AudioTrack.cpp revision 2b691b90507ec45a98636a855c46de5dbe27c84a 
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    if (frameCount == NULL) return BAD_VALUE;
58
59    // default to 0 in case of error
60    *frameCount = 0;
61
62    // FIXME merge with similar code in createTrack_l(), except we're missing
63    //       some information here that is available in createTrack_l():
64    //          audio_io_handle_t output
65    //          audio_format_t format
66    //          audio_channel_mask_t channelMask
67    //          audio_output_flags_t flags
68    int afSampleRate;
69    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
70        return NO_INIT;
71    }
72    int afFrameCount;
73    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
74        return NO_INIT;
75    }
76    uint32_t afLatency;
77    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
78        return NO_INIT;
79    }
80
81    // Ensure that buffer depth covers at least audio hardware latency
82    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
83    if (minBufCount < 2) minBufCount = 2;
84
85    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
86            afFrameCount * minBufCount * sampleRate / afSampleRate;
87    ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
88            *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
89    return NO_ERROR;
90}
91
92// ---------------------------------------------------------------------------
93
94AudioTrack::AudioTrack()
95    : mStatus(NO_INIT),
96      mIsTimed(false),
97      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
98      mPreviousSchedulingGroup(SP_DEFAULT)
99{
100}
101
102AudioTrack::AudioTrack(
103        audio_stream_type_t streamType,
104        uint32_t sampleRate,
105        audio_format_t format,
106        audio_channel_mask_t channelMask,
107        int frameCount,
108        audio_output_flags_t flags,
109        callback_t cbf,
110        void* user,
111        int notificationFrames,
112        int sessionId)
113    : mStatus(NO_INIT),
114      mIsTimed(false),
115      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
116      mPreviousSchedulingGroup(SP_DEFAULT)
117{
118    mStatus = set(streamType, sampleRate, format, channelMask,
119            frameCount, flags, cbf, user, notificationFrames,
120            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
121}
122
123// DEPRECATED
124AudioTrack::AudioTrack(
125        int streamType,
126        uint32_t sampleRate,
127        int format,
128        int channelMask,
129        int frameCount,
130        uint32_t flags,
131        callback_t cbf,
132        void* user,
133        int notificationFrames,
134        int sessionId)
135    : mStatus(NO_INIT),
136      mIsTimed(false),
137      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
138{
139    mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format,
140            (audio_channel_mask_t) channelMask,
141            frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
142            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
143}
144
145AudioTrack::AudioTrack(
146        audio_stream_type_t streamType,
147        uint32_t sampleRate,
148        audio_format_t format,
149        audio_channel_mask_t channelMask,
150        const sp<IMemory>& sharedBuffer,
151        audio_output_flags_t flags,
152        callback_t cbf,
153        void* user,
154        int notificationFrames,
155        int sessionId)
156    : mStatus(NO_INIT),
157      mIsTimed(false),
158      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
159      mPreviousSchedulingGroup(SP_DEFAULT)
160{
161    mStatus = set(streamType, sampleRate, format, channelMask,
162            0 /*frameCount*/, flags, cbf, user, notificationFrames,
163            sharedBuffer, false /*threadCanCallJava*/, sessionId);
164}
165
166AudioTrack::~AudioTrack()
167{
168    ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
169
170    if (mStatus == NO_ERROR) {
171        // Make sure that callback function exits in the case where
172        // it is looping on buffer full condition in obtainBuffer().
173        // Otherwise the callback thread will never exit.
174        stop();
175        if (mAudioTrackThread != 0) {
176            mAudioTrackThread->requestExit();   // see comment in AudioTrack.h
177            mAudioTrackThread->requestExitAndWait();
178            mAudioTrackThread.clear();
179        }
180        mAudioTrack.clear();
181        IPCThreadState::self()->flushCommands();
182        AudioSystem::releaseAudioSessionId(mSessionId);
183    }
184}
185
186status_t AudioTrack::set(
187        audio_stream_type_t streamType,
188        uint32_t sampleRate,
189        audio_format_t format,
190        audio_channel_mask_t channelMask,
191        int frameCount,
192        audio_output_flags_t flags,
193        callback_t cbf,
194        void* user,
195        int notificationFrames,
196        const sp<IMemory>& sharedBuffer,
197        bool threadCanCallJava,
198        int sessionId)
199{
200
201    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
202
203    ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
204
205    AutoMutex lock(mLock);
206    if (mAudioTrack != 0) {
207        ALOGE("Track already in use");
208        return INVALID_OPERATION;
209    }
210
211    // handle default values first.
212    if (streamType == AUDIO_STREAM_DEFAULT) {
213        streamType = AUDIO_STREAM_MUSIC;
214    }
215
216    if (sampleRate == 0) {
217        int afSampleRate;
218        if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
219            return NO_INIT;
220        }
221        sampleRate = afSampleRate;
222    }
223
224    // these below should probably come from the audioFlinger too...
225    if (format == AUDIO_FORMAT_DEFAULT) {
226        format = AUDIO_FORMAT_PCM_16_BIT;
227    }
228    if (channelMask == 0) {
229        channelMask = AUDIO_CHANNEL_OUT_STEREO;
230    }
231
232    // validate parameters
233    if (!audio_is_valid_format(format)) {
234        ALOGE("Invalid format");
235        return BAD_VALUE;
236    }
237
238    // AudioFlinger does not currently support 8-bit data in shared memory
239    if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
240        ALOGE("8-bit data in shared memory is not supported");
241        return BAD_VALUE;
242    }
243
244    // force direct flag if format is not linear PCM
245    if (!audio_is_linear_pcm(format)) {
246        flags = (audio_output_flags_t)
247                // FIXME why can't we allow direct AND fast?
