AudioTrack.cpp revision 847d05dc8fa144dcf8f4f435d6a6ac1727f00937 
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
123AudioTrack::AudioTrack(
124        audio_stream_type_t streamType,
125        uint32_t sampleRate,
126        audio_format_t format,
127        audio_channel_mask_t channelMask,
128        const sp<IMemory>& sharedBuffer,
129        audio_output_flags_t flags,
130        callback_t cbf,
131        void* user,
132        int notificationFrames,
133        int sessionId)
134    : mStatus(NO_INIT),
135      mIsTimed(false),
136      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
137      mPreviousSchedulingGroup(SP_DEFAULT)
138{
139    mStatus = set(streamType, sampleRate, format, channelMask,
140            0 /*frameCount*/, flags, cbf, user, notificationFrames,
141            sharedBuffer, false /*threadCanCallJava*/, sessionId);
142}
143
144AudioTrack::~AudioTrack()
145{
146    ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
147
148    if (mStatus == NO_ERROR) {
149        // Make sure that callback function exits in the case where
150        // it is looping on buffer full condition in obtainBuffer().
151        // Otherwise the callback thread will never exit.
152        stop();
153        if (mAudioTrackThread != 0) {
154            mAudioTrackThread->requestExit();   // see comment in AudioTrack.h
155            mAudioTrackThread->requestExitAndWait();
156            mAudioTrackThread.clear();
157        }
158        mAudioTrack.clear();
159        IPCThreadState::self()->flushCommands();
160        AudioSystem::releaseAudioSessionId(mSessionId);
161    }
162}
163
164status_t AudioTrack::set(
165        audio_stream_type_t streamType,
166        uint32_t sampleRate,
167        audio_format_t format,
168        audio_channel_mask_t channelMask,
169        int frameCount,
170        audio_output_flags_t flags,
171        callback_t cbf,
172        void* user,
173        int notificationFrames,
174        const sp<IMemory>& sharedBuffer,
175        bool threadCanCallJava,
176        int sessionId)
177{
178
179    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
180            sharedBuffer->size());
181
182    ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
183
184    AutoMutex lock(mLock);
185    if (mAudioTrack != 0) {
186        ALOGE("Track already in use");
187        return INVALID_OPERATION;
188    }
189
190    // handle default values first.
191    if (streamType == AUDIO_STREAM_DEFAULT) {
192        streamType = AUDIO_STREAM_MUSIC;
193    }
194
195    if (sampleRate == 0) {
196        int afSampleRate;
197        if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
198            return NO_INIT;
199        }
200        sampleRate = afSampleRate;
201    }
202
203    // these below should probably come from the audioFlinger too...
204    if (format == AUDIO_FORMAT_DEFAULT) {
205        format = AUDIO_FORMAT_PCM_16_BIT;
206    }
207    if (channelMask == 0) {
208        channelMask = AUDIO_CHANNEL_OUT_STEREO;
209    }
210
211    // validate parameters
212    if (!audio_is_valid_format(format)) {
213        ALOGE("Invalid format");
214        return BAD_VALUE;
215    }
216
217    // AudioFlinger does not currently support 8-bit data in shared memory
218    if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
219        ALOGE("8-bit data in shared memory is not supported");
220        return BAD_VALUE;
221    }
222
223    // force direct flag if format is not linear PCM
224    if (!audio_is_linear_pcm(format)) {
225        flags = (audio_output_flags_t)
226                // FIXME why can't we allow direct AND fast?
