AudioTrack.cpp revision b1cf75c4935001f61057989ee3cf27bbf09ecd9c 
1/* frameworks/base/media/libmedia/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 <utils/Atomic.h>
39
40#include <cutils/bitops.h>
41#include <cutils/compiler.h>
42
43#include <system/audio.h>
44#include <system/audio_policy.h>
45
46namespace android {
47// ---------------------------------------------------------------------------
48
49// static
50status_t AudioTrack::getMinFrameCount(
51        int* frameCount,
52        audio_stream_type_t streamType,
53        uint32_t sampleRate)
54{
55    int afSampleRate;
56    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
57        return NO_INIT;
58    }
59    int afFrameCount;
60    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
61        return NO_INIT;
62    }
63    uint32_t afLatency;
64    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
65        return NO_INIT;
66    }
67
68    // Ensure that buffer depth covers at least audio hardware latency
69    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
70    if (minBufCount < 2) minBufCount = 2;
71
72    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
73              afFrameCount * minBufCount * sampleRate / afSampleRate;
74    return NO_ERROR;
75}
76
77// ---------------------------------------------------------------------------
78
79AudioTrack::AudioTrack()
80    : mStatus(NO_INIT),
81      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
82{
83}
84
85AudioTrack::AudioTrack(
86        audio_stream_type_t streamType,
87        uint32_t sampleRate,
88        audio_format_t format,
89        int channelMask,
90        int frameCount,
91        uint32_t flags,
92        callback_t cbf,
93        void* user,
94        int notificationFrames,
95        int sessionId)
96    : mStatus(NO_INIT),
97      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
98{
99    mStatus = set(streamType, sampleRate, format, channelMask,
100            frameCount, flags, cbf, user, notificationFrames,
101            0, false, sessionId);
102}
103
104AudioTrack::AudioTrack(
105        audio_stream_type_t streamType,
106        uint32_t sampleRate,
107        audio_format_t format,
108        int channelMask,
109        const sp<IMemory>& sharedBuffer,
110        uint32_t flags,
111        callback_t cbf,
112        void* user,
113        int notificationFrames,
114        int sessionId)
115    : mStatus(NO_INIT),
116      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
117{
118    mStatus = set(streamType, sampleRate, format, channelMask,
119            0, flags, cbf, user, notificationFrames,
120            sharedBuffer, false, sessionId);
121}
122
123AudioTrack::~AudioTrack()
124{
125    ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
126
127    if (mStatus == NO_ERROR) {
128        // Make sure that callback function exits in the case where
129        // it is looping on buffer full condition in obtainBuffer().
130        // Otherwise the callback thread will never exit.
131        stop();
132        if (mAudioTrackThread != 0) {
133            mAudioTrackThread->requestExitAndWait();
134            mAudioTrackThread.clear();
135        }
136        mAudioTrack.clear();
137        IPCThreadState::self()->flushCommands();
138        AudioSystem::releaseAudioSessionId(mSessionId);
139    }
140}
141
142status_t AudioTrack::set(
143        audio_stream_type_t streamType,
144        uint32_t sampleRate,
145        audio_format_t format,
146        int channelMask,
147        int frameCount,
148        uint32_t flags,
149        callback_t cbf,
150        void* user,
151        int notificationFrames,
152        const sp<IMemory>& sharedBuffer,
153        bool threadCanCallJava,
154        int sessionId)
155{
156
157    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
158
159    AutoMutex lock(mLock);
160    if (mAudioTrack != 0) {
161        ALOGE("Track already in use");
162        return INVALID_OPERATION;
163    }
164
165    int afSampleRate;
166    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
167        return NO_INIT;
168    }
169    uint32_t afLatency;
170    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
171        return NO_INIT;
172    }
173
174    // handle default values first.
175    if (streamType == AUDIO_STREAM_DEFAULT) {
176        streamType = AUDIO_STREAM_MUSIC;
177    }
178    if (sampleRate == 0) {
179        sampleRate = afSampleRate;
180    }
181    // these below should probably come from the audioFlinger too...
