AudioTrack.cpp revision 480b46802bef1371d5caa16ad5454fce04769c57
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
48namespace android {
49// ---------------------------------------------------------------------------
50
51// static
52status_t AudioTrack::getMinFrameCount(
53        int* frameCount,
54        audio_stream_type_t streamType,
55        uint32_t sampleRate)
56{
57    int afSampleRate;
58    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
59        return NO_INIT;
60    }
61    int afFrameCount;
62    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
63        return NO_INIT;
64    }
65    uint32_t afLatency;
66    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
67        return NO_INIT;
68    }
69
70    // Ensure that buffer depth covers at least audio hardware latency
71    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
72    if (minBufCount < 2) minBufCount = 2;
73
74    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
75              afFrameCount * minBufCount * sampleRate / afSampleRate;
76    return NO_ERROR;
77}
78
79// ---------------------------------------------------------------------------
80
81AudioTrack::AudioTrack()
82    : mStatus(NO_INIT),
83      mIsTimed(false),
84      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
85      mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
86{
87}
88
89AudioTrack::AudioTrack(
90        audio_stream_type_t streamType,
91        uint32_t sampleRate,
92        audio_format_t format,
93        int channelMask,
94        int frameCount,
95        uint32_t flags,
96        callback_t cbf,
97        void* user,
98        int notificationFrames,
99        int sessionId)
100    : mStatus(NO_INIT),
101      mIsTimed(false),
102      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
103      mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
104{
105    mStatus = set(streamType, sampleRate, format, channelMask,
106            frameCount, flags, cbf, user, notificationFrames,
107            0, false, sessionId);
108}
109
110AudioTrack::AudioTrack(
111        int streamType,
112        uint32_t sampleRate,
113        int format,
114        int channelMask,
115        int frameCount,
116        uint32_t flags,
117        callback_t cbf,
118        void* user,
119        int notificationFrames,
120        int sessionId)
121    : mStatus(NO_INIT),
122      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
123{
124    mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
125            frameCount, flags, cbf, user, notificationFrames,
126            0, false, sessionId);
127}
128
129AudioTrack::AudioTrack(
130        audio_stream_type_t streamType,
131        uint32_t sampleRate,
132        audio_format_t format,
133        int channelMask,
134        const sp<IMemory>& sharedBuffer,
135        uint32_t flags,
136        callback_t cbf,
137        void* user,
138        int notificationFrames,
139        int sessionId)
140    : mStatus(NO_INIT),
141      mIsTimed(false),
142      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
143      mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
144{
145    mStatus = set(streamType, sampleRate, format, channelMask,
146            0, flags, cbf, user, notificationFrames,
147            sharedBuffer, false, sessionId);
148}
149
150AudioTrack::~AudioTrack()
151{
152    ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
153
154    if (mStatus == NO_ERROR) {
155        // Make sure that callback function exits in the case where
156        // it is looping on buffer full condition in obtainBuffer().
157        // Otherwise the callback thread will never exit.
158        stop();
159        if (mAudioTrackThread != 0) {
160            mAudioTrackThread->requestExitAndWait();
161            mAudioTrackThread.clear();
162        }
163        mAudioTrack.clear();
164        IPCThreadState::self()->flushCommands();
165        AudioSystem::releaseAudioSessionId(mSessionId);
166    }
167}
168
169status_t AudioTrack::set(
170        audio_stream_type_t streamType,
171        uint32_t sampleRate,
172        audio_format_t format,
173        int channelMask,
174        int frameCount,
175        uint32_t flags,
176        callback_t cbf,
177        void* user,
178        int notificationFrames,
179        const sp<IMemory>& sharedBuffer,
180        bool threadCanCallJava,
181        int sessionId)
182{
183
184    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
185
186    AutoMutex lock(mLock);
187    if (mAudioTrack != 0) {
188        ALOGE("Track already in use");
189        return INVALID_OPERATION;
190    }
191
192    int afSampleRate;
193    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
194        return NO_INIT;
195    }
196    uint32_t afLatency;
197    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
198        return NO_INIT;
199    }
200
201    // handle default values first.
202    if (streamType == AUDIO_STREAM_DEFAULT) {
203        streamType = AUDIO_STREAM_MUSIC;
204    }
205    if (sampleRate == 0) {
206        sampleRate = afSampleRate;
207    }
208    // these below should probably come from the audioFlinger too...
