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1 /*
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 
46 #include <audio_utils/primitives.h>
47 
48 namespace android {
49 // ---------------------------------------------------------------------------
50 
51 // static
getMinFrameCount(int * frameCount,audio_stream_type_t streamType,uint32_t sampleRate)52 status_t AudioTrack::getMinFrameCount(
53         int* frameCount,
54         audio_stream_type_t streamType,
55         uint32_t sampleRate)
56 {
57     // FIXME merge with similar code in createTrack_l(), except we're missing
58     //       some information here that is available in createTrack_l():
59     //          audio_io_handle_t output
60     //          audio_format_t format
61     //          audio_channel_mask_t channelMask
62     //          audio_output_flags_t flags
63     int afSampleRate;
64     if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
65         return NO_INIT;
66     }
67     int afFrameCount;
68     if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
69         return NO_INIT;
70     }
71     uint32_t afLatency;
72     if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
73         return NO_INIT;
74     }
75 
76     // Ensure that buffer depth covers at least audio hardware latency
77     uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
78     if (minBufCount < 2) minBufCount = 2;
79 
80     *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
81             afFrameCount * minBufCount * sampleRate / afSampleRate;
82     ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
83             *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
84     return NO_ERROR;
85 }
86 
87 // ---------------------------------------------------------------------------
88 
AudioTrack()89 AudioTrack::AudioTrack()
90     : mStatus(NO_INIT),
91       mIsTimed(false),
92       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
93       mPreviousSchedulingGroup(SP_DEFAULT)
94 {
95 }
96 
AudioTrack(audio_stream_type_t streamType,uint32_t sampleRate,audio_format_t format,int channelMask,int frameCount,audio_output_flags_t flags,callback_t cbf,void * user,int notificationFrames,int sessionId)97 AudioTrack::AudioTrack(
98         audio_stream_type_t streamType,
99         uint32_t sampleRate,
100         audio_format_t format,
101         int channelMask,
102         int frameCount,
103         audio_output_flags_t flags,
104         callback_t cbf,
105         void* user,
106         int notificationFrames,
107         int sessionId)
108     : mStatus(NO_INIT),
109       mIsTimed(false),
110       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
111       mPreviousSchedulingGroup(SP_DEFAULT)
112 {
113     mStatus = set(streamType, sampleRate, format, channelMask,
114             frameCount, flags, cbf, user, notificationFrames,
115             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
116 }
117 
118 // DEPRECATED
AudioTrack(int streamType,uint32_t sampleRate,int format,int channelMask,int frameCount,uint32_t flags,callback_t cbf,void * user,int notificationFrames,int sessionId)119 AudioTrack::AudioTrack(
120         int streamType,
121         uint32_t sampleRate,
122         int format,
123         int channelMask,
124         int frameCount,
125         uint32_t flags,
126         callback_t cbf,
127         void* user,
128         int notificationFrames,
129         int sessionId)
130     : mStatus(NO_INIT),
131       mIsTimed(false),
132       mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
133 {
134     mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
135             frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
136             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
137 }
138 
AudioTrack(audio_stream_type_t streamType,uint32_t sampleRate,audio_format_t format,int channelMask,const sp<IMemory> & sharedBuffer,audio_output_flags_t flags,callback_t cbf,void * user,int notificationFrames,int sessionId)139 AudioTrack::AudioTrack(
140         audio_stream_type_t streamType,
141         uint32_t sampleRate,
142         audio_format_t format,
143         int channelMask,
144         const sp<IMemory>& sharedBuffer,
145         audio_output_flags_t flags,
146         callback_t cbf,
147         void* user,
148         int notificationFrames,
149         int sessionId)
150     : mStatus(NO_INIT),
151       mIsTimed(false),
152       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
153       mPreviousSchedulingGroup(SP_DEFAULT)
154 {
155     mStatus = set(streamType, sampleRate, format, channelMask,
156             0 /*frameCount*/, flags, cbf, user, notificationFrames,
157             sharedBuffer, false /*threadCanCallJava*/, sessionId);
158 }
159 
~AudioTrack()160 AudioTrack::~AudioTrack()
161 {
162     ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
163 
164     if (mStatus == NO_ERROR) {
165         // Make sure that callback function exits in the case where
166         // it is looping on buffer full condition in obtainBuffer().
167         // Otherwise the callback thread will never exit.
168         stop();
169         if (mAudioTrackThread != 0) {
170             mAudioTrackThread->requestExit();   // see comment in AudioTrack.h
171             mAudioTrackThread->requestExitAndWait();
172             mAudioTrackThread.clear();
173         }
174         mAudioTrack.clear();
175         IPCThreadState::self()->flushCommands();
176         AudioSystem::releaseAudioSessionId(mSessionId);
177     }
178 }
179 
set(audio_stream_type_t streamType,uint32_t sampleRate,audio_format_t format,int channelMask,int frameCount,audio_output_flags_t flags,callback_t cbf,void * user,int notificationFrames,const sp<IMemory> & sharedBuffer,bool threadCanCallJava,int sessionId)180 status_t AudioTrack::set(
181         audio_stream_type_t streamType,
182         uint32_t sampleRate,
183         audio_format_t format,
184         int channelMask,
185         int frameCount,
186         audio_output_flags_t flags,
187         callback_t cbf,
188         void* user,
189         int notificationFrames,
190         const sp<IMemory>& sharedBuffer,
191         bool threadCanCallJava,
192         int sessionId)
193 {
194 
195     ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
196 
197     ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
198 
199     AutoMutex lock(mLock);
200     if (mAudioTrack != 0) {
201         ALOGE("Track already in use");
202         return INVALID_OPERATION;
203     }
204 
205     // handle default values first.
206     if (streamType == AUDIO_STREAM_DEFAULT) {
207         streamType = AUDIO_STREAM_MUSIC;
208     }
209 
210     if (sampleRate == 0) {
211         int afSampleRate;
212         if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
213             return NO_INIT;
214         }
215         sampleRate = afSampleRate;
216     }
217 
218     // these below should probably come from the audioFlinger too...
219     if (format == AUDIO_FORMAT_DEFAULT) {
220         format = AUDIO_FORMAT_PCM_16_BIT;
221     }
222     if (channelMask == 0) {
223         channelMask = AUDIO_CHANNEL_OUT_STEREO;
224     }
225 
226     // validate parameters
227     if (!audio_is_valid_format(format)) {
228         ALOGE("Invalid format");
229         return BAD_VALUE;
230     }
231 
232     // AudioFlinger does not currently support 8-bit data in shared memory
233     if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
234         ALOGE("8-bit data in shared memory is not supported");
235         return BAD_VALUE;
236     }
237 
238     // force direct flag if format is not linear PCM
239     if (!audio_is_linear_pcm(format)) {
240         flags = (audio_output_flags_t)
241                 // FIXME why can't we allow direct AND fast?
