AudioTrack.cpp revision 44d9848d6656777a18019223e0d35f2fcc67719a 
1/* //device/extlibs/pv/android/AudioTrack.cpp
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19//#define LOG_NDEBUG 0
20#define LOG_TAG "AudioTrack"
21
22#include <stdint.h>
23#include <sys/types.h>
24#include <limits.h>
25
26#include <sched.h>
27#include <sys/resource.h>
28
29#include <private/media/AudioTrackShared.h>
30
31#include <media/AudioSystem.h>
32#include <media/AudioTrack.h>
33
34#include <utils/Log.h>
35#include <binder/Parcel.h>
36#include <binder/IPCThreadState.h>
37#include <utils/Timers.h>
38#include <cutils/atomic.h>
39
40#define LIKELY( exp )       (__builtin_expect( (exp) != 0, true  ))
41#define UNLIKELY( exp )     (__builtin_expect( (exp) != 0, false ))
42
43namespace android {
44// ---------------------------------------------------------------------------
45
46// static
47status_t AudioTrack::getMinFrameCount(
48        int* frameCount,
49        int streamType,
50        uint32_t sampleRate)
51{
52    int afSampleRate;
53    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
54        return NO_INIT;
55    }
56    int afFrameCount;
57    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
58        return NO_INIT;
59    }
60    uint32_t afLatency;
61    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
62        return NO_INIT;
63    }
64
65    // Ensure that buffer depth covers at least audio hardware latency
66    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
67    if (minBufCount < 2) minBufCount = 2;
68
69    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
70              afFrameCount * minBufCount * sampleRate / afSampleRate;
71    return NO_ERROR;
72}
73
74// ---------------------------------------------------------------------------
75
76AudioTrack::AudioTrack()
77    : mStatus(NO_INIT)
78{
79}
80
81AudioTrack::AudioTrack(
82        int streamType,
83        uint32_t sampleRate,
84        int format,
85        int channels,
86        int frameCount,
87        uint32_t flags,
88        callback_t cbf,
89        void* user,
90        int notificationFrames,
91        int sessionId)
92    : mStatus(NO_INIT)
93{
94    mStatus = set(streamType, sampleRate, format, channels,
95            frameCount, flags, cbf, user, notificationFrames,
96            0, false, sessionId);
97}
98
99AudioTrack::AudioTrack(
100        int streamType,
101        uint32_t sampleRate,
102        int format,
103        int channels,
104        const sp<IMemory>& sharedBuffer,
105        uint32_t flags,
106        callback_t cbf,
107        void* user,
108        int notificationFrames,
109        int sessionId)
110    : mStatus(NO_INIT)
111{
112    mStatus = set(streamType, sampleRate, format, channels,
113            0, flags, cbf, user, notificationFrames,
114            sharedBuffer, false, sessionId);
115}
116
117AudioTrack::~AudioTrack()
118{
119    LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
120
121    if (mStatus == NO_ERROR) {
122        // Make sure that callback function exits in the case where
123        // it is looping on buffer full condition in obtainBuffer().
124        // Otherwise the callback thread will never exit.
125        stop();
126        if (mAudioTrackThread != 0) {
127            mAudioTrackThread->requestExitAndWait();
128            mAudioTrackThread.clear();
129        }
130        mAudioTrack.clear();
131        IPCThreadState::self()->flushCommands();
132    }
133}
134
135status_t AudioTrack::set(
136        int streamType,
137        uint32_t sampleRate,
138        int format,
139        int channels,
140        int frameCount,
141        uint32_t flags,
142        callback_t cbf,
143        void* user,
144        int notificationFrames,
145        const sp<IMemory>& sharedBuffer,
146        bool threadCanCallJava,
147        int sessionId)
148{
149
150    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
151
152    if (mAudioTrack != 0) {
153        LOGE("Track already in use");
154        return INVALID_OPERATION;
155    }
156
157    int afSampleRate;
158    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
159        return NO_INIT;
160    }
161    uint32_t afLatency;
162    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
163        return NO_INIT;
164    }
165
166    // handle default values first.
167    if (streamType == AudioSystem::DEFAULT) {
168        streamType = AudioSystem::MUSIC;
169    }
170    if (sampleRate == 0) {
171        sampleRate = afSampleRate;
172    }
173    // these below should probably come from the audioFlinger too...
