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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 <linux/futex.h>
24#include <math.h>
25#include <sys/syscall.h>
26#include <utils/Log.h>
27
28#include <private/media/AudioTrackShared.h>
29
30#include "AudioMixer.h"
31#include "AudioFlinger.h"
32#include "ServiceUtilities.h"
33
34#include <media/nbaio/Pipe.h>
35#include <media/nbaio/PipeReader.h>
36#include <audio_utils/minifloat.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
53// TODO move to a common header  (Also shared with AudioTrack.cpp)
54#define NANOS_PER_SECOND    1000000000
55#define TIME_TO_NANOS(time) ((uint64_t)time.tv_sec * NANOS_PER_SECOND + time.tv_nsec)
56
57namespace android {
58
59// ----------------------------------------------------------------------------
60//      TrackBase
61// ----------------------------------------------------------------------------
62
63static volatile int32_t nextTrackId = 55;
64
65// TrackBase constructor must be called with AudioFlinger::mLock held
66AudioFlinger::ThreadBase::TrackBase::TrackBase(
67            ThreadBase *thread,
68            const sp<Client>& client,
69            uint32_t sampleRate,
70            audio_format_t format,
71            audio_channel_mask_t channelMask,
72            size_t frameCount,
73            void *buffer,
74            audio_session_t sessionId,
75            int clientUid,
76            IAudioFlinger::track_flags_t flags,
77            bool isOut,
78            alloc_type alloc,
79            track_type type)
80    :   RefBase(),
81        mThread(thread),
82        mClient(client),
83        mCblk(NULL),
84        // mBuffer
85        mState(IDLE),
86        mSampleRate(sampleRate),
87        mFormat(format),
88        mChannelMask(channelMask),
89        mChannelCount(isOut ?
90                audio_channel_count_from_out_mask(channelMask) :
91                audio_channel_count_from_in_mask(channelMask)),
92        mFrameSize(audio_has_proportional_frames(format) ?
93                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
94        mFrameCount(frameCount),
95        mSessionId(sessionId),
96        mFlags(flags),
97        mIsOut(isOut),
98        mServerProxy(NULL),
99        mId(android_atomic_inc(&nextTrackId)),
100        mTerminated(false),
101        mType(type),
102        mThreadIoHandle(thread->id())
103{
104    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
105    if (!isTrustedCallingUid(callingUid) || clientUid == -1) {
106        ALOGW_IF(clientUid != -1 && clientUid != (int)callingUid,
107                "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
108        clientUid = (int)callingUid;
109    }
110    // clientUid contains the uid of the app that is responsible for this track, so we can blame
111    // battery usage on it.
112    mUid = clientUid;
113
114    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
115    size_t size = sizeof(audio_track_cblk_t);
116    size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize;
117    if (buffer == NULL && alloc == ALLOC_CBLK) {
118        size += bufferSize;
119    }
120
121    if (client != 0) {
122        mCblkMemory = client->heap()->allocate(size);
123        if (mCblkMemory == 0 ||
124                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
125            ALOGE("not enough memory for AudioTrack size=%zu", size);
126            client->heap()->dump("AudioTrack");
127            mCblkMemory.clear();
128            return;
129        }
130    } else {
131        // this syntax avoids calling the audio_track_cblk_t constructor twice
132        mCblk = (audio_track_cblk_t *) new uint8_t[size];
133        // assume mCblk != NULL
134    }
135
136    // construct the shared structure in-place.
137    if (mCblk != NULL) {
138        new(mCblk) audio_track_cblk_t();
139        switch (alloc) {
140        case ALLOC_READONLY: {
141            const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
142            if (roHeap == 0 ||
143                    (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
144                    (mBuffer = mBufferMemory->pointer()) == NULL) {
145                ALOGE("not enough memory for read-only buffer size=%zu", bufferSize);
146                if (roHeap != 0) {
147                    roHeap->dump("buffer");
148                }
149                mCblkMemory.clear();
150                mBufferMemory.clear();
151                return;
152            }
153            memset(mBuffer, 0, bufferSize);
154            } break;
155        case ALLOC_PIPE:
156            mBufferMemory = thread->pipeMemory();
157            // mBuffer is the virtual address as seen from current process (mediaserver),
158            // and should normally be coming from mBufferMemory->pointer().
159            // However in this case the TrackBase does not reference the buffer directly.
160            // It should references the buffer via the pipe.
161            // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
162            mBuffer = NULL;
163            break;
164        case ALLOC_CBLK:
165            // clear all buffers
166            if (buffer == NULL) {
167                mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
168                memset(mBuffer, 0, bufferSize);
169            } else {
170                mBuffer = buffer;
171#if 0
172                mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
173#endif
174            }
175            break;
176        case ALLOC_LOCAL:
177            mBuffer = calloc(1, bufferSize);
178            break;
179        case ALLOC_NONE:
180            mBuffer = buffer;
181            break;
182        }
183
184#ifdef TEE_SINK
185        if (mTeeSinkTrackEnabled) {
186            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat);
187            if (Format_isValid(pipeFormat)) {
188                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
189                size_t numCounterOffers = 0;
190                const NBAIO_Format offers[1] = {pipeFormat};
191                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
192                ALOG_ASSERT(index == 0);
193                PipeReader *pipeReader = new PipeReader(*pipe);
194                numCounterOffers = 0;
195                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
196                ALOG_ASSERT(index == 0);
197                mTeeSink = pipe;
198                mTeeSource = pipeReader;
199            }
200        }
201#endif
202
203    }
204}
205
206status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const
207{
208    status_t status;
209    if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) {
210        status = cblk() != NULL ? NO_ERROR : NO_MEMORY;
211    } else {
212        status = getCblk() != 0 ? NO_ERROR : NO_MEMORY;
213    }
214    return status;
215}
216
217AudioFlinger::ThreadBase::TrackBase::~TrackBase()
218{
219#ifdef TEE_SINK
220    dumpTee(-1, mTeeSource, mId);
221#endif
222    // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
223    delete mServerProxy;
224    if (mCblk != NULL) {
225        if (mClient == 0) {
226            delete mCblk;
227        } else {
228            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
229        }
230    }
231    mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
232    if (mClient != 0) {
233        // Client destructor must run with AudioFlinger client mutex locked
234        Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
235        // If the client's reference count drops to zero, the associated destructor
236        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
237        // relying on the automatic clear() at end of scope.