248                ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
249    }
250    // only allow deep buffering for music stream type
251    if (streamType != AUDIO_STREAM_MUSIC) {
252        flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
253    }
254
255    if (!audio_is_output_channel(channelMask)) {
256        ALOGE("Invalid channel mask %#x", channelMask);
257        return BAD_VALUE;
258    }
259    uint32_t channelCount = popcount(channelMask);
260
261    audio_io_handle_t output = AudioSystem::getOutput(
262                                    streamType,
263                                    sampleRate, format, channelMask,
264                                    flags);
265
266    if (output == 0) {
267        ALOGE("Could not get audio output for stream type %d", streamType);
268        return BAD_VALUE;
269    }
270
271    mVolume[LEFT] = 1.0f;
272    mVolume[RIGHT] = 1.0f;
273    mSendLevel = 0.0f;
274    mFrameCount = frameCount;
275    mNotificationFramesReq = notificationFrames;
276    mSessionId = sessionId;
277    mAuxEffectId = 0;
278    mFlags = flags;
279    mCbf = cbf;
280
281    if (cbf != NULL) {
282        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
283        mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
284    }
285
286    // create the IAudioTrack
287    status_t status = createTrack_l(streamType,
288                                  sampleRate,
289                                  format,
290                                  channelMask,
291                                  frameCount,
292                                  flags,
293                                  sharedBuffer,
294                                  output);
295
296    if (status != NO_ERROR) {
297        if (mAudioTrackThread != 0) {
298            mAudioTrackThread->requestExit();
299            mAudioTrackThread.clear();
300        }
301        return status;
302    }
303
304    mStatus = NO_ERROR;
305
306    mStreamType = streamType;
307    mFormat = format;
308    mChannelMask = channelMask;
309    mChannelCount = channelCount;
310    mSharedBuffer = sharedBuffer;
311    mMuted = false;
312    mActive = false;
313    mUserData = user;
314    mLoopCount = 0;
315    mMarkerPosition = 0;
316    mMarkerReached = false;
317    mNewPosition = 0;
318    mUpdatePeriod = 0;
319    mFlushed = false;
320    AudioSystem::acquireAudioSessionId(mSessionId);
321    mRestoreStatus = NO_ERROR;
322    return NO_ERROR;
323}
324
325status_t AudioTrack::initCheck() const
326{
327    return mStatus;
328}
329
330// -------------------------------------------------------------------------
331
332uint32_t AudioTrack::latency() const
333{
334    return mLatency;
335}
336
337audio_stream_type_t AudioTrack::streamType() const
338{
339    return mStreamType;
340}
341
342audio_format_t AudioTrack::format() const
343{
344    return mFormat;
345}
346
347int AudioTrack::channelCount() const
348{
349    return mChannelCount;
350}
351
352uint32_t AudioTrack::frameCount() const
353{
354    return mCblk->frameCount;
355}
356
357size_t AudioTrack::frameSize() const
358{
359    if (audio_is_linear_pcm(mFormat)) {
360        return channelCount()*audio_bytes_per_sample(mFormat);
361    } else {
362        return sizeof(uint8_t);
363    }
364}
365
366sp<IMemory>& AudioTrack::sharedBuffer()
367{
368    return mSharedBuffer;
369}
370
371// -------------------------------------------------------------------------
372
373void AudioTrack::start()
374{
375    sp<AudioTrackThread> t = mAudioTrackThread;
376
377    ALOGV("start %p", this);
378
379    AutoMutex lock(mLock);
380    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
381    // while we are accessing the cblk
382    sp<IAudioTrack> audioTrack = mAudioTrack;
383    sp<IMemory> iMem = mCblkMemory;
384    audio_track_cblk_t* cblk = mCblk;
385
386    if (!mActive) {
387        mFlushed = false;
388        mActive = true;
389        mNewPosition = cblk->server + mUpdatePeriod;
390        cblk->lock.lock();
391        cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
392        cblk->waitTimeMs = 0;
393        android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
394        if (t != 0) {
395            t->resume();
396        } else {
397            mPreviousPriority = getpriority(PRIO_PROCESS, 0);
398            get_sched_policy(0, &mPreviousSchedulingGroup);
399            androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
400        }
401
402        ALOGV("start %p before lock cblk %p", this, mCblk);
403        status_t status = NO_ERROR;
404        if (!(cblk->flags & CBLK_INVALID_MSK)) {
405            cblk->lock.unlock();
406            ALOGV("mAudioTrack->start()");
407            status = mAudioTrack->start();
408            cblk->lock.lock();
409            if (status == DEAD_OBJECT) {
410                android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
411            }
412        }
413        if (cblk->flags & CBLK_INVALID_MSK) {
414            status = restoreTrack_l(cblk, true);
415        }
416        cblk->lock.unlock();
417        if (status != NO_ERROR) {
418            ALOGV("start() failed");
419            mActive = false;
420            if (t != 0) {
421                t->pause();
422            } else {
423                setpriority(PRIO_PROCESS, 0, mPreviousPriority);
424                set_sched_policy(0, mPreviousSchedulingGroup);
425            }
426        }
427    }
428
429}
430
431void AudioTrack::stop()
432{
433    sp<AudioTrackThread> t = mAudioTrackThread;
434
435    ALOGV("stop %p", this);
436
437    AutoMutex lock(mLock);
438    if (mActive) {
439        mActive = false;
440        mCblk->cv.signal();
441        mAudioTrack->stop();
442        // Cancel loops (If we are in the middle of a loop, playback
443        // would not stop until loopCount reaches 0).
444        setLoop_l(0, 0, 0);
445        // the playback head position will reset to 0, so if a marker is set, we need
446        // to activate it again
447        mMarkerReached = false;
448        // Force flush if a shared buffer is used otherwise audioflinger
449        // will not stop before end of buffer is reached.