209    if (format == AUDIO_FORMAT_DEFAULT) {
210        format = AUDIO_FORMAT_PCM_16_BIT;
211    }
212    if (channelMask == 0) {
213        channelMask = AUDIO_CHANNEL_OUT_STEREO;
214    }
215
216    // validate parameters
217    if (!audio_is_valid_format(format)) {
218        ALOGE("Invalid format");
219        return BAD_VALUE;
220    }
221
222    // force direct flag if format is not linear PCM
223    if (!audio_is_linear_pcm(format)) {
224        flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT;
225    }
226
227    if (!audio_is_output_channel(channelMask)) {
228        ALOGE("Invalid channel mask");
229        return BAD_VALUE;
230    }
231    uint32_t channelCount = popcount(channelMask);
232
233    audio_io_handle_t output = AudioSystem::getOutput(
234                                    streamType,
235                                    sampleRate, format, channelMask,
236                                    (audio_policy_output_flags_t)flags);
237
238    if (output == 0) {
239        ALOGE("Could not get audio output for stream type %d", streamType);
240        return BAD_VALUE;
241    }
242
243    mVolume[LEFT] = 1.0f;
244    mVolume[RIGHT] = 1.0f;
245    mSendLevel = 0.0f;
246    mFrameCount = frameCount;
247    mNotificationFramesReq = notificationFrames;
248    mSessionId = sessionId;
249    mAuxEffectId = 0;
250
251    // create the IAudioTrack
252    status_t status = createTrack_l(streamType,
253                                  sampleRate,
254                                  format,
255                                  (uint32_t)channelMask,
256                                  frameCount,
257                                  flags,
258                                  sharedBuffer,
259                                  output,
260                                  true);
261
262    if (status != NO_ERROR) {
263        return status;
264    }
265
266    if (cbf != NULL) {
267        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
268    }
269
270    mStatus = NO_ERROR;
271
272    mStreamType = streamType;
273    mFormat = format;
274    mChannelMask = (uint32_t)channelMask;
275    mChannelCount = channelCount;
276    mSharedBuffer = sharedBuffer;
277    mMuted = false;
278    mActive = false;
279    mCbf = cbf;
280    mUserData = user;
281    mLoopCount = 0;
282    mMarkerPosition = 0;
283    mMarkerReached = false;
284    mNewPosition = 0;
285    mUpdatePeriod = 0;
286    mFlushed = false;
287    mFlags = flags;
288    AudioSystem::acquireAudioSessionId(mSessionId);
289    mRestoreStatus = NO_ERROR;
290    return NO_ERROR;
291}
292
293status_t AudioTrack::initCheck() const
294{
295    return mStatus;
296}
297
298// -------------------------------------------------------------------------
299
300uint32_t AudioTrack::latency() const
301{
302    return mLatency;
303}
304
305audio_stream_type_t AudioTrack::streamType() const
306{
307    return mStreamType;
308}
309
310audio_format_t AudioTrack::format() const
311{
312    return mFormat;
313}
314
315int AudioTrack::channelCount() const
316{
317    return mChannelCount;
318}
319
320uint32_t AudioTrack::frameCount() const
321{
322    return mCblk->frameCount;
323}
324
325size_t AudioTrack::frameSize() const
326{
327    if (audio_is_linear_pcm(mFormat)) {
328        return channelCount()*audio_bytes_per_sample(mFormat);
329    } else {
330        return sizeof(uint8_t);
331    }
332}
333
334sp<IMemory>& AudioTrack::sharedBuffer()
335{
336    return mSharedBuffer;
337}
338
339// -------------------------------------------------------------------------
340
341void AudioTrack::start()
342{
343    sp<AudioTrackThread> t = mAudioTrackThread;
344    status_t status = NO_ERROR;
345
346    ALOGV("start %p", this);
347    if (t != 0) {
348        if (t->exitPending()) {
349            if (t->requestExitAndWait() == WOULD_BLOCK) {
350                ALOGE("AudioTrack::start called from thread");
351                return;
352            }
353        }
354     }
355
356    AutoMutex lock(mLock);
357    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
358    // while we are accessing the cblk
359    sp <IAudioTrack> audioTrack = mAudioTrack;
360    sp <IMemory> iMem = mCblkMemory;
361    audio_track_cblk_t* cblk = mCblk;
362
363    if (!mActive) {
364        mFlushed = false;
365        mActive = true;
366        mNewPosition = cblk->server + mUpdatePeriod;
367        cblk->lock.lock();
368        cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
369        cblk->waitTimeMs = 0;
370        android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
371        pid_t tid;
372        if (t != 0) {
373            t->run("AudioTrack", ANDROID_PRIORITY_AUDIO);
374            tid = t->getTid();  // pid_t is unknown until run()
375            ALOGV("getTid=%d", tid);
376            if (tid == -1) {
377                tid = 0;
378            }
379        } else {
380            mPreviousPriority = getpriority(PRIO_PROCESS, 0);
381            mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
382            androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
383            tid = 0;    // not gettid()
384        }
385
386        ALOGV("start %p before lock cblk %p", this, mCblk);
387        if (!(cblk->flags & CBLK_INVALID_MSK)) {
388            cblk->lock.unlock();
389            ALOGV("mAudioTrack->start(tid=%d)", tid);
390            status = mAudioTrack->start(tid);
391            cblk->lock.lock();
392            if (status == DEAD_OBJECT) {
393                android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
394            }
395        }
396        if (cblk->flags & CBLK_INVALID_MSK) {
397            status = restoreTrack_l(cblk, true);
398        }
399        cblk->lock.unlock();
400        if (status != NO_ERROR) {
401            ALOGV("start() failed");
402            mActive = false;
403            if (t != 0) {
404                t->requestExit();
405            } else {
406                setpriority(PRIO_PROCESS, 0, mPreviousPriority);
407                androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
408            }
409        }
410    }
411
412}
413
414void AudioTrack::stop()
415{
416    sp<AudioTrackThread> t = mAudioTrackThread;
417
418    ALOGV("stop %p", this);
419
420    AutoMutex lock(mLock);
421    if (mActive) {
422        mActive = false;
423        mCblk->cv.signal();
424        mAudioTrack->stop();
425        // Cancel loops (If we are in the middle of a loop, playback
426        // would not stop until loopCount reaches 0).
