AudioTrack.cpp revision 2beeb50b1bba9e92f6cacfeca37fe9fa9d36ead1 
1/* //device/extlibs/pv/android/AudioTrack.cpp
2**
3** Copyright 2007, 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_NDEBUG 0
20#define LOG_TAG "AudioTrack"
21
22#include <stdint.h>
23#include <sys/types.h>
24#include <limits.h>
25
26#include <sched.h>
27#include <sys/resource.h>
28
29#include <private/media/AudioTrackShared.h>
30
31#include <media/AudioSystem.h>
32#include <media/AudioTrack.h>
33
34#include <utils/Log.h>
35#include <binder/Parcel.h>
36#include <binder/IPCThreadState.h>
37#include <utils/Timers.h>
38#include <cutils/atomic.h>
39
40#define LIKELY( exp )       (__builtin_expect( (exp) != 0, true  ))
41#define UNLIKELY( exp )     (__builtin_expect( (exp) != 0, false ))
42
43namespace android {
44// ---------------------------------------------------------------------------
45
46// static
47status_t AudioTrack::getMinFrameCount(
48        int* frameCount,
49        int streamType,
50        uint32_t sampleRate)
51{
52    int afSampleRate;
53    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
54        return NO_INIT;
55    }
56    int afFrameCount;
57    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
58        return NO_INIT;
59    }
60    uint32_t afLatency;
61    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
62        return NO_INIT;
63    }
64
65    // Ensure that buffer depth covers at least audio hardware latency
66    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
67    if (minBufCount < 2) minBufCount = 2;
68
69    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
70              afFrameCount * minBufCount * sampleRate / afSampleRate;
71    return NO_ERROR;
72}
73
74// ---------------------------------------------------------------------------
75
76AudioTrack::AudioTrack()
77    : mStatus(NO_INIT)
78{
79}
80
81AudioTrack::AudioTrack(
82        int streamType,
83        uint32_t sampleRate,
84        int format,
85        int channels,
86        int frameCount,
87        uint32_t flags,
88        callback_t cbf,
89        void* user,
90        int notificationFrames,
91        int sessionId)
92    : mStatus(NO_INIT)
93{
94    mStatus = set(streamType, sampleRate, format, channels,
95            frameCount, flags, cbf, user, notificationFrames,
96            0, false, sessionId);
97}
98
99AudioTrack::AudioTrack(
100        int streamType,
101        uint32_t sampleRate,
102        int format,
103        int channels,
104        const sp<IMemory>& sharedBuffer,
105        uint32_t flags,
106        callback_t cbf,
107        void* user,
108        int notificationFrames,
109        int sessionId)
110    : mStatus(NO_INIT)
111{
112    mStatus = set(streamType, sampleRate, format, channels,
113            0, flags, cbf, user, notificationFrames,
114            sharedBuffer, false, sessionId);
115}
116
117AudioTrack::~AudioTrack()
118{
119    LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
120
121    if (mStatus == NO_ERROR) {
122        // Make sure that callback function exits in the case where
123        // it is looping on buffer full condition in obtainBuffer().
124        // Otherwise the callback thread will never exit.
125        stop();
126        if (mAudioTrackThread != 0) {
127            mAudioTrackThread->requestExitAndWait();
128            mAudioTrackThread.clear();
129        }
130        mAudioTrack.clear();
131        IPCThreadState::self()->flushCommands();
132    }
133}
134
135status_t AudioTrack::set(
136        int streamType,
137        uint32_t sampleRate,
138        int format,
139        int channels,
140        int frameCount,
141        uint32_t flags,
142        callback_t cbf,
143        void* user,
144        int notificationFrames,
145        const sp<IMemory>& sharedBuffer,
146        bool threadCanCallJava,
147        int sessionId)
148{
149
150    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
151
152    if (mAudioTrack != 0) {
153        LOGE("Track already in use");
154        return INVALID_OPERATION;
155    }
156
157    int afSampleRate;
158    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
159        return NO_INIT;
160    }
161    uint32_t afLatency;
162    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
163        return NO_INIT;
164    }
165
166    // handle default values first.
167    if (streamType == AudioSystem::DEFAULT) {
168        streamType = AudioSystem::MUSIC;
169    }
170    if (sampleRate == 0) {
171        sampleRate = afSampleRate;
172    }
173    // these below should probably come from the audioFlinger too...