238        mClient.clear();
239    }
240    // flush the binder command buffer
241    IPCThreadState::self()->flushCommands();
242}
243
244// AudioBufferProvider interface
245// getNextBuffer() = 0;
246// This implementation of releaseBuffer() is used by Track and RecordTrack
247void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
248{
249#ifdef TEE_SINK
250    if (mTeeSink != 0) {
251        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
252    }
253#endif
254
255    ServerProxy::Buffer buf;
256    buf.mFrameCount = buffer->frameCount;
257    buf.mRaw = buffer->raw;
258    buffer->frameCount = 0;
259    buffer->raw = NULL;
260    mServerProxy->releaseBuffer(&buf);
261}
262
263status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
264{
265    mSyncEvents.add(event);
266    return NO_ERROR;
267}
268
269// ----------------------------------------------------------------------------
270//      Playback
271// ----------------------------------------------------------------------------
272
273AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
274    : BnAudioTrack(),
275      mTrack(track)
276{
277}
278
279AudioFlinger::TrackHandle::~TrackHandle() {
280    // just stop the track on deletion, associated resources
281    // will be freed from the main thread once all pending buffers have
282    // been played. Unless it's not in the active track list, in which
283    // case we free everything now...
284    mTrack->destroy();
285}
286
287sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
288    return mTrack->getCblk();
289}
290
291status_t AudioFlinger::TrackHandle::start() {
292    return mTrack->start();
293}
294
295void AudioFlinger::TrackHandle::stop() {
296    mTrack->stop();
297}
298
299void AudioFlinger::TrackHandle::flush() {
300    mTrack->flush();
301}
302
303void AudioFlinger::TrackHandle::pause() {
304    mTrack->pause();
305}
306
307status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
308{
309    return mTrack->attachAuxEffect(EffectId);
310}
311
312status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
313    return mTrack->setParameters(keyValuePairs);
314}
315
316status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
317{
318    return mTrack->getTimestamp(timestamp);
319}
320
321
322void AudioFlinger::TrackHandle::signal()
323{
324    return mTrack->signal();
325}
326
327status_t AudioFlinger::TrackHandle::onTransact(
328    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
329{
330    return BnAudioTrack::onTransact(code, data, reply, flags);
331}
332
333// ----------------------------------------------------------------------------
334
335// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
336AudioFlinger::PlaybackThread::Track::Track(
337            PlaybackThread *thread,
338            const sp<Client>& client,
339            audio_stream_type_t streamType,
340            uint32_t sampleRate,
341            audio_format_t format,
342            audio_channel_mask_t channelMask,
343            size_t frameCount,
344            void *buffer,
345            const sp<IMemory>& sharedBuffer,
346            audio_session_t sessionId,
347            int uid,
348            IAudioFlinger::track_flags_t flags,
349            track_type type)
350    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
351                  (sharedBuffer != 0) ? sharedBuffer->pointer() : buffer,
352                  sessionId, uid, flags, true /*isOut*/,
353                  (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
354                  type),
355    mFillingUpStatus(FS_INVALID),
356    // mRetryCount initialized later when needed
357    mSharedBuffer(sharedBuffer),
358    mStreamType(streamType),
359    mName(-1),  // see note below
360    mMainBuffer(thread->mixBuffer()),
361    mAuxBuffer(NULL),
362    mAuxEffectId(0), mHasVolumeController(false),
363    mPresentationCompleteFrames(0),
364    mFrameMap(16 /* sink-frame-to-track-frame map memory */),
365    // mSinkTimestamp
366    mFastIndex(-1),
367    mCachedVolume(1.0),
368    mIsInvalid(false),
369    mAudioTrackServerProxy(NULL),
370    mResumeToStopping(false),
371    mFlushHwPending(false)
372{
373    // client == 0 implies sharedBuffer == 0
374    ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
375
376    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(),
377            sharedBuffer->size());
378
379    if (mCblk == NULL) {
380        return;
381    }
382
383    if (sharedBuffer == 0) {
384        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
385                mFrameSize, !isExternalTrack(), sampleRate);
386    } else {
387        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
388                mFrameSize);
389    }
390    mServerProxy = mAudioTrackServerProxy;
391
392    mName = thread->getTrackName_l(channelMask, format, sessionId);
393    if (mName < 0) {
394        ALOGE("no more track names available");
395        return;
396    }
397    // only allocate a fast track index if we were able to allocate a normal track name
398    if (flags & IAudioFlinger::TRACK_FAST) {
399        // FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential
400        // race with setSyncEvent(). However, if we call it, we cannot properly start
401        // static fast tracks (SoundPool) immediately after stopping.
402        //mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
403        ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
404        int i = __builtin_ctz(thread->mFastTrackAvailMask);
405        ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
406        // FIXME This is too eager.  We allocate a fast track index before the
407        //       fast track becomes active.  Since fast tracks are a scarce resource,
408        //       this means we are potentially denying other more important fast tracks from
409        //       being created.  It would be better to allocate the index dynamically.
410        mFastIndex = i;
411        thread->mFastTrackAvailMask &= ~(1 << i);
412    }
413}
414
415AudioFlinger::PlaybackThread::Track::~Track()
416{
417    ALOGV("PlaybackThread::Track destructor");
418
419    // The destructor would clear mSharedBuffer,
420    // but it will not push the decremented reference count,
421    // leaving the client's IMemory dangling indefinitely.
422    // This prevents that leak.
423    if (mSharedBuffer != 0) {
424        mSharedBuffer.clear();
425    }
426}
427
428status_t AudioFlinger::PlaybackThread::Track::initCheck() const
429{
430    status_t status = TrackBase::initCheck();
431    if (status == NO_ERROR && mName < 0) {
432        status = NO_MEMORY;
433    }
434    return status;
435}
436
437void AudioFlinger::PlaybackThread::Track::destroy()
438{
439    // NOTE: destroyTrack_l() can remove a strong reference to this Track
440    // by removing it from mTracks vector, so there is a risk that this Tracks's
441    // destructor is called. As the destructor needs to lock mLock,
442    // we must acquire a strong reference on this Track before locking mLock
443    // here so that the destructor is called only when exiting this function.
444    // On the other hand, as long as Track::destroy() is only called by
445    // TrackHandle destructor, the TrackHandle still holds a strong ref on
446    // this Track with its member mTrack.