450        if (mSharedBuffer != 0) {
451            flush_l();
452        }
453        if (t != 0) {
454            t->pause();
455        } else {
456            setpriority(PRIO_PROCESS, 0, mPreviousPriority);
457            set_sched_policy(0, mPreviousSchedulingGroup);
458        }
459    }
460
461}
462
463bool AudioTrack::stopped() const
464{
465    AutoMutex lock(mLock);
466    return stopped_l();
467}
468
469void AudioTrack::flush()
470{
471    AutoMutex lock(mLock);
472    flush_l();
473}
474
475// must be called with mLock held
476void AudioTrack::flush_l()
477{
478    ALOGV("flush");
479
480    // clear playback marker and periodic update counter
481    mMarkerPosition = 0;
482    mMarkerReached = false;
483    mUpdatePeriod = 0;
484
485    if (!mActive) {
486        mFlushed = true;
487        mAudioTrack->flush();
488        // Release AudioTrack callback thread in case it was waiting for new buffers
489        // in AudioTrack::obtainBuffer()
490        mCblk->cv.signal();
491    }
492}
493
494void AudioTrack::pause()
495{
496    ALOGV("pause");
497    AutoMutex lock(mLock);
498    if (mActive) {
499        mActive = false;
500        mCblk->cv.signal();
501        mAudioTrack->pause();
502    }
503}
504
505void AudioTrack::mute(bool e)
506{
507    mAudioTrack->mute(e);
508    mMuted = e;
509}
510
511bool AudioTrack::muted() const
512{
513    return mMuted;
514}
515
516status_t AudioTrack::setVolume(float left, float right)
517{
518    if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
519        return BAD_VALUE;
520    }
521
522    AutoMutex lock(mLock);
523    mVolume[LEFT] = left;
524    mVolume[RIGHT] = right;
525
526    mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
527
528    return NO_ERROR;
529}
530
531void AudioTrack::getVolume(float* left, float* right) const
532{
533    if (left != NULL) {
534        *left  = mVolume[LEFT];
535    }
536    if (right != NULL) {
537        *right = mVolume[RIGHT];
538    }
539}
540
541status_t AudioTrack::setAuxEffectSendLevel(float level)
542{
543    ALOGV("setAuxEffectSendLevel(%f)", level);
544    if (level < 0.0f || level > 1.0f) {
545        return BAD_VALUE;
546    }
547    AutoMutex lock(mLock);
548
549    mSendLevel = level;
550
551    mCblk->setSendLevel(level);
552
553    return NO_ERROR;
554}
555
556void AudioTrack::getAuxEffectSendLevel(float* level) const
557{
558    if (level != NULL) {
559        *level  = mSendLevel;
560    }
561}
562
563status_t AudioTrack::setSampleRate(int rate)
564{
565    int afSamplingRate;
566
567    if (mIsTimed) {
568        return INVALID_OPERATION;
569    }
570
571    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
572        return NO_INIT;
573    }
574    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
575    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
576
577    AutoMutex lock(mLock);
578    mCblk->sampleRate = rate;
579    return NO_ERROR;
580}
581
582uint32_t AudioTrack::getSampleRate() const
583{
584    if (mIsTimed) {
585        return INVALID_OPERATION;
586    }
587
588    AutoMutex lock(mLock);
589    return mCblk->sampleRate;
590}
591
592status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
593{
594    AutoMutex lock(mLock);
595    return setLoop_l(loopStart, loopEnd, loopCount);
596}
597
598// must be called with mLock held
599status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
600{
601    audio_track_cblk_t* cblk = mCblk;
602
603    Mutex::Autolock _l(cblk->lock);
604
605    if (loopCount == 0) {
606        cblk->loopStart = UINT_MAX;
607        cblk->loopEnd = UINT_MAX;
608        cblk->loopCount = 0;
609        mLoopCount = 0;
610        return NO_ERROR;
611    }
612
613    if (mIsTimed) {
614        return INVALID_OPERATION;
615    }
616
617    if (loopStart >= loopEnd ||
618        loopEnd - loopStart > cblk->frameCount ||
619        cblk->server > loopStart) {
620        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
621        return BAD_VALUE;
622    }
623
624    if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
625        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
626            loopStart, loopEnd, cblk->frameCount);
627        return BAD_VALUE;
628    }
629
630    cblk->loopStart = loopStart;
631    cblk->loopEnd = loopEnd;
632    cblk->loopCount = loopCount;
633    mLoopCount = loopCount;
634
635    return NO_ERROR;
636}
637
638status_t AudioTrack::setMarkerPosition(uint32_t marker)
639{
640    if (mCbf == NULL) return INVALID_OPERATION;
641
642    mMarkerPosition = marker;
643    mMarkerReached = false;
644
645    return NO_ERROR;
646}
647
648status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
649{
650    if (marker == NULL) return BAD_VALUE;
651
652    *marker = mMarkerPosition;
653
654    return NO_ERROR;
655}
656
657status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
658{
659    if (mCbf == NULL) return INVALID_OPERATION;
660
661    uint32_t curPosition;
662    getPosition(&curPosition);
663    mNewPosition = curPosition + updatePeriod;
664    mUpdatePeriod = updatePeriod;
665
666    return NO_ERROR;
667}
668
669status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
670{
671    if (updatePeriod == NULL) return BAD_VALUE;
672
673    *updatePeriod = mUpdatePeriod;
674
675    return NO_ERROR;
676}
677
678status_t AudioTrack::setPosition(uint32_t position)
679{
680    if (mIsTimed) return INVALID_OPERATION;
681
682    AutoMutex lock(mLock);
683
684    if (!stopped_l()) return INVALID_OPERATION;
685
686    Mutex::Autolock _l(mCblk->lock);
687
688    if (position > mCblk->user) return BAD_VALUE;
689
690    mCblk->server = position;
691    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
692
693    return NO_ERROR;
694}
695
696status_t AudioTrack::getPosition(uint32_t *position)
697{
698    if (position == NULL) return BAD_VALUE;
699    AutoMutex lock(mLock);
700    *position = mFlushed ? 0 : mCblk->server;
701
702    return NO_ERROR;
703}
704
705status_t AudioTrack::reload()
706{
707    AutoMutex lock(mLock);
708
709    if (!stopped_l()) return INVALID_OPERATION;
710
711    flush_l();
712
713    mCblk->stepUser(mCblk->frameCount);
714
715    return NO_ERROR;
716}
717
718audio_io_handle_t AudioTrack::getOutput()
719{
720    AutoMutex lock(mLock);
721    return getOutput_l();
722}
723
724// must be called with mLock held
725audio_io_handle_t AudioTrack::getOutput_l()
726{
727    return AudioSystem::getOutput(mStreamType,
728            mCblk->sampleRate, mFormat, mChannelMask, mFlags);
729}
730
731int AudioTrack::getSessionId() const
732{
733    return mSessionId;
734}
735
736status_t AudioTrack::attachAuxEffect(int effectId)
737{
738    ALOGV("attachAuxEffect(%d)", effectId);
739    status_t status = mAudioTrack->attachAuxEffect(effectId);
740    if (status == NO_ERROR) {
741        mAuxEffectId = effectId;
742    }
743    return status;
744}
745
746// -------------------------------------------------------------------------
747
748// must be called with mLock held
749status_t AudioTrack::createTrack_l(
750        audio_stream_type_t streamType,
751        uint32_t sampleRate,
752        audio_format_t format,
753        audio_channel_mask_t channelMask,
754        int frameCount,
755        audio_output_flags_t flags,
756        const sp<IMemory>& sharedBuffer,
757        audio_io_handle_t output)
758{
759    status_t status;
760    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
761    if (audioFlinger == 0) {