182    if (format == AUDIO_FORMAT_DEFAULT) {
183        format = AUDIO_FORMAT_PCM_16_BIT;
184    }
185    if (channelMask == 0) {
186        channelMask = AUDIO_CHANNEL_OUT_STEREO;
187    }
188
189    // validate parameters
190    if (!audio_is_valid_format(format)) {
191        ALOGE("Invalid format");
192        return BAD_VALUE;
193    }
194
195    // force direct flag if format is not linear PCM
196    if (!audio_is_linear_pcm(format)) {
197        flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT;
198    }
199
200    if (!audio_is_output_channel(channelMask)) {
201        ALOGE("Invalid channel mask");
202        return BAD_VALUE;
203    }
204    uint32_t channelCount = popcount(channelMask);
205
206    audio_io_handle_t output = AudioSystem::getOutput(
207                                    streamType,
208                                    sampleRate, format, channelMask,
209                                    (audio_policy_output_flags_t)flags);
210
211    if (output == 0) {
212        ALOGE("Could not get audio output for stream type %d", streamType);
213        return BAD_VALUE;
214    }
215
216    mVolume[LEFT] = 1.0f;
217    mVolume[RIGHT] = 1.0f;
218    mSendLevel = 0.0f;
219    mFrameCount = frameCount;
220    mNotificationFramesReq = notificationFrames;
221    mSessionId = sessionId;
222    mAuxEffectId = 0;
223
224    // create the IAudioTrack
225    status_t status = createTrack_l(streamType,
226                                  sampleRate,
227                                  format,
228                                  (uint32_t)channelMask,
229                                  frameCount,
230                                  flags,
231                                  sharedBuffer,
232                                  output,
233                                  true);
234
235    if (status != NO_ERROR) {
236        return status;
237    }
238
239    if (cbf != 0) {
240        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
241    }
242
243    mStatus = NO_ERROR;
244
245    mStreamType = streamType;
246    mFormat = format;
247    mChannelMask = (uint32_t)channelMask;
248    mChannelCount = channelCount;
249    mSharedBuffer = sharedBuffer;
250    mMuted = false;
251    mActive = false;
252    mCbf = cbf;
253    mUserData = user;
254    mLoopCount = 0;
255    mMarkerPosition = 0;
256    mMarkerReached = false;
257    mNewPosition = 0;
258    mUpdatePeriod = 0;
259    mFlushed = false;
260    mFlags = flags;
261    AudioSystem::acquireAudioSessionId(mSessionId);
262    mRestoreStatus = NO_ERROR;
263    return NO_ERROR;
264}
265
266status_t AudioTrack::initCheck() const
267{
268    return mStatus;
269}
270
271// -------------------------------------------------------------------------
272
273uint32_t AudioTrack::latency() const
274{
275    return mLatency;
276}
277
278audio_stream_type_t AudioTrack::streamType() const
279{
280    return mStreamType;
281}
282
283audio_format_t AudioTrack::format() const
284{
285    return mFormat;
286}
287
288int AudioTrack::channelCount() const
289{
290    return mChannelCount;
291}
292
293uint32_t AudioTrack::frameCount() const
294{
295    return mCblk->frameCount;
296}
297
298size_t AudioTrack::frameSize() const
299{
300    if (audio_is_linear_pcm(mFormat)) {
301        return channelCount()*audio_bytes_per_sample(mFormat);
302    } else {
303        return sizeof(uint8_t);
304    }
305}
306
307sp<IMemory>& AudioTrack::sharedBuffer()
308{
309    return mSharedBuffer;
310}
311
312// -------------------------------------------------------------------------
313
314void AudioTrack::start()
315{
316    sp<AudioTrackThread> t = mAudioTrackThread;
317    status_t status = NO_ERROR;
318
319    ALOGV("start %p", this);
320    if (t != 0) {
321        if (t->exitPending()) {
322            if (t->requestExitAndWait() == WOULD_BLOCK) {
323                ALOGE("AudioTrack::start called from thread");
324                return;
325            }
326        }
327        t->mLock.lock();
328     }
329
330    AutoMutex lock(mLock);
331    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
332    // while we are accessing the cblk
333    sp <IAudioTrack> audioTrack = mAudioTrack;
334    sp <IMemory> iMem = mCblkMemory;
335    audio_track_cblk_t* cblk = mCblk;
336
337    if (!mActive) {
338        mFlushed = false;
339        mActive = true;
340        mNewPosition = cblk->server + mUpdatePeriod;
341        cblk->lock.lock();
342        cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
343        cblk->waitTimeMs = 0;
344        android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
345        if (t != 0) {
346            t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO);
347        } else {
348            mPreviousPriority = getpriority(PRIO_PROCESS, 0);
349            mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
350            androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
351        }
352
353        ALOGV("start %p before lock cblk %p", this, mCblk);
354        if (!(cblk->flags & CBLK_INVALID_MSK)) {
355            cblk->lock.unlock();
356            status = mAudioTrack->start();
357            cblk->lock.lock();
358            if (status == DEAD_OBJECT) {
359                android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
360            }
361        }
362        if (cblk->flags & CBLK_INVALID_MSK) {
363            status = restoreTrack_l(cblk, true);
364        }
365        cblk->lock.unlock();
366        if (status != NO_ERROR) {
367            ALOGV("start() failed");
368            mActive = false;
369            if (t != 0) {
370                t->requestExit();
371            } else {
372                setpriority(PRIO_PROCESS, 0, mPreviousPriority);
373                androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
374            }
375        }
376    }
377
378    if (t != 0) {
379        t->mLock.unlock();
380    }
381}
382
383void AudioTrack::stop()
384{
385    sp<AudioTrackThread> t = mAudioTrackThread;
386
387    ALOGV("stop %p", this);
388    if (t != 0) {
389        t->mLock.lock();
390    }
391
392    AutoMutex lock(mLock);
393    if (mActive) {
394        mActive = false;
395        mCblk->cv.signal();
396        mAudioTrack->stop();
397        // Cancel loops (If we are in the middle of a loop, playback
398        // would not stop until loopCount reaches 0).
