frames;
1562            pInBuffer->i16 = pInBuffer->mBuffer;
1563            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1564            mBufferQueue.add(pInBuffer);
1565        } else if (mActive) {
1566            stop();
1567        }
1568    }
1569
1570    return outputBufferFull;
1571}
1572
1573status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1574        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1575{
1576    ClientProxy::Buffer buf;
1577    buf.mFrameCount = buffer->frameCount;
1578    struct timespec timeout;
1579    timeout.tv_sec = waitTimeMs / 1000;
1580    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1581    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1582    buffer->frameCount = buf.mFrameCount;
1583    buffer->raw = buf.mRaw;
1584    return status;
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");
1657    if (mCblk != NULL) {
1658        mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
1659                mFrameSize);
1660        mServerProxy = mAudioRecordServerProxy;
1661    }
1662}
1663
1664AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1665{
1666    ALOGV("%s", __func__);
1667}
1668
1669// AudioBufferProvider interface
1670status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1671        int64_t pts)
1672{
1673    ServerProxy::Buffer buf;
1674    buf.mFrameCount = buffer->frameCount;
1675    status_t status = mServerProxy->obtainBuffer(&buf);
1676    buffer->frameCount = buf.mFrameCount;
1677    buffer->raw = buf.mRaw;
1678    if (buf.mFrameCount == 0) {
1679        // FIXME also wake futex so that overrun is noticed more quickly
1680        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1681    }
1682    return status;
1683}
1684
1685status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1686                                                        int triggerSession)
1687{
1688    sp<ThreadBase> thread = mThread.promote();
1689    if (thread != 0) {
1690        RecordThread *recordThread = (RecordThread *)thread.get();
1691        return recordThread->start(this, event, triggerSession);
1692    } else {
1693        return BAD_VALUE;
1694    }
1695}
1696
1697void AudioFlinger::RecordThread::RecordTrack::stop()
1698{
1699    sp<ThreadBase> thread = mThread.promote();
1700    if (thread != 0) {
1701        RecordThread *recordThread = (RecordThread *)thread.get();
1702        if (recordThread->stop(this)) {
1703            AudioSystem::stopInput(recordThread->id());
1704        }
1705    }
1706}
1707
1708void AudioFlinger::RecordThread::RecordTrack::destroy()
1709{
1710    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1711    sp<RecordTrack> keep(this);
1712    {
1713        sp<ThreadBase> thread = mThread.promote();
1714        if (thread != 0) {
1715            if (mState == ACTIVE || mState == RESUMING) {
1716                AudioSystem::stopInput(thread->id());
1717            }
1718            AudioSystem::releaseInput(thread->id());
1719            Mutex::Autolock _l(thread->mLock);
1720            RecordThread *recordThread = (RecordThread *) thread.get();
1721            recordThread->destroyTrack_l(this);
1722        }
1723    }
1724}
1725
1726
1727/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1728{
1729    result.append("Client Fmt Chn mask Session S   Server fCount\n");
1730}
1731
1732void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
1733{
1734    snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n",
1735            (mClient == 0) ? getpid_cached : mClient->pid(),
1736            mFormat,
1737            mChannelMask,
1738            mSessionId,
1739            mState,
1740            mCblk->mServer,
1741            mFrameCount);
1742}
1743
1744}; // namespace android
1745