762        ALOGE("Could not get audioflinger");
763        return NO_INIT;
764    }
765
766    uint32_t afLatency;
767    if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
768        return NO_INIT;
769    }
770
771    // Client decides whether the track is TIMED (see below), but can only express a preference
772    // for FAST.  Server will perform additional tests.
773    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
774            // either of these use cases:
775            // use case 1: shared buffer
776            (sharedBuffer != 0) ||
777            // use case 2: callback handler
778            (mCbf != NULL))) {
779        ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
780        // once denied, do not request again if IAudioTrack is re-created
781        flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
782        mFlags = flags;
783    }
784    ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
785
786    mNotificationFramesAct = mNotificationFramesReq;
787
788    if (!audio_is_linear_pcm(format)) {
789
790        if (sharedBuffer != 0) {
791            // Same comment as below about ignoring frameCount parameter for set()
792            frameCount = sharedBuffer->size();
793        } else if (frameCount == 0) {
794            int afFrameCount;
795            if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
796                return NO_INIT;
797            }
798            frameCount = afFrameCount;
799        }
800
801    } else if (sharedBuffer != 0) {
802
803        // Ensure that buffer alignment matches channelCount
804        int channelCount = popcount(channelMask);
805        // 8-bit data in shared memory is not currently supported by AudioFlinger
806        size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
807        if (channelCount > 1) {
808            // More than 2 channels does not require stronger alignment than stereo
809            alignment <<= 1;
810        }
811        if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
812            ALOGE("Invalid buffer alignment: address %p, channelCount %d",
813                    sharedBuffer->pointer(), channelCount);
814            return BAD_VALUE;
815        }
816
817        // When initializing a shared buffer AudioTrack via constructors,
818        // there's no frameCount parameter.
819        // But when initializing a shared buffer AudioTrack via set(),
820        // there _is_ a frameCount parameter.  We silently ignore it.
821        frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
822
823    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
824
825        // FIXME move these calculations and associated checks to server
826        int afSampleRate;
827        if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
828            return NO_INIT;
829        }
830        int afFrameCount;
831        if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
832            return NO_INIT;
833        }
834
835        // Ensure that buffer depth covers at least audio hardware latency
836        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
837        if (minBufCount < 2) minBufCount = 2;
838
839        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
840        ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
841                ", afLatency=%d",
842                minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
843
844        if (frameCount == 0) {
845            frameCount = minFrameCount;
846        }
847        if (mNotificationFramesAct == 0) {
848            mNotificationFramesAct = frameCount/2;
849        }
850        // Make sure that application is notified with sufficient margin
851        // before underrun
852        if (mNotificationFramesAct > (uint32_t)frameCount/2) {
853            mNotificationFramesAct = frameCount/2;
854        }
855        if (frameCount < minFrameCount) {
856            // not ALOGW because it happens all the time when playing key clicks over A2DP
857            ALOGV("Minimum buffer size corrected from %d to %d",
858                     frameCount, minFrameCount);
859            frameCount = minFrameCount;
860        }
861
862    } else {
863        // For fast tracks, the frame count calculations and checks are done by server
864    }
865
866    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
867    if (mIsTimed) {
868        trackFlags |= IAudioFlinger::TRACK_TIMED;
869    }
870
871    pid_t tid = -1;
872    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
873        trackFlags |= IAudioFlinger::TRACK_FAST;
874        if (mAudioTrackThread != 0) {
875            tid = mAudioTrackThread->getTid();
876        }
877    }
878
879    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
880                                                      streamType,
881                                                      sampleRate,
882                                                      format,
883                                                      channelMask,
884                                                      frameCount,
885                                                      trackFlags,
886                                                      sharedBuffer,
887                                                      output,
888                                                      tid,
889                                                      &mSessionId,
890                                                      &status);
891
892    if (track == 0) {
893        ALOGE("AudioFlinger could not create track, status: %d", status);
894        return status;
895    }
896    sp<IMemory> cblk = track->getCblk();
897    if (cblk == 0) {
898        ALOGE("Could not get control block");
899        return NO_INIT;
900    }
901    mAudioTrack = track;
902    mCblkMemory = cblk;
903    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
904    // old has the previous value of mCblk->flags before the "or" operation
905    int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
906    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
907        if (old & CBLK_FAST) {
908            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount);
909        } else {
910            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount);
911            // once denied, do not request again if IAudioTrack is re-created
912            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
913            mFlags = flags;
914        }
915        if (sharedBuffer == 0) {
916            mNotificationFramesAct = mCblk->frameCount/2;
917        }
918    }
919    if (sharedBuffer == 0) {
920        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
921    } else {
922        mCblk->buffers = sharedBuffer->pointer();
923        // Force buffer full condition as data is already present in shared memory
924        mCblk->stepUser(mCblk->frameCount);
925    }
926
927    mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
928    mCblk->setSendLevel(mSendLevel);
929    mAudioTrack->attachAuxEffect(mAuxEffectId);
930    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
931    mCblk->waitTimeMs = 0;
932    mRemainingFrames = mNotificationFramesAct;
933    // FIXME don't believe this lie
934    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
935    // If IAudioTrack is re-created, don't let the requested frameCount
936    // decrease.  This can confuse clients that cache frameCount().
