Tracks.cpp revision c56f3426099a3cf2d07ccff8886050c7fbce140f 
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#include <audio_utils/minifloat.h>
38
39// ----------------------------------------------------------------------------
40
41// Note: the following macro is used for extremely verbose logging message.  In
42// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
43// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
44// are so verbose that we want to suppress them even when we have ALOG_ASSERT
45// turned on.  Do not uncomment the #def below unless you really know what you
46// are doing and want to see all of the extremely verbose messages.
47//#define VERY_VERY_VERBOSE_LOGGING
48#ifdef VERY_VERY_VERBOSE_LOGGING
49#define ALOGVV ALOGV
50#else
51#define ALOGVV(a...) do { } while(0)
52#endif
53
54namespace android {
55
56// ----------------------------------------------------------------------------
57//      TrackBase
58// ----------------------------------------------------------------------------
59
60static volatile int32_t nextTrackId = 55;
61
62// TrackBase constructor must be called with AudioFlinger::mLock held
63AudioFlinger::ThreadBase::TrackBase::TrackBase(
64            ThreadBase *thread,
65            const sp<Client>& client,
66            uint32_t sampleRate,
67            audio_format_t format,
68            audio_channel_mask_t channelMask,
69            size_t frameCount,
70            const sp<IMemory>& sharedBuffer,
71            int sessionId,
72            int clientUid,
73            IAudioFlinger::track_flags_t flags,
74            bool isOut,
75            bool useReadOnlyHeap)
76    :   RefBase(),
77        mThread(thread),
78        mClient(client),
79        mCblk(NULL),
80        // mBuffer
81        mState(IDLE),
82        mSampleRate(sampleRate),
83        mFormat(format),
84        mChannelMask(channelMask),
85        mChannelCount(popcount(channelMask)),
86        mFrameSize(audio_is_linear_pcm(format) ?
87                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
88        mFrameCount(frameCount),
89        mSessionId(sessionId),
90        mFlags(flags),
91        mIsOut(isOut),
92        mServerProxy(NULL),
93        mId(android_atomic_inc(&nextTrackId)),
94        mTerminated(false)
95{
96    // if the caller is us, trust the specified uid
97    if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
98        int newclientUid = IPCThreadState::self()->getCallingUid();
99        if (clientUid != -1 && clientUid != newclientUid) {
100            ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
101        }
102        clientUid = newclientUid;
103    }
104    // clientUid contains the uid of the app that is responsible for this track, so we can blame
105    // battery usage on it.
106    mUid = clientUid;
107
108    // client == 0 implies sharedBuffer == 0
109    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
110
111    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
112            sharedBuffer->size());
113
114    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
115    size_t size = sizeof(audio_track_cblk_t);
116    size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
117    if (sharedBuffer == 0 && !useReadOnlyHeap) {
118        size += bufferSize;
119    }
120
121    if (client != 0) {
122        mCblkMemory = client->heap()->allocate(size);
123        if (mCblkMemory == 0 ||
124                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
125            ALOGE("not enough memory for AudioTrack size=%u", size);
126            client->heap()->dump("AudioTrack");
127            mCblkMemory.clear();
128            return;
129        }
130    } else {
131        // this syntax avoids calling the audio_track_cblk_t constructor twice
132        mCblk = (audio_track_cblk_t *) new uint8_t[size];
133        // assume mCblk != NULL
134    }
135
136    // construct the shared structure in-place.
137    if (mCblk != NULL) {
138        new(mCblk) audio_track_cblk_t();
139        if (useReadOnlyHeap) {
140            const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
141            if (roHeap == 0 ||
142                    (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
143                    (mBuffer = mBufferMemory->pointer()) == NULL) {
144                ALOGE("not enough memory for read-only buffer size=%zu", bufferSize);
145                if (roHeap != 0) {
146                    roHeap->dump("buffer");
147                }
148                mCblkMemory.clear();
149                mBufferMemory.clear();
150                return;
151            }
152            memset(mBuffer, 0, bufferSize);
153        } else {
154            // clear all buffers
155            if (sharedBuffer == 0) {
156                mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
157                memset(mBuffer, 0, bufferSize);
158            } else {
159                mBuffer = sharedBuffer->pointer();
160#if 0
161                mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
162#endif
163            }
164        }
165
166#ifdef TEE_SINK
167        if (mTeeSinkTrackEnabled) {
168            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
169            if (Format_isValid(pipeFormat)) {
170                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
171                size_t numCounterOffers = 0;
172                const NBAIO_Format offers[1] = {pipeFormat};
173                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
174                ALOG_ASSERT(index == 0);
175                PipeReader *pipeReader = new PipeReader(*pipe);
176                numCounterOffers = 0;
177                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
178                ALOG_ASSERT(index == 0);
179                mTeeSink = pipe;
180                mTeeSource = pipeReader;
181            }
182        }
183#endif
184
185    }
186}
187
188AudioFlinger::ThreadBase::TrackBase::~TrackBase()
189{
190#ifdef TEE_SINK
191    dumpTee(-1, mTeeSource, mId);
192#endif
193    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
194    delete mServerProxy;
195    if (mCblk != NULL) {
196        if (mClient == 0) {
197            delete mCblk;
198        } else {
199            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
200        }
201    }
202    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
203    if (mClient != 0) {
204        // Client destructor must run with AudioFlinger client mutex locked
205        Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
206        // If the client's reference count drops to zero, the associated destructor
207        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
208        // relying on the automatic clear() at end of scope.
209        mClient.clear();
210    }
211}
212
213// AudioBufferProvider interface
214// getNextBuffer() = 0;
215// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
216void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
217{
218#ifdef TEE_SINK
219    if (mTeeSink != 0) {
220        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
221    }
222#endif
223
224    ServerProxy::Buffer buf;
225    buf.mFrameCount = buffer->frameCount;
226    buf.mRaw = buffer->raw;
227    buffer->frameCount = 0;
228    buffer->raw = NULL;
229    mServerProxy->releaseBuffer(&buf);
230}
231
232status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
233{
234    mSyncEvents.add(event);
235    return NO_ERROR;
236}
237
238// ----------------------------------------------------------------------------
239//      Playback
240// ----------------------------------------------------------------------------
241
242AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
243    : BnAudioTrack(),
244      mTrack(track)
245{
246}
247
248AudioFlinger::TrackHandle::~TrackHandle() {
249    // just stop the track on deletion, associated resources
250    // will be freed from the main thread once all pending buffers have
251    // been played. Unless it's not in the active track list, in which
252    // case we free everything now...