242                 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
243     }
244     // only allow deep buffering for music stream type
245     if (streamType != AUDIO_STREAM_MUSIC) {
246         flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
247     }
248 
249     if (!audio_is_output_channel(channelMask)) {
250         ALOGE("Invalid channel mask");
251         return BAD_VALUE;
252     }
253     uint32_t channelCount = popcount(channelMask);
254 
255     audio_io_handle_t output = AudioSystem::getOutput(
256                                     streamType,
257                                     sampleRate, format, channelMask,
258                                     flags);
259 
260     if (output == 0) {
261         ALOGE("Could not get audio output for stream type %d", streamType);
262         return BAD_VALUE;
263     }
264 
265     mVolume[LEFT] = 1.0f;
266     mVolume[RIGHT] = 1.0f;
267     mSendLevel = 0.0f;
268     mFrameCount = frameCount;
269     mNotificationFramesReq = notificationFrames;
270     mSessionId = sessionId;
271     mAuxEffectId = 0;
272     mFlags = flags;
273     mCbf = cbf;
274 
275     if (cbf != NULL) {
276         mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
277         mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
278     }
279 
280     // create the IAudioTrack
281     status_t status = createTrack_l(streamType,
282                                   sampleRate,
283                                   format,
284                                   (uint32_t)channelMask,
285                                   frameCount,
286                                   flags,
287                                   sharedBuffer,
288                                   output);
289 
290     if (status != NO_ERROR) {
291         if (mAudioTrackThread != 0) {
292             mAudioTrackThread->requestExit();
293             mAudioTrackThread.clear();
294         }
295         return status;
296     }
297 
298     mStatus = NO_ERROR;
299 
300     mStreamType = streamType;
301     mFormat = format;
302     mChannelMask = (uint32_t)channelMask;
303     mChannelCount = channelCount;
304     mSharedBuffer = sharedBuffer;
305     mMuted = false;
306     mActive = false;
307     mUserData = user;
308     mLoopCount = 0;
309     mMarkerPosition = 0;
310     mMarkerReached = false;
311     mNewPosition = 0;
312     mUpdatePeriod = 0;
313     mFlushed = false;
314     AudioSystem::acquireAudioSessionId(mSessionId);
315     mRestoreStatus = NO_ERROR;
316     return NO_ERROR;
317 }
318 
initCheck() const319 status_t AudioTrack::initCheck() const
320 {
321     return mStatus;
322 }
323 
324 // -------------------------------------------------------------------------
325 
latency() const326 uint32_t AudioTrack::latency() const
327 {
328     return mLatency;
329 }
330 
streamType() const331 audio_stream_type_t AudioTrack::streamType() const
332 {
333     return mStreamType;
334 }
335 
format() const336 audio_format_t AudioTrack::format() const
337 {
338     return mFormat;
339 }
340 
channelCount() const341 int AudioTrack::channelCount() const
342 {
343     return mChannelCount;
344 }
345 
frameCount() const346 uint32_t AudioTrack::frameCount() const
347 {
348     return mCblk->frameCount;
349 }
350 
frameSize() const351 size_t AudioTrack::frameSize() const
352 {
353     if (audio_is_linear_pcm(mFormat)) {
354         return channelCount()*audio_bytes_per_sample(mFormat);
355     } else {
356         return sizeof(uint8_t);
357     }
358 }
359 
sharedBuffer()360 sp<IMemory>& AudioTrack::sharedBuffer()
361 {
362     return mSharedBuffer;
363 }
364 
365 // -------------------------------------------------------------------------
366 
start()367 void AudioTrack::start()
368 {
369     sp<AudioTrackThread> t = mAudioTrackThread;
370     status_t status = NO_ERROR;
371 
372     ALOGV("start %p", this);
373 
374     AutoMutex lock(mLock);
375     // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
376     // while we are accessing the cblk
377     sp<IAudioTrack> audioTrack = mAudioTrack;
378     sp<IMemory> iMem = mCblkMemory;
379     audio_track_cblk_t* cblk = mCblk;
380 
381     if (!mActive) {
382         mFlushed = false;
383         mActive = true;
384         mNewPosition = cblk->server + mUpdatePeriod;
385         cblk->lock.lock();
386         cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
387         cblk->waitTimeMs = 0;
388         android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
389         if (t != 0) {
390             t->resume();
391         } else {
392             mPreviousPriority = getpriority(PRIO_PROCESS, 0);
393             get_sched_policy(0, &mPreviousSchedulingGroup);
394             androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
395         }
396 
397         ALOGV("start %p before lock cblk %p", this, mCblk);
398         if (!(cblk->flags & CBLK_INVALID_MSK)) {
399             cblk->lock.unlock();
400             ALOGV("mAudioTrack->start()");
401             status = mAudioTrack->start();
402             cblk->lock.lock();
403             if (status == DEAD_OBJECT) {
404                 android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
405             }
406         }
407         if (cblk->flags & CBLK_INVALID_MSK) {
408             status = restoreTrack_l(cblk, true);
409         }
410         cblk->lock.unlock();
411         if (status != NO_ERROR) {
412             ALOGV("start() failed");
413             mActive = false;
414             if (t != 0) {
415                 t->pause();
416             } else {
417                 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
418                 set_sched_policy(0, mPreviousSchedulingGroup);
419             }
420         }
421     }
422 
423 }
424 
stop()425 void AudioTrack::stop()
426 {
427     sp<AudioTrackThread> t = mAudioTrackThread;
428 
429     ALOGV("stop %p", this);
430 
431     AutoMutex lock(mLock);
432     if (mActive) {
433         mActive = false;
434         mCblk->cv.signal();
435         mAudioTrack->stop();
436         // Cancel loops (If we are in the middle of a loop, playback
437         // would not stop until loopCount reaches 0).
438         setLoop_l(0, 0, 0);
439         // the playback head position will reset to 0, so if a marker is set, we need
440         // to activate it again
441         mMarkerReached = false;
442         // Force flush if a shared buffer is used otherwise audioflinger
443         // will not stop before end of buffer is reached.