174    if (format == 0) {
175        format = AudioSystem::PCM_16_BIT;
176    }
177    if (channels == 0) {
178        channels = AudioSystem::CHANNEL_OUT_STEREO;
179    }
180
181    // validate parameters
182    if (!AudioSystem::isValidFormat(format)) {
183        LOGE("Invalid format");
184        return BAD_VALUE;
185    }
186
187    // force direct flag if format is not linear PCM
188    if (!AudioSystem::isLinearPCM(format)) {
189        flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
190    }
191
192    if (!AudioSystem::isOutputChannel(channels)) {
193        LOGE("Invalid channel mask");
194        return BAD_VALUE;
195    }
196    uint32_t channelCount = AudioSystem::popCount(channels);
197
198    audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
199            sampleRate, format, channels, (AudioSystem::output_flags)flags);
200
201    if (output == 0) {
202        LOGE("Could not get audio output for stream type %d", streamType);
203        return BAD_VALUE;
204    }
205
206    mVolume[LEFT] = 1.0f;
207    mVolume[RIGHT] = 1.0f;
208    mSendLevel = 0;
209    mFrameCount = frameCount;
210    mNotificationFramesReq = notificationFrames;
211    mSessionId = sessionId;
212    mAuxEffectId = 0;
213
214    // create the IAudioTrack
215    status_t status = createTrack(streamType, sampleRate, format, channelCount,
216                                  frameCount, flags, sharedBuffer, output, true);
217
218    if (status != NO_ERROR) {
219        return status;
220    }
221
222    if (cbf != 0) {
223        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
224        if (mAudioTrackThread == 0) {
225          LOGE("Could not create callback thread");
226          return NO_INIT;
227        }
228    }
229
230    mStatus = NO_ERROR;
231
232    mStreamType = streamType;
233    mFormat = format;
234    mChannels = channels;
235    mChannelCount = channelCount;
236    mSharedBuffer = sharedBuffer;
237    mMuted = false;
238    mActive = 0;
239    mCbf = cbf;
240    mUserData = user;
241    mLoopCount = 0;
242    mMarkerPosition = 0;
243    mMarkerReached = false;
244    mNewPosition = 0;
245    mUpdatePeriod = 0;
246    mFlags = flags;
247
248    return NO_ERROR;
249}
250
251status_t AudioTrack::initCheck() const
252{
253    return mStatus;
254}
255
256// -------------------------------------------------------------------------
257
258uint32_t AudioTrack::latency() const
259{
260    return mLatency;
261}
262
263int AudioTrack::streamType() const
264{
265    return mStreamType;
266}
267
268int AudioTrack::format() const
269{
270    return mFormat;
271}
272
273int AudioTrack::channelCount() const
274{
275    return mChannelCount;
276}
277
278uint32_t AudioTrack::frameCount() const
279{
280    return mCblk->frameCount;
281}
282
283int AudioTrack::frameSize() const
284{
285    if (AudioSystem::isLinearPCM(mFormat)) {
286        return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
287    } else {
288        return sizeof(uint8_t);
289    }
290}
291
292sp<IMemory>& AudioTrack::sharedBuffer()
293{
294    return mSharedBuffer;
295}
296
297// -------------------------------------------------------------------------
298
299void AudioTrack::start()
300{
301    sp<AudioTrackThread> t = mAudioTrackThread;
302    status_t status;
303
304    LOGV("start %p", this);
305    if (t != 0) {
306        if (t->exitPending()) {
307            if (t->requestExitAndWait() == WOULD_BLOCK) {
308                LOGE("AudioTrack::start called from thread");
309                return;
310            }
311        }
312        t->mLock.lock();
313     }
314
315    if (android_atomic_or(1, &mActive) == 0) {
316        mNewPosition = mCblk->server + mUpdatePeriod;
317        mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
318        mCblk->waitTimeMs = 0;
319        if (t != 0) {
320           t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
321        } else {
322            setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
323        }
324
325        if (mCblk->flags & CBLK_INVALID_MSK) {
326            LOGW("start() track %p invalidated, creating a new one", this);
327            // no need to clear the invalid flag as this cblk will not be used anymore
328            // force new track creation
329            status = DEAD_OBJECT;
330        } else {
331            status = mAudioTrack->start();
332        }
333        if (status == DEAD_OBJECT) {
334            LOGV("start() dead IAudioTrack: creating a new one");
335            status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
336                                 mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
337            if (status == NO_ERROR) {
338                status = mAudioTrack->start();
339                if (status == NO_ERROR) {
340                    mNewPosition = mCblk->server + mUpdatePeriod;
341                }
342            }
343        }
344        if (status != NO_ERROR) {
345            LOGV("start() failed");
346            android_atomic_and(~1, &mActive);
347            if (t != 0) {
348                t->requestExit();
349            } else {
350                setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
351            }
352        }
353    }
354
355    if (t != 0) {
356        t->mLock.unlock();
357    }
358}
359
360void AudioTrack::stop()
361{
362    sp<AudioTrackThread> t = mAudioTrackThread;
363
364    LOGV("stop %p", this);
365    if (t != 0) {
366        t->mLock.lock();
367    }
368
369    if (android_atomic_and(~1, &mActive) == 1) {
370        mCblk->cv.signal();
371        mAudioTrack->stop();
372        // Cancel loops (If we are in the middle of a loop, playback
373        // would not stop until loopCount reaches 0).