227                ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
228    }
229    // only allow deep buffering for music stream type
230    if (streamType != AUDIO_STREAM_MUSIC) {
231        flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
232    }
233
234    if (!audio_is_output_channel(channelMask)) {
235        ALOGE("Invalid channel mask %#x", channelMask);
236        return BAD_VALUE;
237    }
238    uint32_t channelCount = popcount(channelMask);
239
240    audio_io_handle_t output = AudioSystem::getOutput(
241                                    streamType,
242                                    sampleRate, format, channelMask,
243                                    flags);
244
245    if (output == 0) {
246        ALOGE("Could not get audio output for stream type %d", streamType);
247        return BAD_VALUE;
248    }
249
250    mVolume[LEFT] = 1.0f;
251    mVolume[RIGHT] = 1.0f;
252    mSendLevel = 0.0f;
253    mFrameCount = frameCount;
254    mNotificationFramesReq = notificationFrames;
255    mSessionId = sessionId;
256    mAuxEffectId = 0;
257    mFlags = flags;
258    mCbf = cbf;
259
260    if (cbf != NULL) {
261        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
262        mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
263    }
264
265    // create the IAudioTrack
266    status_t status = createTrack_l(streamType,
267                                  sampleRate,
268                                  format,
269                                  channelMask,
270                                  frameCount,
271                                  flags,
272                                  sharedBuffer,
273                                  output);
274
275    if (status != NO_ERROR) {
276        if (mAudioTrackThread != 0) {
277            mAudioTrackThread->requestExit();
278            mAudioTrackThread.clear();
279        }
280        return status;
281    }
282
283    mStatus = NO_ERROR;
284
285    mStreamType = streamType;
286    mFormat = format;
287    mChannelMask = channelMask;
288    mChannelCount = channelCount;
289    mSharedBuffer = sharedBuffer;
290    mMuted = false;
291    mActive = false;
292    mUserData = user;
293    mLoopCount = 0;
294    mMarkerPosition = 0;
295    mMarkerReached = false;
296    mNewPosition = 0;
297    mUpdatePeriod = 0;
298    mFlushed = false;
299    AudioSystem::acquireAudioSessionId(mSessionId);
300    mRestoreStatus = NO_ERROR;
301    return NO_ERROR;
302}
303
304status_t AudioTrack::initCheck() const
305{
306    return mStatus;
307}
308
309// -------------------------------------------------------------------------
310
311uint32_t AudioTrack::latency() const
312{
313    return mLatency;
314}
315
316audio_stream_type_t AudioTrack::streamType() const
317{
318    return mStreamType;
319}
320
321audio_format_t AudioTrack::format() const
322{
323    return mFormat;
324}
325
326int AudioTrack::channelCount() const
327{
328    return mChannelCount;
329}
330
331uint32_t AudioTrack::frameCount() const
332{
333    return mCblk->frameCount;
334}
335
336size_t AudioTrack::frameSize() const
337{
338    if (audio_is_linear_pcm(mFormat)) {
339        return channelCount()*audio_bytes_per_sample(mFormat);
340    } else {
341        return sizeof(uint8_t);
342    }
343}
344
345sp<IMemory>& AudioTrack::sharedBuffer()
346{
347    return mSharedBuffer;
348}
349
350// -------------------------------------------------------------------------
351
352void AudioTrack::start()
353{
354    sp<AudioTrackThread> t = mAudioTrackThread;
355
356    ALOGV("start %p", this);
357
358    AutoMutex lock(mLock);
359    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
360    // while we are accessing the cblk
361    sp<IAudioTrack> audioTrack = mAudioTrack;
362    sp<IMemory> iMem = mCblkMemory;
363    audio_track_cblk_t* cblk = mCblk;
364
365    if (!mActive) {
366        mFlushed = false;
367        mActive = true;
368        mNewPosition = cblk->server + mUpdatePeriod;
369        cblk->lock.lock();
370        cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
371        cblk->waitTimeMs = 0;
372        android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
373        if (t != 0) {
374            t->resume();
375        } else {
376            mPreviousPriority = getpriority(PRIO_PROCESS, 0);
377            get_sched_policy(0, &mPreviousSchedulingGroup);
378            androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
379        }
380
381        ALOGV("start %p before lock cblk %p", this, mCblk);
382        status_t status = NO_ERROR;
383        if (!(cblk->flags & CBLK_INVALID_MSK)) {
384            cblk->lock.unlock();
385            ALOGV("mAudioTrack->start()");
386            status = mAudioTrack->start();
387            cblk->lock.lock();
388            if (status == DEAD_OBJECT) {
389                android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
390            }
391        }
392        if (cblk->flags & CBLK_INVALID_MSK) {
393            status = restoreTrack_l(cblk, true);
394        }
395        cblk->lock.unlock();
396        if (status != NO_ERROR) {
397            ALOGV("start() failed");
398            mActive = false;
399            if (t != 0) {
400                t->pause();
401            } else {
402                setpriority(PRIO_PROCESS, 0, mPreviousPriority);
403                set_sched_policy(0, mPreviousSchedulingGroup);
404            }
405        }
406    }
407
408}
409
410void AudioTrack::stop()
411{
412    sp<AudioTrackThread> t = mAudioTrackThread;
413
414    ALOGV("stop %p", this);
415
416    AutoMutex lock(mLock);
417    if (mActive) {
418        mActive = false;
419        mCblk->cv.signal();
420        mAudioTrack->stop();
421        // Cancel loops (If we are in the middle of a loop, playback
422        // would not stop until loopCount reaches 0).
423        setLoop_l(0, 0, 0);
424        // the playback head position will reset to 0, so if a marker is set, we need
425        // to activate it again
426        mMarkerReached = false;
427        // Force flush if a shared buffer is used otherwise audioflinger
428        // will not stop before end of buffer is reached.