447    sp<Track> keep(this);
448    { // scope for mLock
449        bool wasActive = false;
450        sp<ThreadBase> thread = mThread.promote();
451        if (thread != 0) {
452            Mutex::Autolock _l(thread->mLock);
453            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
454            wasActive = playbackThread->destroyTrack_l(this);
455        }
456        if (isExternalTrack() && !wasActive) {
457            AudioSystem::releaseOutput(mThreadIoHandle, mStreamType, mSessionId);
458        }
459    }
460}
461
462/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
463{
464    result.append("    Name Active Client Type      Fmt Chn mask Session fCount S F SRate  "
465                  "L dB  R dB    Server Main buf  Aux Buf Flags UndFrmCnt\n");
466}
467
468void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
469{
470    gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
471    if (isFastTrack()) {
472        sprintf(buffer, "    F %2d", mFastIndex);
473    } else if (mName >= AudioMixer::TRACK0) {
474        sprintf(buffer, "    %4d", mName - AudioMixer::TRACK0);
475    } else {
476        sprintf(buffer, "    none");
477    }
478    track_state state = mState;
479    char stateChar;
480    if (isTerminated()) {
481        stateChar = 'T';
482    } else {
483        switch (state) {
484        case IDLE:
485            stateChar = 'I';
486            break;
487        case STOPPING_1:
488            stateChar = 's';
489            break;
490        case STOPPING_2:
491            stateChar = '5';
492            break;
493        case STOPPED:
494            stateChar = 'S';
495            break;
496        case RESUMING:
497            stateChar = 'R';
498            break;
499        case ACTIVE:
500            stateChar = 'A';
501            break;
502        case PAUSING:
503            stateChar = 'p';
504            break;
505        case PAUSED:
506            stateChar = 'P';
507            break;
508        case FLUSHED:
509            stateChar = 'F';
510            break;
511        default:
512            stateChar = '?';
513            break;
514        }
515    }
516    char nowInUnderrun;
517    switch (mObservedUnderruns.mBitFields.mMostRecent) {
518    case UNDERRUN_FULL:
519        nowInUnderrun = ' ';
520        break;
521    case UNDERRUN_PARTIAL:
522        nowInUnderrun = '<';
523        break;
524    case UNDERRUN_EMPTY:
525        nowInUnderrun = '*';
526        break;
527    default:
528        nowInUnderrun = '?';
529        break;
530    }
531    snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g  "
532                                 "%08X %p %p 0x%03X %9u%c\n",
533            active ? "yes" : "no",
534            (mClient == 0) ? getpid_cached : mClient->pid(),
535            mStreamType,
536            mFormat,
537            mChannelMask,
538            mSessionId,
539            mFrameCount,
540            stateChar,
541            mFillingUpStatus,
542            mAudioTrackServerProxy->getSampleRate(),
543            20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
544            20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
545            mCblk->mServer,
546            mMainBuffer,
547            mAuxBuffer,
548            mCblk->mFlags,
549            mAudioTrackServerProxy->getUnderrunFrames(),
550            nowInUnderrun);
551}
552
553uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
554    return mAudioTrackServerProxy->getSampleRate();
555}
556
557// AudioBufferProvider interface
558status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
559        AudioBufferProvider::Buffer* buffer)
560{
561    ServerProxy::Buffer buf;
562    size_t desiredFrames = buffer->frameCount;
563    buf.mFrameCount = desiredFrames;
564    status_t status = mServerProxy->obtainBuffer(&buf);
565    buffer->frameCount = buf.mFrameCount;
566    buffer->raw = buf.mRaw;
567    if (buf.mFrameCount == 0) {
568        mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
569    } else {
570        mAudioTrackServerProxy->tallyUnderrunFrames(0);
571    }
572
573    return status;
574}
575
576// releaseBuffer() is not overridden
577
578// ExtendedAudioBufferProvider interface
579
580// framesReady() may return an approximation of the number of frames if called
581// from a different thread than the one calling Proxy->obtainBuffer() and
582// Proxy->releaseBuffer(). Also note there is no mutual exclusion in the
583// AudioTrackServerProxy so be especially careful calling with FastTracks.
584size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
585    if (mSharedBuffer != 0 && (isStopped() || isStopping())) {
586        // Static tracks return zero frames immediately upon stopping (for FastTracks).
587        // The remainder of the buffer is not drained.
588        return 0;
589    }
590    return mAudioTrackServerProxy->framesReady();
591}
592
593int64_t AudioFlinger::PlaybackThread::Track::framesReleased() const
594{
595    return mAudioTrackServerProxy->framesReleased();
596}
597
598void AudioFlinger::PlaybackThread::Track::onTimestamp(const ExtendedTimestamp &timestamp)
599{
600    // This call comes from a FastTrack and should be kept lockless.
601    // The server side frames are already translated to client frames.
602    mAudioTrackServerProxy->setTimestamp(timestamp);
603
604    // We do not set drained here, as FastTrack timestamp may not go to very last frame.
605}
606
607// Don't call for fast tracks; the framesReady() could result in priority inversion
608bool AudioFlinger::PlaybackThread::Track::isReady() const {
609    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
610        return true;
611    }
612
613    if (isStopping()) {
614        if (framesReady() > 0) {
615            mFillingUpStatus = FS_FILLED;
616        }
617        return true;
618    }
619
620    if (framesReady() >= mServerProxy->getBufferSizeInFrames() ||
621            (mCblk->mFlags & CBLK_FORCEREADY)) {
622        mFillingUpStatus = FS_FILLED;
623        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
624        return true;
625    }
626    return false;
627}
628
629status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
630                                                    audio_session_t triggerSession __unused)
631{
632    status_t status = NO_ERROR;
633    ALOGV("start(%d), calling pid %d session %d",
634            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
635
636    sp<ThreadBase> thread = mThread.promote();
637    if (thread != 0) {
638        if (isOffloaded()) {
639            Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
640            Mutex::Autolock _lth(thread->mLock);
641            sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
642            if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
643                    (ec != 0 && ec->isNonOffloadableEnabled())) {
644                invalidate();
645                return PERMISSION_DENIED;
646            }
647        }
648        Mutex::Autolock _lth(thread->mLock);
649        track_state state = mState;
650        // here the track could be either new, or restarted
651        // in both cases "unstop" the track
652
653        // initial state-stopping. next state-pausing.
654        // What if resume is called ?
655
656        if (state == PAUSED || state == PAUSING) {
657            if (mResumeToStopping) {
658                // happened we need to resume to STOPPING_1
659                mState = TrackBase::STOPPING_1;
660                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
661            } else {
662                mState = TrackBase::RESUMING;
663                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
664            }
665        } else {
666            mState = TrackBase::ACTIVE;
667            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
668        }
669
670        // states to reset position info for non-offloaded/direct tracks
671        if (!isOffloaded() && !isDirect()
672                && (state == IDLE || state == STOPPED || state == FLUSHED)) {
673            mFrameMap.reset();
674        }
675        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
676        if (isFastTrack()) {
677            // refresh fast track underruns on start because that field is never cleared
678            // by the fast mixer; furthermore, the same track can be recycled, i.e. start
679            // after stop.
680            mObservedUnderruns = playbackThread->getFastTrackUnderruns(mFastIndex);
681        }
682        status = playbackThread->addTrack_l(this);
683        if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
684            triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
685            //  restore previous state if start was rejected by policy manager
686            if (status == PERMISSION_DENIED) {
687                mState = state;
688            }
689        }
690        // track was already in the active list, not a problem
691        if (status == ALREADY_EXISTS) {
692            status = NO_ERROR;
693        } else {
694            // Acknowledge any pending flush(), so that subsequent new data isn't discarded.