174    if (format == 0) {
175        format = AudioSystem::PCM_16_BIT;
176    }
177    if (channels == 0) {
178        channels = AudioSystem::CHANNEL_OUT_STEREO;
179    }
180
181    // validate parameters
182    if (!AudioSystem::isValidFormat(format)) {
183        LOGE("Invalid format");
184        return BAD_VALUE;
185    }
186
187    // force direct flag if format is not linear PCM
188    if (!AudioSystem::isLinearPCM(format)) {
189        flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
190    }
191
192    if (!AudioSystem::isOutputChannel(channels)) {
193        LOGE("Invalid channel mask");
194        return BAD_VALUE;
195    }
196    uint32_t channelCount = AudioSystem::popCount(channels);
197
198    audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
199            sampleRate, format, channels, (AudioSystem::output_flags)flags);
200
201    if (output == 0) {
202        LOGE("Could not get audio output for stream type %d", streamType);
203        return BAD_VALUE;
204    }
205
206    mVolume[LEFT] = 1.0f;
207    mVolume[RIGHT] = 1.0f;
208    mSendLevel = 0;
209    mFrameCount = frameCount;
210    mNotificationFramesReq = notificationFrames;
211    mSessionId = sessionId;
212    mAuxEffectId = 0;
213
214    // create the IAudioTrack
215    status_t status = createTrack(streamType, sampleRate, format, channelCount,
216                                  frameCount, flags, sharedBuffer, output, true);
217
218    if (status != NO_ERROR) {
219        return status;
220    }
221
222    if (cbf != 0) {
223        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
224        if (mAudioTrackThread == 0) {
225          LOGE("Could not create callback thread");
226          return NO_INIT;
227        }
228    }
229
230    mStatus = NO_ERROR;
231
232    mStreamType = streamType;
233    mFormat = format;
234    mChannels = channels;
235    mChannelCount = channelCount;
236    mSharedBuffer = sharedBuffer;
237    mMuted = false;
238    mActive = 0;
239    mCbf = cbf;
240    mUserData = user;
241    mLoopCount = 0;
242    mMarkerPosition = 0;
243    mMarkerReached = false;
244    mNewPosition = 0;
245    mUpdatePeriod = 0;
246    mFlags = flags;
247
248    return NO_ERROR;
249}
250
251status_t AudioTrack::initCheck() const
252{
253    return mStatus;
254}
255
256// -------------------------------------------------------------------------
257
258uint32_t AudioTrack::latency() const
259{
260    return mLatency;
261}
262
263int AudioTrack::streamType() const
264{
265    return mStreamType;
266}
267
268int AudioTrack::format() const
269{
270    return mFormat;
271}
272
273int AudioTrack::channelCount() const
274{
275    return mChannelCount;
276}
277
278uint32_t AudioTrack::frameCount() const
279{
280    return mCblk->frameCount;
281}
282
283int AudioTrack::frameSize() const
284{
285    if (AudioSystem::isLinearPCM(mFormat)) {
286        return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
287    } else {
288        return sizeof(uint8_t);
289    }
290}
291
292sp<IMemory>& AudioTrack::sharedBuffer()
293{
294    return mSharedBuffer;
295}
296
297// -------------------------------------------------------------------------
298
299void AudioTrack::start()
300{
301    sp<AudioTrackThread> t = mAudioTrackThread;
302    status_t status;
303
304    LOGV("start %p", this);
305    if (t != 0) {
306        if (t->exitPending()) {
307            if (t->requestExitAndWait() == WOULD_BLOCK) {
308                LOGE("AudioTrack::start called from thread");
309                return;
310            }
311        }
312        t->mLock.lock();
313     }
314
315    if (android_atomic_or(1, &mActive) == 0) {
316        mNewPosition = mCblk->server + mUpdatePeriod;
317        mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
318        mCblk->waitTimeMs = 0;
319        mCblk->flags &= ~CBLK_DISABLED_ON;
320        if (t != 0) {
321           t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
322        } else {
323            setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
324        }
325
326        if (mCblk->flags & CBLK_INVALID_MSK) {
327            LOGW("start() track %p invalidated, creating a new one", this);
328            // no need to clear the invalid flag as this cblk will not be used anymore
329            // force new track creation
330            status = DEAD_OBJECT;
331        } else {
332            status = mAudioTrack->start();
333        }
334        if (status == DEAD_OBJECT) {
335            LOGV("start() dead IAudioTrack: creating a new one");
336            status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
337                                 mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
338            if (status == NO_ERROR) {
339                status = mAudioTrack->start();
340                if (status == NO_ERROR) {
341                    mNewPosition = mCblk->server + mUpdatePeriod;
342                }
343            }
344        }
345        if (status != NO_ERROR) {
346            LOGV("start() failed");
347            android_atomic_and(~1, &mActive);
348            if (t != 0) {
349                t->requestExit();
350            } else {
351                setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
352            }
353        }
354    }
355
356    if (t != 0) {
357        t->mLock.unlock();
358    }
359}
360
361void AudioTrack::stop()
362{
363    sp<AudioTrackThread> t = mAudioTrackThread;
364
365    LOGV("stop %p", this);
366    if (t != 0) {
367        t->mLock.lock();
368    }
369
370    if (android_atomic_and(~1, &mActive) == 1) {
371        mCblk->cv.signal();
372        mAudioTrack->stop();
373        // Cancel loops (If we are in the middle of a loop, playback
374        // would not stop until loopCount reaches 0).