937    if (mCblk->frameCount > mFrameCount) {
938        mFrameCount = mCblk->frameCount;
939    }
940    return NO_ERROR;
941}
942
943status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
944{
945    AutoMutex lock(mLock);
946    bool active;
947    status_t result = NO_ERROR;
948    audio_track_cblk_t* cblk = mCblk;
949    uint32_t framesReq = audioBuffer->frameCount;
950    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
951
952    audioBuffer->frameCount  = 0;
953    audioBuffer->size = 0;
954
955    uint32_t framesAvail = cblk->framesAvailable();
956
957    cblk->lock.lock();
958    if (cblk->flags & CBLK_INVALID_MSK) {
959        goto create_new_track;
960    }
961    cblk->lock.unlock();
962
963    if (framesAvail == 0) {
964        cblk->lock.lock();
965        goto start_loop_here;
966        while (framesAvail == 0) {
967            active = mActive;
968            if (CC_UNLIKELY(!active)) {
969                ALOGV("Not active and NO_MORE_BUFFERS");
970                cblk->lock.unlock();
971                return NO_MORE_BUFFERS;
972            }
973            if (CC_UNLIKELY(!waitCount)) {
974                cblk->lock.unlock();
975                return WOULD_BLOCK;
976            }
977            if (!(cblk->flags & CBLK_INVALID_MSK)) {
978                mLock.unlock();
979                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
980                cblk->lock.unlock();
981                mLock.lock();
982                if (!mActive) {
983                    return status_t(STOPPED);
984                }
985                cblk->lock.lock();
986            }
987
988            if (cblk->flags & CBLK_INVALID_MSK) {
989                goto create_new_track;
990            }
991            if (CC_UNLIKELY(result != NO_ERROR)) {
992                cblk->waitTimeMs += waitTimeMs;
993                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
994                    // timing out when a loop has been set and we have already written upto loop end
995                    // is a normal condition: no need to wake AudioFlinger up.
996                    if (cblk->user < cblk->loopEnd) {
997                        ALOGW(   "obtainBuffer timed out (is the CPU pegged?) %p name=%#x"
998                                "user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server);
999                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
1000                        cblk->lock.unlock();
1001                        result = mAudioTrack->start();
1002                        cblk->lock.lock();
1003                        if (result == DEAD_OBJECT) {
1004                            android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
1005create_new_track:
1006                            result = restoreTrack_l(cblk, false);
1007                        }
1008                        if (result != NO_ERROR) {
1009                            ALOGW("obtainBuffer create Track error %d", result);
1010                            cblk->lock.unlock();
1011                            return result;
1012                        }
1013                    }
1014                    cblk->waitTimeMs = 0;
1015                }
1016
1017                if (--waitCount == 0) {
1018                    cblk->lock.unlock();
1019                    return TIMED_OUT;
1020                }
1021            }
1022            // read the server count again
1023        start_loop_here:
1024            framesAvail = cblk->framesAvailable_l();
1025        }
1026        cblk->lock.unlock();
1027    }
1028
1029    cblk->waitTimeMs = 0;
1030
1031    if (framesReq > framesAvail) {
1032        framesReq = framesAvail;
1033    }
1034
1035    uint32_t u = cblk->user;
1036    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
1037
1038    if (framesReq > bufferEnd - u) {
1039        framesReq = bufferEnd - u;
1040    }
1041
1042    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
1043    audioBuffer->channelCount = mChannelCount;
1044    audioBuffer->frameCount = framesReq;
1045    audioBuffer->size = framesReq * cblk->frameSize;
1046    if (audio_is_linear_pcm(mFormat)) {
1047        audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
1048    } else {
1049        audioBuffer->format = mFormat;
1050    }
1051    audioBuffer->raw = (int8_t *)cblk->buffer(u);
1052    active = mActive;
1053    return active ? status_t(NO_ERROR) : status_t(STOPPED);
1054}
1055
1056void AudioTrack::releaseBuffer(Buffer* audioBuffer)
1057{
1058    AutoMutex lock(mLock);
1059    mCblk->stepUser(audioBuffer->frameCount);
1060    if (audioBuffer->frameCount > 0) {
1061        // restart track if it was disabled by audioflinger due to previous underrun
1062        if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
1063            android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
1064            ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName);
1065            mAudioTrack->start();
1066        }
1067    }
1068}
1069
1070// -------------------------------------------------------------------------
1071
1072ssize_t AudioTrack::write(const void* buffer, size_t userSize)
1073{
1074
1075    if (mSharedBuffer != 0) return INVALID_OPERATION;
1076    if (mIsTimed) return INVALID_OPERATION;
1077
1078    if (ssize_t(userSize) < 0) {
1079        // Sanity-check: user is most-likely passing an error code, and it would
1080        // make the return value ambiguous (actualSize vs error).