695            // It is usually unsafe to access the server proxy from a binder thread.
696            // But in this case we know the mixer thread (whether normal mixer or fast mixer)
697            // isn't looking at this track yet:  we still hold the normal mixer thread lock,
698            // and for fast tracks the track is not yet in the fast mixer thread's active set.
699            // For static tracks, this is used to acknowledge change in position or loop.
700            ServerProxy::Buffer buffer;
701            buffer.mFrameCount = 1;
702            (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
703        }
704    } else {
705        status = BAD_VALUE;
706    }
707    return status;
708}
709
710void AudioFlinger::PlaybackThread::Track::stop()
711{
712    ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
713    sp<ThreadBase> thread = mThread.promote();
714    if (thread != 0) {
715        Mutex::Autolock _l(thread->mLock);
716        track_state state = mState;
717        if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
718            // If the track is not active (PAUSED and buffers full), flush buffers
719            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
720            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
721                reset();
722                mState = STOPPED;
723            } else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
724                mState = STOPPED;
725            } else {
726                // For fast tracks prepareTracks_l() will set state to STOPPING_2
727                // presentation is complete
728                // For an offloaded track this starts a drain and state will
729                // move to STOPPING_2 when drain completes and then STOPPED
730                mState = STOPPING_1;
731                if (isOffloaded()) {
732                    mRetryCount = PlaybackThread::kMaxTrackStopRetriesOffload;
733                }
734            }
735            playbackThread->broadcast_l();
736            ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
737                    playbackThread);
738        }
739    }
740}
741
742void AudioFlinger::PlaybackThread::Track::pause()
743{
744    ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
745    sp<ThreadBase> thread = mThread.promote();
746    if (thread != 0) {
747        Mutex::Autolock _l(thread->mLock);
748        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
749        switch (mState) {
750        case STOPPING_1:
751        case STOPPING_2:
752            if (!isOffloaded()) {
753                /* nothing to do if track is not offloaded */
754                break;
755            }
756
757            // Offloaded track was draining, we need to carry on draining when resumed
758            mResumeToStopping = true;
759            // fall through...
760        case ACTIVE:
761        case RESUMING:
762            mState = PAUSING;
763            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
764            playbackThread->broadcast_l();
765            break;
766
767        default:
768            break;
769        }
770    }
771}
772
773void AudioFlinger::PlaybackThread::Track::flush()
774{
775    ALOGV("flush(%d)", mName);
776    sp<ThreadBase> thread = mThread.promote();
777    if (thread != 0) {
778        Mutex::Autolock _l(thread->mLock);
779        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
780
781        if (isOffloaded()) {
782            // If offloaded we allow flush during any state except terminated
783            // and keep the track active to avoid problems if user is seeking
784            // rapidly and underlying hardware has a significant delay handling
785            // a pause
786            if (isTerminated()) {
787                return;
788            }
789
790            ALOGV("flush: offload flush");
791            reset();
792
793            if (mState == STOPPING_1 || mState == STOPPING_2) {
794                ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE");
795                mState = ACTIVE;
796            }
797
798            mFlushHwPending = true;
799            mResumeToStopping = false;
800        } else {
801            if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
802                    mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
803                return;
804            }
805            // No point remaining in PAUSED state after a flush => go to
806            // FLUSHED state
807            mState = FLUSHED;
808            // do not reset the track if it is still in the process of being stopped or paused.
809            // this will be done by prepareTracks_l() when the track is stopped.
810            // prepareTracks_l() will see mState == FLUSHED, then
811            // remove from active track list, reset(), and trigger presentation complete
812            if (isDirect()) {
813                mFlushHwPending = true;
814            }
815            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
816                reset();
817            }
818        }
819        // Prevent flush being lost if the track is flushed and then resumed
820        // before mixer thread can run. This is important when offloading
821        // because the hardware buffer could hold a large amount of audio
822        playbackThread->broadcast_l();
823    }
824}
825
826// must be called with thread lock held
827void AudioFlinger::PlaybackThread::Track::flushAck()
828{
829    if (!isOffloaded() && !isDirect())
830        return;
831
832    mFlushHwPending = false;
833}
834
835void AudioFlinger::PlaybackThread::Track::reset()
836{
837    // Do not reset twice to avoid discarding data written just after a flush and before
838    // the audioflinger thread detects the track is stopped.
839    if (!mResetDone) {
840        // Force underrun condition to avoid false underrun callback until first data is
841        // written to buffer
842        android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
843        mFillingUpStatus = FS_FILLING;
844        mResetDone = true;
845        if (mState == FLUSHED) {
846            mState = IDLE;
847        }
848    }
849}
850
851status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
852{
853    sp<ThreadBase> thread = mThread.promote();
854    if (thread == 0) {
855        ALOGE("thread is dead");
856        return FAILED_TRANSACTION;
857    } else if ((thread->type() == ThreadBase::DIRECT) ||
858                    (thread->type() == ThreadBase::OFFLOAD)) {
859        return thread->setParameters(keyValuePairs);
860    } else {
861        return PERMISSION_DENIED;
862    }
863}
864
865status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
866{
867    if (!isOffloaded() && !isDirect()) {
868        return INVALID_OPERATION; // normal tracks handled through SSQ
869    }
870    sp<ThreadBase> thread = mThread.promote();
871    if (thread == 0) {
872        return INVALID_OPERATION;
873    }
874
875    Mutex::Autolock _l(thread->mLock);
876    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
877    return playbackThread->getTimestamp_l(timestamp);
878}
879
880status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
881{
882    status_t status = DEAD_OBJECT;
883    sp<ThreadBase> thread = mThread.promote();
884    if (thread != 0) {
885        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
886        sp<AudioFlinger> af = mClient->audioFlinger();
887
888        Mutex::Autolock _l(af->mLock);
889
890        sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
891
892        if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
893            Mutex::Autolock _dl(playbackThread->mLock);
894            Mutex::Autolock _sl(srcThread->mLock);
895            sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
896            if (chain == 0) {
897                return INVALID_OPERATION;
898            }
899
900            sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
901            if (effect == 0) {
902                return INVALID_OPERATION;
903            }
904            srcThread->removeEffect_l(effect);
905            status = playbackThread->addEffect_l(effect);
906            if (status != NO_ERROR) {
907                srcThread->addEffect_l(effect);
908                return INVALID_OPERATION;
909            }
910            // removeEffect_l() has stopped the effect if it was active so it must be restarted
911            if (effect->state() == EffectModule::ACTIVE ||
912                    effect->state() == EffectModule::STOPPING) {
913                effect->start();
914            }
915
916            sp<EffectChain> dstChain = effect->chain().promote();
917            if (dstChain == 0) {
918                srcThread->addEffect_l(effect);
919                return INVALID_OPERATION;
920            }
921            AudioSystem::unregisterEffect(effect->id());
922            AudioSystem::registerEffect(&effect->desc(),
923                                        srcThread->id(),
924                                        dstChain->strategy(),
925                                        AUDIO_SESSION_OUTPUT_MIX,
926                                        effect->id());
927            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
928        }
929        status = playbackThread->attachAuxEffect(this, EffectId);
930    }
931    return status;
932}
933
934void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
935{
936    mAuxEffectId = EffectId;
937    mAuxBuffer = buffer;
938}
939
940bool AudioFlinger::PlaybackThread::Track::presentationComplete(
941        int64_t framesWritten, size_t audioHalFrames)
942{
943    // TODO: improve this based on FrameMap if it exists, to ensure full drain.