375        setLoop(0, 0, 0);
376        // the playback head position will reset to 0, so if a marker is set, we need
377        // to activate it again
378        mMarkerReached = false;
379        // Force flush if a shared buffer is used otherwise audioflinger
380        // will not stop before end of buffer is reached.
381        if (mSharedBuffer != 0) {
382            flush();
383        }
384        if (t != 0) {
385            t->requestExit();
386        } else {
387            setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
388        }
389    }
390
391    if (t != 0) {
392        t->mLock.unlock();
393    }
394}
395
396bool AudioTrack::stopped() const
397{
398    return !mActive;
399}
400
401void AudioTrack::flush()
402{
403    LOGV("flush");
404
405    // clear playback marker and periodic update counter
406    mMarkerPosition = 0;
407    mMarkerReached = false;
408    mUpdatePeriod = 0;
409
410
411    if (!mActive) {
412        mAudioTrack->flush();
413        // Release AudioTrack callback thread in case it was waiting for new buffers
414        // in AudioTrack::obtainBuffer()
415        mCblk->cv.signal();
416    }
417}
418
419void AudioTrack::pause()
420{
421    LOGV("pause");
422    if (android_atomic_and(~1, &mActive) == 1) {
423        mAudioTrack->pause();
424    }
425}
426
427void AudioTrack::mute(bool e)
428{
429    mAudioTrack->mute(e);
430    mMuted = e;
431}
432
433bool AudioTrack::muted() const
434{
435    return mMuted;
436}
437
438status_t AudioTrack::setVolume(float left, float right)
439{
440    if (left > 1.0f || right > 1.0f) {
441        return BAD_VALUE;
442    }
443
444    mVolume[LEFT] = left;
445    mVolume[RIGHT] = right;
446
447    // write must be atomic
448    mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
449
450    return NO_ERROR;
451}
452
453void AudioTrack::getVolume(float* left, float* right)
454{
455    if (left != NULL) {
456        *left  = mVolume[LEFT];
457    }
458    if (right != NULL) {
459        *right = mVolume[RIGHT];
460    }
461}
462
463status_t AudioTrack::setAuxEffectSendLevel(float level)
464{
465    LOGV("setAuxEffectSendLevel(%f)", level);
466    if (level > 1.0f) {
467        return BAD_VALUE;
468    }
469
470    mSendLevel = level;
471
472    mCblk->sendLevel = uint16_t(level * 0x1000);
473
474    return NO_ERROR;
475}
476
477void AudioTrack::getAuxEffectSendLevel(float* level)
478{
479    if (level != NULL) {
480        *level  = mSendLevel;
481    }
482}
483
484status_t AudioTrack::setSampleRate(int rate)
485{
486    int afSamplingRate;
487
488    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
489        return NO_INIT;
490    }
491    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
492    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
493
494    mCblk->sampleRate = rate;
495    return NO_ERROR;
496}
497
498uint32_t AudioTrack::getSampleRate()
499{
500    return mCblk->sampleRate;
501}
502
503status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
504{
505    audio_track_cblk_t* cblk = mCblk;
506
507    Mutex::Autolock _l(cblk->lock);
508
509    if (loopCount == 0) {
510        cblk->loopStart = UINT_MAX;
511        cblk->loopEnd = UINT_MAX;
512        cblk->loopCount = 0;
513        mLoopCount = 0;
514        return NO_ERROR;
515    }
516
517    if (loopStart >= loopEnd ||
518        loopEnd - loopStart > cblk->frameCount) {
519        LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
520        return BAD_VALUE;
521    }
522
523    if ((mSharedBuffer != 0) && (loopEnd   > cblk->frameCount)) {
524        LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
525            loopStart, loopEnd, cblk->frameCount);
526        return BAD_VALUE;
527    }
528
529    cblk->loopStart = loopStart;
530    cblk->loopEnd = loopEnd;
531    cblk->loopCount = loopCount;
532    mLoopCount = loopCount;
533
534    return NO_ERROR;
535}
536
537status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
538{
539    if (loopStart != 0) {
540        *loopStart = mCblk->loopStart;
541    }
542    if (loopEnd != 0) {
543        *loopEnd = mCblk->loopEnd;
544    }
545    if (loopCount != 0) {
546        if (mCblk->loopCount < 0) {
547            *loopCount = -1;
548        } else {
549            *loopCount = mCblk->loopCount;
550        }
551    }