444         if (mSharedBuffer != 0) {
445             flush_l();
446         }
447         if (t != 0) {
448             t->pause();
449         } else {
450             setpriority(PRIO_PROCESS, 0, mPreviousPriority);
451             set_sched_policy(0, mPreviousSchedulingGroup);
452         }
453     }
454 
455 }
456 
stopped() const457 bool AudioTrack::stopped() const
458 {
459     AutoMutex lock(mLock);
460     return stopped_l();
461 }
462 
flush()463 void AudioTrack::flush()
464 {
465     AutoMutex lock(mLock);
466     flush_l();
467 }
468 
469 // must be called with mLock held
flush_l()470 void AudioTrack::flush_l()
471 {
472     ALOGV("flush");
473 
474     // clear playback marker and periodic update counter
475     mMarkerPosition = 0;
476     mMarkerReached = false;
477     mUpdatePeriod = 0;
478 
479     if (!mActive) {
480         mFlushed = true;
481         mAudioTrack->flush();
482         // Release AudioTrack callback thread in case it was waiting for new buffers
483         // in AudioTrack::obtainBuffer()
484         mCblk->cv.signal();
485     }
486 }
487 
pause()488 void AudioTrack::pause()
489 {
490     ALOGV("pause");
491     AutoMutex lock(mLock);
492     if (mActive) {
493         mActive = false;
494         mCblk->cv.signal();
495         mAudioTrack->pause();
496     }
497 }
498 
mute(bool e)499 void AudioTrack::mute(bool e)
500 {
501     mAudioTrack->mute(e);
502     mMuted = e;
503 }
504 
muted() const505 bool AudioTrack::muted() const
506 {
507     return mMuted;
508 }
509 
setVolume(float left,float right)510 status_t AudioTrack::setVolume(float left, float right)
511 {
512     if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
513         return BAD_VALUE;
514     }
515 
516     AutoMutex lock(mLock);
517     mVolume[LEFT] = left;
518     mVolume[RIGHT] = right;
519 
520     mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
521 
522     return NO_ERROR;
523 }
524 
getVolume(float * left,float * right) const525 void AudioTrack::getVolume(float* left, float* right) const
526 {
527     if (left != NULL) {
528         *left  = mVolume[LEFT];
529     }
530     if (right != NULL) {
531         *right = mVolume[RIGHT];
532     }
533 }
534 
setAuxEffectSendLevel(float level)535 status_t AudioTrack::setAuxEffectSendLevel(float level)
536 {
537     ALOGV("setAuxEffectSendLevel(%f)", level);
538     if (level < 0.0f || level > 1.0f) {
539         return BAD_VALUE;
540     }
541     AutoMutex lock(mLock);
542 
543     mSendLevel = level;
544 
545     mCblk->setSendLevel(level);
546 
547     return NO_ERROR;
548 }
549 
getAuxEffectSendLevel(float * level) const550 void AudioTrack::getAuxEffectSendLevel(float* level) const
551 {
552     if (level != NULL) {
553         *level  = mSendLevel;
554     }
555 }
556 
setSampleRate(int rate)557 status_t AudioTrack::setSampleRate(int rate)
558 {
559     int afSamplingRate;
560 
561     if (mIsTimed) {
562         return INVALID_OPERATION;
563     }
564 
565     if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
566         return NO_INIT;
567     }
568     // Resampler implementation limits input sampling rate to 2 x output sampling rate.
569     if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
570 
571     AutoMutex lock(mLock);
572     mCblk->sampleRate = rate;
573     return NO_ERROR;
574 }
575 
getSampleRate() const576 uint32_t AudioTrack::getSampleRate() const
577 {
578     if (mIsTimed) {
579         return INVALID_OPERATION;
580     }
581 
582     AutoMutex lock(mLock);
583     return mCblk->sampleRate;
584 }
585 
setLoop(uint32_t loopStart,uint32_t loopEnd,int loopCount)586 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
587 {
588     AutoMutex lock(mLock);
589     return setLoop_l(loopStart, loopEnd, loopCount);
590 }
591 
592 // must be called with mLock held
setLoop_l(uint32_t loopStart,uint32_t loopEnd,int loopCount)593 status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
594 {
595     audio_track_cblk_t* cblk = mCblk;
596 
597     Mutex::Autolock _l(cblk->lock);
598 
599     if (loopCount == 0) {
600         cblk->loopStart = UINT_MAX;
601         cblk->loopEnd = UINT_MAX;
602         cblk->loopCount = 0;
603         mLoopCount = 0;
604         return NO_ERROR;
605     }
606 
607     if (mIsTimed) {
608         return INVALID_OPERATION;
609     }
610 
611     if (loopStart >= loopEnd ||
612         loopEnd - loopStart > cblk->frameCount ||
613         cblk->server > loopStart) {
614         ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
615         return BAD_VALUE;
616     }
617 
618     if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
619         ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
620             loopStart, loopEnd, cblk->frameCount);
621         return BAD_VALUE;
622     }
623 
624     cblk->loopStart = loopStart;
625     cblk->loopEnd = loopEnd;
626     cblk->loopCount = loopCount;
627     mLoopCount = loopCount;
628 
629     return NO_ERROR;
630 }
631 
setMarkerPosition(uint32_t marker)632 status_t AudioTrack::setMarkerPosition(uint32_t marker)
633 {
634     if (mCbf == NULL) return INVALID_OPERATION;
635 
636     mMarkerPosition = marker;
637     mMarkerReached = false;
638 
639     return NO_ERROR;
640 }
641 
getMarkerPosition(uint32_t * marker) const642 status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
643 {
644     if (marker == NULL) return BAD_VALUE;
645 
646     *marker = mMarkerPosition;
647 
648     return NO_ERROR;
649 }
650 
setPositionUpdatePeriod(uint32_t updatePeriod)651 status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
652 {
653     if (mCbf == NULL) return INVALID_OPERATION;
654 
655     uint32_t curPosition;
656     getPosition(&curPosition);
657     mNewPosition = curPosition + updatePeriod;
658     mUpdatePeriod = updatePeriod;
659 
660     return NO_ERROR;
661 }
662 
getPositionUpdatePeriod(uint32_t * updatePeriod) const663 status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
664 {
665     if (updatePeriod == NULL) return BAD_VALUE;
666 
667     *updatePeriod = mUpdatePeriod;
668 
669     return NO_ERROR;
670 }
671 
setPosition(uint32_t position)672 status_t AudioTrack::setPosition(uint32_t position)
673 {
674     if (mIsTimed) return INVALID_OPERATION;
675 
676     AutoMutex lock(mLock);
677 
678     if (!stopped_l()) return INVALID_OPERATION;
679 
680     Mutex::Autolock _l(mCblk->lock);
681 
682     if (position > mCblk->user) return BAD_VALUE;
683 
684     mCblk->server = position;
685     android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
686 
687     return NO_ERROR;
688 }
689 
getPosition(uint32_t * position)690 status_t AudioTrack::getPosition(uint32_t *position)
691 {
692     if (position == NULL) return BAD_VALUE;
693     AutoMutex lock(mLock);
694     *position = mFlushed ? 0 : mCblk->server;
695 
696     return NO_ERROR;
697 }
698 
reload()699 status_t AudioTrack::reload()
700 {
701     AutoMutex lock(mLock);
702 
703     if (!stopped_l()) return INVALID_OPERATION;
704 
705     flush_l();
706 
707     mCblk->stepUser(mCblk->frameCount);
708 
709     return NO_ERROR;
710 }
711 
getOutput()712 audio_io_handle_t AudioTrack::getOutput()
713 {
714     AutoMutex lock(mLock);
715     return getOutput_l();
716 }
717 
718 // must be called with mLock held
getOutput_l()719 audio_io_handle_t AudioTrack::getOutput_l()
720 {
721     return AudioSystem::getOutput(mStreamType,
722             mCblk->sampleRate, mFormat, mChannelMask, mFlags);
723 }
724 
getSessionId() const725 int AudioTrack::getSessionId() const
726 {
727     return mSessionId;
728 }
729 
attachAuxEffect(int effectId)730 status_t AudioTrack::attachAuxEffect(int effectId)
731 {
732     ALOGV("attachAuxEffect(%d)", effectId);
733     status_t status = mAudioTrack->attachAuxEffect(effectId);
734     if (status == NO_ERROR) {
735         mAuxEffectId = effectId;
736     }
737     return status;
738 }
739 
740 // -------------------------------------------------------------------------
741 
742 // must be called with mLock held
createTrack_l(audio_stream_type_t streamType,uint32_t sampleRate,audio_format_t format,uint32_t channelMask,int frameCount,audio_output_flags_t flags,const sp<IMemory> & sharedBuffer,audio_io_handle_t output)743 status_t AudioTrack::createTrack_l(