944    // This assists in proper timestamp computation as well as wakelock management.
945
946    // a track is considered presented when the total number of frames written to audio HAL
947    // corresponds to the number of frames written when presentationComplete() is called for the
948    // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
949    // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
950    // to detect when all frames have been played. In this case framesWritten isn't
951    // useful because it doesn't always reflect whether there is data in the h/w
952    // buffers, particularly if a track has been paused and resumed during draining
953    ALOGV("presentationComplete() mPresentationCompleteFrames %lld framesWritten %lld",
954            (long long)mPresentationCompleteFrames, (long long)framesWritten);
955    if (mPresentationCompleteFrames == 0) {
956        mPresentationCompleteFrames = framesWritten + audioHalFrames;
957        ALOGV("presentationComplete() reset: mPresentationCompleteFrames %lld audioHalFrames %zu",
958                (long long)mPresentationCompleteFrames, audioHalFrames);
959    }
960
961    bool complete;
962    if (isOffloaded()) {
963        complete = true;
964    } else if (isDirect() || isFastTrack()) { // these do not go through linear map
965        complete = framesWritten >= (int64_t) mPresentationCompleteFrames;
966    } else {  // Normal tracks, OutputTracks, and PatchTracks
967        complete = framesWritten >= (int64_t) mPresentationCompleteFrames
968                && mAudioTrackServerProxy->isDrained();
969    }
970
971    if (complete) {
972        triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
973        mAudioTrackServerProxy->setStreamEndDone();
974        return true;
975    }
976    return false;
977}
978
979void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
980{
981    for (size_t i = 0; i < mSyncEvents.size(); i++) {
982        if (mSyncEvents[i]->type() == type) {
983            mSyncEvents[i]->trigger();
984            mSyncEvents.removeAt(i);
985            i--;
986        }
987    }
988}
989
990// implement VolumeBufferProvider interface
991
992gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
993{
994    // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
995    ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
996    gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
997    float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
998    float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
999    // track volumes come from shared memory, so can't be trusted and must be clamped
1000    if (vl > GAIN_FLOAT_UNITY) {
1001        vl = GAIN_FLOAT_UNITY;
1002    }
1003    if (vr > GAIN_FLOAT_UNITY) {
1004        vr = GAIN_FLOAT_UNITY;
1005    }
1006    // now apply the cached master volume and stream type volume;
1007    // this is trusted but lacks any synchronization or barrier so may be stale
1008    float v = mCachedVolume;
1009    vl *= v;
1010    vr *= v;
1011    // re-combine into packed minifloat
1012    vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
1013    // FIXME look at mute, pause, and stop flags
1014    return vlr;
1015}
1016
1017status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
1018{
1019    if (isTerminated() || mState == PAUSED ||
1020            ((framesReady() == 0) && ((mSharedBuffer != 0) ||
1021                                      (mState == STOPPED)))) {
1022        ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %zu",
1023              mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
1024        event->cancel();
1025        return INVALID_OPERATION;
1026    }
1027    (void) TrackBase::setSyncEvent(event);
1028    return NO_ERROR;
1029}
1030
1031void AudioFlinger::PlaybackThread::Track::invalidate()
1032{
1033    signalClientFlag(CBLK_INVALID);
1034    mIsInvalid = true;
1035}
1036
1037void AudioFlinger::PlaybackThread::Track::disable()
1038{
1039    signalClientFlag(CBLK_DISABLED);
1040}
1041
1042void AudioFlinger::PlaybackThread::Track::signalClientFlag(int32_t flag)
1043{
1044    // FIXME should use proxy, and needs work
1045    audio_track_cblk_t* cblk = mCblk;
1046    android_atomic_or(flag, &cblk->mFlags);
1047    android_atomic_release_store(0x40000000, &cblk->mFutex);
1048    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1049    (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
1050}
1051
1052void AudioFlinger::PlaybackThread::Track::signal()
1053{
1054    sp<ThreadBase> thread = mThread.promote();
1055    if (thread != 0) {
1056        PlaybackThread *t = (PlaybackThread *)thread.get();
1057        Mutex::Autolock _l(t->mLock);
1058        t->broadcast_l();
1059    }
1060}
1061
1062//To be called with thread lock held
1063bool AudioFlinger::PlaybackThread::Track::isResumePending() {
1064
1065    if (mState == RESUMING)
1066        return true;
1067    /* Resume is pending if track was stopping before pause was called */
1068    if (mState == STOPPING_1 &&
1069        mResumeToStopping)
1070        return true;
1071
1072    return false;
1073}
1074
1075//To be called with thread lock held
1076void AudioFlinger::PlaybackThread::Track::resumeAck() {
1077
1078
1079    if (mState == RESUMING)
1080        mState = ACTIVE;
1081
1082    // Other possibility of  pending resume is stopping_1 state
1083    // Do not update the state from stopping as this prevents
1084    // drain being called.
1085    if (mState == STOPPING_1) {
1086        mResumeToStopping = false;
1087    }
1088}
1089
1090//To be called with thread lock held
1091void AudioFlinger::PlaybackThread::Track::updateTrackFrameInfo(
1092        int64_t trackFramesReleased, int64_t sinkFramesWritten,
1093        const ExtendedTimestamp &timeStamp) {
1094    //update frame map
1095    mFrameMap.push(trackFramesReleased, sinkFramesWritten);
1096
1097    // adjust server times and set drained state.
1098    //
1099    // Our timestamps are only updated when the track is on the Thread active list.
1100    // We need to ensure that tracks are not removed before full drain.
1101    ExtendedTimestamp local = timeStamp;
1102    bool checked = false;
1103    for (int i = ExtendedTimestamp::LOCATION_MAX - 1;
1104            i >= ExtendedTimestamp::LOCATION_SERVER; --i) {
1105        // Lookup the track frame corresponding to the sink frame position.