1081        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
1082                buffer, userSize, userSize);
1083        return BAD_VALUE;
1084    }
1085
1086    ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
1087
1088    if (userSize == 0) {
1089        return 0;
1090    }
1091
1092    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1093    // while we are accessing the cblk
1094    mLock.lock();
1095    sp<IAudioTrack> audioTrack = mAudioTrack;
1096    sp<IMemory> iMem = mCblkMemory;
1097    mLock.unlock();
1098
1099    ssize_t written = 0;
1100    const int8_t *src = (const int8_t *)buffer;
1101    Buffer audioBuffer;
1102    size_t frameSz = frameSize();
1103
1104    do {
1105        audioBuffer.frameCount = userSize/frameSz;
1106
1107        status_t err = obtainBuffer(&audioBuffer, -1);
1108        if (err < 0) {
1109            // out of buffers, return #bytes written
1110            if (err == status_t(NO_MORE_BUFFERS))
1111                break;
1112            return ssize_t(err);
1113        }
1114
1115        size_t toWrite;
1116
1117        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1118            // Divide capacity by 2 to take expansion into account
1119            toWrite = audioBuffer.size>>1;
1120            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
1121        } else {
1122            toWrite = audioBuffer.size;
1123            memcpy(audioBuffer.i8, src, toWrite);
1124            src += toWrite;
1125        }
1126        userSize -= toWrite;
1127        written += toWrite;
1128
1129        releaseBuffer(&audioBuffer);
1130    } while (userSize >= frameSz);
1131
1132    return written;
1133}
1134
1135// -------------------------------------------------------------------------
1136
1137TimedAudioTrack::TimedAudioTrack() {
1138    mIsTimed = true;
1139}
1140
1141status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
1142{
1143    status_t result = UNKNOWN_ERROR;
1144
1145    // If the track is not invalid already, try to allocate a buffer.  alloc
1146    // fails indicating that the server is dead, flag the track as invalid so
1147    // we can attempt to restore in just a bit.
1148    if (!(mCblk->flags & CBLK_INVALID_MSK)) {
1149        result = mAudioTrack->allocateTimedBuffer(size, buffer);
1150        if (result == DEAD_OBJECT) {
1151            android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
1152        }
1153    }
1154
1155    // If the track is invalid at this point, attempt to restore it. and try the
1156    // allocation one more time.
1157    if (mCblk->flags & CBLK_INVALID_MSK) {
1158        mCblk->lock.lock();
1159        result = restoreTrack_l(mCblk, false);
1160        mCblk->lock.unlock();
1161
1162        if (result == OK)
1163            result = mAudioTrack->allocateTimedBuffer(size, buffer);
1164    }
1165
1166    return result;
1167}
1168
1169status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
1170                                           int64_t pts)
1171{
1172    status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
1173    {
1174        AutoMutex lock(mLock);
1175        // restart track if it was disabled by audioflinger due to previous underrun
1176        if (buffer->size() != 0 && status == NO_ERROR &&
1177                mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
1178            android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
1179            ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
1180            mAudioTrack->start();
1181        }
1182    }
1183    return status;
1184}
1185
1186status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
1187                                                TargetTimeline target)
1188{
1189    return mAudioTrack->setMediaTimeTransform(xform, target);
1190}
1191
1192// -------------------------------------------------------------------------
1193
1194bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
1195{
1196    Buffer audioBuffer;
1197    uint32_t frames;
1198    size_t writtenSize;
1199
1200    mLock.lock();
1201    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1202    // while we are accessing the cblk
1203    sp<IAudioTrack> audioTrack = mAudioTrack;
1204    sp<IMemory> iMem = mCblkMemory;
1205    audio_track_cblk_t* cblk = mCblk;
1206    bool active = mActive;
1207    mLock.unlock();
1208
1209    // Manage underrun callback
1210    if (active && (cblk->framesAvailable() == cblk->frameCount)) {
1211        ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
1212        if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
1213            mCbf(EVENT_UNDERRUN, mUserData, 0);
1214            if (cblk->server == cblk->frameCount) {
1215                mCbf(EVENT_BUFFER_END, mUserData, 0);
1216            }
1217            if (mSharedBuffer != 0) return false;
1218        }
1219    }
1220
1221    // Manage loop end callback
1222    while (mLoopCount > cblk->loopCount) {
1223        int loopCount = -1;
1224        mLoopCount--;
1225        if (mLoopCount >= 0) loopCount = mLoopCount;
1226
1227        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
1228    }
1229
1230    // Manage marker callback
1231    if (!mMarkerReached && (mMarkerPosition > 0)) {
1232        if (cblk->server >= mMarkerPosition) {
1233            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
1234            mMarkerReached = true;
1235        }
1236    }
1237
1238    // Manage new position callback
1239    if (mUpdatePeriod > 0) {
1240        while (cblk->server >= mNewPosition) {
1241            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
1242            mNewPosition += mUpdatePeriod;
1243        }
1244    }
1245
1246    // If Shared buffer is used, no data is requested from client.
1247    if (mSharedBuffer != 0) {
1248        frames = 0;
1249    } else {
1250        frames = mRemainingFrames;
1251    }
1252
1253    // See description of waitCount parameter at declaration of obtainBuffer().
1254    // The logic below prevents us from being stuck below at obtainBuffer()
1255    // not being able to handle timed events (position, markers, loops).