552
553    return NO_ERROR;
554}
555
556status_t AudioTrack::setMarkerPosition(uint32_t marker)
557{
558    if (mCbf == 0) return INVALID_OPERATION;
559
560    mMarkerPosition = marker;
561    mMarkerReached = false;
562
563    return NO_ERROR;
564}
565
566status_t AudioTrack::getMarkerPosition(uint32_t *marker)
567{
568    if (marker == 0) return BAD_VALUE;
569
570    *marker = mMarkerPosition;
571
572    return NO_ERROR;
573}
574
575status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
576{
577    if (mCbf == 0) return INVALID_OPERATION;
578
579    uint32_t curPosition;
580    getPosition(&curPosition);
581    mNewPosition = curPosition + updatePeriod;
582    mUpdatePeriod = updatePeriod;
583
584    return NO_ERROR;
585}
586
587status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
588{
589    if (updatePeriod == 0) return BAD_VALUE;
590
591    *updatePeriod = mUpdatePeriod;
592
593    return NO_ERROR;
594}
595
596status_t AudioTrack::setPosition(uint32_t position)
597{
598    Mutex::Autolock _l(mCblk->lock);
599
600    if (!stopped()) return INVALID_OPERATION;
601
602    if (position > mCblk->user) return BAD_VALUE;
603
604    mCblk->server = position;
605    mCblk->flags |= CBLK_FORCEREADY_ON;
606
607    return NO_ERROR;
608}
609
610status_t AudioTrack::getPosition(uint32_t *position)
611{
612    if (position == 0) return BAD_VALUE;
613
614    *position = mCblk->server;
615
616    return NO_ERROR;
617}
618
619status_t AudioTrack::reload()
620{
621    if (!stopped()) return INVALID_OPERATION;
622
623    flush();
624
625    mCblk->stepUser(mCblk->frameCount);
626
627    return NO_ERROR;
628}
629
630audio_io_handle_t AudioTrack::getOutput()
631{
632    return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
633            mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
634}
635
636int AudioTrack::getSessionId()
637{
638    return mSessionId;
639}
640
641status_t AudioTrack::attachAuxEffect(int effectId)
642{
643    LOGV("attachAuxEffect(%d)", effectId);
644    status_t status = mAudioTrack->attachAuxEffect(effectId);
645    if (status == NO_ERROR) {
646        mAuxEffectId = effectId;
647    }
648    return status;
649}
650
651// -------------------------------------------------------------------------
652
653status_t AudioTrack::createTrack(
654        int streamType,
655        uint32_t sampleRate,
656        int format,
657        int channelCount,
658        int frameCount,
659        uint32_t flags,
660        const sp<IMemory>& sharedBuffer,
661        audio_io_handle_t output,
662        bool enforceFrameCount)
663{
664    status_t status;
665    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
666    if (audioFlinger == 0) {
667       LOGE("Could not get audioflinger");
668       return NO_INIT;
669    }
670
671    int afSampleRate;
672    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
673        return NO_INIT;
674    }
675    int afFrameCount;
676    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
677        return NO_INIT;
678    }
679    uint32_t afLatency;
680    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
681        return NO_INIT;
682    }
683
684    mNotificationFramesAct = mNotificationFramesReq;
685    if (!AudioSystem::isLinearPCM(format)) {
686        if (sharedBuffer != 0) {
687            frameCount = sharedBuffer->size();
688        }
689    } else {
690        // Ensure that buffer depth covers at least audio hardware latency
691        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
692        if (minBufCount < 2) minBufCount = 2;
693
694        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
695
696        if (sharedBuffer == 0) {
697            if (frameCount == 0) {
698                frameCount = minFrameCount;
699            }
700            if (mNotificationFramesAct == 0) {
701                mNotificationFramesAct = frameCount/2;
702            }
703            // Make sure that application is notified with sufficient margin
704            // before underrun
705            if (mNotificationFramesAct > (uint32_t)frameCount/2) {
706                mNotificationFramesAct = frameCount/2;
707            }
708            if (frameCount < minFrameCount) {
709                if (enforceFrameCount) {
710                    LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