399        setLoop_l(0, 0, 0);
400        // the playback head position will reset to 0, so if a marker is set, we need
401        // to activate it again
402        mMarkerReached = false;
403        // Force flush if a shared buffer is used otherwise audioflinger
404        // will not stop before end of buffer is reached.
405        if (mSharedBuffer != 0) {
406            flush_l();
407        }
408        if (t != 0) {
409            t->requestExit();
410        } else {
411            setpriority(PRIO_PROCESS, 0, mPreviousPriority);
412            androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
413        }
414    }
415
416    if (t != 0) {
417        t->mLock.unlock();
418    }
419}
420
421bool AudioTrack::stopped() const
422{
423    AutoMutex lock(mLock);
424    return stopped_l();
425}
426
427void AudioTrack::flush()
428{
429    AutoMutex lock(mLock);
430    flush_l();
431}
432
433// must be called with mLock held
434void AudioTrack::flush_l()
435{
436    ALOGV("flush");
437
438    // clear playback marker and periodic update counter
439    mMarkerPosition = 0;
440    mMarkerReached = false;
441    mUpdatePeriod = 0;
442
443    if (!mActive) {
444        mFlushed = true;
445        mAudioTrack->flush();
446        // Release AudioTrack callback thread in case it was waiting for new buffers
447        // in AudioTrack::obtainBuffer()
448        mCblk->cv.signal();
449    }
450}
451
452void AudioTrack::pause()
453{
454    ALOGV("pause");
455    AutoMutex lock(mLock);
456    if (mActive) {
457        mActive = false;
458        mAudioTrack->pause();
459    }
460}
461
462void AudioTrack::mute(bool e)
463{
464    mAudioTrack->mute(e);
465    mMuted = e;
466}
467
468bool AudioTrack::muted() const
469{
470    return mMuted;
471}
472
473status_t AudioTrack::setVolume(float left, float right)
474{
475    if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
476        return BAD_VALUE;
477    }
478
479    AutoMutex lock(mLock);
480    mVolume[LEFT] = left;
481    mVolume[RIGHT] = right;
482
483    mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
484
485    return NO_ERROR;
486}
487
488void AudioTrack::getVolume(float* left, float* right)
489{
490    if (left != NULL) {
491        *left  = mVolume[LEFT];
492    }
493    if (right != NULL) {
494        *right = mVolume[RIGHT];
495    }
496}
497
498status_t AudioTrack::setAuxEffectSendLevel(float level)
499{
500    ALOGV("setAuxEffectSendLevel(%f)", level);
501    if (level < 0.0f || level > 1.0f) {
502        return BAD_VALUE;
503    }
504    AutoMutex lock(mLock);
505
506    mSendLevel = level;
507
508    mCblk->setSendLevel(level);
509
510    return NO_ERROR;
511}
512
513void AudioTrack::getAuxEffectSendLevel(float* level)
514{
515    if (level != NULL) {
516        *level  = mSendLevel;
517    }
518}
519
520status_t AudioTrack::setSampleRate(int rate)
521{
522    int afSamplingRate;
523
524    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
525        return NO_INIT;
526    }
527    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
528    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
529
530    AutoMutex lock(mLock);
531    mCblk->sampleRate = rate;
532    return NO_ERROR;
533}
534
535uint32_t AudioTrack::getSampleRate()
536{
537    AutoMutex lock(mLock);
538    return mCblk->sampleRate;
539}
540
541status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
542{
543    AutoMutex lock(mLock);
544    return setLoop_l(loopStart, loopEnd, loopCount);
545}
546
547// must be called with mLock held
548status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
549{
550    audio_track_cblk_t* cblk = mCblk;
551
552    Mutex::Autolock _l(cblk->lock);
553
554    if (loopCount == 0) {
555        cblk->loopStart = UINT_MAX;
556        cblk->loopEnd = UINT_MAX;
557        cblk->loopCount = 0;
558        mLoopCount = 0;
559        return NO_ERROR;
560    }
561
562    if (loopStart >= loopEnd ||
563        loopEnd - loopStart > cblk->frameCount ||
564        cblk->server > loopStart) {
565        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
566        return BAD_VALUE;
567    }
568
569    if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
570        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
571            loopStart, loopEnd, cblk->frameCount);
572        return BAD_VALUE;
573    }
574
575    cblk->loopStart = loopStart;
576    cblk->loopEnd = loopEnd;
577    cblk->loopCount = loopCount;
578    mLoopCount = loopCount;
579
580    return NO_ERROR;
581}
582
583status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
584{
585    AutoMutex lock(mLock);
586    if (loopStart != 0) {
587        *loopStart = mCblk->loopStart;
588    }
589    if (loopEnd != 0) {
590        *loopEnd = mCblk->loopEnd;
591    }
592    if (loopCount != 0) {
593        if (mCblk->loopCount < 0) {
594            *loopCount = -1;
595        } else {
596            *loopCount = mCblk->loopCount;
597        }
598    }
599
600    return NO_ERROR;
601}
602
603status_t AudioTrack::setMarkerPosition(uint32_t marker)
604{
605    if (mCbf == 0) return INVALID_OPERATION;
606
607    mMarkerPosition = marker;
608    mMarkerReached = false;
609
610    return NO_ERROR;
611}
612
613status_t AudioTrack::getMarkerPosition(uint32_t *marker)
614{
615    if (marker == 0) return BAD_VALUE;
616
617    *marker = mMarkerPosition;
618
619    return NO_ERROR;
620}
621
622status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
623{
624    if (mCbf == 0) return INVALID_OPERATION;
625
626    uint32_t curPosition;
627    getPosition(&curPosition);
628    mNewPosition = curPosition + updatePeriod;
629    mUpdatePeriod = updatePeriod;
630
631    return NO_ERROR;
632}
633
634status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