374        setLoop(0, 0, 0);
375        // the playback head position will reset to 0, so if a marker is set, we need
376        // to activate it again
377        mMarkerReached = false;
378        // Force flush if a shared buffer is used otherwise audioflinger
379        // will not stop before end of buffer is reached.
380        if (mSharedBuffer != 0) {
381            flush();
382        }
383        if (t != 0) {
384            t->requestExit();
385        } else {
386            setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
387        }
388    }
389
390    if (t != 0) {
391        t->mLock.unlock();
392    }
393}
394
395bool AudioTrack::stopped() const
396{
397    return !mActive;
398}
399
400void AudioTrack::flush()
401{
402    LOGV("flush");
403
404    // clear playback marker and periodic update counter
405    mMarkerPosition = 0;
406    mMarkerReached = false;
407    mUpdatePeriod = 0;
408
409
410    if (!mActive) {
411        mAudioTrack->flush();
412        // Release AudioTrack callback thread in case it was waiting for new buffers
413        // in AudioTrack::obtainBuffer()
414        mCblk->cv.signal();
415    }
416}
417
418void AudioTrack::pause()
419{
420    LOGV("pause");
421    if (android_atomic_and(~1, &mActive) == 1) {
422        mAudioTrack->pause();
423    }
424}
425
426void AudioTrack::mute(bool e)
427{
428    mAudioTrack->mute(e);
429    mMuted = e;
430}
431
432bool AudioTrack::muted() const
433{
434    return mMuted;
435}
436
437status_t AudioTrack::setVolume(float left, float right)
438{
439    if (left > 1.0f || right > 1.0f) {
440        return BAD_VALUE;
441    }
442
443    mVolume[LEFT] = left;
444    mVolume[RIGHT] = right;
445
446    // write must be atomic
447    mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
448
449    return NO_ERROR;
450}
451
452void AudioTrack::getVolume(float* left, float* right)
453{
454    if (left != NULL) {
455        *left  = mVolume[LEFT];
456    }
457    if (right != NULL) {
458        *right = mVolume[RIGHT];
459    }
460}
461
462status_t AudioTrack::setAuxEffectSendLevel(float level)
463{
464    LOGV("setAuxEffectSendLevel(%f)", level);
465    if (level > 1.0f) {
466        return BAD_VALUE;
467    }
468
469    mSendLevel = level;
470
471    mCblk->sendLevel = uint16_t(level * 0x1000);
472
473    return NO_ERROR;
474}
475
476void AudioTrack::getAuxEffectSendLevel(float* level)
477{
478    if (level != NULL) {
479        *level  = mSendLevel;
480    }
481}
482
483status_t AudioTrack::setSampleRate(int rate)
484{
485    int afSamplingRate;
486
487    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
488        return NO_INIT;
489    }
490    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
491    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
492
493    mCblk->sampleRate = rate;
494    return NO_ERROR;
495}
496
497uint32_t AudioTrack::getSampleRate()
498{
499    return mCblk->sampleRate;
500}
501
502status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
503{
504    audio_track_cblk_t* cblk = mCblk;
505
506    Mutex::Autolock _l(cblk->lock);
507
508    if (loopCount == 0) {
509        cblk->loopStart = UINT_MAX;
510        cblk->loopEnd = UINT_MAX;
511        cblk->loopCount = 0;
512        mLoopCount = 0;
513        return NO_ERROR;
514    }
515
516    if (loopStart >= loopEnd ||
517        loopEnd - loopStart > cblk->frameCount) {
518        LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
519        return BAD_VALUE;
520    }
521
522    if ((mSharedBuffer != 0) && (loopEnd   > cblk->frameCount)) {
523        LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
524            loopStart, loopEnd, cblk->frameCount);
525        return BAD_VALUE;
526    }
527
528    cblk->loopStart = loopStart;
529    cblk->loopEnd = loopEnd;
530    cblk->loopCount = loopCount;
531    mLoopCount = loopCount;
532
533    return NO_ERROR;
534}
535
536status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
537{
538    if (loopStart != 0) {
539        *loopStart = mCblk->loopStart;
540    }
541    if (loopEnd != 0) {
542        *loopEnd = mCblk->loopEnd;
543    }
544    if (loopCount != 0) {
545        if (mCblk->loopCount < 0) {
546            *loopCount = -1;
547        } else {
548            *loopCount = mCblk->loopCount;
549        }
550    }