1106        if (local.mTimeNs[i] > 0) {
1107            local.mPosition[i] = mFrameMap.findX(local.mPosition[i]);
1108            // check drain state from the latest stage in the pipeline.
1109            if (!checked && i <= ExtendedTimestamp::LOCATION_KERNEL) {
1110                mAudioTrackServerProxy->setDrained(
1111                        local.mPosition[i] >= mAudioTrackServerProxy->framesReleased());
1112                checked = true;
1113            }
1114        }
1115    }
1116    if (!checked) { // no server info, assume drained.
1117        mAudioTrackServerProxy->setDrained(true);
1118    }
1119    // Set correction for flushed frames that are not accounted for in released.
1120    local.mFlushed = mAudioTrackServerProxy->framesFlushed();
1121    mServerProxy->setTimestamp(local);
1122}
1123
1124// ----------------------------------------------------------------------------
1125
1126AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1127            PlaybackThread *playbackThread,
1128            DuplicatingThread *sourceThread,
1129            uint32_t sampleRate,
1130            audio_format_t format,
1131            audio_channel_mask_t channelMask,
1132            size_t frameCount,
1133            int uid)
1134    :   Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
1135              sampleRate, format, channelMask, frameCount,
1136              NULL, 0, AUDIO_SESSION_NONE, uid, IAudioFlinger::TRACK_DEFAULT,
1137              TYPE_OUTPUT),
1138    mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
1139{
1140
1141    if (mCblk != NULL) {
1142        mOutBuffer.frameCount = 0;
1143        playbackThread->mTracks.add(this);
1144        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1145                "frameCount %zu, mChannelMask 0x%08x",
1146                mCblk, mBuffer,
1147                frameCount, mChannelMask);
1148        // since client and server are in the same process,
1149        // the buffer has the same virtual address on both sides
1150        mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
1151                true /*clientInServer*/);
1152        mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
1153        mClientProxy->setSendLevel(0.0);
1154        mClientProxy->setSampleRate(sampleRate);
1155    } else {
1156        ALOGW("Error creating output track on thread %p", playbackThread);
1157    }
1158}
1159
1160AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1161{
1162    clearBufferQueue();
1163    delete mClientProxy;
1164    // superclass destructor will now delete the server proxy and shared memory both refer to
1165}
1166
1167status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1168                                                          audio_session_t triggerSession)
1169{
1170    status_t status = Track::start(event, triggerSession);
1171    if (status != NO_ERROR) {
1172        return status;
1173    }
1174
1175    mActive = true;
1176    mRetryCount = 127;
1177    return status;
1178}
1179
1180void AudioFlinger::PlaybackThread::OutputTrack::stop()
1181{
1182    Track::stop();
1183    clearBufferQueue();
1184    mOutBuffer.frameCount = 0;
1185    mActive = false;
1186}
1187
1188bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
1189{
1190    Buffer *pInBuffer;
1191    Buffer inBuffer;
1192    bool outputBufferFull = false;
1193    inBuffer.frameCount = frames;
1194    inBuffer.raw = data;
1195
1196    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1197
1198    if (!mActive && frames != 0) {
1199        (void) start();
1200    }
1201
1202    while (waitTimeLeftMs) {
1203        // First write pending buffers, then new data
1204        if (mBufferQueue.size()) {
1205            pInBuffer = mBufferQueue.itemAt(0);
1206        } else {
1207            pInBuffer = &inBuffer;
1208        }
1209
1210        if (pInBuffer->frameCount == 0) {
1211            break;
1212        }
1213
1214        if (mOutBuffer.frameCount == 0) {
1215            mOutBuffer.frameCount = pInBuffer->frameCount;
1216            nsecs_t startTime = systemTime();
1217            status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
1218            if (status != NO_ERROR && status != NOT_ENOUGH_DATA) {
1219                ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this,
1220                        mThread.unsafe_get(), status);
1221                outputBufferFull = true;
1222                break;
1223            }
1224            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1225            if (waitTimeLeftMs >= waitTimeMs) {
1226                waitTimeLeftMs -= waitTimeMs;
1227            } else {
1228                waitTimeLeftMs = 0;
1229            }
1230            if (status == NOT_ENOUGH_DATA) {
1231                restartIfDisabled();
1232                continue;
1233            }
1234        }
1235
1236        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1237                pInBuffer->frameCount;
1238        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize);
1239        Proxy::Buffer buf;
1240        buf.mFrameCount = outFrames;
1241        buf.mRaw = NULL;
1242        mClientProxy->releaseBuffer(&buf);
1243        restartIfDisabled();
1244        pInBuffer->frameCount -= outFrames;
1245        pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize;
1246        mOutBuffer.frameCount -= outFrames;
1247        mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize;
1248
1249        if (pInBuffer->frameCount == 0) {
1250            if (mBufferQueue.size()) {
1251                mBufferQueue.removeAt(0);
1252                free(pInBuffer->mBuffer);
1253                delete pInBuffer;
1254                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %zu", this,
1255                        mThread.unsafe_get(), mBufferQueue.size());
1256            } else {
1257                break;
1258            }
1259        }
1260    }
1261
1262    // If we could not write all frames, allocate a buffer and queue it for next time.
1263    if (inBuffer.frameCount) {
1264        sp<ThreadBase> thread = mThread.promote();
1265        if (thread != 0 && !thread->standby()) {
1266            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1267                pInBuffer = new Buffer;
1268                pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize);
1269                pInBuffer->frameCount = inBuffer.frameCount;
1270                pInBuffer->raw = pInBuffer->mBuffer;
1271                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
1272                mBufferQueue.add(pInBuffer);
1273                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %zu", this,
1274                        mThread.unsafe_get(), mBufferQueue.size());
1275            } else {
1276                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1277                        mThread.unsafe_get(), this);
1278            }
1279        }
1280    }
1281
1282    // Calling write() with a 0 length buffer means that no more data will be written:
1283    // We rely on stop() to set the appropriate flags to allow the remaining frames to play out.