635{
636    if (updatePeriod == 0) return BAD_VALUE;
637
638    *updatePeriod = mUpdatePeriod;
639
640    return NO_ERROR;
641}
642
643status_t AudioTrack::setPosition(uint32_t position)
644{
645    AutoMutex lock(mLock);
646
647    if (!stopped_l()) return INVALID_OPERATION;
648
649    Mutex::Autolock _l(mCblk->lock);
650
651    if (position > mCblk->user) return BAD_VALUE;
652
653    mCblk->server = position;
654    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
655
656    return NO_ERROR;
657}
658
659status_t AudioTrack::getPosition(uint32_t *position)
660{
661    if (position == 0) return BAD_VALUE;
662    AutoMutex lock(mLock);
663    *position = mFlushed ? 0 : mCblk->server;
664
665    return NO_ERROR;
666}
667
668status_t AudioTrack::reload()
669{
670    AutoMutex lock(mLock);
671
672    if (!stopped_l()) return INVALID_OPERATION;
673
674    flush_l();
675
676    mCblk->stepUser(mCblk->frameCount);
677
678    return NO_ERROR;
679}
680
681audio_io_handle_t AudioTrack::getOutput()
682{
683    AutoMutex lock(mLock);
684    return getOutput_l();
685}
686
687// must be called with mLock held
688audio_io_handle_t AudioTrack::getOutput_l()
689{
690    return AudioSystem::getOutput(mStreamType,
691            mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
692}
693
694int AudioTrack::getSessionId()
695{
696    return mSessionId;
697}
698
699status_t AudioTrack::attachAuxEffect(int effectId)
700{
701    ALOGV("attachAuxEffect(%d)", effectId);
702    status_t status = mAudioTrack->attachAuxEffect(effectId);
703    if (status == NO_ERROR) {
704        mAuxEffectId = effectId;
705    }
706    return status;
707}
708
709// -------------------------------------------------------------------------
710
711// must be called with mLock held
712status_t AudioTrack::createTrack_l(
713        audio_stream_type_t streamType,
714        uint32_t sampleRate,
715        audio_format_t format,
716        uint32_t channelMask,
717        int frameCount,
718        uint32_t flags,
719        const sp<IMemory>& sharedBuffer,
720        audio_io_handle_t output,
721        bool enforceFrameCount)
722{
723    status_t status;
724    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
725    if (audioFlinger == 0) {
726       ALOGE("Could not get audioflinger");
727       return NO_INIT;
728    }
729
730    int afSampleRate;
731    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
732        return NO_INIT;
733    }
734    int afFrameCount;
735    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
736        return NO_INIT;
737    }
738    uint32_t afLatency;
739    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
740        return NO_INIT;
741    }
742
743    mNotificationFramesAct = mNotificationFramesReq;
744    if (!audio_is_linear_pcm(format)) {
745        if (sharedBuffer != 0) {
746            frameCount = sharedBuffer->size();
747        }
748    } else {
749        // Ensure that buffer depth covers at least audio hardware latency
750        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
751        if (minBufCount < 2) minBufCount = 2;
752
753        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
754
755        if (sharedBuffer == 0) {
756            if (frameCount == 0) {
757                frameCount = minFrameCount;
758            }
759            if (mNotificationFramesAct == 0) {
760                mNotificationFramesAct = frameCount/2;
761            }
762            // Make sure that application is notified with sufficient margin
763            // before underrun
764            if (mNotificationFramesAct > (uint32_t)frameCount/2) {
765                mNotificationFramesAct = frameCount/2;
766            }
767            if (frameCount < minFrameCount) {
768                if (enforceFrameCount) {
769                    ALOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
770                    return BAD_VALUE;
771                } else {
772                    frameCount = minFrameCount;
773                }
774            }
775        } else {
776            // Ensure that buffer alignment matches channelcount
777            int channelCount = popcount(channelMask);
778            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
779                ALOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
780                return BAD_VALUE;
781            }
782            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
783        }
784    }
785
786    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
787                                                      streamType,
788                                                      sampleRate,
789                                                      format,
790                                                      channelMask,
791                                                      frameCount,
792                                                      ((uint16_t)flags) << 16,
793                                                      sharedBuffer,
794                                                      output,
795                                                      &mSessionId,
796                                                      &status);
797
798    if (track == 0) {
799        ALOGE("AudioFlinger could not create track, status: %d", status);
800        return status;
801    }
802    sp<IMemory> cblk = track->getCblk();
803    if (cblk == 0) {
804        ALOGE("Could not get control block");
805        return NO_INIT;
806    }
807    mAudioTrack = track;
808    mCblkMemory = cblk;
809    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
810    android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
811    if (sharedBuffer == 0) {