253    mTrack->destroy();
254}
255
256sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
257    return mTrack->getCblk();
258}
259
260status_t AudioFlinger::TrackHandle::start() {
261    return mTrack->start();
262}
263
264void AudioFlinger::TrackHandle::stop() {
265    mTrack->stop();
266}
267
268void AudioFlinger::TrackHandle::flush() {
269    mTrack->flush();
270}
271
272void AudioFlinger::TrackHandle::pause() {
273    mTrack->pause();
274}
275
276status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
277{
278    return mTrack->attachAuxEffect(EffectId);
279}
280
281status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
282                                                         sp<IMemory>* buffer) {
283    if (!mTrack->isTimedTrack())
284        return INVALID_OPERATION;
285
286    PlaybackThread::TimedTrack* tt =
287            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
288    return tt->allocateTimedBuffer(size, buffer);
289}
290
291status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
292                                                     int64_t pts) {
293    if (!mTrack->isTimedTrack())
294        return INVALID_OPERATION;
295
296    if (buffer == 0 || buffer->pointer() == NULL) {
297        ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
298        return BAD_VALUE;
299    }
300
301    PlaybackThread::TimedTrack* tt =
302            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
303    return tt->queueTimedBuffer(buffer, pts);
304}
305
306status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
307    const LinearTransform& xform, int target) {
308
309    if (!mTrack->isTimedTrack())
310        return INVALID_OPERATION;
311
312    PlaybackThread::TimedTrack* tt =
313            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
314    return tt->setMediaTimeTransform(
315        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
316}
317
318status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
319    return mTrack->setParameters(keyValuePairs);
320}
321
322status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
323{
324    return mTrack->getTimestamp(timestamp);
325}
326
327
328void AudioFlinger::TrackHandle::signal()
329{
330    return mTrack->signal();
331}
332
333status_t AudioFlinger::TrackHandle::onTransact(
334    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
335{
336    return BnAudioTrack::onTransact(code, data, reply, flags);
337}
338
339// ----------------------------------------------------------------------------
340
341// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
342AudioFlinger::PlaybackThread::Track::Track(
343            PlaybackThread *thread,
344            const sp<Client>& client,
345            audio_stream_type_t streamType,
346            uint32_t sampleRate,
347            audio_format_t format,
348            audio_channel_mask_t channelMask,
349            size_t frameCount,
350            const sp<IMemory>& sharedBuffer,
351            int sessionId,
352            int uid,
353            IAudioFlinger::track_flags_t flags)
354    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
355            sessionId, uid, flags, true /*isOut*/),
356    mFillingUpStatus(FS_INVALID),
357    // mRetryCount initialized later when needed
358    mSharedBuffer(sharedBuffer),
359    mStreamType(streamType),
360    mName(-1),  // see note below
361    mMainBuffer(thread->mixBuffer()),
362    mAuxBuffer(NULL),
363    mAuxEffectId(0), mHasVolumeController(false),
364    mPresentationCompleteFrames(0),
365    mFastIndex(-1),
366    mCachedVolume(1.0),
367    mIsInvalid(false),
368    mAudioTrackServerProxy(NULL),
369    mResumeToStopping(false),
370    mFlushHwPending(false)
371{
372    if (mCblk == NULL) {
373        return;
374    }
375
376    if (sharedBuffer == 0) {
377        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
378                mFrameSize);
379    } else {
380        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
381                mFrameSize);
382    }
383    mServerProxy = mAudioTrackServerProxy;
384
385    mName = thread->getTrackName_l(channelMask, sessionId);
386    if (mName < 0) {
387        ALOGE("no more track names available");
388        return;
389    }
390    // only allocate a fast track index if we were able to allocate a normal track name
391    if (flags & IAudioFlinger::TRACK_FAST) {
392        mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
393        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
394        int i = __builtin_ctz(thread->mFastTrackAvailMask);
395        ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
396        // FIXME This is too eager.  We allocate a fast track index before the
397        //       fast track becomes active.  Since fast tracks are a scarce resource,
398        //       this means we are potentially denying other more important fast tracks from
399        //       being created.  It would be better to allocate the index dynamically.
400        mFastIndex = i;
401        // Read the initial underruns because this field is never cleared by the fast mixer
402        mObservedUnderruns = thread->getFastTrackUnderruns(i);
403        thread->mFastTrackAvailMask &= ~(1 << i);
404    }
405}
406
407AudioFlinger::PlaybackThread::Track::~Track()
408{
409    ALOGV("PlaybackThread::Track destructor");
410
411    // The destructor would clear mSharedBuffer,
412    // but it will not push the decremented reference count,
413    // leaving the client's IMemory dangling indefinitely.
414    // This prevents that leak.
415    if (mSharedBuffer != 0) {
416        mSharedBuffer.clear();
417        // flush the binder command buffer
418        IPCThreadState::self()->flushCommands();
419    }
420}
421
422status_t AudioFlinger::PlaybackThread::Track::initCheck() const
423{
424    status_t status = TrackBase::initCheck();
425    if (status == NO_ERROR && mName < 0) {
426        status = NO_MEMORY;
427    }
428    return status;
429}
430
431void AudioFlinger::PlaybackThread::Track::destroy()
432{
433    // NOTE: destroyTrack_l() can remove a strong reference to this Track
434    // by removing it from mTracks vector, so there is a risk that this Tracks's
435    // destructor is called. As the destructor needs to lock mLock,
436    // we must acquire a strong reference on this Track before locking mLock
437    // here so that the destructor is called only when exiting this function.
438    // On the other hand, as long as Track::destroy() is only called by
439    // TrackHandle destructor, the TrackHandle still holds a strong ref on
440    // this Track with its member mTrack.
441    sp<Track> keep(this);
442    { // scope for mLock
443        sp<ThreadBase> thread = mThread.promote();
444        if (thread != 0) {
445            Mutex::Autolock _l(thread->mLock);
446            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
447            bool wasActive = playbackThread->destroyTrack_l(this);
448            if (!isOutputTrack() && !wasActive) {
449                AudioSystem::releaseOutput(thread->id());
450            }
451        }
452    }
453}
454
455/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
456{
457    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
458                  "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
459}
460
461void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
462{
463    gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
464    if (isFastTrack()) {
465        sprintf(buffer, "    F %2d", mFastIndex);
466    } else if (mName >= AudioMixer::TRACK0) {
467        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
468    } else {
469        sprintf(buffer, "    none");
470    }
471    track_state state = mState;
472    char stateChar;
473    if (isTerminated()) {
474        stateChar = 'T';
475    } else {
476        switch (state) {
477        case IDLE:
478            stateChar = 'I';
479            break;
480        case STOPPING_1:
481            stateChar = 's';
482            break;
483        case STOPPING_2:
484            stateChar = '5';
485            break;
486        case STOPPED:
487            stateChar = 'S';
488            break;
489        case RESUMING:
490            stateChar = 'R';
491            break;
492        case ACTIVE:
493            stateChar = 'A';
494            break;
495        case PAUSING:
496            stateChar = 'p';
497            break;
498        case PAUSED:
499            stateChar = 'P';
500            break;
501        case FLUSHED:
502            stateChar = 'F';
503            break;
504        default:
505            stateChar = '?';
506            break;
507        }
508    }
509    char nowInUnderrun;
510    switch (mObservedUnderruns.mBitFields.mMostRecent) {
511    case UNDERRUN_FULL:
512        nowInUnderrun = ' ';
513        break;
514    case UNDERRUN_PARTIAL:
515        nowInUnderrun = '<';
516        break;
517    case UNDERRUN_EMPTY:
518        nowInUnderrun = '*';
519        break;
520    default:
521        nowInUnderrun = '?';
522        break;
523    }
524    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
525                                 "%08X %p %p 0x%03X %9u%c\n",
526            active ? "yes" : "no",
527            (mClient == 0) ? getpid_cached : mClient->pid(),
528            mStreamType,
529            mFormat,
530            mChannelMask,
531            mSessionId,
532            mFrameCount,
533            stateChar,
534            mFillingUpStatus,
535            mAudioTrackServerProxy->getSampleRate(),
536            20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
537            20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
538            mCblk->mServer,
539            mMainBuffer,
540            mAuxBuffer,
541            mCblk->mFlags,
542            mAudioTrackServerProxy->getUnderrunFrames(),
543            nowInUnderrun);
544}
545
546uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
547    return mAudioTrackServerProxy->getSampleRate();
548}
549
550// AudioBufferProvider interface
551status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
552        AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
553{
554    ServerProxy::Buffer buf;
555    size_t desiredFrames = buffer->frameCount;
556    buf.mFrameCount = desiredFrames;
557    status_t status = mServerProxy->obtainBuffer(&buf);
558    buffer->frameCount = buf.mFrameCount;
559    buffer->raw = buf.mRaw;
560    if (buf.mFrameCount == 0) {
561        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
562    }
563    return status;
564}
565
566// releaseBuffer() is not overridden
567
568// ExtendedAudioBufferProvider interface
569
570// Note that framesReady() takes a mutex on the control block using tryLock().