551
552    return NO_ERROR;
553}
554
555status_t AudioTrack::setMarkerPosition(uint32_t marker)
556{
557    if (mCbf == 0) return INVALID_OPERATION;
558
559    mMarkerPosition = marker;
560    mMarkerReached = false;
561
562    return NO_ERROR;
563}
564
565status_t AudioTrack::getMarkerPosition(uint32_t *marker)
566{
567    if (marker == 0) return BAD_VALUE;
568
569    *marker = mMarkerPosition;
570
571    return NO_ERROR;
572}
573
574status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
575{
576    if (mCbf == 0) return INVALID_OPERATION;
577
578    uint32_t curPosition;
579    getPosition(&curPosition);
580    mNewPosition = curPosition + updatePeriod;
581    mUpdatePeriod = updatePeriod;
582
583    return NO_ERROR;
584}
585
586status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
587{
588    if (updatePeriod == 0) return BAD_VALUE;
589
590    *updatePeriod = mUpdatePeriod;
591
592    return NO_ERROR;
593}
594
595status_t AudioTrack::setPosition(uint32_t position)
596{
597    Mutex::Autolock _l(mCblk->lock);
598
599    if (!stopped()) return INVALID_OPERATION;
600
601    if (position > mCblk->user) return BAD_VALUE;
602
603    mCblk->server = position;
604    mCblk->flags |= CBLK_FORCEREADY_ON;
605
606    return NO_ERROR;
607}
608
609status_t AudioTrack::getPosition(uint32_t *position)
610{
611    if (position == 0) return BAD_VALUE;
612
613    *position = mCblk->server;
614
615    return NO_ERROR;
616}
617
618status_t AudioTrack::reload()
619{
620    if (!stopped()) return INVALID_OPERATION;
621
622    flush();
623
624    mCblk->stepUser(mCblk->frameCount);
625
626    return NO_ERROR;
627}
628
629audio_io_handle_t AudioTrack::getOutput()
630{
631    return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
632            mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
633}
634
635int AudioTrack::getSessionId()
636{
637    return mSessionId;
638}
639
640status_t AudioTrack::attachAuxEffect(int effectId)
641{
642    LOGV("attachAuxEffect(%d)", effectId);
643    status_t status = mAudioTrack->attachAuxEffect(effectId);
644    if (status == NO_ERROR) {
645        mAuxEffectId = effectId;
646    }
647    return status;
648}
649
650// -------------------------------------------------------------------------
651
652status_t AudioTrack::createTrack(
653        int streamType,
654        uint32_t sampleRate,
655        int format,
656        int channelCount,
657        int frameCount,
658        uint32_t flags,
659        const sp<IMemory>& sharedBuffer,
660        audio_io_handle_t output,
661        bool enforceFrameCount)
662{
663    status_t status;
664    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
665    if (audioFlinger == 0) {
666       LOGE("Could not get audioflinger");
667       return NO_INIT;
668    }
669
670    int afSampleRate;
671    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
672        return NO_INIT;
673    }
674    int afFrameCount;
675    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
676        return NO_INIT;
677    }
678    uint32_t afLatency;
679    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
680        return NO_INIT;
681    }
682
683    mNotificationFramesAct = mNotificationFramesReq;
684    if (!AudioSystem::isLinearPCM(format)) {
685        if (sharedBuffer != 0) {
686            frameCount = sharedBuffer->size();
687        }
688    } else {
689        // Ensure that buffer depth covers at least audio hardware latency
690        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
691        if (minBufCount < 2) minBufCount = 2;
692
693        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
694
695        if (sharedBuffer == 0) {
696            if (frameCount == 0) {
697                frameCount = minFrameCount;
698            }
699            if (mNotificationFramesAct == 0) {
700                mNotificationFramesAct = frameCount/2;
701            }
702            // Make sure that application is notified with sufficient margin
703            // before underrun
704            if (mNotificationFramesAct > (uint32_t)frameCount/2) {
705                mNotificationFramesAct = frameCount/2;
706            }
707            if (frameCount < minFrameCount) {
708                if (enforceFrameCount) {
709                    LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