427        setLoop_l(0, 0, 0);
428        // the playback head position will reset to 0, so if a marker is set, we need
429        // to activate it again
430        mMarkerReached = false;
431        // Force flush if a shared buffer is used otherwise audioflinger
432        // will not stop before end of buffer is reached.
433        if (mSharedBuffer != 0) {
434            flush_l();
435        }
436        if (t != 0) {
437            t->requestExit();
438        } else {
439            setpriority(PRIO_PROCESS, 0, mPreviousPriority);
440            androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
441        }
442    }
443
444}
445
446bool AudioTrack::stopped() const
447{
448    AutoMutex lock(mLock);
449    return stopped_l();
450}
451
452void AudioTrack::flush()
453{
454    AutoMutex lock(mLock);
455    flush_l();
456}
457
458// must be called with mLock held
459void AudioTrack::flush_l()
460{
461    ALOGV("flush");
462
463    // clear playback marker and periodic update counter
464    mMarkerPosition = 0;
465    mMarkerReached = false;
466    mUpdatePeriod = 0;
467
468    if (!mActive) {
469        mFlushed = true;
470        mAudioTrack->flush();
471        // Release AudioTrack callback thread in case it was waiting for new buffers
472        // in AudioTrack::obtainBuffer()
473        mCblk->cv.signal();
474    }
475}
476
477void AudioTrack::pause()
478{
479    ALOGV("pause");
480    AutoMutex lock(mLock);
481    if (mActive) {
482        mActive = false;
483        mAudioTrack->pause();
484    }
485}
486
487void AudioTrack::mute(bool e)
488{
489    mAudioTrack->mute(e);
490    mMuted = e;
491}
492
493bool AudioTrack::muted() const
494{
495    return mMuted;
496}
497
498status_t AudioTrack::setVolume(float left, float right)
499{
500    if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
501        return BAD_VALUE;
502    }
503
504    AutoMutex lock(mLock);
505    mVolume[LEFT] = left;
506    mVolume[RIGHT] = right;
507
508    mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
509
510    return NO_ERROR;
511}
512
513void AudioTrack::getVolume(float* left, float* right) const
514{
515    if (left != NULL) {
516        *left  = mVolume[LEFT];
517    }
518    if (right != NULL) {
519        *right = mVolume[RIGHT];
520    }
521}
522
523status_t AudioTrack::setAuxEffectSendLevel(float level)
524{
525    ALOGV("setAuxEffectSendLevel(%f)", level);
526    if (level < 0.0f || level > 1.0f) {
527        return BAD_VALUE;
528    }
529    AutoMutex lock(mLock);
530
531    mSendLevel = level;
532
533    mCblk->setSendLevel(level);
534
535    return NO_ERROR;
536}
537
538void AudioTrack::getAuxEffectSendLevel(float* level) const
539{
540    if (level != NULL) {
541        *level  = mSendLevel;
542    }
543}
544
545status_t AudioTrack::setSampleRate(int rate)
546{
547    int afSamplingRate;
548
549    if (mIsTimed) {
550        return INVALID_OPERATION;
551    }
552
553    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
554        return NO_INIT;
555    }
556    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
557    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
558
559    AutoMutex lock(mLock);
560    mCblk->sampleRate = rate;
561    return NO_ERROR;
562}
563
564uint32_t AudioTrack::getSampleRate() const
565{
566    if (mIsTimed) {
567        return INVALID_OPERATION;
568    }
569
570    AutoMutex lock(mLock);
571    return mCblk->sampleRate;
572}
573
574status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
575{
576    AutoMutex lock(mLock);
577    return setLoop_l(loopStart, loopEnd, loopCount);
578}
579
580// must be called with mLock held
581status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
582{
583    audio_track_cblk_t* cblk = mCblk;
584
585    Mutex::Autolock _l(cblk->lock);
586
587    if (loopCount == 0) {
588        cblk->loopStart = UINT_MAX;
589        cblk->loopEnd = UINT_MAX;
590        cblk->loopCount = 0;
591        mLoopCount = 0;
592        return NO_ERROR;
593    }
594
595    if (mIsTimed) {
596        return INVALID_OPERATION;
597    }
598
599    if (loopStart >= loopEnd ||
600        loopEnd - loopStart > cblk->frameCount ||
601        cblk->server > loopStart) {
602        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
603        return BAD_VALUE;
604    }
605
606    if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
607        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
608            loopStart, loopEnd, cblk->frameCount);
609        return BAD_VALUE;
610    }
611
612    cblk->loopStart = loopStart;
613    cblk->loopEnd = loopEnd;
614    cblk->loopCount = loopCount;
615    mLoopCount = loopCount;
616
617    return NO_ERROR;
618}
619
620status_t AudioTrack::setMarkerPosition(uint32_t marker)
621{
622    if (mCbf == NULL) return INVALID_OPERATION;
623
624    mMarkerPosition = marker;
625    mMarkerReached = false;
626
627    return NO_ERROR;
628}
629
630status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
631{
632    if (marker == NULL) return BAD_VALUE;
633
634    *marker = mMarkerPosition;
635
636    return NO_ERROR;
637}
638
639status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
640{
641    if (mCbf == NULL) return INVALID_OPERATION;
642
643    uint32_t curPosition;
644    getPosition(&curPosition);
645    mNewPosition = curPosition + updatePeriod;
646    mUpdatePeriod = updatePeriod;
647
648    return NO_ERROR;
649}
650
651status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
652{
653    if (updatePeriod == NULL) return BAD_VALUE;
654
655    *updatePeriod = mUpdatePeriod;
656
657    return NO_ERROR;
658}
659
660status_t AudioTrack::setPosition(uint32_t position)
661{
662    if (mIsTimed) return INVALID_OPERATION;
663
664    AutoMutex lock(mLock);