571// This could result in priority inversion if framesReady() is called by the normal mixer,
572// as the normal mixer thread runs at lower
573// priority than the client's callback thread:  there is a short window within framesReady()
574// during which the normal mixer could be preempted, and the client callback would block.
575// Another problem can occur if framesReady() is called by the fast mixer:
576// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
577// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
578size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
579    return mAudioTrackServerProxy->framesReady();
580}
581
582size_t AudioFlinger::PlaybackThread::Track::framesReleased() const
583{
584    return mAudioTrackServerProxy->framesReleased();
585}
586
587// Don't call for fast tracks; the framesReady() could result in priority inversion
588bool AudioFlinger::PlaybackThread::Track::isReady() const {
589    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
590        return true;
591    }
592
593    if (isStopping()) {
594        if (framesReady() > 0) {
595            mFillingUpStatus = FS_FILLED;
596        }
597        return true;
598    }
599
600    if (framesReady() >= mFrameCount ||
601            (mCblk->mFlags & CBLK_FORCEREADY)) {
602        mFillingUpStatus = FS_FILLED;
603        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
604        return true;
605    }
606    return false;
607}
608
609status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
610                                                    int triggerSession __unused)
611{
612    status_t status = NO_ERROR;
613    ALOGV("start(%d), calling pid %d session %d",
614            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
615
616    sp<ThreadBase> thread = mThread.promote();
617    if (thread != 0) {
618        if (isOffloaded()) {
619            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
620            Mutex::Autolock _lth(thread->mLock);
621            sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
622            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
623                    (ec != 0 && ec->isNonOffloadableEnabled())) {
624                invalidate();
625                return PERMISSION_DENIED;
626            }
627        }
628        Mutex::Autolock _lth(thread->mLock);
629        track_state state = mState;
630        // here the track could be either new, or restarted
631        // in both cases "unstop" the track
632
633        // initial state-stopping. next state-pausing.
634        // What if resume is called ?
635
636        if (state == PAUSED || state == PAUSING) {
637            if (mResumeToStopping) {
638                // happened we need to resume to STOPPING_1
639                mState = TrackBase::STOPPING_1;
640                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
641            } else {
642                mState = TrackBase::RESUMING;
643                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
644            }
645        } else {
646            mState = TrackBase::ACTIVE;
647            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
648        }
649
650        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
651        status = playbackThread->addTrack_l(this);
652        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
653            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
654            //  restore previous state if start was rejected by policy manager
655            if (status == PERMISSION_DENIED) {
656                mState = state;
657            }
658        }
659        // track was already in the active list, not a problem
660        if (status == ALREADY_EXISTS) {
661            status = NO_ERROR;
662        } else {
663            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
664            // It is usually unsafe to access the server proxy from a binder thread.
665            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
666            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
667            // and for fast tracks the track is not yet in the fast mixer thread's active set.
668            ServerProxy::Buffer buffer;
669            buffer.mFrameCount = 1;
670            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
671        }
672    } else {
673        status = BAD_VALUE;
674    }
675    return status;
676}
677
678void AudioFlinger::PlaybackThread::Track::stop()
679{
680    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
681    sp<ThreadBase> thread = mThread.promote();
682    if (thread != 0) {
683        Mutex::Autolock _l(thread->mLock);
684        track_state state = mState;
685        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
686            // If the track is not active (PAUSED and buffers full), flush buffers
687            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
688            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
689                reset();
690                mState = STOPPED;
691            } else if (!isFastTrack() && !isOffloaded()) {
692                mState = STOPPED;
693            } else {
694                // For fast tracks prepareTracks_l() will set state to STOPPING_2
695                // presentation is complete
696                // For an offloaded track this starts a drain and state will
697                // move to STOPPING_2 when drain completes and then STOPPED
698                mState = STOPPING_1;
699            }
700            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
701                    playbackThread);
702        }
703    }
704}
705
706void AudioFlinger::PlaybackThread::Track::pause()
707{
708    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
709    sp<ThreadBase> thread = mThread.promote();
710    if (thread != 0) {
711        Mutex::Autolock _l(thread->mLock);
712        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
713        switch (mState) {
714        case STOPPING_1:
715        case STOPPING_2:
716            if (!isOffloaded()) {
717                /* nothing to do if track is not offloaded */
718                break;
719            }
720
721            // Offloaded track was draining, we need to carry on draining when resumed
722            mResumeToStopping = true;
723            // fall through...
724        case ACTIVE:
725        case RESUMING:
726            mState = PAUSING;
727            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
728            playbackThread->broadcast_l();
729            break;
730
731        default:
732            break;
733        }
734    }
735}
736
737void AudioFlinger::PlaybackThread::Track::flush()
738{
739    ALOGV("flush(%d)", mName);
740    sp<ThreadBase> thread = mThread.promote();
741    if (thread != 0) {
742        Mutex::Autolock _l(thread->mLock);
743        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
744
745        if (isOffloaded()) {
746            // If offloaded we allow flush during any state except terminated
747            // and keep the track active to avoid problems if user is seeking
748            // rapidly and underlying hardware has a significant delay handling
749            // a pause
750            if (isTerminated()) {
751                return;
752            }
753
754            ALOGV("flush: offload flush");
755            reset();
756
757            if (mState == STOPPING_1 || mState == STOPPING_2) {
758                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
759                mState = ACTIVE;
760            }
761
762            if (mState == ACTIVE) {
763                ALOGV("flush called in active state, resetting buffer time out retry count");
764                mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
765            }
766
767            mFlushHwPending = true;
768            mResumeToStopping = false;
769        } else {
770            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
771                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
772                return;
773            }
774            // No point remaining in PAUSED state after a flush => go to
775            // FLUSHED state
776            mState = FLUSHED;
777            // do not reset the track if it is still in the process of being stopped or paused.
778            // this will be done by prepareTracks_l() when the track is stopped.
779            // prepareTracks_l() will see mState == FLUSHED, then
780            // remove from active track list, reset(), and trigger presentation complete
781            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
782                reset();
783            }
784        }
785        // Prevent flush being lost if the track is flushed and then resumed
786        // before mixer thread can run. This is important when offloading
787        // because the hardware buffer could hold a large amount of audio
788        playbackThread->broadcast_l();
789    }
790}
791
792// must be called with thread lock held
793void AudioFlinger::PlaybackThread::Track::flushAck()
794{
795    if (!isOffloaded())
796        return;
797
798    mFlushHwPending = false;
799}
800
801void AudioFlinger::PlaybackThread::Track::reset()
802{
803    // Do not reset twice to avoid discarding data written just after a flush and before
804    // the audioflinger thread detects the track is stopped.