1256    int32_t waitCount = -1;
1257    if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
1258        waitCount = 1;
1259    }
1260
1261    do {
1262
1263        audioBuffer.frameCount = frames;
1264
1265        status_t err = obtainBuffer(&audioBuffer, waitCount);
1266        if (err < NO_ERROR) {
1267            if (err != TIMED_OUT) {
1268                ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
1269                return false;
1270            }
1271            break;
1272        }
1273        if (err == status_t(STOPPED)) return false;
1274
1275        // Divide buffer size by 2 to take into account the expansion
1276        // due to 8 to 16 bit conversion: the callback must fill only half
1277        // of the destination buffer
1278        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1279            audioBuffer.size >>= 1;
1280        }
1281
1282        size_t reqSize = audioBuffer.size;
1283        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1284        writtenSize = audioBuffer.size;
1285
1286        // Sanity check on returned size
1287        if (ssize_t(writtenSize) <= 0) {
1288            // The callback is done filling buffers
1289            // Keep this thread going to handle timed events and
1290            // still try to get more data in intervals of WAIT_PERIOD_MS
1291            // but don't just loop and block the CPU, so wait
1292            usleep(WAIT_PERIOD_MS*1000);
1293            break;
1294        }
1295
1296        if (writtenSize > reqSize) writtenSize = reqSize;
1297
1298        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1299            // 8 to 16 bit conversion, note that source and destination are the same address
1300            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
1301            writtenSize <<= 1;
1302        }
1303
1304        audioBuffer.size = writtenSize;
1305        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
1306        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
1307        // 16 bit.
1308        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
1309
1310        frames -= audioBuffer.frameCount;
1311
1312        releaseBuffer(&audioBuffer);
1313    }
1314    while (frames);
1315
1316    if (frames == 0) {
1317        mRemainingFrames = mNotificationFramesAct;
1318    } else {
1319        mRemainingFrames = frames;
1320    }
1321    return true;
1322}
1323
1324// must be called with mLock and cblk.lock held. Callers must also hold strong references on
1325// the IAudioTrack and IMemory in case they are recreated here.
1326// If the IAudioTrack is successfully restored, the cblk pointer is updated
1327status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
1328{
1329    status_t result;
1330
1331    if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
1332        ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
1333            fromStart ? "start()" : "obtainBuffer()", gettid());
1334
1335        // signal old cblk condition so that other threads waiting for available buffers stop
1336        // waiting now
1337        cblk->cv.broadcast();
1338        cblk->lock.unlock();
1339
1340        // refresh the audio configuration cache in this process to make sure we get new
1341        // output parameters in getOutput_l() and createTrack_l()
1342        AudioSystem::clearAudioConfigCache();
1343
1344        // if the new IAudioTrack is created, createTrack_l() will modify the
1345        // following member variables: mAudioTrack, mCblkMemory and mCblk.
1346        // It will also delete the strong references on previous IAudioTrack and IMemory
1347        result = createTrack_l(mStreamType,
1348                               cblk->sampleRate,
1349                               mFormat,
1350                               mChannelMask,
1351                               mFrameCount,
1352                               mFlags,
1353                               mSharedBuffer,
1354                               getOutput_l());
1355
1356        if (result == NO_ERROR) {
1357            uint32_t user = cblk->user;
1358            uint32_t server = cblk->server;
1359            // restore write index and set other indexes to reflect empty buffer status
1360            mCblk->user = user;
1361            mCblk->server = user;
1362            mCblk->userBase = user;
1363            mCblk->serverBase = user;
1364            // restore loop: this is not guaranteed to succeed if new frame count is not
1365            // compatible with loop length
1366            setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
1367            if (!fromStart) {
1368                mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1369                // Make sure that a client relying on callback events indicating underrun or
1370                // the actual amount of audio frames played (e.g SoundPool) receives them.
1371                if (mSharedBuffer == 0) {
1372                    uint32_t frames = 0;
1373                    if (user > server) {
1374                        frames = ((user - server) > mCblk->frameCount) ?
1375                                mCblk->frameCount : (user - server);
1376                        memset(mCblk->buffers, 0, frames * mCblk->frameSize);
1377                    }
1378                    // restart playback even if buffer is not completely filled.
1379                    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
1380                    // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
1381                    // the client
1382                    mCblk->stepUser(frames);
1383                }
1384            }
1385            if (mSharedBuffer != 0) {
1386                mCblk->stepUser(mCblk->frameCount);
1387            }
1388            if (mActive) {
1389                result = mAudioTrack->start();
1390                ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
1391            }
1392            if (fromStart && result == NO_ERROR) {
1393                mNewPosition = mCblk->server + mUpdatePeriod;
1394            }
1395        }
1396        if (result != NO_ERROR) {
1397            android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
1398            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
1399        }
1400        mRestoreStatus = result;
1401        // signal old cblk condition for other threads waiting for restore completion
1402        android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
1403        cblk->cv.broadcast();
1404    } else {
1405        if (!(cblk->flags & CBLK_RESTORED_MSK)) {
1406            ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid());
1407            mLock.unlock();
1408            result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
1409            if (result == NO_ERROR) {
1410                result = mRestoreStatus;
1411            }
1412            cblk->lock.unlock();
1413            mLock.lock();
1414        } else {
1415            ALOGW("dead IAudioTrack, already restored TID %d", gettid());
1416            result = mRestoreStatus;
1417            cblk->lock.unlock();
1418        }
1419    }
1420    ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
1421        result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
1422
1423    if (result == NO_ERROR) {
1424        // from now on we switch to the newly created cblk
1425        cblk = mCblk;
1426    }