710                    return BAD_VALUE;
711                } else {
712                    frameCount = minFrameCount;
713                }
714            }
715        } else {
716            // Ensure that buffer alignment matches channelcount
717            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
718                LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
719                return BAD_VALUE;
720            }
721            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
722        }
723    }
724
725    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
726                                                      streamType,
727                                                      sampleRate,
728                                                      format,
729                                                      channelCount,
730                                                      frameCount,
731                                                      ((uint16_t)flags) << 16,
732                                                      sharedBuffer,
733                                                      output,
734                                                      &mSessionId,
735                                                      &status);
736
737    if (track == 0) {
738        LOGE("AudioFlinger could not create track, status: %d", status);
739        return status;
740    }
741    sp<IMemory> cblk = track->getCblk();
742    if (cblk == 0) {
743        LOGE("Could not get control block");
744        return NO_INIT;
745    }
746    mAudioTrack.clear();
747    mAudioTrack = track;
748    mCblkMemory.clear();
749    mCblkMemory = cblk;
750    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
751    mCblk->flags |= CBLK_DIRECTION_OUT;
752    if (sharedBuffer == 0) {
753        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
754    } else {
755        mCblk->buffers = sharedBuffer->pointer();
756         // Force buffer full condition as data is already present in shared memory
757        mCblk->stepUser(mCblk->frameCount);
758    }
759
760    mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
761    mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
762    mAudioTrack->attachAuxEffect(mAuxEffectId);
763    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
764    mCblk->waitTimeMs = 0;
765    mRemainingFrames = mNotificationFramesAct;
766    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
767    return NO_ERROR;
768}
769
770status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
771{
772    int active;
773    status_t result;
774    audio_track_cblk_t* cblk = mCblk;
775    uint32_t framesReq = audioBuffer->frameCount;
776    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
777
778    audioBuffer->frameCount  = 0;
779    audioBuffer->size = 0;
780
781    uint32_t framesAvail = cblk->framesAvailable();
782
783    if (framesAvail == 0) {
784        cblk->lock.lock();
785        goto start_loop_here;
786        while (framesAvail == 0) {
787            active = mActive;
788            if (UNLIKELY(!active)) {
789                LOGV("Not active and NO_MORE_BUFFERS");
790                cblk->lock.unlock();
791                return NO_MORE_BUFFERS;
792            }
793            if (UNLIKELY(!waitCount)) {
794                cblk->lock.unlock();
795                return WOULD_BLOCK;
796            }
797            if (!(cblk->flags & CBLK_INVALID_MSK)) {
798                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
799            }
800            if (cblk->flags & CBLK_INVALID_MSK) {
801                LOGW("obtainBuffer() track %p invalidated, creating a new one", this);
802                // no need to clear the invalid flag as this cblk will not be used anymore
803                cblk->lock.unlock();
804                goto create_new_track;
805            }
806            if (__builtin_expect(result!=NO_ERROR, false)) {
807                cblk->waitTimeMs += waitTimeMs;
808                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
809                    // timing out when a loop has been set and we have already written upto loop end
810                    // is a normal condition: no need to wake AudioFlinger up.