805    if (!mResetDone) {
806        // Force underrun condition to avoid false underrun callback until first data is
807        // written to buffer
808        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
809        mFillingUpStatus = FS_FILLING;
810        mResetDone = true;
811        if (mState == FLUSHED) {
812            mState = IDLE;
813        }
814    }
815}
816
817status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
818{
819    sp<ThreadBase> thread = mThread.promote();
820    if (thread == 0) {
821        ALOGE("thread is dead");
822        return FAILED_TRANSACTION;
823    } else if ((thread->type() == ThreadBase::DIRECT) ||
824                    (thread->type() == ThreadBase::OFFLOAD)) {
825        return thread->setParameters(keyValuePairs);
826    } else {
827        return PERMISSION_DENIED;
828    }
829}
830
831status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
832{
833    // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
834    if (isFastTrack()) {
835        return INVALID_OPERATION;
836    }
837    sp<ThreadBase> thread = mThread.promote();
838    if (thread == 0) {
839        return INVALID_OPERATION;
840    }
841    Mutex::Autolock _l(thread->mLock);
842    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
843    if (!isOffloaded()) {
844        if (!playbackThread->mLatchQValid) {
845            return INVALID_OPERATION;
846        }
847        uint32_t unpresentedFrames =
848                ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
849                playbackThread->mSampleRate;
850        uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
851        if (framesWritten < unpresentedFrames) {
852            return INVALID_OPERATION;
853        }
854        timestamp.mPosition = framesWritten - unpresentedFrames;
855        timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
856        return NO_ERROR;
857    }
858
859    return playbackThread->getTimestamp_l(timestamp);
860}
861
862status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
863{
864    status_t status = DEAD_OBJECT;
865    sp<ThreadBase> thread = mThread.promote();
866    if (thread != 0) {
867        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
868        sp<AudioFlinger> af = mClient->audioFlinger();
869
870        Mutex::Autolock _l(af->mLock);
871
872        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
873
874        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
875            Mutex::Autolock _dl(playbackThread->mLock);
876            Mutex::Autolock _sl(srcThread->mLock);
877            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
878            if (chain == 0) {
879                return INVALID_OPERATION;
880            }
881
882            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
883            if (effect == 0) {
884                return INVALID_OPERATION;
885            }
886            srcThread->removeEffect_l(effect);
887            status = playbackThread->addEffect_l(effect);
888            if (status != NO_ERROR) {
889                srcThread->addEffect_l(effect);
890                return INVALID_OPERATION;
891            }
892            // removeEffect_l() has stopped the effect if it was active so it must be restarted
893            if (effect->state() == EffectModule::ACTIVE ||
894                    effect->state() == EffectModule::STOPPING) {
895                effect->start();
896            }
897
898            sp<EffectChain> dstChain = effect->chain().promote();
899            if (dstChain == 0) {
900                srcThread->addEffect_l(effect);
901                return INVALID_OPERATION;
902            }
903            AudioSystem::unregisterEffect(effect->id());
904            AudioSystem::registerEffect(&effect->desc(),
905                                        srcThread->id(),
906                                        dstChain->strategy(),
907                                        AUDIO_SESSION_OUTPUT_MIX,
908                                        effect->id());
909            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
910        }
911        status = playbackThread->attachAuxEffect(this, EffectId);
912    }
913    return status;
914}
915
916void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
917{
918    mAuxEffectId = EffectId;
919    mAuxBuffer = buffer;
920}
921
922bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
923                                                         size_t audioHalFrames)
924{
925    // a track is considered presented when the total number of frames written to audio HAL
926    // corresponds to the number of frames written when presentationComplete() is called for the
927    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
928    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
929    // to detect when all frames have been played. In this case framesWritten isn't
930    // useful because it doesn't always reflect whether there is data in the h/w
931    // buffers, particularly if a track has been paused and resumed during draining
932    ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d",
933                      mPresentationCompleteFrames, framesWritten);
934    if (mPresentationCompleteFrames == 0) {
935        mPresentationCompleteFrames = framesWritten + audioHalFrames;
936        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
937                  mPresentationCompleteFrames, audioHalFrames);
938    }
939
940    if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
941        ALOGV("presentationComplete() session %d complete: framesWritten %d",
942                  mSessionId, framesWritten);
943        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
944        mAudioTrackServerProxy->setStreamEndDone();
945        return true;
946    }
947    return false;
948}
949
950void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
951{
952    for (size_t i = 0; i < mSyncEvents.size(); i++) {
953        if (mSyncEvents[i]->type() == type) {
954            mSyncEvents[i]->trigger();
955            mSyncEvents.removeAt(i);
956            i--;
957        }
958    }
959}
960
961// implement VolumeBufferProvider interface
962
963gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
964{
965    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
966    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
967    gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
968    float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
969    float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
970    // track volumes come from shared memory, so can't be trusted and must be clamped
971    if (vl > GAIN_FLOAT_UNITY) {
972        vl = GAIN_FLOAT_UNITY;
973    }
974    if (vr > GAIN_FLOAT_UNITY) {
975        vr = GAIN_FLOAT_UNITY;
976    }
977    // now apply the cached master volume and stream type volume;
978    // this is trusted but lacks any synchronization or barrier so may be stale
979    float v = mCachedVolume;
980    vl *= v;
981    vr *= v;
982    // re-combine into packed minifloat
983    vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
984    // FIXME look at mute, pause, and stop flags
985    return vlr;
986}
987
988status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
989{
990    if (isTerminated() || mState == PAUSED ||
991            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
992                                      (mState == STOPPED)))) {
993        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
994              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
995        event->cancel();
996        return INVALID_OPERATION;
997    }
998    (void) TrackBase::setSyncEvent(event);
999    return NO_ERROR;
1000}
1001
1002void AudioFlinger::PlaybackThread::Track::invalidate()
1003{
1004    // FIXME should use proxy, and needs work
1005    audio_track_cblk_t* cblk = mCblk;
1006    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1007    android_atomic_release_store(0x40000000, &cblk->mFutex);
1008    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1009    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1010    mIsInvalid = true;
1011}
1012
1013void AudioFlinger::PlaybackThread::Track::signal()
1014{
1015    sp<ThreadBase> thread = mThread.promote();
1016    if (thread != 0) {
1017        PlaybackThread *t = (PlaybackThread *)thread.get();
1018        Mutex::Autolock _l(t->mLock);
1019        t->broadcast_l();
1020    }
1021}
1022
1023//To be called with thread lock held
1024bool AudioFlinger::PlaybackThread::Track::isResumePending() {
1025
1026    if (mState == RESUMING)
1027        return true;
1028    /* Resume is pending if track was stopping before pause was called */
1029    if (mState == STOPPING_1 &&
1030        mResumeToStopping)
1031        return true;
1032
1033    return false;
1034}
1035
1036//To be called with thread lock held
1037void AudioFlinger::PlaybackThread::Track::resumeAck() {
1038
1039
1040    if (mState == RESUMING)
1041        mState = ACTIVE;
1042
1043    // Other possibility of  pending resume is stopping_1 state
1044    // Do not update the state from stopping as this prevents
1045    // drain being called.