711                    return BAD_VALUE;
712                } else {
713                    frameCount = minFrameCount;
714                }
715            }
716        } else {
717            // Ensure that buffer alignment matches channelcount
718            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
719                LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
720                return BAD_VALUE;
721            }
722            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
723        }
724    }
725
726    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
727                                                      streamType,
728                                                      sampleRate,
729                                                      format,
730                                                      channelCount,
731                                                      frameCount,
732                                                      ((uint16_t)flags) << 16,
733                                                      sharedBuffer,
734                                                      output,
735                                                      &mSessionId,
736                                                      &status);
737
738    if (track == 0) {
739        LOGE("AudioFlinger could not create track, status: %d", status);
740        return status;
741    }
742    sp<IMemory> cblk = track->getCblk();
743    if (cblk == 0) {
744        LOGE("Could not get control block");
745        return NO_INIT;
746    }
747    mAudioTrack.clear();
748    mAudioTrack = track;
749    mCblkMemory.clear();
750    mCblkMemory = cblk;
751    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
752    mCblk->flags |= CBLK_DIRECTION_OUT;
753    if (sharedBuffer == 0) {
754        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
755    } else {
756        mCblk->buffers = sharedBuffer->pointer();
757         // Force buffer full condition as data is already present in shared memory
758        mCblk->stepUser(mCblk->frameCount);
759    }
760
761    mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
762    mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
763    mAudioTrack->attachAuxEffect(mAuxEffectId);
764    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
765    mCblk->waitTimeMs = 0;
766    mRemainingFrames = mNotificationFramesAct;
767    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
768    return NO_ERROR;
769}
770
771status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
772{
773    int active;
774    status_t result;
775    audio_track_cblk_t* cblk = mCblk;
776    uint32_t framesReq = audioBuffer->frameCount;
777    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
778
779    audioBuffer->frameCount  = 0;
780    audioBuffer->size = 0;
781
782    uint32_t framesAvail = cblk->framesAvailable();
783
784    if (framesAvail == 0) {
785        cblk->lock.lock();
786        goto start_loop_here;
787        while (framesAvail == 0) {
788            active = mActive;
789            if (UNLIKELY(!active)) {
790                LOGV("Not active and NO_MORE_BUFFERS");
791                cblk->lock.unlock();
792                return NO_MORE_BUFFERS;
793            }
794            if (UNLIKELY(!waitCount)) {
795                cblk->lock.unlock();
796                return WOULD_BLOCK;
797            }
798            if (!(cblk->flags & CBLK_INVALID_MSK)) {
799                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
800            }
801            if (cblk->flags & CBLK_INVALID_MSK) {
802                LOGW("obtainBuffer() track %p invalidated, creating a new one", this);
803                // no need to clear the invalid flag as this cblk will not be used anymore
804                cblk->lock.unlock();
805                goto create_new_track;
806            }
807            if (__builtin_expect(result!=NO_ERROR, false)) {
808                cblk->waitTimeMs += waitTimeMs;
809                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
810                    // timing out when a loop has been set and we have already written upto loop end
811                    // is a normal condition: no need to wake AudioFlinger up.