744         audio_stream_type_t streamType,
745         uint32_t sampleRate,
746         audio_format_t format,
747         uint32_t channelMask,
748         int frameCount,
749         audio_output_flags_t flags,
750         const sp<IMemory>& sharedBuffer,
751         audio_io_handle_t output)
752 {
753     status_t status;
754     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
755     if (audioFlinger == 0) {
756         ALOGE("Could not get audioflinger");
757         return NO_INIT;
758     }
759 
760     uint32_t afLatency;
761     if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
762         return NO_INIT;
763     }
764 
765     // Client decides whether the track is TIMED (see below), but can only express a preference
766     // for FAST.  Server will perform additional tests.
767     if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
768             // either of these use cases:
769             // use case 1: shared buffer
770             (sharedBuffer != 0) ||
771             // use case 2: callback handler
772             (mCbf != NULL))) {
773         ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
774         // once denied, do not request again if IAudioTrack is re-created
775         flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
776         mFlags = flags;
777     }
778     ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
779 
780     mNotificationFramesAct = mNotificationFramesReq;
781 
782     if (!audio_is_linear_pcm(format)) {
783 
784         if (sharedBuffer != 0) {
785             // Same comment as below about ignoring frameCount parameter for set()
786             frameCount = sharedBuffer->size();
787         } else if (frameCount == 0) {
788             int afFrameCount;
789             if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
790                 return NO_INIT;
791             }
792             frameCount = afFrameCount;
793         }
794 
795     } else if (sharedBuffer != 0) {
796 
797         // Ensure that buffer alignment matches channelCount
798         int channelCount = popcount(channelMask);
799         // 8-bit data in shared memory is not currently supported by AudioFlinger
800         size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
801         if (channelCount > 1) {
802             // More than 2 channels does not require stronger alignment than stereo
803             alignment <<= 1;
804         }
805         if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
806             ALOGE("Invalid buffer alignment: address %p, channelCount %d",
807                     sharedBuffer->pointer(), channelCount);
808             return BAD_VALUE;
809         }
810 
811         // When initializing a shared buffer AudioTrack via constructors,
812         // there's no frameCount parameter.
813         // But when initializing a shared buffer AudioTrack via set(),
814         // there _is_ a frameCount parameter.  We silently ignore it.
815         frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
816 
817     } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
818 
819         // FIXME move these calculations and associated checks to server
820         int afSampleRate;
821         if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
822             return NO_INIT;
823         }
824         int afFrameCount;
825         if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
826             return NO_INIT;
827         }
828 
829         // Ensure that buffer depth covers at least audio hardware latency
830         uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
831         if (minBufCount < 2) minBufCount = 2;
832 
833         int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
834         ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
835                 ", afLatency=%d",
836                 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
837 
838         if (frameCount == 0) {
839             frameCount = minFrameCount;
840         }
841         if (mNotificationFramesAct == 0) {
842             mNotificationFramesAct = frameCount/2;
843         }
844         // Make sure that application is notified with sufficient margin
845         // before underrun
846         if (mNotificationFramesAct > (uint32_t)frameCount/2) {
847             mNotificationFramesAct = frameCount/2;
848         }
849         if (frameCount < minFrameCount) {
850             // not ALOGW because it happens all the time when playing key clicks over A2DP
851             ALOGV("Minimum buffer size corrected from %d to %d",
852                      frameCount, minFrameCount);
853             frameCount = minFrameCount;
854         }
855 
856     } else {
857         // For fast tracks, the frame count calculations and checks are done by server
858     }
859 
860     IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
861     if (mIsTimed) {
862         trackFlags |= IAudioFlinger::TRACK_TIMED;
863     }
864 
865     pid_t tid = -1;
866     if (flags & AUDIO_OUTPUT_FLAG_FAST) {
867         trackFlags |= IAudioFlinger::TRACK_FAST;
868         if (mAudioTrackThread != 0) {
869             tid = mAudioTrackThread->getTid();
870         }
871     }
872 
873     sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
874                                                       streamType,
875                                                       sampleRate,
876                                                       format,
877                                                       channelMask,
878                                                       frameCount,
879                                                       trackFlags,
880                                                       sharedBuffer,
881                                                       output,
882                                                       tid,
883                                                       &mSessionId,
884                                                       &status);
885 
886     if (track == 0) {
887         ALOGE("AudioFlinger could not create track, status: %d", status);
888         return status;
889     }
890     sp<IMemory> cblk = track->getCblk();
891     if (cblk == 0) {
892         ALOGE("Could not get control block");
893         return NO_INIT;
894     }
895     mAudioTrack = track;
896     mCblkMemory = cblk;
897     mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
898     // old has the previous value of mCblk->flags before the "or" operation
899     int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
900     if (flags & AUDIO_OUTPUT_FLAG_FAST) {
901         if (old & CBLK_FAST) {
902             ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount);
903         } else {
904             ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount);
905             // once denied, do not request again if IAudioTrack is re-created
906             flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
907             mFlags = flags;
908         }
909         if (sharedBuffer == 0) {
910             mNotificationFramesAct = mCblk->frameCount/2;
911         }
912     }
913     if (sharedBuffer == 0) {
914         mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
915     } else {
916         mCblk->buffers = sharedBuffer->pointer();
917         // Force buffer full condition as data is already present in shared memory
918         mCblk->stepUser(mCblk->frameCount);
919     }
920 
921     mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
922     mCblk->setSendLevel(mSendLevel);
923     mAudioTrack->attachAuxEffect(mAuxEffectId);
924     mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
925     mCblk->waitTimeMs = 0;
926     mRemainingFrames = mNotificationFramesAct;
927     // FIXME don't believe this lie
928     mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
929     // If IAudioTrack is re-created, don't let the requested frameCount
930     // decrease.  This can confuse clients that cache frameCount().