429        if (mSharedBuffer != 0) {
430            flush_l();
431        }
432        if (t != 0) {
433            t->pause();
434        } else {
435            setpriority(PRIO_PROCESS, 0, mPreviousPriority);
436            set_sched_policy(0, mPreviousSchedulingGroup);
437        }
438    }
439
440}
441
442bool AudioTrack::stopped() const
443{
444    AutoMutex lock(mLock);
445    return stopped_l();
446}
447
448void AudioTrack::flush()
449{
450    AutoMutex lock(mLock);
451    flush_l();
452}
453
454// must be called with mLock held
455void AudioTrack::flush_l()
456{
457    ALOGV("flush");
458
459    // clear playback marker and periodic update counter
460    mMarkerPosition = 0;
461    mMarkerReached = false;
462    mUpdatePeriod = 0;
463
464    if (!mActive) {
465        mFlushed = true;
466        mAudioTrack->flush();
467        // Release AudioTrack callback thread in case it was waiting for new buffers
468        // in AudioTrack::obtainBuffer()
469        mCblk->cv.signal();
470    }
471}
472
473void AudioTrack::pause()
474{
475    ALOGV("pause");
476    AutoMutex lock(mLock);
477    if (mActive) {
478        mActive = false;
479        mCblk->cv.signal();
480        mAudioTrack->pause();
481    }
482}
483
484void AudioTrack::mute(bool e)
485{
486    mAudioTrack->mute(e);
487    mMuted = e;
488}
489
490bool AudioTrack::muted() const
491{
492    return mMuted;
493}
494
495status_t AudioTrack::setVolume(float left, float right)
496{
497    if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
498        return BAD_VALUE;
499    }
500
501    AutoMutex lock(mLock);
502    mVolume[LEFT] = left;
503    mVolume[RIGHT] = right;
504
505    mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
506
507    return NO_ERROR;
508}
509
510void AudioTrack::getVolume(float* left, float* right) const
511{
512    if (left != NULL) {
513        *left  = mVolume[LEFT];
514    }
515    if (right != NULL) {
516        *right = mVolume[RIGHT];
517    }
518}
519
520status_t AudioTrack::setAuxEffectSendLevel(float level)
521{
522    ALOGV("setAuxEffectSendLevel(%f)", level);
523    if (level < 0.0f || level > 1.0f) {
524        return BAD_VALUE;
525    }
526    AutoMutex lock(mLock);
527
528    mSendLevel = level;
529
530    mCblk->setSendLevel(level);
531
532    return NO_ERROR;
533}
534
535void AudioTrack::getAuxEffectSendLevel(float* level) const
536{
537    if (level != NULL) {
538        *level  = mSendLevel;
539    }
540}
541
542status_t AudioTrack::setSampleRate(int rate)
543{
544    int afSamplingRate;
545
546    if (mIsTimed) {
547        return INVALID_OPERATION;
548    }
549
550    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
551        return NO_INIT;
552    }
553    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
554    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
555
556    AutoMutex lock(mLock);
557    mCblk->sampleRate = rate;
558    return NO_ERROR;
559}
560
561uint32_t AudioTrack::getSampleRate() const
562{
563    if (mIsTimed) {
564        return INVALID_OPERATION;
565    }
566
567    AutoMutex lock(mLock);
568    return mCblk->sampleRate;
569}
570
571status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
572{
573    AutoMutex lock(mLock);
574    return setLoop_l(loopStart, loopEnd, loopCount);
575}
576
577// must be called with mLock held
578status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
579{
580    audio_track_cblk_t* cblk = mCblk;
581
582    Mutex::Autolock _l(cblk->lock);
583
584    if (loopCount == 0) {
585        cblk->loopStart = UINT_MAX;
586        cblk->loopEnd = UINT_MAX;
587        cblk->loopCount = 0;
588        mLoopCount = 0;
589        return NO_ERROR;
590    }
591
592    if (mIsTimed) {
593        return INVALID_OPERATION;
594    }
595
596    if (loopStart >= loopEnd ||
597        loopEnd - loopStart > cblk->frameCount ||
598        cblk->server > loopStart) {
599        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, "
600              "user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
601        return BAD_VALUE;
602    }
603
604    if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
605        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, "
606            "framecount %d",
607            loopStart, loopEnd, cblk->frameCount);
608        return BAD_VALUE;
609    }
610
611    cblk->loopStart = loopStart;
612    cblk->loopEnd = loopEnd;
613    cblk->loopCount = loopCount;
614    mLoopCount = loopCount;
615
616    return NO_ERROR;
617}
618
619status_t AudioTrack::setMarkerPosition(uint32_t marker)
620{
621    if (mCbf == NULL) return INVALID_OPERATION;
622