1046    if (mState == STOPPING_1) {
1047        mResumeToStopping = false;
1048    }
1049}
1050// ----------------------------------------------------------------------------
1051
1052sp<AudioFlinger::PlaybackThread::TimedTrack>
1053AudioFlinger::PlaybackThread::TimedTrack::create(
1054            PlaybackThread *thread,
1055            const sp<Client>& client,
1056            audio_stream_type_t streamType,
1057            uint32_t sampleRate,
1058            audio_format_t format,
1059            audio_channel_mask_t channelMask,
1060            size_t frameCount,
1061            const sp<IMemory>& sharedBuffer,
1062            int sessionId,
1063            int uid)
1064{
1065    if (!client->reserveTimedTrack())
1066        return 0;
1067
1068    return new TimedTrack(
1069        thread, client, streamType, sampleRate, format, channelMask, frameCount,
1070        sharedBuffer, sessionId, uid);
1071}
1072
1073AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
1074            PlaybackThread *thread,
1075            const sp<Client>& client,
1076            audio_stream_type_t streamType,
1077            uint32_t sampleRate,
1078            audio_format_t format,
1079            audio_channel_mask_t channelMask,
1080            size_t frameCount,
1081            const sp<IMemory>& sharedBuffer,
1082            int sessionId,
1083            int uid)
1084    : Track(thread, client, streamType, sampleRate, format, channelMask,
1085            frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
1086      mQueueHeadInFlight(false),
1087      mTrimQueueHeadOnRelease(false),
1088      mFramesPendingInQueue(0),
1089      mTimedSilenceBuffer(NULL),
1090      mTimedSilenceBufferSize(0),
1091      mTimedAudioOutputOnTime(false),
1092      mMediaTimeTransformValid(false)
1093{
1094    LocalClock lc;
1095    mLocalTimeFreq = lc.getLocalFreq();
1096
1097    mLocalTimeToSampleTransform.a_zero = 0;
1098    mLocalTimeToSampleTransform.b_zero = 0;
1099    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
1100    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
1101    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
1102                            &mLocalTimeToSampleTransform.a_to_b_denom);
1103
1104    mMediaTimeToSampleTransform.a_zero = 0;
1105    mMediaTimeToSampleTransform.b_zero = 0;
1106    mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
1107    mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
1108    LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
1109                            &mMediaTimeToSampleTransform.a_to_b_denom);
1110}
1111
1112AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
1113    mClient->releaseTimedTrack();
1114    delete [] mTimedSilenceBuffer;
1115}
1116
1117status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
1118    size_t size, sp<IMemory>* buffer) {
1119
1120    Mutex::Autolock _l(mTimedBufferQueueLock);
1121
1122    trimTimedBufferQueue_l();
1123
1124    // lazily initialize the shared memory heap for timed buffers
1125    if (mTimedMemoryDealer == NULL) {
1126        const int kTimedBufferHeapSize = 512 << 10;
1127
1128        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
1129                                              "AudioFlingerTimed");
1130        if (mTimedMemoryDealer == NULL) {
1131            return NO_MEMORY;
1132        }
1133    }
1134
1135    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
1136    if (newBuffer == 0 || newBuffer->pointer() == NULL) {
1137        return NO_MEMORY;
1138    }
1139
1140    *buffer = newBuffer;
1141    return NO_ERROR;
1142}
1143
1144// caller must hold mTimedBufferQueueLock
1145void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
1146    int64_t mediaTimeNow;
1147    {
1148        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1149        if (!mMediaTimeTransformValid)
1150            return;
1151
1152        int64_t targetTimeNow;
1153        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
1154            ? mCCHelper.getCommonTime(&targetTimeNow)
1155            : mCCHelper.getLocalTime(&targetTimeNow);
1156
1157        if (OK != res)
1158            return;
1159
1160        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
1161                                                    &mediaTimeNow)) {
1162            return;
1163        }
1164    }
1165
1166    size_t trimEnd;
1167    for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
1168        int64_t bufEnd;
1169
1170        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
1171            // We have a next buffer.  Just use its PTS as the PTS of the frame
1172            // following the last frame in this buffer.  If the stream is sparse
1173            // (ie, there are deliberate gaps left in the stream which should be
1174            // filled with silence by the TimedAudioTrack), then this can result
1175            // in one extra buffer being left un-trimmed when it could have
1176            // been.  In general, this is not typical, and we would rather
1177            // optimized away the TS calculation below for the more common case
1178            // where PTSes are contiguous.
1179            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
1180        } else {
1181            // We have no next buffer.  Compute the PTS of the frame following
1182            // the last frame in this buffer by computing the duration of of
1183            // this frame in media time units and adding it to the PTS of the
1184            // buffer.
1185            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
1186                               / mFrameSize;
1187
1188            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
1189                                                                &bufEnd)) {
1190                ALOGE("Failed to convert frame count of %lld to media time"
1191                      " duration" " (scale factor %d/%u) in %s",
1192                      frameCount,
1193                      mMediaTimeToSampleTransform.a_to_b_numer,
1194                      mMediaTimeToSampleTransform.a_to_b_denom,
1195                      __PRETTY_FUNCTION__);
1196                break;
1197            }
1198            bufEnd += mTimedBufferQueue[trimEnd].pts();
1199        }
1200
1201        if (bufEnd > mediaTimeNow)
1202            break;
1203
1204        // Is the buffer we want to use in the middle of a mix operation right
1205        // now?  If so, don't actually trim it.  Just wait for the releaseBuffer
1206        // from the mixer which should be coming back shortly.
1207        if (!trimEnd && mQueueHeadInFlight) {
1208            mTrimQueueHeadOnRelease = true;
1209        }
1210    }
1211
1212    size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1213    if (trimStart < trimEnd) {
1214        // Update the bookkeeping for framesReady()
1215        for (size_t i = trimStart; i < trimEnd; ++i) {
1216            updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1217        }
1218
1219        // Now actually remove the buffers from the queue.