665
666    if (!stopped_l()) return INVALID_OPERATION;
667
668    Mutex::Autolock _l(mCblk->lock);
669
670    if (position > mCblk->user) return BAD_VALUE;
671
672    mCblk->server = position;
673    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
674
675    return NO_ERROR;
676}
677
678status_t AudioTrack::getPosition(uint32_t *position)
679{
680    if (position == NULL) return BAD_VALUE;
681    AutoMutex lock(mLock);
682    *position = mFlushed ? 0 : mCblk->server;
683
684    return NO_ERROR;
685}
686
687status_t AudioTrack::reload()
688{
689    AutoMutex lock(mLock);
690
691    if (!stopped_l()) return INVALID_OPERATION;
692
693    flush_l();
694
695    mCblk->stepUser(mCblk->frameCount);
696
697    return NO_ERROR;
698}
699
700audio_io_handle_t AudioTrack::getOutput()
701{
702    AutoMutex lock(mLock);
703    return getOutput_l();
704}
705
706// must be called with mLock held
707audio_io_handle_t AudioTrack::getOutput_l()
708{
709    return AudioSystem::getOutput(mStreamType,
710            mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
711}
712
713int AudioTrack::getSessionId() const
714{
715    return mSessionId;
716}
717
718status_t AudioTrack::attachAuxEffect(int effectId)
719{
720    ALOGV("attachAuxEffect(%d)", effectId);
721    status_t status = mAudioTrack->attachAuxEffect(effectId);
722    if (status == NO_ERROR) {
723        mAuxEffectId = effectId;
724    }
725    return status;
726}
727
728// -------------------------------------------------------------------------
729
730// must be called with mLock held
731status_t AudioTrack::createTrack_l(
732        audio_stream_type_t streamType,
733        uint32_t sampleRate,
734        audio_format_t format,
735        uint32_t channelMask,
736        int frameCount,
737        uint32_t flags,
738        const sp<IMemory>& sharedBuffer,
739        audio_io_handle_t output,
740        bool enforceFrameCount)
741{
742    status_t status;
743    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
744    if (audioFlinger == 0) {
745       ALOGE("Could not get audioflinger");
746       return NO_INIT;
747    }
748
749    int afSampleRate;
750    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
751        return NO_INIT;
752    }
753    int afFrameCount;
754    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
755        return NO_INIT;
756    }
757    uint32_t afLatency;
758    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
759        return NO_INIT;
760    }
761
762    mNotificationFramesAct = mNotificationFramesReq;
763    if (!audio_is_linear_pcm(format)) {
764        if (sharedBuffer != 0) {
765            frameCount = sharedBuffer->size();
766        }
767    } else {
768        // Ensure that buffer depth covers at least audio hardware latency
769        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
770        if (minBufCount < 2) minBufCount = 2;
771
772        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
773
774        if (sharedBuffer == 0) {
775            if (frameCount == 0) {
776                frameCount = minFrameCount;
777            }
778            if (mNotificationFramesAct == 0) {
779                mNotificationFramesAct = frameCount/2;
780            }
781            // Make sure that application is notified with sufficient margin
782            // before underrun
783            if (mNotificationFramesAct > (uint32_t)frameCount/2) {
784                mNotificationFramesAct = frameCount/2;
785            }
786            if (frameCount < minFrameCount) {
787                if (enforceFrameCount) {
788                    ALOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
789                    return BAD_VALUE;
790                } else {
791                    frameCount = minFrameCount;
792                }
793            }
794        } else {
795            // Ensure that buffer alignment matches channelCount
796            int channelCount = popcount(channelMask);
797            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
798                ALOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
799                return BAD_VALUE;
800            }
801            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
802        }
803    }
804
805    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
806                                                      streamType,
807                                                      sampleRate,
808                                                      format,
809                                                      channelMask,
810                                                      frameCount,
811                                                      ((uint16_t)flags) << 16,
812                                                      sharedBuffer,
813                                                      output,
814                                                      mIsTimed,
815                                                      &mSessionId,
816                                                      &status);
817
818    if (track == 0) {
819        ALOGE("AudioFlinger could not create track, status: %d", status);
820        return status;
821    }
822    sp<IMemory> cblk = track->getCblk();
823    if (cblk == 0) {
824        ALOGE("Could not get control block");
825        return NO_INIT;
826    }
827    mAudioTrack = track;
828    mCblkMemory = cblk;
829    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
830    android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
831    if (sharedBuffer == 0) {
832        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
833    } else {
834        mCblk->buffers = sharedBuffer->pointer();