811                    if (cblk->user < cblk->loopEnd) {
812                        LOGW(   "obtainBuffer timed out (is the CPU pegged?) %p "
813                                "user=%08x, server=%08x", this, cblk->user, cblk->server);
814                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
815                        cblk->lock.unlock();
816                        result = mAudioTrack->start();
817                        if (result == DEAD_OBJECT) {
818                            LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
819create_new_track:
820                            result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
821                                                 mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
822                            if (result == NO_ERROR) {
823                                cblk = mCblk;
824                                cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
825                                mAudioTrack->start();
826                            }
827                        }
828                        cblk->lock.lock();
829                    }
830                    cblk->waitTimeMs = 0;
831                }
832
833                if (--waitCount == 0) {
834                    cblk->lock.unlock();
835                    return TIMED_OUT;
836                }
837            }
838            // read the server count again
839        start_loop_here:
840            framesAvail = cblk->framesAvailable_l();
841        }
842        cblk->lock.unlock();
843    }
844
845    cblk->waitTimeMs = 0;
846
847    if (framesReq > framesAvail) {
848        framesReq = framesAvail;
849    }
850
851    uint32_t u = cblk->user;
852    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
853
854    if (u + framesReq > bufferEnd) {
855        framesReq = bufferEnd - u;
856    }
857
858    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
859    audioBuffer->channelCount = mChannelCount;
860    audioBuffer->frameCount = framesReq;
861    audioBuffer->size = framesReq * cblk->frameSize;
862    if (AudioSystem::isLinearPCM(mFormat)) {
863        audioBuffer->format = AudioSystem::PCM_16_BIT;
864    } else {
865        audioBuffer->format = mFormat;
866    }
867    audioBuffer->raw = (int8_t *)cblk->buffer(u);
868    active = mActive;
869    return active ? status_t(NO_ERROR) : status_t(STOPPED);
870}
871
872void AudioTrack::releaseBuffer(Buffer* audioBuffer)
873{
874    audio_track_cblk_t* cblk = mCblk;
875    cblk->stepUser(audioBuffer->frameCount);
876}
877
878// -------------------------------------------------------------------------
879
880ssize_t AudioTrack::write(const void* buffer, size_t userSize)
881{
882
883    if (mSharedBuffer != 0) return INVALID_OPERATION;
884
885    if (ssize_t(userSize) < 0) {
886        // sanity-check. user is most-likely passing an error code.
887        LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
888                buffer, userSize, userSize);
889        return BAD_VALUE;
890    }
891
892    LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
893
894    ssize_t written = 0;
895    const int8_t *src = (const int8_t *)buffer;
896    Buffer audioBuffer;
897
898    do {
899        audioBuffer.frameCount = userSize/frameSize();
900
901        // Calling obtainBuffer() with a negative wait count causes
902        // an (almost) infinite wait time.
903        status_t err = obtainBuffer(&audioBuffer, -1);
904        if (err < 0) {
905            // out of buffers, return #bytes written
906            if (err == status_t(NO_MORE_BUFFERS))
907                break;
908            return ssize_t(err);
909        }
910
911        size_t toWrite;
912
913        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
914            // Divide capacity by 2 to take expansion into account
915            toWrite = audioBuffer.size>>1;
916            // 8 to 16 bit conversion
917            int count = toWrite;
918            int16_t *dst = (int16_t *)(audioBuffer.i8);
919            while(count--) {
920                *dst++ = (int16_t)(*src++^0x80) << 8;
921            }
922        } else {
923            toWrite = audioBuffer.size;
924            memcpy(audioBuffer.i8, src, toWrite);
925            src += toWrite;
926        }
927        userSize -= toWrite;
928        written += toWrite;
929
930        releaseBuffer(&audioBuffer);
931    } while (userSize);
932
933    return written;
934}
935
936// -------------------------------------------------------------------------
937
938bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
939{
940    Buffer audioBuffer;
941    uint32_t frames;
942    size_t writtenSize;
943
944    // Manage underrun callback
945    if (mActive && (mCblk->framesReady() == 0)) {
946        LOGV("Underrun user: %x, server: %x, flags %04x", mCblk->user, mCblk->server, mCblk->flags);
947        if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) {
948            mCbf(EVENT_UNDERRUN, mUserData, 0);
949            if (mCblk->server == mCblk->frameCount) {
950                mCbf(EVENT_BUFFER_END, mUserData, 0);
951            }
952            mCblk->flags |= CBLK_UNDERRUN_ON;
953            if (mSharedBuffer != 0) return false;
954        }
955    }
956
957    // Manage loop end callback
958    while (mLoopCount > mCblk->loopCount) {
959        int loopCount = -1;
960        mLoopCount--;
961        if (mLoopCount >= 0) loopCount = mLoopCount;
962
963        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
964    }
965
966    // Manage marker callback
967    if (!mMarkerReached && (mMarkerPosition > 0)) {
968        if (mCblk->server >= mMarkerPosition) {
969            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
970            mMarkerReached = true;
971        }
972    }
973
974    // Manage new position callback
975    if (mUpdatePeriod > 0) {
976        while (mCblk->server >= mNewPosition) {
977            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
978            mNewPosition += mUpdatePeriod;
979        }
980    }
981
982    // If Shared buffer is used, no data is requested from client.