812        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
813    } else {
814        mCblk->buffers = sharedBuffer->pointer();
815         // Force buffer full condition as data is already present in shared memory
816        mCblk->stepUser(mCblk->frameCount);
817    }
818
819    mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
820    mCblk->setSendLevel(mSendLevel);
821    mAudioTrack->attachAuxEffect(mAuxEffectId);
822    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
823    mCblk->waitTimeMs = 0;
824    mRemainingFrames = mNotificationFramesAct;
825    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
826    return NO_ERROR;
827}
828
829status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
830{
831    AutoMutex lock(mLock);
832    bool active;
833    status_t result = NO_ERROR;
834    audio_track_cblk_t* cblk = mCblk;
835    uint32_t framesReq = audioBuffer->frameCount;
836    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
837
838    audioBuffer->frameCount  = 0;
839    audioBuffer->size = 0;
840
841    uint32_t framesAvail = cblk->framesAvailable();
842
843    cblk->lock.lock();
844    if (cblk->flags & CBLK_INVALID_MSK) {
845        goto create_new_track;
846    }
847    cblk->lock.unlock();
848
849    if (framesAvail == 0) {
850        cblk->lock.lock();
851        goto start_loop_here;
852        while (framesAvail == 0) {
853            active = mActive;
854            if (CC_UNLIKELY(!active)) {
855                ALOGV("Not active and NO_MORE_BUFFERS");
856                cblk->lock.unlock();
857                return NO_MORE_BUFFERS;
858            }
859            if (CC_UNLIKELY(!waitCount)) {
860                cblk->lock.unlock();
861                return WOULD_BLOCK;
862            }
863            if (!(cblk->flags & CBLK_INVALID_MSK)) {
864                mLock.unlock();
865                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
866                cblk->lock.unlock();
867                mLock.lock();
868                if (!mActive) {
869                    return status_t(STOPPED);
870                }
871                cblk->lock.lock();
872            }
873
874            if (cblk->flags & CBLK_INVALID_MSK) {
875                goto create_new_track;
876            }
877            if (CC_UNLIKELY(result != NO_ERROR)) {
878                cblk->waitTimeMs += waitTimeMs;
879                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
880                    // timing out when a loop has been set and we have already written upto loop end
881                    // is a normal condition: no need to wake AudioFlinger up.
882                    if (cblk->user < cblk->loopEnd) {
883                        ALOGW(   "obtainBuffer timed out (is the CPU pegged?) %p "
884                                "user=%08x, server=%08x", this, cblk->user, cblk->server);
885                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
886                        cblk->lock.unlock();
887                        result = mAudioTrack->start();
888                        cblk->lock.lock();
889                        if (result == DEAD_OBJECT) {
890                            android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
891create_new_track:
892                            result = restoreTrack_l(cblk, false);
893                        }
894                        if (result != NO_ERROR) {
895                            ALOGW("obtainBuffer create Track error %d", result);
896                            cblk->lock.unlock();
897                            return result;
898                        }
899                    }
900                    cblk->waitTimeMs = 0;
901                }
902
903                if (--waitCount == 0) {
904                    cblk->lock.unlock();
905                    return TIMED_OUT;
906                }
907            }
908            // read the server count again
909        start_loop_here:
910            framesAvail = cblk->framesAvailable_l();
911        }
912        cblk->lock.unlock();
913    }
914
915    // restart track if it was disabled by audioflinger due to previous underrun
916    if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) {
917        android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
918        ALOGW("obtainBuffer() track %p disabled, restarting", this);
919        mAudioTrack->start();
920    }
921
922    cblk->waitTimeMs = 0;
923
924    if (framesReq > framesAvail) {
925        framesReq = framesAvail;
926    }
927
928    uint32_t u = cblk->user;
929    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
930
931    if (u + framesReq > bufferEnd) {
932        framesReq = bufferEnd - u;
933    }
934
935    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
936    audioBuffer->channelCount = mChannelCount;
937    audioBuffer->frameCount = framesReq;
938    audioBuffer->size = framesReq * cblk->frameSize;
939    if (audio_is_linear_pcm(mFormat)) {
940        audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
941    } else {
942        audioBuffer->format = mFormat;
943    }
944    audioBuffer->raw = (int8_t *)cblk->buffer(u);
945    active = mActive;
946    return active ? status_t(NO_ERROR) : status_t(STOPPED);
947}
948
949void AudioTrack::releaseBuffer(Buffer* audioBuffer)
950{
951    AutoMutex lock(mLock);
952    mCblk->stepUser(audioBuffer->frameCount);
953}
954
955// -------------------------------------------------------------------------
956
957ssize_t AudioTrack::write(const void* buffer, size_t userSize)
958{
959
960    if (mSharedBuffer != 0) return INVALID_OPERATION;
961
962    if (ssize_t(userSize) < 0) {
963        // sanity-check. user is most-likely passing an error code.