835         // Force buffer full condition as data is already present in shared memory
836        mCblk->stepUser(mCblk->frameCount);
837    }
838
839    mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
840    mCblk->setSendLevel(mSendLevel);
841    mAudioTrack->attachAuxEffect(mAuxEffectId);
842    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
843    mCblk->waitTimeMs = 0;
844    mRemainingFrames = mNotificationFramesAct;
845    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
846    return NO_ERROR;
847}
848
849status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
850{
851    AutoMutex lock(mLock);
852    bool active;
853    status_t result = NO_ERROR;
854    audio_track_cblk_t* cblk = mCblk;
855    uint32_t framesReq = audioBuffer->frameCount;
856    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
857
858    audioBuffer->frameCount  = 0;
859    audioBuffer->size = 0;
860
861    uint32_t framesAvail = cblk->framesAvailable();
862
863    cblk->lock.lock();
864    if (cblk->flags & CBLK_INVALID_MSK) {
865        goto create_new_track;
866    }
867    cblk->lock.unlock();
868
869    if (framesAvail == 0) {
870        cblk->lock.lock();
871        goto start_loop_here;
872        while (framesAvail == 0) {
873            active = mActive;
874            if (CC_UNLIKELY(!active)) {
875                ALOGV("Not active and NO_MORE_BUFFERS");
876                cblk->lock.unlock();
877                return NO_MORE_BUFFERS;
878            }
879            if (CC_UNLIKELY(!waitCount)) {
880                cblk->lock.unlock();
881                return WOULD_BLOCK;
882            }
883            if (!(cblk->flags & CBLK_INVALID_MSK)) {
884                mLock.unlock();
885                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
886                cblk->lock.unlock();
887                mLock.lock();
888                if (!mActive) {
889                    return status_t(STOPPED);
890                }
891                cblk->lock.lock();
892            }
893
894            if (cblk->flags & CBLK_INVALID_MSK) {
895                goto create_new_track;
896            }
897            if (CC_UNLIKELY(result != NO_ERROR)) {
898                cblk->waitTimeMs += waitTimeMs;
899                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
900                    // timing out when a loop has been set and we have already written upto loop end
901                    // is a normal condition: no need to wake AudioFlinger up.
902                    if (cblk->user < cblk->loopEnd) {
903                        ALOGW(   "obtainBuffer timed out (is the CPU pegged?) %p "
904                                "user=%08x, server=%08x", this, cblk->user, cblk->server);
905                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
906                        cblk->lock.unlock();
907                        result = mAudioTrack->start(0); // callback thread hasn't changed
908                        cblk->lock.lock();
909                        if (result == DEAD_OBJECT) {
910                            android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
911create_new_track:
912                            result = restoreTrack_l(cblk, false);
913                        }
914                        if (result != NO_ERROR) {
915                            ALOGW("obtainBuffer create Track error %d", result);
916                            cblk->lock.unlock();
917                            return result;
918                        }
919                    }
920                    cblk->waitTimeMs = 0;
921                }
922
923                if (--waitCount == 0) {
924                    cblk->lock.unlock();
925                    return TIMED_OUT;
926                }
927            }
928            // read the server count again
929        start_loop_here:
930            framesAvail = cblk->framesAvailable_l();
931        }
932        cblk->lock.unlock();
933    }
934
935    // restart track if it was disabled by audioflinger due to previous underrun
936    if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) {
937        android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
938        ALOGW("obtainBuffer() track %p disabled, restarting", this);
939        mAudioTrack->start(0);  // callback thread hasn't changed
940    }
941
942    cblk->waitTimeMs = 0;
943
944    if (framesReq > framesAvail) {
945        framesReq = framesAvail;
946    }
947
948    uint32_t u = cblk->user;
949    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
950
951    if (u + framesReq > bufferEnd) {
952        framesReq = bufferEnd - u;
953    }
954
955    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
956    audioBuffer->channelCount = mChannelCount;
957    audioBuffer->frameCount = framesReq;
958    audioBuffer->size = framesReq * cblk->frameSize;
959    if (audio_is_linear_pcm(mFormat)) {
960        audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
961    } else {
962        audioBuffer->format = mFormat;
963    }
964    audioBuffer->raw = (int8_t *)cblk->buffer(u);
965    active = mActive;
966    return active ? status_t(NO_ERROR) : status_t(STOPPED);
967}
968
969void AudioTrack::releaseBuffer(Buffer* audioBuffer)
970{
971    AutoMutex lock(mLock);
972    mCblk->stepUser(audioBuffer->frameCount);
973}
974
975// -------------------------------------------------------------------------
976
977ssize_t AudioTrack::write(const void* buffer, size_t userSize)
978{
979
980    if (mSharedBuffer != 0) return INVALID_OPERATION;
981    if (mIsTimed) return INVALID_OPERATION;
982
983    if (ssize_t(userSize) < 0) {
984        // Sanity-check: user is most-likely passing an error code, and it would
985        // make the return value ambiguous (actualSize vs error).