931     if (mCblk->frameCount > mFrameCount) {
932         mFrameCount = mCblk->frameCount;
933     }
934     return NO_ERROR;
935 }
936 
obtainBuffer(Buffer * audioBuffer,int32_t waitCount)937 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
938 {
939     AutoMutex lock(mLock);
940     bool active;
941     status_t result = NO_ERROR;
942     audio_track_cblk_t* cblk = mCblk;
943     uint32_t framesReq = audioBuffer->frameCount;
944     uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
945 
946     audioBuffer->frameCount  = 0;
947     audioBuffer->size = 0;
948 
949     uint32_t framesAvail = cblk->framesAvailable();
950 
951     cblk->lock.lock();
952     if (cblk->flags & CBLK_INVALID_MSK) {
953         goto create_new_track;
954     }
955     cblk->lock.unlock();
956 
957     if (framesAvail == 0) {
958         cblk->lock.lock();
959         goto start_loop_here;
960         while (framesAvail == 0) {
961             active = mActive;
962             if (CC_UNLIKELY(!active)) {
963                 ALOGV("Not active and NO_MORE_BUFFERS");
964                 cblk->lock.unlock();
965                 return NO_MORE_BUFFERS;
966             }
967             if (CC_UNLIKELY(!waitCount)) {
968                 cblk->lock.unlock();
969                 return WOULD_BLOCK;
970             }
971             if (!(cblk->flags & CBLK_INVALID_MSK)) {
972                 mLock.unlock();
973                 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
974                 cblk->lock.unlock();
975                 mLock.lock();
976                 if (!mActive) {
977                     return status_t(STOPPED);
978                 }
979                 cblk->lock.lock();
980             }
981 
982             if (cblk->flags & CBLK_INVALID_MSK) {
983                 goto create_new_track;
984             }
985             if (CC_UNLIKELY(result != NO_ERROR)) {
986                 cblk->waitTimeMs += waitTimeMs;
987                 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
988                     // timing out when a loop has been set and we have already written upto loop end
989                     // is a normal condition: no need to wake AudioFlinger up.
990                     if (cblk->user < cblk->loopEnd) {
991                         ALOGW(   "obtainBuffer timed out (is the CPU pegged?) %p name=%#x"
992                                 "user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server);
993                         //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
994                         cblk->lock.unlock();
995                         result = mAudioTrack->start();
996                         cblk->lock.lock();
997                         if (result == DEAD_OBJECT) {
998                             android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
999 create_new_track:
1000                             result = restoreTrack_l(cblk, false);
1001                         }
1002                         if (result != NO_ERROR) {
1003                             ALOGW("obtainBuffer create Track error %d", result);
1004                             cblk->lock.unlock();
1005                             return result;
1006                         }
1007                     }
1008                     cblk->waitTimeMs = 0;
1009                 }
1010 
1011                 if (--waitCount == 0) {
1012                     cblk->lock.unlock();
1013                     return TIMED_OUT;
1014                 }
1015             }
1016             // read the server count again
1017         start_loop_here:
1018             framesAvail = cblk->framesAvailable_l();
1019         }
1020         cblk->lock.unlock();
1021     }
1022 
1023     cblk->waitTimeMs = 0;
1024 
1025     if (framesReq > framesAvail) {
1026         framesReq = framesAvail;
1027     }
1028 
1029     uint32_t u = cblk->user;
1030     uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
1031 
1032     if (framesReq > bufferEnd - u) {
1033         framesReq = bufferEnd - u;
1034     }
1035 
1036     audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
1037     audioBuffer->channelCount = mChannelCount;
1038     audioBuffer->frameCount = framesReq;
1039     audioBuffer->size = framesReq * cblk->frameSize;
1040     if (audio_is_linear_pcm(mFormat)) {
1041         audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
1042     } else {
1043         audioBuffer->format = mFormat;
1044     }
1045     audioBuffer->raw = (int8_t *)cblk->buffer(u);
1046     active = mActive;
1047     return active ? status_t(NO_ERROR) : status_t(STOPPED);
1048 }
1049 
releaseBuffer(Buffer * audioBuffer)1050 void AudioTrack::releaseBuffer(Buffer* audioBuffer)
1051 {
1052     AutoMutex lock(mLock);
1053     mCblk->stepUser(audioBuffer->frameCount);
1054     if (audioBuffer->frameCount > 0) {
1055         // restart track if it was disabled by audioflinger due to previous underrun
1056         if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
1057             android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
1058             ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName);
1059             mAudioTrack->start();
1060         }
1061     }
1062 }
1063 
1064 // -------------------------------------------------------------------------
1065 
write(const void * buffer,size_t userSize)1066 ssize_t AudioTrack::write(const void* buffer, size_t userSize)
1067 {
1068 
1069     if (mSharedBuffer != 0) return INVALID_OPERATION;
1070     if (mIsTimed) return INVALID_OPERATION;
1071 
1072     if (ssize_t(userSize) < 0) {
1073         // Sanity-check: user is most-likely passing an error code, and it would
1074         // make the return value ambiguous (actualSize vs error).