1220        mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1221    }
1222}
1223
1224void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1225        const char* logTag) {
1226    ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1227                "%s called (reason \"%s\"), but timed buffer queue has no"
1228                " elements to trim.", __FUNCTION__, logTag);
1229
1230    updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1231    mTimedBufferQueue.removeAt(0);
1232}
1233
1234void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1235        const TimedBuffer& buf,
1236        const char* logTag __unused) {
1237    uint32_t bufBytes        = buf.buffer()->size();
1238    uint32_t consumedAlready = buf.position();
1239
1240    ALOG_ASSERT(consumedAlready <= bufBytes,
1241                "Bad bookkeeping while updating frames pending.  Timed buffer is"
1242                " only %u bytes long, but claims to have consumed %u"
1243                " bytes.  (update reason: \"%s\")",
1244                bufBytes, consumedAlready, logTag);
1245
1246    uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1247    ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1248                "Bad bookkeeping while updating frames pending.  Should have at"
1249                " least %u queued frames, but we think we have only %u.  (update"
1250                " reason: \"%s\")",
1251                bufFrames, mFramesPendingInQueue, logTag);
1252
1253    mFramesPendingInQueue -= bufFrames;
1254}
1255
1256status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1257    const sp<IMemory>& buffer, int64_t pts) {
1258
1259    {
1260        Mutex::Autolock mttLock(mMediaTimeTransformLock);
1261        if (!mMediaTimeTransformValid)
1262            return INVALID_OPERATION;
1263    }
1264
1265    Mutex::Autolock _l(mTimedBufferQueueLock);
1266
1267    uint32_t bufFrames = buffer->size() / mFrameSize;
1268    mFramesPendingInQueue += bufFrames;
1269    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1270
1271    return NO_ERROR;
1272}
1273
1274status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1275    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1276
1277    ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1278           xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1279           target);
1280
1281    if (!(target == TimedAudioTrack::LOCAL_TIME ||
1282          target == TimedAudioTrack::COMMON_TIME)) {
1283        return BAD_VALUE;
1284    }
1285
1286    Mutex::Autolock lock(mMediaTimeTransformLock);
1287    mMediaTimeTransform = xform;
1288    mMediaTimeTransformTarget = target;
1289    mMediaTimeTransformValid = true;
1290
1291    return NO_ERROR;
1292}
1293
1294#define min(a, b) ((a) < (b) ? (a) : (b))
1295
1296// implementation of getNextBuffer for tracks whose buffers have timestamps
1297status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1298    AudioBufferProvider::Buffer* buffer, int64_t pts)
1299{
1300    if (pts == AudioBufferProvider::kInvalidPTS) {
1301        buffer->raw = NULL;
1302        buffer->frameCount = 0;
1303        mTimedAudioOutputOnTime = false;
1304        return INVALID_OPERATION;
1305    }
1306
1307    Mutex::Autolock _l(mTimedBufferQueueLock);
1308
1309    ALOG_ASSERT(!mQueueHeadInFlight,
1310                "getNextBuffer called without releaseBuffer!");
1311
1312    while (true) {
1313
1314        // if we have no timed buffers, then fail
1315        if (mTimedBufferQueue.isEmpty()) {
1316            buffer->raw = NULL;
1317            buffer->frameCount = 0;
1318            return NOT_ENOUGH_DATA;
1319        }
1320
1321        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1322
1323        // calculate the PTS of the head of the timed buffer queue expressed in
1324        // local time
1325        int64_t headLocalPTS;
1326        {
1327            Mutex::Autolock mttLock(mMediaTimeTransformLock);
1328
1329            ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1330
1331            if (mMediaTimeTransform.a_to_b_denom == 0) {
1332                // the transform represents a pause, so yield silence
1333                timedYieldSilence_l(buffer->frameCount, buffer);
1334                return NO_ERROR;
1335            }
1336
1337            int64_t transformedPTS;
1338            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1339                                                        &transformedPTS)) {
1340                // the transform failed.  this shouldn't happen, but if it does
1341                // then just drop this buffer
1342                ALOGW("timedGetNextBuffer transform failed");
1343                buffer->raw = NULL;
1344                buffer->frameCount = 0;
1345                trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1346                return NO_ERROR;
1347            }
1348
1349            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1350                if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1351                                                          &headLocalPTS)) {
1352                    buffer->raw = NULL;
1353                    buffer->frameCount = 0;
1354                    return INVALID_OPERATION;
1355                }
1356            } else {
1357                headLocalPTS = transformedPTS;
1358            }
1359        }
1360
1361        uint32_t sr = sampleRate();
1362
1363        // adjust the head buffer's PTS to reflect the portion of the head buffer
1364        // that has already been consumed
1365        int64_t effectivePTS = headLocalPTS +
1366                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
1367
1368        // Calculate the delta in samples between the head of the input buffer
1369        // queue and the start of the next output buffer that will be written.
1370        // If the transformation fails because of over or underflow, it means
1371        // that the sample's position in the output stream is so far out of
1372        // whack that it should just be dropped.
1373        int64_t sampleDelta;
1374        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1375            ALOGV("*** head buffer is too far from PTS: dropped buffer");
1376            trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1377                                       " mix");
1378            continue;
1379        }
1380        if (!mLocalTimeToSampleTransform.doForwardTransform(
1381                (effectivePTS - pts) << 32, &sampleDelta)) {
1382            ALOGV("*** too late during sample rate transform: dropped buffer");
1383            trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1384            continue;
1385        }
1386
1387        ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1388               " sampleDelta=[%d.%08x]",
1389               head.pts(), head.position(), pts,
1390               static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1391                   + (sampleDelta >> 32)),
1392               static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1393
1394        // if the delta between the ideal placement for the next input sample and
1395        // the current output position is within this threshold, then we will
1396        // concatenate the next input samples to the previous output
1397        const int64_t kSampleContinuityThreshold =
1398                (static_cast<int64_t>(sr) << 32) / 250;
1399
1400        // if this is the first buffer of audio that we're emitting from this track
1401        // then it should be almost exactly on time.
1402        const int64_t kSampleStartupThreshold = 1LL << 32;
1403
1404        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1405           (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1406            // the next input is close enough to being on time, so concatenate it
1407            // with the last output
1408            timedYieldSamples_l(buffer);
1409
1410            ALOGVV("*** on time: head.pos=%d frameCount=%u",
1411                    head.position(), buffer->frameCount);
1412            return NO_ERROR;
1413        }
1414
1415        // Looks like our output is not on time.  Reset our on timed status.
1416        // Next time we mix samples from our input queue, then should be within
1417        // the StartupThreshold.
1418        mTimedAudioOutputOnTime = false;
1419        if (sampleDelta > 0) {
1420            // the gap between the current output position and the proper start of
1421            // the next input sample is too big, so fill it with silence
1422            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1423
1424            timedYieldSilence_l(framesUntilNextInput, buffer);
1425            ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1426            return NO_ERROR;
1427        } else {
1428            // the next input sample is late
1429            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1430            size_t onTimeSamplePosition =
1431                    head.position() + lateFrames * mFrameSize;
1432
1433            if (onTimeSamplePosition > head.buffer()->size()) {
1434                // all the remaining samples in the head are too late, so
1435                // drop it and move on
1436                ALOGV("*** too late: dropped buffer");
1437                trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1438                continue;
1439            } else {
1440                // skip over the late samples
1441                head.setPosition(onTimeSamplePosition);
1442
1443                // yield the available samples
1444                timedYieldSamples_l(buffer);
1445
1446                ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1447                return NO_ERROR;
1448            }
1449        }
1450    }
1451}
1452
1453// Yield samples from the timed buffer queue head up to the given output
1454// buffer's capacity.
1455//
1456// Caller must hold mTimedBufferQueueLock
1457void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1458    AudioBufferProvider::Buffer* buffer) {
1459
1460    const TimedBuffer& head = mTimedBufferQueue[0];
1461
1462    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1463                   head.position());
1464
1465    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1466                                 mFrameSize);
1467    size_t framesRequested = buffer->frameCount;
1468    buffer->frameCount = min(framesLeftInHead, framesRequested);
1469
1470    mQueueHeadInFlight = true;
1471    mTimedAudioOutputOnTime = true;
1472}
1473
1474// Yield samples of silence up to the given output buffer's capacity
1475//
1476// Caller must hold mTimedBufferQueueLock
1477void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1478    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1479
1480    // lazily allocate a buffer filled with silence
1481    if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1482        delete [] mTimedSilenceBuffer;
1483        mTimedSilenceBufferSize = numFrames * mFrameSize;
1484        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1485        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1486    }
1487
1488    buffer->raw = mTimedSilenceBuffer;
1489    size_t framesRequested = buffer->frameCount;
1490    buffer->frameCount = min(numFrames, framesRequested);
1491
1492    mTimedAudioOutputOnTime = false;
1493}
1494
1495// AudioBufferProvider interface
1496void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1497    AudioBufferProvider::Buffer* buffer) {
1498
1499    Mutex::Autolock _l(mTimedBufferQueueLock);
1500
1501    // If the buffer which was just released is part of the buffer at the head
1502    // of the queue, be sure to update the amt of the buffer which has been
1503    // consumed.  If the buffer being returned is not part of the head of the
1504    // queue, its either because the buffer is part of the silence buffer, or
1505    // because the head of the timed queue was trimmed after the mixer called
1506    // getNextBuffer but before the mixer called releaseBuffer.