986        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
987                buffer, userSize, userSize);
988        return BAD_VALUE;
989    }
990
991    ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
992
993    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
994    // while we are accessing the cblk
995    mLock.lock();
996    sp <IAudioTrack> audioTrack = mAudioTrack;
997    sp <IMemory> iMem = mCblkMemory;
998    mLock.unlock();
999
1000    ssize_t written = 0;
1001    const int8_t *src = (const int8_t *)buffer;
1002    Buffer audioBuffer;
1003    size_t frameSz = frameSize();
1004
1005    do {
1006        audioBuffer.frameCount = userSize/frameSz;
1007
1008        status_t err = obtainBuffer(&audioBuffer, -1);
1009        if (err < 0) {
1010            // out of buffers, return #bytes written
1011            if (err == status_t(NO_MORE_BUFFERS))
1012                break;
1013            return ssize_t(err);
1014        }
1015
1016        size_t toWrite;
1017
1018        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
1019            // Divide capacity by 2 to take expansion into account
1020            toWrite = audioBuffer.size>>1;
1021            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
1022        } else {
1023            toWrite = audioBuffer.size;
1024            memcpy(audioBuffer.i8, src, toWrite);
1025            src += toWrite;
1026        }
1027        userSize -= toWrite;
1028        written += toWrite;
1029
1030        releaseBuffer(&audioBuffer);
1031    } while (userSize >= frameSz);
1032
1033    return written;
1034}
1035
1036// -------------------------------------------------------------------------
1037
1038TimedAudioTrack::TimedAudioTrack() {
1039    mIsTimed = true;
1040}
1041
1042status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
1043{
1044    status_t result = UNKNOWN_ERROR;
1045
1046    // If the track is not invalid already, try to allocate a buffer.  alloc
1047    // fails indicating that the server is dead, flag the track as invalid so
1048    // we can attempt to restore in in just a bit.
1049    if (!(mCblk->flags & CBLK_INVALID_MSK)) {
1050        result = mAudioTrack->allocateTimedBuffer(size, buffer);
1051        if (result == DEAD_OBJECT) {
1052            android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
1053        }
1054    }
1055
1056    // If the track is invalid at this point, attempt to restore it. and try the
1057    // allocation one more time.
1058    if (mCblk->flags & CBLK_INVALID_MSK) {
1059        mCblk->lock.lock();
1060        result = restoreTrack_l(mCblk, false);
1061        mCblk->lock.unlock();
1062
1063        if (result == OK)
1064            result = mAudioTrack->allocateTimedBuffer(size, buffer);
1065    }
1066
1067    return result;
1068}
1069
1070status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
1071                                           int64_t pts)
1072{
1073    // restart track if it was disabled by audioflinger due to previous underrun
1074    if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
1075        android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
1076        ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
1077        mAudioTrack->start(0);
1078    }
1079
1080    return mAudioTrack->queueTimedBuffer(buffer, pts);
1081}
1082
1083status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
1084                                                TargetTimeline target)
1085{
1086    return mAudioTrack->setMediaTimeTransform(xform, target);
1087}
1088
1089// -------------------------------------------------------------------------
1090
1091bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
1092{
1093    Buffer audioBuffer;
1094    uint32_t frames;
1095    size_t writtenSize;
1096
1097    mLock.lock();
1098    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1099    // while we are accessing the cblk
1100    sp <IAudioTrack> audioTrack = mAudioTrack;
1101    sp <IMemory> iMem = mCblkMemory;
1102    audio_track_cblk_t* cblk = mCblk;
1103    bool active = mActive;
1104    mLock.unlock();
1105
1106    // Manage underrun callback
1107    if (active && (cblk->framesAvailable() == cblk->frameCount)) {
1108        ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
1109        if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
1110            mCbf(EVENT_UNDERRUN, mUserData, 0);
1111            if (cblk->server == cblk->frameCount) {
1112                mCbf(EVENT_BUFFER_END, mUserData, 0);
1113            }
1114            if (mSharedBuffer != 0) return false;
1115        }
1116    }
1117
1118    // Manage loop end callback
1119    while (mLoopCount > cblk->loopCount) {
1120        int loopCount = -1;
1121        mLoopCount--;
1122        if (mLoopCount >= 0) loopCount = mLoopCount;
1123
1124        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
1125    }
1126
1127    // Manage marker callback
1128    if (!mMarkerReached && (mMarkerPosition > 0)) {
1129        if (cblk->server >= mMarkerPosition) {
1130            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
1131            mMarkerReached = true;
1132        }
1133    }
1134
1135    // Manage new position callback
1136    if (mUpdatePeriod > 0) {
1137        while (cblk->server >= mNewPosition) {
1138            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
1139            mNewPosition += mUpdatePeriod;
1140        }
1141    }
1142
1143    // If Shared buffer is used, no data is requested from client.
1144    if (mSharedBuffer != 0) {
1145        frames = 0;
1146    } else {
1147        frames = mRemainingFrames;
1148    }
1149
1150    // See description of waitCount parameter at declaration of obtainBuffer().
1151    // The logic below prevents us from being stuck below at obtainBuffer()
1152    // not being able to handle timed events (position, markers, loops).