1075         ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
1076                 buffer, userSize, userSize);
1077         return BAD_VALUE;
1078     }
1079 
1080     ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
1081 
1082     if (userSize == 0) {
1083         return 0;
1084     }
1085 
1086     // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1087     // while we are accessing the cblk
1088     mLock.lock();
1089     sp<IAudioTrack> audioTrack = mAudioTrack;
1090     sp<IMemory> iMem = mCblkMemory;
1091     mLock.unlock();
1092 
1093     ssize_t written = 0;
1094     const int8_t *src = (const int8_t *)buffer;
1095     Buffer audioBuffer;
1096     size_t frameSz = frameSize();
1097 
1098     do {
1099         audioBuffer.frameCount = userSize/frameSz;
1100 
1101         status_t err = obtainBuffer(&audioBuffer, -1);
1102         if (err < 0) {
1103             // out of buffers, return #bytes written
1104             if (err == status_t(NO_MORE_BUFFERS))
1105                 break;
1106             return ssize_t(err);
1107         }
1108 
1109         size_t toWrite;
1110 
1111         if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1112             // Divide capacity by 2 to take expansion into account
1113             toWrite = audioBuffer.size>>1;
1114             memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
1115         } else {
1116             toWrite = audioBuffer.size;
1117             memcpy(audioBuffer.i8, src, toWrite);
1118             src += toWrite;
1119         }
1120         userSize -= toWrite;
1121         written += toWrite;
1122 
1123         releaseBuffer(&audioBuffer);
1124     } while (userSize >= frameSz);
1125 
1126     return written;
1127 }
1128 
1129 // -------------------------------------------------------------------------
1130 
TimedAudioTrack()1131 TimedAudioTrack::TimedAudioTrack() {
1132     mIsTimed = true;
1133 }
1134 
allocateTimedBuffer(size_t size,sp<IMemory> * buffer)1135 status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
1136 {
1137     status_t result = UNKNOWN_ERROR;
1138 
1139     // If the track is not invalid already, try to allocate a buffer.  alloc
1140     // fails indicating that the server is dead, flag the track as invalid so
1141     // we can attempt to restore in in just a bit.
1142     if (!(mCblk->flags & CBLK_INVALID_MSK)) {
1143         result = mAudioTrack->allocateTimedBuffer(size, buffer);
1144         if (result == DEAD_OBJECT) {
1145             android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
1146         }
1147     }
1148 
1149     // If the track is invalid at this point, attempt to restore it. and try the
1150     // allocation one more time.
1151     if (mCblk->flags & CBLK_INVALID_MSK) {
1152         mCblk->lock.lock();
1153         result = restoreTrack_l(mCblk, false);
1154         mCblk->lock.unlock();
1155 
1156         if (result == OK)
1157             result = mAudioTrack->allocateTimedBuffer(size, buffer);
1158     }
1159 
1160     return result;
1161 }
1162 
queueTimedBuffer(const sp<IMemory> & buffer,int64_t pts)1163 status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
1164                                            int64_t pts)
1165 {
1166     status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
1167     {
1168         AutoMutex lock(mLock);
1169         // restart track if it was disabled by audioflinger due to previous underrun
1170         if (buffer->size() != 0 && status == NO_ERROR &&
1171                 mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
1172             android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
1173             ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
1174             mAudioTrack->start();
1175         }
1176     }
1177     return status;
1178 }
1179 
setMediaTimeTransform(const LinearTransform & xform,TargetTimeline target)1180 status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
1181                                                 TargetTimeline target)
1182 {
1183     return mAudioTrack->setMediaTimeTransform(xform, target);
1184 }
1185 
1186 // -------------------------------------------------------------------------
1187 
processAudioBuffer(const sp<AudioTrackThread> & thread)1188 bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
1189 {
1190     Buffer audioBuffer;
1191     uint32_t frames;
1192     size_t writtenSize;
1193 
1194     mLock.lock();
1195     // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1196     // while we are accessing the cblk
1197     sp<IAudioTrack> audioTrack = mAudioTrack;
1198     sp<IMemory> iMem = mCblkMemory;
1199     audio_track_cblk_t* cblk = mCblk;
1200     bool active = mActive;
1201     mLock.unlock();
1202 
1203     // Manage underrun callback
1204     if (active && (cblk->framesAvailable() == cblk->frameCount)) {
1205         ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
1206         if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
1207             mCbf(EVENT_UNDERRUN, mUserData, 0);
1208             if (cblk->server == cblk->frameCount) {
1209                 mCbf(EVENT_BUFFER_END, mUserData, 0);
1210             }
1211             if (mSharedBuffer != 0) return false;
1212         }
1213     }
1214 
1215     // Manage loop end callback
1216     while (mLoopCount > cblk->loopCount) {
1217         int loopCount = -1;
1218         mLoopCount--;
1219         if (mLoopCount >= 0) loopCount = mLoopCount;
1220 
1221         mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
1222     }
1223 
1224     // Manage marker callback
1225     if (!mMarkerReached && (mMarkerPosition > 0)) {
1226         if (cblk->server >= mMarkerPosition) {
1227             mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
1228             mMarkerReached = true;
1229         }
1230     }
1231 
1232     // Manage new position callback
1233     if (mUpdatePeriod > 0) {
1234         while (cblk->server >= mNewPosition) {
1235             mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
1236             mNewPosition += mUpdatePeriod;
1237         }
1238     }
1239 
1240     // If Shared buffer is used, no data is requested from client.
1241     if (mSharedBuffer != 0) {
1242         frames = 0;
1243     } else {
1244         frames = mRemainingFrames;
1245     }
1246 
1247     // See description of waitCount parameter at declaration of obtainBuffer().
1248     // The logic below prevents us from being stuck below at obtainBuffer()
1249     // not being able to handle timed events (position, markers, loops).