983    if (mSharedBuffer != 0) {
984        frames = 0;
985    } else {
986        frames = mRemainingFrames;
987    }
988
989    do {
990
991        audioBuffer.frameCount = frames;
992
993        // Calling obtainBuffer() with a wait count of 1
994        // limits wait time to WAIT_PERIOD_MS. This prevents from being
995        // stuck here not being able to handle timed events (position, markers, loops).
996        status_t err = obtainBuffer(&audioBuffer, 1);
997        if (err < NO_ERROR) {
998            if (err != TIMED_OUT) {
999                LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
1000                return false;
1001            }
1002            break;
1003        }
1004        if (err == status_t(STOPPED)) return false;
1005
1006        // Divide buffer size by 2 to take into account the expansion
1007        // due to 8 to 16 bit conversion: the callback must fill only half
1008        // of the destination buffer
1009        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
1010            audioBuffer.size >>= 1;
1011        }
1012
1013        size_t reqSize = audioBuffer.size;
1014        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1015        writtenSize = audioBuffer.size;
1016
1017        // Sanity check on returned size
1018        if (ssize_t(writtenSize) <= 0) {
1019            // The callback is done filling buffers
1020            // Keep this thread going to handle timed events and
1021            // still try to get more data in intervals of WAIT_PERIOD_MS
1022            // but don't just loop and block the CPU, so wait
1023            usleep(WAIT_PERIOD_MS*1000);
1024            break;
1025        }
1026        if (writtenSize > reqSize) writtenSize = reqSize;
1027
1028        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
1029            // 8 to 16 bit conversion
1030            const int8_t *src = audioBuffer.i8 + writtenSize-1;
1031            int count = writtenSize;
1032            int16_t *dst = audioBuffer.i16 + writtenSize-1;
1033            while(count--) {
1034                *dst-- = (int16_t)(*src--^0x80) << 8;
1035            }
1036            writtenSize <<= 1;
1037        }
1038
1039        audioBuffer.size = writtenSize;
1040        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
1041        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sampel size of
1042        // 16 bit.