1153    int32_t waitCount = -1;
1154    if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
1155        waitCount = 1;
1156    }
1157
1158    do {
1159
1160        audioBuffer.frameCount = frames;
1161
1162        status_t err = obtainBuffer(&audioBuffer, waitCount);
1163        if (err < NO_ERROR) {
1164            if (err != TIMED_OUT) {
1165                ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
1166                return false;
1167            }
1168            break;
1169        }
1170        if (err == status_t(STOPPED)) return false;
1171
1172        // Divide buffer size by 2 to take into account the expansion
1173        // due to 8 to 16 bit conversion: the callback must fill only half
1174        // of the destination buffer
1175        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
1176            audioBuffer.size >>= 1;
1177        }
1178
1179        size_t reqSize = audioBuffer.size;
1180        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1181        writtenSize = audioBuffer.size;
1182
1183        // Sanity check on returned size
1184        if (ssize_t(writtenSize) <= 0) {
1185            // The callback is done filling buffers
1186            // Keep this thread going to handle timed events and
1187            // still try to get more data in intervals of WAIT_PERIOD_MS
1188            // but don't just loop and block the CPU, so wait
1189            usleep(WAIT_PERIOD_MS*1000);
1190            break;
1191        }
1192        if (writtenSize > reqSize) writtenSize = reqSize;
1193
1194        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
1195            // 8 to 16 bit conversion, note that source and destination are the same address
1196            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
1197            writtenSize <<= 1;
1198        }
1199
1200        audioBuffer.size = writtenSize;
1201        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
1202        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
1203        // 16 bit.
1204        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
1205
1206        frames -= audioBuffer.frameCount;
1207
1208        releaseBuffer(&audioBuffer);
1209    }
1210    while (frames);
1211
1212    if (frames == 0) {
1213        mRemainingFrames = mNotificationFramesAct;
1214    } else {
1215        mRemainingFrames = frames;
1216    }
1217    return true;
1218}
1219
1220// must be called with mLock and cblk.lock held. Callers must also hold strong references on
1221// the IAudioTrack and IMemory in case they are recreated here.
1222// If the IAudioTrack is successfully restored, the cblk pointer is updated
1223status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
1224{
1225    status_t result;
1226
1227    if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
1228        ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
1229             fromStart ? "start()" : "obtainBuffer()", gettid());
1230
1231        // signal old cblk condition so that other threads waiting for available buffers stop
1232        // waiting now
1233        cblk->cv.broadcast();
1234        cblk->lock.unlock();
1235
1236        // refresh the audio configuration cache in this process to make sure we get new
1237        // output parameters in getOutput_l() and createTrack_l()
1238        AudioSystem::clearAudioConfigCache();
1239
1240        // if the new IAudioTrack is created, createTrack_l() will modify the
1241        // following member variables: mAudioTrack, mCblkMemory and mCblk.
1242        // It will also delete the strong references on previous IAudioTrack and IMemory
1243        result = createTrack_l(mStreamType,
1244                               cblk->sampleRate,
1245                               mFormat,
1246                               mChannelMask,
1247                               mFrameCount,
1248                               mFlags,
1249                               mSharedBuffer,
1250                               getOutput_l(),
1251                               false);
1252
1253        if (result == NO_ERROR) {
1254            uint32_t user = cblk->user;
1255            uint32_t server = cblk->server;
1256            // restore write index and set other indexes to reflect empty buffer status
1257            mCblk->user = user;
1258            mCblk->server = user;
1259            mCblk->userBase = user;
1260            mCblk->serverBase = user;
1261            // restore loop: this is not guaranteed to succeed if new frame count is not
1262            // compatible with loop length
1263            setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
1264            if (!fromStart) {
1265                mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1266                // Make sure that a client relying on callback events indicating underrun or
1267                // the actual amount of audio frames played (e.g SoundPool) receives them.
1268                if (mSharedBuffer == 0) {
1269                    uint32_t frames = 0;
1270                    if (user > server) {
1271                        frames = ((user - server) > mCblk->frameCount) ?
1272                                mCblk->frameCount : (user - server);
1273                        memset(mCblk->buffers, 0, frames * mCblk->frameSize);
1274                    }
1275                    // restart playback even if buffer is not completely filled.
1276                    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
1277                    // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
1278                    // the client
1279                    mCblk->stepUser(frames);
1280                }
1281            }
1282            if (mActive) {
1283                result = mAudioTrack->start(0); // callback thread hasn't changed
1284                ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
1285            }
1286            if (fromStart && result == NO_ERROR) {
1287                mNewPosition = mCblk->server + mUpdatePeriod;
1288            }
1289        }
1290        if (result != NO_ERROR) {
1291            android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
1292            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
1293        }
1294        mRestoreStatus = result;
1295        // signal old cblk condition for other threads waiting for restore completion
1296        android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
1297        cblk->cv.broadcast();
1298    } else {
1299        if (!(cblk->flags & CBLK_RESTORED_MSK)) {
1300            ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid());
1301            mLock.unlock();
1302            result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
1303            if (result == NO_ERROR) {