1250     int32_t waitCount = -1;
1251     if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
1252         waitCount = 1;
1253     }
1254 
1255     do {
1256 
1257         audioBuffer.frameCount = frames;
1258 
1259         status_t err = obtainBuffer(&audioBuffer, waitCount);
1260         if (err < NO_ERROR) {
1261             if (err != TIMED_OUT) {
1262                 ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
1263                 return false;
1264             }
1265             break;
1266         }
1267         if (err == status_t(STOPPED)) return false;
1268 
1269         // Divide buffer size by 2 to take into account the expansion
1270         // due to 8 to 16 bit conversion: the callback must fill only half
1271         // of the destination buffer
1272         if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1273             audioBuffer.size >>= 1;
1274         }
1275 
1276         size_t reqSize = audioBuffer.size;
1277         mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1278         writtenSize = audioBuffer.size;
1279 
1280         // Sanity check on returned size
1281         if (ssize_t(writtenSize) <= 0) {
1282             // The callback is done filling buffers
1283             // Keep this thread going to handle timed events and
1284             // still try to get more data in intervals of WAIT_PERIOD_MS
1285             // but don't just loop and block the CPU, so wait
1286             usleep(WAIT_PERIOD_MS*1000);
1287             break;
1288         }
1289 
1290         if (writtenSize > reqSize) writtenSize = reqSize;
1291 
1292         if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1293             // 8 to 16 bit conversion, note that source and destination are the same address
1294             memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
1295             writtenSize <<= 1;
1296         }
1297 
1298         audioBuffer.size = writtenSize;
1299         // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
1300         // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
1301         // 16 bit.
1302         audioBuffer.frameCount = writtenSize/mCblk->frameSize;
1303 
1304         frames -= audioBuffer.frameCount;
1305 
1306         releaseBuffer(&audioBuffer);
1307     }
1308     while (frames);
1309 
1310     if (frames == 0) {
1311         mRemainingFrames = mNotificationFramesAct;
1312     } else {
1313         mRemainingFrames = frames;
1314     }
1315     return true;
1316 }
1317 
1318 // must be called with mLock and cblk.lock held. Callers must also hold strong references on
1319 // the IAudioTrack and IMemory in case they are recreated here.
1320 // If the IAudioTrack is successfully restored, the cblk pointer is updated
restoreTrack_l(audio_track_cblk_t * & cblk,bool fromStart)1321 status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
1322 {
1323     status_t result;
1324 
1325     if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
1326         ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
1327             fromStart ? "start()" : "obtainBuffer()", gettid());
1328 
1329         // signal old cblk condition so that other threads waiting for available buffers stop
1330         // waiting now
1331         cblk->cv.broadcast();
1332         cblk->lock.unlock();
1333 
1334         // refresh the audio configuration cache in this process to make sure we get new
1335         // output parameters in getOutput_l() and createTrack_l()
1336         AudioSystem::clearAudioConfigCache();
1337 
1338         // if the new IAudioTrack is created, createTrack_l() will modify the
1339         // following member variables: mAudioTrack, mCblkMemory and mCblk.
1340         // It will also delete the strong references on previous IAudioTrack and IMemory
1341         result = createTrack_l(mStreamType,
1342                                cblk->sampleRate,
1343                                mFormat,
1344                                mChannelMask,
1345                                mFrameCount,
1346                                mFlags,
1347                                mSharedBuffer,
1348                                getOutput_l());
1349 
1350         if (result == NO_ERROR) {
1351             uint32_t user = cblk->user;
1352             uint32_t server = cblk->server;
1353             // restore write index and set other indexes to reflect empty buffer status
1354             mCblk->user = user;
1355             mCblk->server = user;
1356             mCblk->userBase = user;
1357             mCblk->serverBase = user;
1358             // restore loop: this is not guaranteed to succeed if new frame count is not
1359             // compatible with loop length
1360             setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
1361             if (!fromStart) {
1362                 mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1363                 // Make sure that a client relying on callback events indicating underrun or
1364                 // the actual amount of audio frames played (e.g SoundPool) receives them.
1365                 if (mSharedBuffer == 0) {
1366                     uint32_t frames = 0;
1367                     if (user > server) {
1368                         frames = ((user - server) > mCblk->frameCount) ?
1369                                 mCblk->frameCount : (user - server);
1370                         memset(mCblk->buffers, 0, frames * mCblk->frameSize);
1371                     }
1372                     // restart playback even if buffer is not completely filled.
1373                     android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
1374                     // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
1375                     // the client
1376                     mCblk->stepUser(frames);
1377                 }
1378             }
1379             if (mSharedBuffer != 0) {
1380                 mCblk->stepUser(mCblk->frameCount);
1381             }
1382             if (mActive) {
1383                 result = mAudioTrack->start();
1384                 ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
1385             }
1386             if (fromStart && result == NO_ERROR) {
1387                 mNewPosition = mCblk->server + mUpdatePeriod;
1388             }
1389         }
1390         if (result != NO_ERROR) {
1391             android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
1392             ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
1393         }
1394         mRestoreStatus = result;
1395         // signal old cblk condition for other threads waiting for restore completion
1396         android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
1397         cblk->cv.broadcast();
1398     } else {
1399         if (!(cblk->flags & CBLK_RESTORED_MSK)) {
1400             ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid());
1401             mLock.unlock();
1402             result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
1403             if (result == NO_ERROR) {
1404                 result = mRestoreStatus;
1405             }
1406             cblk->lock.unlock();
1407             mLock.lock();
1408         } else {
1409             ALOGW("dead IAudioTrack, already restored TID %d", gettid());
1410             result = mRestoreStatus;
1411             cblk->lock.unlock();
1412         }
1413     }
1414     ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
1415         result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
1416 
1417     if (result == NO_ERROR) {
1418         // from now on we switch to the newly created cblk
1419         cblk = mCblk;
1420     }
1421     cblk->lock.lock();
1422 
1423     ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
1424 
1425     return result;
1426 }
1427 
dump(int fd,const Vector<String16> & args) const1428 status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1429 {
1430 