1427    cblk->lock.lock();
1428
1429    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
1430
1431    return result;
1432}
1433
1434status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1435{
1436
1437    const size_t SIZE = 256;
1438    char buffer[SIZE];
1439    String8 result;
1440
1441    result.append(" AudioTrack::dump\n");
1442    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
1443    result.append(buffer);
1444    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, (mCblk == 0) ? 0 : mCblk->frameCount);
1445    result.append(buffer);
1446    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
1447    result.append(buffer);
1448    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1449    result.append(buffer);
1450    ::write(fd, result.string(), result.size());
1451    return NO_ERROR;
1452}
1453
1454// =========================================================================
1455
1456AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1457    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
1458{
1459}
1460
1461AudioTrack::AudioTrackThread::~AudioTrackThread()
1462{
1463}
1464
1465bool AudioTrack::AudioTrackThread::threadLoop()
1466{
1467    {
1468        AutoMutex _l(mMyLock);
1469        if (mPaused) {
1470            mMyCond.wait(mMyLock);
1471            // caller will check for exitPending()
1472            return true;
1473        }
1474    }
1475    if (!mReceiver.processAudioBuffer(this)) {
1476        pause();
1477    }
1478    return true;
1479}
1480
1481void AudioTrack::AudioTrackThread::requestExit()
1482{
1483    // must be in this order to avoid a race condition
1484    Thread::requestExit();
1485    resume();
1486}
1487
1488void AudioTrack::AudioTrackThread::pause()
1489{
1490    AutoMutex _l(mMyLock);
1491    mPaused = true;
1492}
1493
1494void AudioTrack::AudioTrackThread::resume()
1495{
1496    AutoMutex _l(mMyLock);
1497    if (mPaused) {
1498        mPaused = false;
1499        mMyCond.signal();
1500    }
1501}
1502
1503// =========================================================================
1504
1505
1506audio_track_cblk_t::audio_track_cblk_t()
1507    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1508    userBase(0), serverBase(0), buffers(NULL), frameCount(0),
1509    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
1510    mSendLevel(0), flags(0)
1511{
1512}
1513
1514uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1515{
1516    ALOGV("stepuser %08x %08x %d", user, server, frameCount);
1517
1518    uint32_t u = user;
1519    u += frameCount;
1520    // Ensure that user is never ahead of server for AudioRecord
1521    if (flags & CBLK_DIRECTION_MSK) {
1522        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1523        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1524            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1525        }
1526    } else if (u > server) {
1527        ALOGW("stepUser occurred after track reset");
1528        u = server;
1529    }
1530
1531    uint32_t fc = this->frameCount;
1532    if (u >= fc) {
1533        // common case, user didn't just wrap
1534        if (u - fc >= userBase ) {
1535            userBase += fc;
1536        }
1537    } else if (u >= userBase + fc) {
1538        // user just wrapped
1539        userBase += fc;
1540    }
1541
1542    user = u;
1543
1544    // Clear flow control error condition as new data has been written/read to/from buffer.
1545    if (flags & CBLK_UNDERRUN_MSK) {
1546        android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
1547    }
1548
1549    return u;
1550}
1551
1552bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1553{
1554    ALOGV("stepserver %08x %08x %d", user, server, frameCount);
1555
1556    if (!tryLock()) {
1557        ALOGW("stepServer() could not lock cblk");
1558        return false;
1559    }
1560
1561    uint32_t s = server;
1562    bool flushed = (s == user);
1563
1564    s += frameCount;
1565    if (flags & CBLK_DIRECTION_MSK) {
1566        // Mark that we have read the first buffer so that next time stepUser() is called
1567        // we switch to normal obtainBuffer() timeout period
1568        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1569            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1570        }
1571        // It is possible that we receive a flush()
1572        // while the mixer is processing a block: in this case,
1573        // stepServer() is called After the flush() has reset u & s and
1574        // we have s > u
1575        if (flushed) {
1576            ALOGW("stepServer occurred after track reset");
1577            s = user;
1578        }
1579    }
1580
1581    if (s >= loopEnd) {
1582        ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1583        s = loopStart;
1584        if (--loopCount == 0) {
1585            loopEnd = UINT_MAX;
1586            loopStart = UINT_MAX;
1587        }
1588    }
1589
1590    uint32_t fc = this->frameCount;
1591    if (s >= fc) {
1592        // common case, server didn't just wrap
1593        if (s - fc >= serverBase ) {
1594            serverBase += fc;
1595        }
1596    } else if (s >= serverBase + fc) {
1597        // server just wrapped
1598        serverBase += fc;
1599    }
1600
1601    server = s;
1602
1603    if (!(flags & CBLK_INVALID_MSK)) {
1604        cv.signal();
1605    }
1606    lock.unlock();
1607    return true;
1608}
1609
1610void* audio_track_cblk_t::buffer(uint32_t offset) const
1611{
1612    return (int8_t *)buffers + (offset - userBase) * frameSize;
1613}
1614
1615uint32_t audio_track_cblk_t::framesAvailable()
1616{
1617    Mutex::Autolock _l(lock);
1618    return framesAvailable_l();
1619}
1620
1621uint32_t audio_track_cblk_t::framesAvailable_l()
1622{
1623    uint32_t u = user;
1624    uint32_t s = server;
1625
1626    if (flags & CBLK_DIRECTION_MSK) {
1627        uint32_t limit = (s < loopStart) ? s : loopStart;
1628        return limit + frameCount - u;
1629    } else {
1630        return frameCount + u - s;
1631    }
1632}
1633
1634uint32_t audio_track_cblk_t::framesReady()
1635{
1636    uint32_t u = user;
1637    uint32_t s = server;
1638
1639    if (flags & CBLK_DIRECTION_MSK) {
1640        if (u < loopEnd) {
1641            return u - s;
1642        } else {
1643            // do not block on mutex shared with client on AudioFlinger side
1644            if (!tryLock()) {
1645                ALOGW("framesReady() could not lock cblk");
1646                return 0;
1647            }
1648            uint32_t frames = UINT_MAX;
1649            if (loopCount >= 0) {
1650                frames = (loopEnd - loopStart)*loopCount + u - s;
1651            }
1652            lock.unlock();
1653            return frames;
1654        }
1655    } else {
1656        return s - u;
1657    }
1658}
1659
1660bool audio_track_cblk_t::tryLock()
1661{
1662    // the code below simulates lock-with-timeout
1663    // we MUST do this to protect the AudioFlinger server
1664    // as this lock is shared with the client.
1665    status_t err;
1666
1667    err = lock.tryLock();
1668    if (err == -EBUSY) { // just wait a bit
1669        usleep(1000);
1670        err = lock.tryLock();
1671    }
1672    if (err != NO_ERROR) {
1673        // probably, the client just died.
1674        return false;
1675    }
1676    return true;
1677}
1678
1679// -------------------------------------------------------------------------
1680
1681}; // namespace android
1682