1284    if (frames == 0 && mBufferQueue.size() == 0 && mActive) {
1285        stop();
1286    }
1287
1288    return outputBufferFull;
1289}
1290
1291status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1292        AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1293{
1294    ClientProxy::Buffer buf;
1295    buf.mFrameCount = buffer->frameCount;
1296    struct timespec timeout;
1297    timeout.tv_sec = waitTimeMs / 1000;
1298    timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
1299    status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
1300    buffer->frameCount = buf.mFrameCount;
1301    buffer->raw = buf.mRaw;
1302    return status;
1303}
1304
1305void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1306{
1307    size_t size = mBufferQueue.size();
1308
1309    for (size_t i = 0; i < size; i++) {
1310        Buffer *pBuffer = mBufferQueue.itemAt(i);
1311        free(pBuffer->mBuffer);
1312        delete pBuffer;
1313    }
1314    mBufferQueue.clear();
1315}
1316
1317void AudioFlinger::PlaybackThread::OutputTrack::restartIfDisabled()
1318{
1319    int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
1320    if (mActive && (flags & CBLK_DISABLED)) {
1321        start();
1322    }
1323}
1324
1325AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread,
1326                                                     audio_stream_type_t streamType,
1327                                                     uint32_t sampleRate,
1328                                                     audio_channel_mask_t channelMask,
1329                                                     audio_format_t format,
1330                                                     size_t frameCount,
1331                                                     void *buffer,
1332                                                     IAudioFlinger::track_flags_t flags)
1333    :   Track(playbackThread, NULL, streamType,
1334              sampleRate, format, channelMask, frameCount,
1335              buffer, 0, AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
1336              mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true))
1337{
1338    uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) /
1339                                                                    playbackThread->sampleRate();
1340    mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
1341    mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
1342
1343    ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec",
1344                                      this, sampleRate,
1345                                      (int)mPeerTimeout.tv_sec,
1346                                      (int)(mPeerTimeout.tv_nsec / 1000000));
1347}
1348
1349AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack()
1350{
1351}
1352
1353status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event,
1354                                                          audio_session_t triggerSession)
1355{
1356    status_t status = Track::start(event, triggerSession);
1357    if (status != NO_ERROR) {
1358        return status;
1359    }
1360    android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
1361    return status;
1362}
1363
1364// AudioBufferProvider interface
1365status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer(
1366        AudioBufferProvider::Buffer* buffer)
1367{
1368    ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy");
1369    Proxy::Buffer buf;
1370    buf.mFrameCount = buffer->frameCount;
1371    status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
1372    ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status);
1373    buffer->frameCount = buf.mFrameCount;
1374    if (buf.mFrameCount == 0) {
1375        return WOULD_BLOCK;
1376    }
1377    status = Track::getNextBuffer(buffer);
1378    return status;
1379}
1380
1381void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer)
1382{
1383    ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy");
1384    Proxy::Buffer buf;
1385    buf.mFrameCount = buffer->frameCount;
1386    buf.mRaw = buffer->raw;
1387    mPeerProxy->releaseBuffer(&buf);
1388    TrackBase::releaseBuffer(buffer);
1389}
1390
1391status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
1392                                                                const struct timespec *timeOut)
1393{
1394    status_t status = NO_ERROR;
1395    static const int32_t kMaxTries = 5;
1396    int32_t tryCounter = kMaxTries;
1397    do {
1398        if (status == NOT_ENOUGH_DATA) {
1399            restartIfDisabled();
1400        }
1401        status = mProxy->obtainBuffer(buffer, timeOut);
1402    } while ((status == NOT_ENOUGH_DATA) && (tryCounter-- > 0));
1403    return status;
1404}
1405
1406void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
1407{
1408    mProxy->releaseBuffer(buffer);
1409    restartIfDisabled();
1410    android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
1411}
1412
1413void AudioFlinger::PlaybackThread::PatchTrack::restartIfDisabled()
1414{
1415    if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) {
1416        ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting");
1417        start();
1418    }
1419}
1420
1421// ----------------------------------------------------------------------------
1422//      Record
1423// ----------------------------------------------------------------------------
1424
1425AudioFlinger::RecordHandle::RecordHandle(
1426        const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1427    : BnAudioRecord(),
1428    mRecordTrack(recordTrack)
1429{
1430}
1431
1432AudioFlinger::RecordHandle::~RecordHandle() {
1433    stop_nonvirtual();
1434    mRecordTrack->destroy();
1435}
1436
1437status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1438        audio_session_t triggerSession) {
1439    ALOGV("RecordHandle::start()");
1440    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1441}
1442
1443void AudioFlinger::RecordHandle::stop() {
1444    stop_nonvirtual();
1445}
1446
1447void AudioFlinger::RecordHandle::stop_nonvirtual() {
1448    ALOGV("RecordHandle::stop()");
1449    mRecordTrack->stop();
1450}
1451
1452status_t AudioFlinger::RecordHandle::onTransact(
1453    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1454{
1455    return BnAudioRecord::onTransact(code, data, reply, flags);
1456}
1457
1458// ----------------------------------------------------------------------------
1459
1460// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
1461AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1462            RecordThread *thread,
1463            const sp<Client>& client,
1464            uint32_t sampleRate,
1465            audio_format_t format,
1466            audio_channel_mask_t channelMask,
1467            size_t frameCount,
1468            void *buffer,
1469            audio_session_t sessionId,
1470            int uid,
1471            IAudioFlinger::track_flags_t flags,
1472            track_type type)
1473    :   TrackBase(thread, client, sampleRate, format,
1474                  channelMask, frameCount, buffer, sessionId, uid,
1475                  flags, false /*isOut*/,
1476                  (type == TYPE_DEFAULT) ?
1477                          ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
1478                          ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
1479                  type),
1480        mOverflow(false),
1481        mFramesToDrop(0),
1482        mResamplerBufferProvider(NULL), // initialize in case of early constructor exit
1483        mRecordBufferConverter(NULL)
1484{
1485    if (mCblk == NULL) {
1486        return;
1487    }
1488
1489    mRecordBufferConverter = new RecordBufferConverter(
1490            thread->mChannelMask, thread->mFormat, thread->mSampleRate,
1491            channelMask, format, sampleRate);
1492    // Check if the RecordBufferConverter construction was successful.
1493    // If not, don't continue with construction.
1494    //
1495    // NOTE: It would be extremely rare that the record track cannot be created
1496    // for the current device, but a pending or future device change would make
1497    // the record track configuration valid.