1043        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
1044
1045        frames -= audioBuffer.frameCount;
1046
1047        releaseBuffer(&audioBuffer);
1048    }
1049    while (frames);
1050
1051    if (frames == 0) {
1052        mRemainingFrames = mNotificationFramesAct;
1053    } else {
1054        mRemainingFrames = frames;
1055    }
1056    return true;
1057}
1058
1059status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1060{
1061
1062    const size_t SIZE = 256;
1063    char buffer[SIZE];
1064    String8 result;
1065
1066    result.append(" AudioTrack::dump\n");
1067    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
1068    result.append(buffer);
1069    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
1070    result.append(buffer);
1071    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
1072    result.append(buffer);
1073    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1074    result.append(buffer);
1075    ::write(fd, result.string(), result.size());
1076    return NO_ERROR;
1077}
1078
1079// =========================================================================
1080
1081AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1082    : Thread(bCanCallJava), mReceiver(receiver)
1083{
1084}
1085
1086bool AudioTrack::AudioTrackThread::threadLoop()
1087{
1088    return mReceiver.processAudioBuffer(this);
1089}
1090
1091status_t AudioTrack::AudioTrackThread::readyToRun()
1092{
1093    return NO_ERROR;
1094}
1095
1096void AudioTrack::AudioTrackThread::onFirstRef()
1097{
1098}
1099
1100// =========================================================================
1101
1102audio_track_cblk_t::audio_track_cblk_t()
1103    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1104    userBase(0), serverBase(0), buffers(0), frameCount(0),
1105    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
1106    flags(0), sendLevel(0)
1107{
1108}
1109
1110uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1111{
1112    uint32_t u = this->user;
1113
1114    u += frameCount;
1115    // Ensure that user is never ahead of server for AudioRecord
1116    if (flags & CBLK_DIRECTION_MSK) {
1117        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1118        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1119            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1120        }
1121    } else if (u > this->server) {
1122        LOGW("stepServer occured after track reset");
1123        u = this->server;
1124    }
1125
1126    if (u >= userBase + this->frameCount) {
1127        userBase += this->frameCount;
1128    }
1129
1130    this->user = u;
1131
1132    // Clear flow control error condition as new data has been written/read to/from buffer.
1133    flags &= ~CBLK_UNDERRUN_MSK;
1134
1135    return u;
1136}
1137
1138bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1139{
1140    // the code below simulates lock-with-timeout
1141    // we MUST do this to protect the AudioFlinger server
1142    // as this lock is shared with the client.
1143    status_t err;
1144
1145    err = lock.tryLock();
1146    if (err == -EBUSY) { // just wait a bit
1147        usleep(1000);
1148        err = lock.tryLock();
1149    }
1150    if (err != NO_ERROR) {
1151        // probably, the client just died.
1152        return false;
1153    }
1154
1155    uint32_t s = this->server;
1156
1157    s += frameCount;
1158    if (flags & CBLK_DIRECTION_MSK) {
1159        // Mark that we have read the first buffer so that next time stepUser() is called
1160        // we switch to normal obtainBuffer() timeout period
1161        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1162            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1163        }
1164        // It is possible that we receive a flush()
1165        // while the mixer is processing a block: in this case,
1166        // stepServer() is called After the flush() has reset u & s and
1167        // we have s > u
1168        if (s > this->user) {
1169            LOGW("stepServer occured after track reset");
1170            s = this->user;
1171        }
1172    }
1173
1174    if (s >= loopEnd) {
1175        LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1176        s = loopStart;
1177        if (--loopCount == 0) {
1178            loopEnd = UINT_MAX;
1179            loopStart = UINT_MAX;
1180        }
1181    }
1182    if (s >= serverBase + this->frameCount) {
1183        serverBase += this->frameCount;
1184    }
1185
1186    this->server = s;
1187
1188    cv.signal();
1189    lock.unlock();
1190    return true;
1191}
1192
1193void* audio_track_cblk_t::buffer(uint32_t offset) const
1194{
1195    return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
1196}
1197
1198uint32_t audio_track_cblk_t::framesAvailable()
1199{
1200    Mutex::Autolock _l(lock);
1201    return framesAvailable_l();
1202}
1203
1204uint32_t audio_track_cblk_t::framesAvailable_l()
1205{
1206    uint32_t u = this->user;
1207    uint32_t s = this->server;
1208
1209    if (flags & CBLK_DIRECTION_MSK) {
1210        uint32_t limit = (s < loopStart) ? s : loopStart;
1211        return limit + frameCount - u;
1212    } else {
1213        return frameCount + u - s;
1214    }
1215}
1216
1217uint32_t audio_track_cblk_t::framesReady()
1218{
1219    uint32_t u = this->user;
1220    uint32_t s = this->server;
1221
1222    if (flags & CBLK_DIRECTION_MSK) {
1223        if (u < loopEnd) {
1224            return u - s;
1225        } else {
1226            Mutex::Autolock _l(lock);
1227            if (loopCount >= 0) {
1228                return (loopEnd - loopStart)*loopCount + u - s;
1229            } else {
1230                return UINT_MAX;
1231            }
1232        }
1233    } else {
1234        return s - u;
1235    }
1236}
1237
1238// -------------------------------------------------------------------------
1239
1240}; // namespace android
1241
1242