964        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
965                buffer, userSize, userSize);
966        return BAD_VALUE;
967    }
968
969    ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
970
971    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
972    // while we are accessing the cblk
973    mLock.lock();
974    sp <IAudioTrack> audioTrack = mAudioTrack;
975    sp <IMemory> iMem = mCblkMemory;
976    mLock.unlock();
977
978    ssize_t written = 0;
979    const int8_t *src = (const int8_t *)buffer;
980    Buffer audioBuffer;
981    size_t frameSz = frameSize();
982
983    do {
984        audioBuffer.frameCount = userSize/frameSz;
985
986        // Calling obtainBuffer() with a negative wait count causes
987        // an (almost) infinite wait time.
988        status_t err = obtainBuffer(&audioBuffer, -1);
989        if (err < 0) {
990            // out of buffers, return #bytes written
991            if (err == status_t(NO_MORE_BUFFERS))
992                break;
993            return ssize_t(err);
994        }
995
996        size_t toWrite;
997
998        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
999            // Divide capacity by 2 to take expansion into account
1000            toWrite = audioBuffer.size>>1;
1001            // 8 to 16 bit conversion
1002            int count = toWrite;
1003            int16_t *dst = (int16_t *)(audioBuffer.i8);
1004            while(count--) {
1005                *dst++ = (int16_t)(*src++^0x80) << 8;
1006            }
1007        } else {
1008            toWrite = audioBuffer.size;
1009            memcpy(audioBuffer.i8, src, toWrite);
1010            src += toWrite;
1011        }
1012        userSize -= toWrite;
1013        written += toWrite;
1014
1015        releaseBuffer(&audioBuffer);
1016    } while (userSize >= frameSz);
1017
1018    return written;
1019}
1020
1021// -------------------------------------------------------------------------
1022
1023bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
1024{
1025    Buffer audioBuffer;
1026    uint32_t frames;
1027    size_t writtenSize;
1028
1029    mLock.lock();
1030    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1031    // while we are accessing the cblk
1032    sp <IAudioTrack> audioTrack = mAudioTrack;
1033    sp <IMemory> iMem = mCblkMemory;
1034    audio_track_cblk_t* cblk = mCblk;
1035    bool active = mActive;
1036    mLock.unlock();
1037
1038    // Manage underrun callback
1039    if (active && (cblk->framesAvailable() == cblk->frameCount)) {
1040        ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
1041        if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
1042            mCbf(EVENT_UNDERRUN, mUserData, 0);
1043            if (cblk->server == cblk->frameCount) {
1044                mCbf(EVENT_BUFFER_END, mUserData, 0);
1045            }
1046            if (mSharedBuffer != 0) return false;
1047        }
1048    }
1049
1050    // Manage loop end callback
1051    while (mLoopCount > cblk->loopCount) {
1052        int loopCount = -1;
1053        mLoopCount--;
1054        if (mLoopCount >= 0) loopCount = mLoopCount;
1055
1056        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
1057    }
1058
1059    // Manage marker callback
1060    if (!mMarkerReached && (mMarkerPosition > 0)) {
1061        if (cblk->server >= mMarkerPosition) {
1062            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
1063            mMarkerReached = true;
1064        }
1065    }
1066
1067    // Manage new position callback
1068    if (mUpdatePeriod > 0) {
1069        while (cblk->server >= mNewPosition) {
1070            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
1071            mNewPosition += mUpdatePeriod;
1072        }
1073    }
1074
1075    // If Shared buffer is used, no data is requested from client.
1076    if (mSharedBuffer != 0) {
1077        frames = 0;
1078    } else {
1079        frames = mRemainingFrames;
1080    }
1081
1082    int32_t waitCount = -1;
1083    if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
1084        waitCount = 1;
1085    }
1086
1087    do {
1088
1089        audioBuffer.frameCount = frames;
1090
1091        // Calling obtainBuffer() with a wait count of 1
1092        // limits wait time to WAIT_PERIOD_MS. This prevents from being
1093        // stuck here not being able to handle timed events (position, markers, loops).
1094        status_t err = obtainBuffer(&audioBuffer, waitCount);
1095        if (err < NO_ERROR) {
1096            if (err != TIMED_OUT) {
1097                ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
1098                return false;
1099            }
1100            break;
1101        }
1102        if (err == status_t(STOPPED)) return false;
1103
1104        // Divide buffer size by 2 to take into account the expansion
1105        // due to 8 to 16 bit conversion: the callback must fill only half
1106        // of the destination buffer
1107        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
1108            audioBuffer.size >>= 1;
1109        }
1110
1111        size_t reqSize = audioBuffer.size;
1112        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1113        writtenSize = audioBuffer.size;
1114
1115        // Sanity check on returned size
1116        if (ssize_t(writtenSize) <= 0) {
1117            // The callback is done filling buffers
1118            // Keep this thread going to handle timed events and
1119            // still try to get more data in intervals of WAIT_PERIOD_MS
1120            // but don't just loop and block the CPU, so wait
1121            usleep(WAIT_PERIOD_MS*1000);
1122            break;
1123        }
1124        if (writtenSize > reqSize) writtenSize = reqSize;
1125
1126        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
1127            // 8 to 16 bit conversion
1128            const int8_t *src = audioBuffer.i8 + writtenSize-1;
1129            int count = writtenSize;
1130            int16_t *dst = audioBuffer.i16 + writtenSize-1;
1131            while(count--) {
1132                *dst-- = (int16_t)(*src--^0x80) << 8;
1133            }
1134            writtenSize <<= 1;
1135        }
1136
1137        audioBuffer.size = writtenSize;
1138        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
1139        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
1140        // 16 bit.