1507    if (buffer->raw == mTimedSilenceBuffer) {
1508        ALOG_ASSERT(!mQueueHeadInFlight,
1509                    "Queue head in flight during release of silence buffer!");
1510        goto done;
1511    }
1512
1513    ALOG_ASSERT(mQueueHeadInFlight,
1514                "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1515                " head in flight.");
1516
1517    if (mTimedBufferQueue.size()) {
1518        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1519
1520        void* start = head.buffer()->pointer();
1521        void* end   = reinterpret_cast<void*>(
1522                        reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1523                        + head.buffer()->size());
1524
1525        ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1526                    "released buffer not within the head of the timed buffer"
1527                    " queue; qHead = [%p, %p], released buffer = %p",
1528                    start, end, buffer->raw);
1529
1530        head.setPosition(head.position() +
1531                (buffer->frameCount * mFrameSize));
1532        mQueueHeadInFlight = false;
1533
1534        ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1535                    "Bad bookkeeping during releaseBuffer!  Should have at"
1536                    " least %u queued frames, but we think we have only %u",
1537                    buffer->frameCount, mFramesPendingInQueue);
1538
1539        mFramesPendingInQueue -= buffer->frameCount;
1540
1541        if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1542            || mTrimQueueHeadOnRelease) {
1543            trimTimedBufferQueueHead_l("releaseBuffer");
1544            mTrimQueueHeadOnRelease = false;
1545        }
1546    } else {
1547        LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1548                  " buffers in the timed buffer queue");
1549    }
1550
1551done:
1552    buffer->raw = 0;
1553    buffer->frameCount = 0;
1554}
1555
1556size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1557    Mutex::Autolock _l(mTimedBufferQueueLock);
1558    return mFramesPendingInQueue;
1559}
1560
1561AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1562        : mPTS(0), mPosition(0) {}
1563
1564AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1565    const sp<IMemory>& buffer, int64_t pts)
1566        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1567
1568
1569// ----------------------------------------------------------------------------
1570
1571AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1572            PlaybackThread *playbackThread,
1573            DuplicatingThread *sourceThread,
1574            uint32_t sampleRate,
1575            audio_format_t format,
1576            audio_channel_mask_t channelMask,
1577            size_t frameCount,
1578            int uid)
1579    :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1580                NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
1581    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1582{
1583
1584    if (mCblk != NULL) {
1585        mOutBuffer.frameCount = 0;
1586        playbackThread->mTracks.add(this);
1587        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1588                "frameCount %u, mChannelMask 0x%08x",
1589                mCblk, mBuffer,
1590                frameCount, mChannelMask);
1591        // since client and server are in the same process,
1592        // the buffer has the same virtual address on both sides
1593        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
1594        mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
1595        mClientProxy->setSendLevel(0.0);
1596        mClientProxy->setSampleRate(sampleRate);
1597        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1598                true /*clientInServer*/);
1599    } else {
1600        ALOGW("Error creating output track on thread %p", playbackThread);
1601    }
1602}
1603
1604AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1605{
1606    clearBufferQueue();
1607    delete mClientProxy;
1608    // superclass destructor will now delete the server proxy and shared memory both refer to
1609}
1610
1611status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1612                                                          int triggerSession)
1613{
1614    status_t status = Track::start(event, triggerSession);
1615    if (status != NO_ERROR) {
1616        return status;
1617    }
1618
1619    mActive = true;
1620    mRetryCount = 127;
1621    return status;
1622}
1623
1624void AudioFlinger::PlaybackThread::OutputTrack::stop()
1625{
1626    Track::stop();
1627    clearBufferQueue();
1628    mOutBuffer.frameCount = 0;
1629    mActive = false;
1630}
1631
1632bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1633{
1634    Buffer *pInBuffer;
1635    Buffer inBuffer;
1636    uint32_t channelCount = mChannelCount;
1637    bool outputBufferFull = false;
1638    inBuffer.frameCount = frames;
1639    inBuffer.i16 = data;
1640
1641    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1642
1643    if (!mActive && frames != 0) {
1644        start();
1645        sp<ThreadBase> thread = mThread.promote();
1646        if (thread != 0) {
1647            MixerThread *mixerThread = (MixerThread *)thread.get();
1648            if (mFrameCount > frames) {
1649                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1650                    uint32_t startFrames = (mFrameCount - frames);
1651                    pInBuffer = new Buffer;
1652                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1653                    pInBuffer->frameCount = startFrames;
1654                    pInBuffer->i16 = pInBuffer->mBuffer;
1655                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1656                    mBufferQueue.add(pInBuffer);
1657                } else {
1658                    ALOGW("OutputTrack::write() %p no more buffers in queue", this);
1659                }
1660            }
1661        }
1662    }
1663
1664    while (waitTimeLeftMs) {
1665        // First write pending buffers, then new data
1666        if (mBufferQueue.size()) {
1667            pInBuffer = mBufferQueue.itemAt(0);
1668        } else {
1669            pInBuffer = &inBuffer;
1670        }
1671
1672        if (pInBuffer->frameCount == 0) {
1673            break;
1674        }
1675
1676        if (mOutBuffer.frameCount == 0) {
1677            mOutBuffer.frameCount = pInBuffer->frameCount;
1678            nsecs_t startTime = systemTime();
1679            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1680            if (status != NO_ERROR) {
1681                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1682                        mThread.unsafe_get(), status);
1683                outputBufferFull = true;
1684                break;
1685            }
1686            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1687            if (waitTimeLeftMs >= waitTimeMs) {
1688                waitTimeLeftMs -= waitTimeMs;
1689            } else {
1690                waitTimeLeftMs = 0;
1691            }
1692        }
1693
1694        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1695                pInBuffer->frameCount;
1696        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
1697        Proxy::Buffer buf;
1698        buf.mFrameCount = outFrames;
1699        buf.mRaw = NULL;
1700        mClientProxy->releaseBuffer(&buf);
1701        pInBuffer->frameCount -= outFrames;
1702        pInBuffer->i16 += outFrames * channelCount;
1703        mOutBuffer.frameCount -= outFrames;
1704        mOutBuffer.i16 += outFrames * channelCount;
1705
1706        if (pInBuffer->frameCount == 0) {
1707            if (mBufferQueue.size()) {
1708                mBufferQueue.removeAt(0);
1709                delete [] pInBuffer->mBuffer;
1710                delete pInBuffer;
1711                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1712                        mThread.unsafe_get(), mBufferQueue.size());
1713            } else {
1714                break;
1715            }
1716        }
1717    }
1718
1719    // If we could not write all frames, allocate a buffer and queue it for next time.
1720    if (inBuffer.frameCount) {
1721        sp<ThreadBase> thread = mThread.promote();
1722        if (thread != 0 && !thread->standby()) {
1723            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1724                pInBuffer = new Buffer;
1725                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1726                pInBuffer->frameCount = inBuffer.frameCount;
1727                pInBuffer->i16 = pInBuffer->mBuffer;
1728                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1729                        sizeof(int16_t));
1730                mBufferQueue.add(pInBuffer);
1731                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1732                        mThread.unsafe_get(), mBufferQueue.size());
1733            } else {
1734                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1735                        mThread.unsafe_get(), this);
1736            }
1737        }
1738    }
1739
1740    // Calling write() with a 0 length buffer, means that no more data will be written:
1741    // If no more buffers are pending, fill output track buffer to make sure it is started
1742    // by output mixer.