1498    if (mRecordBufferConverter->initCheck() != NO_ERROR) {
1499        ALOGE("RecordTrack unable to create record buffer converter");
1500        return;
1501    }
1502
1503    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
1504            mFrameSize, !isExternalTrack());
1505
1506    mResamplerBufferProvider = new ResamplerBufferProvider(this);
1507
1508    if (flags & IAudioFlinger::TRACK_FAST) {
1509        ALOG_ASSERT(thread->mFastTrackAvail);
1510        thread->mFastTrackAvail = false;
1511    }
1512}
1513
1514AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1515{
1516    ALOGV("%s", __func__);
1517    delete mRecordBufferConverter;
1518    delete mResamplerBufferProvider;
1519}
1520
1521status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const
1522{
1523    status_t status = TrackBase::initCheck();
1524    if (status == NO_ERROR && mServerProxy == 0) {
1525        status = BAD_VALUE;
1526    }
1527    return status;
1528}
1529
1530// AudioBufferProvider interface
1531status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
1532{
1533    ServerProxy::Buffer buf;
1534    buf.mFrameCount = buffer->frameCount;
1535    status_t status = mServerProxy->obtainBuffer(&buf);
1536    buffer->frameCount = buf.mFrameCount;
1537    buffer->raw = buf.mRaw;
1538    if (buf.mFrameCount == 0) {
1539        // FIXME also wake futex so that overrun is noticed more quickly
1540        (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
1541    }
1542    return status;
1543}
1544
1545status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1546                                                        audio_session_t triggerSession)
1547{
1548    sp<ThreadBase> thread = mThread.promote();
1549    if (thread != 0) {
1550        RecordThread *recordThread = (RecordThread *)thread.get();
1551        return recordThread->start(this, event, triggerSession);
1552    } else {
1553        return BAD_VALUE;
1554    }
1555}
1556
1557void AudioFlinger::RecordThread::RecordTrack::stop()
1558{
1559    sp<ThreadBase> thread = mThread.promote();
1560    if (thread != 0) {
1561        RecordThread *recordThread = (RecordThread *)thread.get();
1562        if (recordThread->stop(this) && isExternalTrack()) {
1563            AudioSystem::stopInput(mThreadIoHandle, mSessionId);
1564        }
1565    }
1566}
1567
1568void AudioFlinger::RecordThread::RecordTrack::destroy()
1569{
1570    // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1571    sp<RecordTrack> keep(this);
1572    {
1573        if (isExternalTrack()) {
1574            if (mState == ACTIVE || mState == RESUMING) {
1575                AudioSystem::stopInput(mThreadIoHandle, mSessionId);
1576            }
1577            AudioSystem::releaseInput(mThreadIoHandle, mSessionId);
1578        }
1579        sp<ThreadBase> thread = mThread.promote();
1580        if (thread != 0) {
1581            Mutex::Autolock _l(thread->mLock);
1582            RecordThread *recordThread = (RecordThread *) thread.get();
1583            recordThread->destroyTrack_l(this);
1584        }
1585    }
1586}
1587
1588void AudioFlinger::RecordThread::RecordTrack::invalidate()
1589{
1590    // FIXME should use proxy, and needs work
1591    audio_track_cblk_t* cblk = mCblk;
1592    android_atomic_or(CBLK_INVALID, &cblk->mFlags);
1593    android_atomic_release_store(0x40000000, &cblk->mFutex);
1594    // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
1595    (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
1596}
1597
1598
1599/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1600{
1601    result.append("    Active Client Fmt Chn mask Session S   Server fCount SRate\n");
1602}
1603
1604void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
1605{
1606    snprintf(buffer, size, "    %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n",
1607            active ? "yes" : "no",
1608            (mClient == 0) ? getpid_cached : mClient->pid(),
1609            mFormat,
1610            mChannelMask,
1611            mSessionId,
1612            mState,
1613            mCblk->mServer,
1614            mFrameCount,
1615            mSampleRate);
1616
1617}
1618
1619void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
1620{
1621    if (event == mSyncStartEvent) {
1622        ssize_t framesToDrop = 0;
1623        sp<ThreadBase> threadBase = mThread.promote();
1624        if (threadBase != 0) {
1625            // TODO: use actual buffer filling status instead of 2 buffers when info is available
1626            // from audio HAL
1627            framesToDrop = threadBase->mFrameCount * 2;
1628        }
1629        mFramesToDrop = framesToDrop;
1630    }
1631}
1632
1633void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
1634{
1635    if (mSyncStartEvent != 0) {
1636        mSyncStartEvent->cancel();
1637        mSyncStartEvent.clear();
1638    }
1639    mFramesToDrop = 0;
1640}
1641
1642void AudioFlinger::RecordThread::RecordTrack::updateTrackFrameInfo(
1643        int64_t trackFramesReleased, int64_t sourceFramesRead,
1644        uint32_t halSampleRate, const ExtendedTimestamp &timestamp)
1645{
1646    ExtendedTimestamp local = timestamp;
1647
1648    // Convert HAL frames to server-side track frames at track sample rate.
1649    // We use trackFramesReleased and sourceFramesRead as an anchor point.
1650    for (int i = ExtendedTimestamp::LOCATION_SERVER; i < ExtendedTimestamp::LOCATION_MAX; ++i) {
1651        if (local.mTimeNs[i] != 0) {
1652            const int64_t relativeServerFrames = local.mPosition[i] - sourceFramesRead;
1653            const int64_t relativeTrackFrames = relativeServerFrames
1654                    * mSampleRate / halSampleRate; // TODO: potential computation overflow
1655            local.mPosition[i] = relativeTrackFrames + trackFramesReleased;
1656        }
1657    }
1658    mServerProxy->setTimestamp(local);
1659}
1660
1661AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
1662                                                     uint32_t sampleRate,
1663                                                     audio_channel_mask_t channelMask,
1664                                                     audio_format_t format,
1665                                                     size_t frameCount,
1666                                                     void *buffer,
1667                                                     IAudioFlinger::track_flags_t flags)
1668    :   RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount,
1669                buffer, AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
1670                mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true))
1671{
1672    uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) /
1673                                                                recordThread->sampleRate();
1674    mPeerTimeout.tv_sec = mixBufferNs / 1000000000;
1675    mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000);
1676
1677    ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec",
1678                                      this, sampleRate,
1679                                      (int)mPeerTimeout.tv_sec,
1680                                      (int)(mPeerTimeout.tv_nsec / 1000000));
1681}
1682
1683AudioFlinger::RecordThread::PatchRecord::~PatchRecord()
1684{
1685}
1686
1687// AudioBufferProvider interface
1688status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
1689                                                  AudioBufferProvider::Buffer* buffer)
1690{
1691    ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy");
1692    Proxy::Buffer buf;
1693    buf.mFrameCount = buffer->frameCount;
1694    status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
1695    ALOGV_IF(status != NO_ERROR,
1696             "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status);
1697    buffer->frameCount = buf.mFrameCount;
1698    if (buf.mFrameCount == 0) {
1699        return WOULD_BLOCK;
1700    }
1701    status = RecordTrack::getNextBuffer(buffer);
1702    return status;
1703}
1704
1705void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer)
1706{
1707    ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy");
1708    Proxy::Buffer buf;
1709    buf.mFrameCount = buffer->frameCount;
1710    buf.mRaw = buffer->raw;
1711    mPeerProxy->releaseBuffer(&buf);
1712    TrackBase::releaseBuffer(buffer);
1713}
1714
1715status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer,
1716                                                               const struct timespec *timeOut)
1717{
1718    return mProxy->obtainBuffer(buffer, timeOut);
1719}
1720
1721void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer)
1722{
1723    mProxy->releaseBuffer(buffer);
1724}
1725
1726} // namespace android
1727