1431     const size_t SIZE = 256;
1432     char buffer[SIZE];
1433     String8 result;
1434 
1435     result.append(" AudioTrack::dump\n");
1436     snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
1437     result.append(buffer);
1438     snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
1439     result.append(buffer);
1440     snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
1441     result.append(buffer);
1442     snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1443     result.append(buffer);
1444     ::write(fd, result.string(), result.size());
1445     return NO_ERROR;
1446 }
1447 
1448 // =========================================================================
1449 
AudioTrackThread(AudioTrack & receiver,bool bCanCallJava)1450 AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1451     : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
1452 {
1453 }
1454 
~AudioTrackThread()1455 AudioTrack::AudioTrackThread::~AudioTrackThread()
1456 {
1457 }
1458 
threadLoop()1459 bool AudioTrack::AudioTrackThread::threadLoop()
1460 {
1461     {
1462         AutoMutex _l(mMyLock);
1463         if (mPaused) {
1464             mMyCond.wait(mMyLock);
1465             // caller will check for exitPending()
1466             return true;
1467         }
1468     }
1469     if (!mReceiver.processAudioBuffer(this)) {
1470         pause();
1471     }
1472     return true;
1473 }
1474 
readyToRun()1475 status_t AudioTrack::AudioTrackThread::readyToRun()
1476 {
1477     return NO_ERROR;
1478 }
1479 
onFirstRef()1480 void AudioTrack::AudioTrackThread::onFirstRef()
1481 {
1482 }
1483 
requestExit()1484 void AudioTrack::AudioTrackThread::requestExit()
1485 {
1486     // must be in this order to avoid a race condition
1487     Thread::requestExit();
1488     resume();
1489 }
1490 
pause()1491 void AudioTrack::AudioTrackThread::pause()
1492 {
1493     AutoMutex _l(mMyLock);
1494     mPaused = true;
1495 }
1496 
resume()1497 void AudioTrack::AudioTrackThread::resume()
1498 {
1499     AutoMutex _l(mMyLock);
1500     if (mPaused) {
1501         mPaused = false;
1502         mMyCond.signal();
1503     }
1504 }
1505 
1506 // =========================================================================
1507 
1508 
audio_track_cblk_t()1509 audio_track_cblk_t::audio_track_cblk_t()
1510     : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1511     userBase(0), serverBase(0), buffers(NULL), frameCount(0),
1512     loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
1513     mSendLevel(0), flags(0)
1514 {
1515 }
1516 
stepUser(uint32_t frameCount)1517 uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1518 {
1519     ALOGV("stepuser %08x %08x %d", user, server, frameCount);
1520 
1521     uint32_t u = user;
1522     u += frameCount;
1523     // Ensure that user is never ahead of server for AudioRecord
1524     if (flags & CBLK_DIRECTION_MSK) {
1525         // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1526         if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1527             bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1528         }
1529     } else if (u > server) {
1530         ALOGW("stepUser occurred after track reset");
1531         u = server;
1532     }
1533 
1534     uint32_t fc = this->frameCount;
1535     if (u >= fc) {
1536         // common case, user didn't just wrap
1537         if (u - fc >= userBase ) {
1538             userBase += fc;
1539         }
1540     } else if (u >= userBase + fc) {
1541         // user just wrapped
1542         userBase += fc;
1543     }
1544 
1545     user = u;
1546 
1547     // Clear flow control error condition as new data has been written/read to/from buffer.
1548     if (flags & CBLK_UNDERRUN_MSK) {
1549         android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
1550     }
1551 
1552     return u;
1553 }
1554 
stepServer(uint32_t frameCount)1555 bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1556 {
1557     ALOGV("stepserver %08x %08x %d", user, server, frameCount);
1558 
1559     if (!tryLock()) {
1560         ALOGW("stepServer() could not lock cblk");
1561         return false;
1562     }
1563 
1564     uint32_t s = server;
1565     bool flushed = (s == user);
1566 
1567     s += frameCount;
1568     if (flags & CBLK_DIRECTION_MSK) {
1569         // Mark that we have read the first buffer so that next time stepUser() is called
1570         // we switch to normal obtainBuffer() timeout period
1571         if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1572             bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1573         }
1574         // It is possible that we receive a flush()
1575         // while the mixer is processing a block: in this case,
1576         // stepServer() is called After the flush() has reset u & s and
1577         // we have s > u
1578         if (flushed) {
1579             ALOGW("stepServer occurred after track reset");
1580             s = user;
1581         }
1582     }
1583 
1584     if (s >= loopEnd) {
1585         ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1586         s = loopStart;
1587         if (--loopCount == 0) {
1588             loopEnd = UINT_MAX;
1589             loopStart = UINT_MAX;
1590         }
1591     }
1592 
1593     uint32_t fc = this->frameCount;
1594     if (s >= fc) {
1595         // common case, server didn't just wrap
1596         if (s - fc >= serverBase ) {
1597             serverBase += fc;
1598         }
1599     } else if (s >= serverBase + fc) {
1600         // server just wrapped
1601         serverBase += fc;
1602     }
1603 
1604     server = s;
1605 
1606     if (!(flags & CBLK_INVALID_MSK)) {
1607         cv.signal();
1608     }
1609     lock.unlock();
1610     return true;
1611 }
1612 
buffer(uint32_t offset) const1613 void* audio_track_cblk_t::buffer(uint32_t offset) const
1614 {
1615     return (int8_t *)buffers + (offset - userBase) * frameSize;
1616 }
1617 
framesAvailable()1618 uint32_t audio_track_cblk_t::framesAvailable()
1619 {
1620     Mutex::Autolock _l(lock);
1621     return framesAvailable_l();
1622 }
1623 
framesAvailable_l()1624 uint32_t audio_track_cblk_t::framesAvailable_l()
1625 {
1626     uint32_t u = user;
1627     uint32_t s = server;
1628 
1629     if (flags & CBLK_DIRECTION_MSK) {
1630         uint32_t limit = (s < loopStart) ? s : loopStart;
1631         return limit + frameCount - u;
1632     } else {
1633         return frameCount + u - s;
1634     }
1635 }
1636 
framesReady()1637 uint32_t audio_track_cblk_t::framesReady()
1638 {
1639     uint32_t u = user;
1640     uint32_t s = server;
1641 
1642     if (flags & CBLK_DIRECTION_MSK) {
1643         if (u < loopEnd) {
1644             return u - s;
1645         } else {
1646             // do not block on mutex shared with client on AudioFlinger side
1647             if (!tryLock()) {
1648                 ALOGW("framesReady() could not lock cblk");
1649                 return 0;
1650             }
1651             uint32_t frames = UINT_MAX;
1652             if (loopCount >= 0) {
1653                 frames = (loopEnd - loopStart)*loopCount + u - s;
1654             }
1655             lock.unlock();
1656             return frames;
1657         }
1658     } else {
1659         return s - u;
1660     }
1661 }
1662 
tryLock()1663 bool audio_track_cblk_t::tryLock()
1664 {
1665     // the code below simulates lock-with-timeout
1666     // we MUST do this to protect the AudioFlinger server
1667     // as this lock is shared with the client.
1668     status_t err;
1669 
1670     err = lock.tryLock();
1671     if (err == -EBUSY) { // just wait a bit
1672         usleep(1000);
1673         err = lock.tryLock();
1674     }
1675     if (err != NO_ERROR) {
1676         // probably, the client just died.
1677         return false;
1678     }
1679     return true;
1680 }
1681 
1682 // -------------------------------------------------------------------------
1683 
1684 }; // namespace android
1685