1141        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
1142
1143        frames -= audioBuffer.frameCount;
1144
1145        releaseBuffer(&audioBuffer);
1146    }
1147    while (frames);
1148
1149    if (frames == 0) {
1150        mRemainingFrames = mNotificationFramesAct;
1151    } else {
1152        mRemainingFrames = frames;
1153    }
1154    return true;
1155}
1156
1157// must be called with mLock and cblk.lock held. Callers must also hold strong references on
1158// the IAudioTrack and IMemory in case they are recreated here.
1159// If the IAudioTrack is successfully restored, the cblk pointer is updated
1160status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
1161{
1162    status_t result;
1163
1164    if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
1165        ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
1166             fromStart ? "start()" : "obtainBuffer()", gettid());
1167
1168        // signal old cblk condition so that other threads waiting for available buffers stop
1169        // waiting now
1170        cblk->cv.broadcast();
1171        cblk->lock.unlock();
1172
1173        // refresh the audio configuration cache in this process to make sure we get new
1174        // output parameters in getOutput_l() and createTrack_l()
1175        AudioSystem::clearAudioConfigCache();
1176
1177        // if the new IAudioTrack is created, createTrack_l() will modify the
1178        // following member variables: mAudioTrack, mCblkMemory and mCblk.
1179        // It will also delete the strong references on previous IAudioTrack and IMemory
1180        result = createTrack_l(mStreamType,
1181                               cblk->sampleRate,
1182                               mFormat,
1183                               mChannelMask,
1184                               mFrameCount,
1185                               mFlags,
1186                               mSharedBuffer,
1187                               getOutput_l(),
1188                               false);
1189
1190        if (result == NO_ERROR) {
1191            uint32_t user = cblk->user;
1192            uint32_t server = cblk->server;
1193            // restore write index and set other indexes to reflect empty buffer status
1194            mCblk->user = user;
1195            mCblk->server = user;
1196            mCblk->userBase = user;
1197            mCblk->serverBase = user;
1198            // restore loop: this is not guaranteed to succeed if new frame count is not
1199            // compatible with loop length
1200            setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
1201            if (!fromStart) {
1202                mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1203                // Make sure that a client relying on callback events indicating underrun or
1204                // the actual amount of audio frames played (e.g SoundPool) receives them.
1205                if (mSharedBuffer == 0) {
1206                    uint32_t frames = 0;
1207                    if (user > server) {
1208                        frames = ((user - server) > mCblk->frameCount) ?
1209                                mCblk->frameCount : (user - server);
1210                        memset(mCblk->buffers, 0, frames * mCblk->frameSize);
1211                    }
1212                    // restart playback even if buffer is not completely filled.
1213                    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
1214                    // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
1215                    // the client
1216                    mCblk->stepUser(frames);
1217                }
1218            }
1219            if (mActive) {
1220                result = mAudioTrack->start();
1221                ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
1222            }
1223            if (fromStart && result == NO_ERROR) {
1224                mNewPosition = mCblk->server + mUpdatePeriod;
1225            }
1226        }
1227        if (result != NO_ERROR) {
1228            android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
1229            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
1230        }
1231        mRestoreStatus = result;
1232        // signal old cblk condition for other threads waiting for restore completion
1233        android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
1234        cblk->cv.broadcast();
1235    } else {
1236        if (!(cblk->flags & CBLK_RESTORED_MSK)) {
1237            ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid());
1238            mLock.unlock();
1239            result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
1240            if (result == NO_ERROR) {
1241                result = mRestoreStatus;
1242            }
1243            cblk->lock.unlock();
1244            mLock.lock();
1245        } else {
1246            ALOGW("dead IAudioTrack, already restored TID %d", gettid());
1247            result = mRestoreStatus;
1248            cblk->lock.unlock();
1249        }
1250    }
1251    ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
1252         result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
1253
1254    if (result == NO_ERROR) {
1255        // from now on we switch to the newly created cblk
1256        cblk = mCblk;
1257    }
1258    cblk->lock.lock();
1259
1260    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
1261
1262    return result;
1263}
1264
1265status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1266{
1267
1268    const size_t SIZE = 256;
1269    char buffer[SIZE];
1270    String8 result;
1271
1272    result.append(" AudioTrack::dump\n");
1273    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
1274    result.append(buffer);
1275    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
1276    result.append(buffer);
1277    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
1278    result.append(buffer);
1279    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1280    result.append(buffer);
1281    ::write(fd, result.string(), result.size());
1282    return NO_ERROR;
1283}
1284
1285// =========================================================================
1286
1287AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1288    : Thread(bCanCallJava), mReceiver(receiver)
1289{
1290}
1291
1292bool AudioTrack::AudioTrackThread::threadLoop()
1293{
1294    return mReceiver.processAudioBuffer(this);
1295}
1296
1297status_t AudioTrack::AudioTrackThread::readyToRun()
1298{
1299    return NO_ERROR;
1300}
1301
1302void AudioTrack::AudioTrackThread::onFirstRef()
1303{
1304}
1305
1306// =========================================================================
1307
1308
1309audio_track_cblk_t::audio_track_cblk_t()
1310    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1311    userBase(0), serverBase(0), buffers(0), frameCount(0),
1312    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
1313    mSendLevel(0), flags(0)
1314{
1315}
1316
1317uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1318{
1319    uint32_t u = user;
1320
1321    u += frameCount;
1322    // Ensure that user is never ahead of server for AudioRecord
1323    if (flags & CBLK_DIRECTION_MSK) {
1324        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1325        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1326            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1327        }
1328    } else if (u > server) {
1329        ALOGW("stepServer occurred after track reset");
1330        u = server;
1331    }
1332
1333    if (u >= userBase + this->frameCount) {
1334        userBase += this->frameCount;
1335    }
1336
1337    user = u;
1338
1339    // Clear flow control error condition as new data has been written/read to/from buffer.