812                    if (cblk->user < cblk->loopEnd) {
813                        LOGW(   "obtainBuffer timed out (is the CPU pegged?) %p "
814                                "user=%08x, server=%08x", this, cblk->user, cblk->server);
815                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
816                        cblk->lock.unlock();
817                        result = mAudioTrack->start();
818                        if (result == DEAD_OBJECT) {
819                            LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
820create_new_track:
821                            result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
822                                                 mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
823                            if (result == NO_ERROR) {
824                                cblk = mCblk;
825                                cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
826                                mAudioTrack->start();
827                            }
828                        }
829                        cblk->lock.lock();
830                    }
831                    cblk->waitTimeMs = 0;
832                }
833
834                if (--waitCount == 0) {
835                    cblk->lock.unlock();
836                    return TIMED_OUT;
837                }
838            }
839            // read the server count again
840        start_loop_here:
841            framesAvail = cblk->framesAvailable_l();
842        }
843        cblk->lock.unlock();
844    }
845
846    // restart track if it was disabled by audioflinger due to previous underrun
847    if (cblk->flags & CBLK_DISABLED_MSK) {
848        cblk->flags &= ~CBLK_DISABLED_ON;
849        LOGW("obtainBuffer() track %p disabled, restarting", this);
850        mAudioTrack->start();
851    }
852
853    cblk->waitTimeMs = 0;
854
855    if (framesReq > framesAvail) {
856        framesReq = framesAvail;
857    }
858
859    uint32_t u = cblk->user;
860    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
861
862    if (u + framesReq > bufferEnd) {
863        framesReq = bufferEnd - u;
864    }
865
866    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
867    audioBuffer->channelCount = mChannelCount;
868    audioBuffer->frameCount = framesReq;
869    audioBuffer->size = framesReq * cblk->frameSize;
870    if (AudioSystem::isLinearPCM(mFormat)) {
871        audioBuffer->format = AudioSystem::PCM_16_BIT;
872    } else {
873        audioBuffer->format = mFormat;
874    }
875    audioBuffer->raw = (int8_t *)cblk->buffer(u);
876    active = mActive;
877    return active ? status_t(NO_ERROR) : status_t(STOPPED);
878}
879
880void AudioTrack::releaseBuffer(Buffer* audioBuffer)
881{
882    audio_track_cblk_t* cblk = mCblk;
883    cblk->stepUser(audioBuffer->frameCount);
884}
885
886// -------------------------------------------------------------------------
887
888ssize_t AudioTrack::write(const void* buffer, size_t userSize)
889{
890
891    if (mSharedBuffer != 0) return INVALID_OPERATION;
892
893    if (ssize_t(userSize) < 0) {
894        // sanity-check. user is most-likely passing an error code.
895        LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
896                buffer, userSize, userSize);
897        return BAD_VALUE;
898    }
899
900    LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
901
902    ssize_t written = 0;
903    const int8_t *src = (const int8_t *)buffer;
904    Buffer audioBuffer;
905
906    do {
907        audioBuffer.frameCount = userSize/frameSize();
908
909        // Calling obtainBuffer() with a negative wait count causes
910        // an (almost) infinite wait time.