1304                result = mRestoreStatus;
1305            }
1306            cblk->lock.unlock();
1307            mLock.lock();
1308        } else {
1309            ALOGW("dead IAudioTrack, already restored TID %d", gettid());
1310            result = mRestoreStatus;
1311            cblk->lock.unlock();
1312        }
1313    }
1314    ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
1315         result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
1316
1317    if (result == NO_ERROR) {
1318        // from now on we switch to the newly created cblk
1319        cblk = mCblk;
1320    }
1321    cblk->lock.lock();
1322
1323    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
1324
1325    return result;
1326}
1327
1328status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1329{
1330
1331    const size_t SIZE = 256;
1332    char buffer[SIZE];
1333    String8 result;
1334
1335    result.append(" AudioTrack::dump\n");
1336    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
1337    result.append(buffer);
1338    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
1339    result.append(buffer);
1340    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
1341    result.append(buffer);
1342    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1343    result.append(buffer);
1344    ::write(fd, result.string(), result.size());
1345    return NO_ERROR;
1346}
1347
1348// =========================================================================
1349
1350AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1351    : Thread(bCanCallJava), mReceiver(receiver)
1352{
1353}
1354
1355bool AudioTrack::AudioTrackThread::threadLoop()
1356{
1357    return mReceiver.processAudioBuffer(this);
1358}
1359
1360status_t AudioTrack::AudioTrackThread::readyToRun()
1361{
1362    return NO_ERROR;
1363}
1364
1365void AudioTrack::AudioTrackThread::onFirstRef()
1366{
1367}
1368
1369// =========================================================================
1370
1371
1372audio_track_cblk_t::audio_track_cblk_t()
1373    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1374    userBase(0), serverBase(0), buffers(NULL), frameCount(0),
1375    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
1376    mSendLevel(0), flags(0)
1377{
1378}
1379
1380uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1381{
1382    uint32_t u = user;
1383
1384    u += frameCount;
1385    // Ensure that user is never ahead of server for AudioRecord
1386    if (flags & CBLK_DIRECTION_MSK) {
1387        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1388        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1389            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1390        }
1391    } else if (u > server) {
1392        ALOGW("stepServer occurred after track reset");
1393        u = server;
1394    }
1395
1396    if (u >= userBase + this->frameCount) {
1397        userBase += this->frameCount;
1398    }
1399
1400    user = u;
1401
1402    // Clear flow control error condition as new data has been written/read to/from buffer.
1403    if (flags & CBLK_UNDERRUN_MSK) {
1404        android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
1405    }
1406
1407    return u;
1408}
1409
1410bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1411{
1412    if (!tryLock()) {
1413        ALOGW("stepServer() could not lock cblk");
1414        return false;
1415    }
1416
1417    uint32_t s = server;
1418
1419    s += frameCount;
1420    if (flags & CBLK_DIRECTION_MSK) {
1421        // Mark that we have read the first buffer so that next time stepUser() is called
1422        // we switch to normal obtainBuffer() timeout period
1423        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1424            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1425        }
1426        // It is possible that we receive a flush()
1427        // while the mixer is processing a block: in this case,
1428        // stepServer() is called After the flush() has reset u & s and
1429        // we have s > u
1430        if (s > user) {
1431            ALOGW("stepServer occurred after track reset");
1432            s = user;
1433        }
1434    }
1435
1436    if (s >= loopEnd) {
1437        ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1438        s = loopStart;
1439        if (--loopCount == 0) {
1440            loopEnd = UINT_MAX;
1441            loopStart = UINT_MAX;
1442        }
1443    }
1444    if (s >= serverBase + this->frameCount) {
1445        serverBase += this->frameCount;
1446    }
1447
1448    server = s;
1449
1450    if (!(flags & CBLK_INVALID_MSK)) {
1451        cv.signal();
1452    }
1453    lock.unlock();
1454    return true;
1455}
1456
1457void* audio_track_cblk_t::buffer(uint32_t offset) const
1458{
1459    return (int8_t *)buffers + (offset - userBase) * frameSize;
1460}
1461
1462uint32_t audio_track_cblk_t::framesAvailable()
1463{
1464    Mutex::Autolock _l(lock);
1465    return framesAvailable_l();
1466}
1467
1468uint32_t audio_track_cblk_t::framesAvailable_l()
1469{
1470    uint32_t u = user;
1471    uint32_t s = server;
1472
1473    if (flags & CBLK_DIRECTION_MSK) {
1474        uint32_t limit = (s < loopStart) ? s : loopStart;
1475        return limit + frameCount - u;
1476    } else {
1477        return frameCount + u - s;
1478    }
1479}
1480
1481uint32_t audio_track_cblk_t::framesReady()
1482{
1483    uint32_t u = user;
1484    uint32_t s = server;
1485
1486    if (flags & CBLK_DIRECTION_MSK) {
1487        if (u < loopEnd) {
1488            return u - s;
1489        } else {
1490            // do not block on mutex shared with client on AudioFlinger side
1491            if (!tryLock()) {
1492                ALOGW("framesReady() could not lock cblk");
1493                return 0;
1494            }
1495            uint32_t frames = UINT_MAX;
1496            if (loopCount >= 0) {
1497                frames = (loopEnd - loopStart)*loopCount + u - s;
1498            }
1499            lock.unlock();
1500            return frames;
1501        }
1502    } else {
1503        return s - u;
1504    }
1505}
1506
1507bool audio_track_cblk_t::tryLock()
1508{
1509    // the code below simulates lock-with-timeout
1510    // we MUST do this to protect the AudioFlinger server
1511    // as this lock is shared with the client.
1512    status_t err;
1513
1514    err = lock.tryLock();
1515    if (err == -EBUSY) { // just wait a bit
1516        usleep(1000);
1517        err = lock.tryLock();
1518    }
1519    if (err != NO_ERROR) {
1520        // probably, the client just died.
1521        return false;
1522    }
1523    return true;
1524}
1525
1526// -------------------------------------------------------------------------
1527
1528}; // namespace android
1529