1340    if (flags & CBLK_UNDERRUN_MSK) {
1341        android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
1342    }
1343
1344    return u;
1345}
1346
1347bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1348{
1349    if (!tryLock()) {
1350        ALOGW("stepServer() could not lock cblk");
1351        return false;
1352    }
1353
1354    uint32_t s = server;
1355
1356    s += frameCount;
1357    if (flags & CBLK_DIRECTION_MSK) {
1358        // Mark that we have read the first buffer so that next time stepUser() is called
1359        // we switch to normal obtainBuffer() timeout period
1360        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1361            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1362        }
1363        // It is possible that we receive a flush()
1364        // while the mixer is processing a block: in this case,
1365        // stepServer() is called After the flush() has reset u & s and
1366        // we have s > u
1367        if (s > user) {
1368            ALOGW("stepServer occurred after track reset");
1369            s = user;
1370        }
1371    }
1372
1373    if (s >= loopEnd) {
1374        ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1375        s = loopStart;
1376        if (--loopCount == 0) {
1377            loopEnd = UINT_MAX;
1378            loopStart = UINT_MAX;
1379        }
1380    }
1381    if (s >= serverBase + this->frameCount) {
1382        serverBase += this->frameCount;
1383    }
1384
1385    server = s;
1386
1387    if (!(flags & CBLK_INVALID_MSK)) {
1388        cv.signal();
1389    }
1390    lock.unlock();
1391    return true;
1392}
1393
1394void* audio_track_cblk_t::buffer(uint32_t offset) const
1395{
1396    return (int8_t *)buffers + (offset - userBase) * frameSize;
1397}
1398
1399uint32_t audio_track_cblk_t::framesAvailable()
1400{
1401    Mutex::Autolock _l(lock);
1402    return framesAvailable_l();
1403}
1404
1405uint32_t audio_track_cblk_t::framesAvailable_l()
1406{
1407    uint32_t u = user;
1408    uint32_t s = server;
1409
1410    if (flags & CBLK_DIRECTION_MSK) {
1411        uint32_t limit = (s < loopStart) ? s : loopStart;
1412        return limit + frameCount - u;
1413    } else {
1414        return frameCount + u - s;
1415    }
1416}
1417
1418uint32_t audio_track_cblk_t::framesReady()
1419{
1420    uint32_t u = user;
1421    uint32_t s = server;
1422
1423    if (flags & CBLK_DIRECTION_MSK) {
1424        if (u < loopEnd) {
1425            return u - s;
1426        } else {
1427            // do not block on mutex shared with client on AudioFlinger side
1428            if (!tryLock()) {
1429                ALOGW("framesReady() could not lock cblk");
1430                return 0;
1431            }
1432            uint32_t frames = UINT_MAX;
1433            if (loopCount >= 0) {
1434                frames = (loopEnd - loopStart)*loopCount + u - s;
1435            }
1436            lock.unlock();
1437            return frames;
1438        }
1439    } else {
1440        return s - u;
1441    }
1442}
1443
1444bool audio_track_cblk_t::tryLock()
1445{
1446    // the code below simulates lock-with-timeout
1447    // we MUST do this to protect the AudioFlinger server
1448    // as this lock is shared with the client.
1449    status_t err;
1450
1451    err = lock.tryLock();
1452    if (err == -EBUSY) { // just wait a bit
1453        usleep(1000);
1454        err = lock.tryLock();
1455    }
1456    if (err != NO_ERROR) {
1457        // probably, the client just died.
1458        return false;
1459    }
1460    return true;
1461}
1462
1463// -------------------------------------------------------------------------
1464
1465}; // namespace android
1466