911        status_t err = obtainBuffer(&audioBuffer, -1);
912        if (err < 0) {
913            // out of buffers, return #bytes written
914            if (err == status_t(NO_MORE_BUFFERS))
915                break;
916            return ssize_t(err);
917        }
918
919        size_t toWrite;
920
921        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
922            // Divide capacity by 2 to take expansion into account
923            toWrite = audioBuffer.size>>1;
924            // 8 to 16 bit conversion
925            int count = toWrite;
926            int16_t *dst = (int16_t *)(audioBuffer.i8);
927            while(count--) {
928                *dst++ = (int16_t)(*src++^0x80) << 8;
929            }
930        } else {
931            toWrite = audioBuffer.size;
932            memcpy(audioBuffer.i8, src, toWrite);
933            src += toWrite;
934        }
935        userSize -= toWrite;
936        written += toWrite;
937
938        releaseBuffer(&audioBuffer);
939    } while (userSize);
940
941    return written;
942}
943
944// -------------------------------------------------------------------------
945
946bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
947{
948    Buffer audioBuffer;
949    uint32_t frames;
950    size_t writtenSize;
951
952    // Manage underrun callback
953    if (mActive && (mCblk->framesReady() == 0)) {
954        LOGV("Underrun user: %x, server: %x, flags %04x", mCblk->user, mCblk->server, mCblk->flags);
955        if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) {
956            mCbf(EVENT_UNDERRUN, mUserData, 0);
957            if (mCblk->server == mCblk->frameCount) {
958                mCbf(EVENT_BUFFER_END, mUserData, 0);
959            }
960            mCblk->flags |= CBLK_UNDERRUN_ON;
961            if (mSharedBuffer != 0) return false;
962        }
963    }
964
965    // Manage loop end callback
966    while (mLoopCount > mCblk->loopCount) {
967        int loopCount = -1;
968        mLoopCount--;
969        if (mLoopCount >= 0) loopCount = mLoopCount;
970
971        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
972    }
973
974    // Manage marker callback
975    if (!mMarkerReached && (mMarkerPosition > 0)) {
976        if (mCblk->server >= mMarkerPosition) {
977            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
978            mMarkerReached = true;
979        }
980    }
981
982    // Manage new position callback
983    if (mUpdatePeriod > 0) {
984        while (mCblk->server >= mNewPosition) {
985            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
986            mNewPosition += mUpdatePeriod;
987        }
988    }
989
990    // If Shared buffer is used, no data is requested from client.
991    if (mSharedBuffer != 0) {
992        frames = 0;
993    } else {
994        frames = mRemainingFrames;
995    }
996
997    do {
998
999        audioBuffer.frameCount = frames;
1000
1001        // Calling obtainBuffer() with a wait count of 1
1002        // limits wait time to WAIT_PERIOD_MS. This prevents from being
1003        // stuck here not being able to handle timed events (position, markers, loops).
1004        status_t err = obtainBuffer(&audioBuffer, 1);
1005        if (err < NO_ERROR) {
1006            if (err != TIMED_OUT) {
1007                LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
1008                return false;
1009            }
1010            break;
1011        }
1012        if (err == status_t(STOPPED)) return false;
1013
1014        // Divide buffer size by 2 to take into account the expansion
1015        // due to 8 to 16 bit conversion: the callback must fill only half
1016        // of the destination buffer
1017        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
1018            audioBuffer.size >>= 1;
1019        }
1020
1021        size_t reqSize = audioBuffer.size;
1022        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1023        writtenSize = audioBuffer.size;
1024
1025        // Sanity check on returned size
1026        if (ssize_t(writtenSize) <= 0) {
1027            // The callback is done filling buffers
1028            // Keep this thread going to handle timed events and
1029            // still try to get more data in intervals of WAIT_PERIOD_MS
1030            // but don't just loop and block the CPU, so wait
1031            usleep(WAIT_PERIOD_MS*1000);
1032            break;
1033        }
1034        if (writtenSize > reqSize) writtenSize = reqSize;
1035
1036        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
1037            // 8 to 16 bit conversion
1038            const int8_t *src = audioBuffer.i8 + writtenSize-1;
1039            int count = writtenSize;
1040            int16_t *dst = audioBuffer.i16 + writtenSize-1;
1041            while(count--) {
1042                *dst-- = (int16_t)(*src--^0x80) << 8;
1043            }
1044            writtenSize <<= 1;
1045        }
1046
1047        audioBuffer.size = writtenSize;
1048        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
1049        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sampel size of
1050        // 16 bit.