1743    if (frames == 0 && mBufferQueue.size() == 0) {
1744        // FIXME borken, replace by getting framesReady() from proxy
1745        size_t user = 0;    // was mCblk->user
1746        if (user < mFrameCount) {
1747            frames = mFrameCount - user;
1748            pInBuffer = new Buffer;
1749            pInBuffer->mBuffer = new int16_t[frames * channelCount];
1750            pInBuffer->frameCount = frames;
1751            pInBuffer->i16 = pInBuffer->mBuffer;
1752            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1753            mBufferQueue.add(pInBuffer);
1754        } else if (mActive) {
1755            stop();
1756        }
1757    }
1758
1759    return outputBufferFull;
1760}
1761
1762status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1763        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1764{
1765    ClientProxy::Buffer buf;
1766    buf.mFrameCount = buffer->frameCount;
1767    struct timespec timeout;
1768    timeout.tv_sec = waitTimeMs / 1000;
1769    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1770    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1771    buffer->frameCount = buf.mFrameCount;
1772    buffer->raw = buf.mRaw;
1773    return status;
1774}
1775
1776void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1777{
1778    size_t size = mBufferQueue.size();
1779
1780    for (size_t i = 0; i < size; i++) {
1781        Buffer *pBuffer = mBufferQueue.itemAt(i);
1782        delete [] pBuffer->mBuffer;
1783        delete pBuffer;
1784    }
1785    mBufferQueue.clear();
1786}
1787
1788
1789// ----------------------------------------------------------------------------
1790//      Record
1791// ----------------------------------------------------------------------------
1792
1793AudioFlinger::RecordHandle::RecordHandle(
1794        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1795    : BnAudioRecord(),
1796    mRecordTrack(recordTrack)
1797{
1798}
1799
1800AudioFlinger::RecordHandle::~RecordHandle() {
1801    stop_nonvirtual();
1802    mRecordTrack->destroy();
1803}
1804
1805status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1806        int triggerSession) {
1807    ALOGV("RecordHandle::start()");
1808    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1809}
1810
1811void AudioFlinger::RecordHandle::stop() {
1812    stop_nonvirtual();
1813}
1814
1815void AudioFlinger::RecordHandle::stop_nonvirtual() {
1816    ALOGV("RecordHandle::stop()");
1817    mRecordTrack->stop();
1818}
1819
1820status_t AudioFlinger::RecordHandle::onTransact(
1821    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1822{
1823    return BnAudioRecord::onTransact(code, data, reply, flags);
1824}
1825
1826// ----------------------------------------------------------------------------
1827
1828// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
1829AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1830            RecordThread *thread,
1831            const sp<Client>& client,
1832            uint32_t sampleRate,
1833            audio_format_t format,
1834            audio_channel_mask_t channelMask,
1835            size_t frameCount,
1836            int sessionId,
1837            int uid,
1838            IAudioFlinger::track_flags_t flags)
1839    :   TrackBase(thread, client, sampleRate, format,
1840                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid,
1841                  flags, false /*isOut*/,
1842                  (flags & IAudioFlinger::TRACK_FAST) != 0 /*useReadOnlyHeap*/),
1843        mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
1844        // See real initialization of mRsmpInFront at RecordThread::start()
1845        mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
1846{
1847    if (mCblk == NULL) {
1848        return;
1849    }
1850
1851    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
1852
1853    uint32_t channelCount = popcount(channelMask);
1854    // FIXME I don't understand either of the channel count checks
1855    if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
1856            channelCount <= FCC_2) {
1857        // sink SR
1858        mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
1859        // source SR
1860        mResampler->setSampleRate(thread->mSampleRate);
1861        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
1862        mResamplerBufferProvider = new ResamplerBufferProvider(this);
1863    }
1864}
1865
1866AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1867{
1868    ALOGV("%s", __func__);
1869    delete mResampler;
1870    delete[] mRsmpOutBuffer;
1871    delete mResamplerBufferProvider;
1872}
1873
1874// AudioBufferProvider interface
1875status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1876        int64_t pts __unused)
1877{
1878    ServerProxy::Buffer buf;
1879    buf.mFrameCount = buffer->frameCount;
1880    status_t status = mServerProxy->obtainBuffer(&buf);
1881    buffer->frameCount = buf.mFrameCount;
1882    buffer->raw = buf.mRaw;
1883    if (buf.mFrameCount == 0) {
1884        // FIXME also wake futex so that overrun is noticed more quickly
1885        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1886    }
1887    return status;
1888}
1889
1890status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1891                                                        int triggerSession)
1892{
1893    sp<ThreadBase> thread = mThread.promote();
1894    if (thread != 0) {
1895        RecordThread *recordThread = (RecordThread *)thread.get();
1896        return recordThread->start(this, event, triggerSession);
1897    } else {
1898        return BAD_VALUE;
1899    }
1900}
1901
1902void AudioFlinger::RecordThread::RecordTrack::stop()
1903{
1904    sp<ThreadBase> thread = mThread.promote();
1905    if (thread != 0) {
1906        RecordThread *recordThread = (RecordThread *)thread.get();
1907        if (recordThread->stop(this)) {
1908            AudioSystem::stopInput(recordThread->id());
1909        }
1910    }
1911}
1912
1913void AudioFlinger::RecordThread::RecordTrack::destroy()
1914{
1915    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1916    sp<RecordTrack> keep(this);
1917    {
1918        sp<ThreadBase> thread = mThread.promote();
1919        if (thread != 0) {
1920            if (mState == ACTIVE || mState == RESUMING) {
1921                AudioSystem::stopInput(thread->id());
1922            }
1923            AudioSystem::releaseInput(thread->id());
1924            Mutex::Autolock _l(thread->mLock);
1925            RecordThread *recordThread = (RecordThread *) thread.get();
1926            recordThread->destroyTrack_l(this);
1927        }
1928    }
1929}
1930
1931void AudioFlinger::RecordThread::RecordTrack::invalidate()
1932{
1933    // FIXME should use proxy, and needs work
1934    audio_track_cblk_t* cblk = mCblk;
1935    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1936    android_atomic_release_store(0x40000000, &cblk->mFutex);
1937    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1938    (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
1939}
1940
1941
1942/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1943{
1944    result.append("    Active Client Fmt Chn mask Session S   Server fCount Resampling\n");
1945}
1946
1947void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
1948{
1949    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n",
1950            active ? "yes" : "no",
1951            (mClient == 0) ? getpid_cached : mClient->pid(),
1952            mFormat,
1953            mChannelMask,
1954            mSessionId,
1955            mState,
1956            mCblk->mServer,
1957            mFrameCount,
1958            mResampler != NULL);
1959
1960}
1961
1962void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
1963{
1964    if (event == mSyncStartEvent) {
1965        ssize_t framesToDrop = 0;
1966        sp<ThreadBase> threadBase = mThread.promote();
1967        if (threadBase != 0) {
1968            // TODO: use actual buffer filling status instead of 2 buffers when info is available
1969            // from audio HAL
1970            framesToDrop = threadBase->mFrameCount * 2;
1971        }
1972        mFramesToDrop = framesToDrop;
1973    }
1974}
1975
1976void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
1977{
1978    if (mSyncStartEvent != 0) {
1979        mSyncStartEvent->cancel();
1980        mSyncStartEvent.clear();
1981    }
1982    mFramesToDrop = 0;
1983}
1984
1985}; // namespace android
1986