1051        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
1052
1053        frames -= audioBuffer.frameCount;
1054
1055        releaseBuffer(&audioBuffer);
1056    }
1057    while (frames);
1058
1059    if (frames == 0) {
1060        mRemainingFrames = mNotificationFramesAct;
1061    } else {
1062        mRemainingFrames = frames;
1063    }
1064    return true;
1065}
1066
1067status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1068{
1069
1070    const size_t SIZE = 256;
1071    char buffer[SIZE];
1072    String8 result;
1073
1074    result.append(" AudioTrack::dump\n");
1075    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
1076    result.append(buffer);
1077    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
1078    result.append(buffer);
1079    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
1080    result.append(buffer);
1081    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1082    result.append(buffer);
1083    ::write(fd, result.string(), result.size());
1084    return NO_ERROR;
1085}
1086
1087// =========================================================================
1088
1089AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1090    : Thread(bCanCallJava), mReceiver(receiver)
1091{
1092}
1093
1094bool AudioTrack::AudioTrackThread::threadLoop()
1095{
1096    return mReceiver.processAudioBuffer(this);
1097}
1098
1099status_t AudioTrack::AudioTrackThread::readyToRun()
1100{
1101    return NO_ERROR;
1102}
1103
1104void AudioTrack::AudioTrackThread::onFirstRef()
1105{
1106}
1107
1108// =========================================================================
1109
1110audio_track_cblk_t::audio_track_cblk_t()
1111    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1112    userBase(0), serverBase(0), buffers(0), frameCount(0),
1113    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
1114    flags(0), sendLevel(0)
1115{
1116}
1117
1118uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1119{
1120    uint32_t u = this->user;
1121
1122    u += frameCount;
1123    // Ensure that user is never ahead of server for AudioRecord
1124    if (flags & CBLK_DIRECTION_MSK) {
1125        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1126        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1127            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1128        }
1129    } else if (u > this->server) {
1130        LOGW("stepServer occured after track reset");
1131        u = this->server;
1132    }
1133
1134    if (u >= userBase + this->frameCount) {
1135        userBase += this->frameCount;
1136    }
1137
1138    this->user = u;
1139
1140    // Clear flow control error condition as new data has been written/read to/from buffer.
1141    flags &= ~CBLK_UNDERRUN_MSK;
1142
1143    return u;
1144}
1145
1146bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1147{
1148    // the code below simulates lock-with-timeout
1149    // we MUST do this to protect the AudioFlinger server
1150    // as this lock is shared with the client.
1151    status_t err;
1152
1153    err = lock.tryLock();
1154    if (err == -EBUSY) { // just wait a bit
1155        usleep(1000);
1156        err = lock.tryLock();
1157    }
1158    if (err != NO_ERROR) {
1159        // probably, the client just died.
1160        return false;
1161    }
1162
1163    uint32_t s = this->server;
1164
1165    s += frameCount;
1166    if (flags & CBLK_DIRECTION_MSK) {
1167        // Mark that we have read the first buffer so that next time stepUser() is called
1168        // we switch to normal obtainBuffer() timeout period
1169        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1170            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1171        }
1172        // It is possible that we receive a flush()
1173        // while the mixer is processing a block: in this case,
1174        // stepServer() is called After the flush() has reset u & s and
1175        // we have s > u
1176        if (s > this->user) {
1177            LOGW("stepServer occured after track reset");
1178            s = this->user;
1179        }
1180    }
1181
1182    if (s >= loopEnd) {
1183        LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1184        s = loopStart;
1185        if (--loopCount == 0) {
1186            loopEnd = UINT_MAX;
1187            loopStart = UINT_MAX;
1188        }
1189    }
1190    if (s >= serverBase + this->frameCount) {
1191        serverBase += this->frameCount;
1192    }
1193
1194    this->server = s;
1195
1196    cv.signal();
1197    lock.unlock();
1198    return true;
1199}
1200
1201void* audio_track_cblk_t::buffer(uint32_t offset) const
1202{
1203    return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
1204}
1205
1206uint32_t audio_track_cblk_t::framesAvailable()
1207{
1208    Mutex::Autolock _l(lock);
1209    return framesAvailable_l();
1210}
1211
1212uint32_t audio_track_cblk_t::framesAvailable_l()
1213{
1214    uint32_t u = this->user;
1215    uint32_t s = this->server;
1216
1217    if (flags & CBLK_DIRECTION_MSK) {
1218        uint32_t limit = (s < loopStart) ? s : loopStart;
1219        return limit + frameCount - u;
1220    } else {
1221        return frameCount + u - s;
1222    }
1223}
1224
1225uint32_t audio_track_cblk_t::framesReady()
1226{
1227    uint32_t u = this->user;
1228    uint32_t s = this->server;
1229
1230    if (flags & CBLK_DIRECTION_MSK) {
1231        if (u < loopEnd) {
1232            return u - s;
1233        } else {
1234            Mutex::Autolock _l(lock);
1235            if (loopCount >= 0) {
1236                return (loopEnd - loopStart)*loopCount + u - s;
1237            } else {
1238                return UINT_MAX;
1239            }
1240        }
1241    } else {
1242        return s - u;
1243    }
1244}
1245
1246// -------------------------------------------------------------------------
1247
1248}; // namespace android
1249
1250