623    mMarkerPosition = marker;
624    mMarkerReached = false;
625
626    return NO_ERROR;
627}
628
629status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
630{
631    if (marker == NULL) return BAD_VALUE;
632
633    *marker = mMarkerPosition;
634
635    return NO_ERROR;
636}
637
638status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
639{
640    if (mCbf == NULL) return INVALID_OPERATION;
641
642    uint32_t curPosition;
643    getPosition(&curPosition);
644    mNewPosition = curPosition + updatePeriod;
645    mUpdatePeriod = updatePeriod;
646
647    return NO_ERROR;
648}
649
650status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
651{
652    if (updatePeriod == NULL) return BAD_VALUE;
653
654    *updatePeriod = mUpdatePeriod;
655
656    return NO_ERROR;
657}
658
659status_t AudioTrack::setPosition(uint32_t position)
660{
661    if (mIsTimed) return INVALID_OPERATION;
662
663    AutoMutex lock(mLock);
664
665    if (!stopped_l()) return INVALID_OPERATION;
666
667    Mutex::Autolock _l(mCblk->lock);
668
669    if (position > mCblk->user) return BAD_VALUE;
670
671    mCblk->server = position;
672    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
673
674    return NO_ERROR;
675}
676
677status_t AudioTrack::getPosition(uint32_t *position)
678{
679    if (position == NULL) return BAD_VALUE;
680    AutoMutex lock(mLock);
681    *position = mFlushed ? 0 : mCblk->server;
682
683    return NO_ERROR;
684}
685
686status_t AudioTrack::reload()
687{
688    AutoMutex lock(mLock);
689
690    if (!stopped_l()) return INVALID_OPERATION;
691
692    flush_l();
693
694    mCblk->stepUser(mCblk->frameCount);
695
696    return NO_ERROR;
697}
698
699audio_io_handle_t AudioTrack::getOutput()
700{
701    AutoMutex lock(mLock);
702    return getOutput_l();
703}
704
705// must be called with mLock held
706audio_io_handle_t AudioTrack::getOutput_l()
707{
708    return AudioSystem::getOutput(mStreamType,
709            mCblk->sampleRate, mFormat, mChannelMask, mFlags);
710}
711
712int AudioTrack::getSessionId() const
713{
714    return mSessionId;
715}
716
717status_t AudioTrack::attachAuxEffect(int effectId)
718{
719    ALOGV("attachAuxEffect(%d)", effectId);
720    status_t status = mAudioTrack->attachAuxEffect(effectId);
721    if (status == NO_ERROR) {
722        mAuxEffectId = effectId;
723    }
724    return status;
725}
726
727// -------------------------------------------------------------------------
728
729// must be called with mLock held
730status_t AudioTrack::createTrack_l(
731        audio_stream_type_t streamType,
732        uint32_t sampleRate,
733        audio_format_t format,
734        audio_channel_mask_t channelMask,
735        int frameCount,
736        audio_output_flags_t flags,
737        const sp<IMemory>& sharedBuffer,
738        audio_io_handle_t output)
739{
740    status_t status;
741    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
742    if (audioFlinger == 0) {
743        ALOGE("Could not get audioflinger");
744        return NO_INIT;
745    }
746
747    uint32_t afLatency;
748    if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
749        return NO_INIT;
750    }
751
752    // Client decides whether the track is TIMED (see below), but can only express a preference
753    // for FAST.  Server will perform additional tests.
754    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
755            // either of these use cases:
756            // use case 1: shared buffer
757            (sharedBuffer != 0) ||
758            // use case 2: callback handler
759            (mCbf != NULL))) {
760        ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
761        // once denied, do not request again if IAudioTrack is re-created
762        flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
763        mFlags = flags;
764    }
765    ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
766
767    mNotificationFramesAct = mNotificationFramesReq;
768
769    if (!audio_is_linear_pcm(format)) {
770
771        if (sharedBuffer != 0) {
772            // Same comment as below about ignoring frameCount parameter for set()
773            frameCount = sharedBuffer->size();
774        } else if (frameCount == 0) {
775            int afFrameCount;
776            if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
777                return NO_INIT;
778            }
779            frameCount = afFrameCount;
780        }
781
782    } else if (sharedBuffer != 0) {
783
784        // Ensure that buffer alignment matches channelCount
785        int channelCount = popcount(channelMask);
786        // 8-bit data in shared memory is not currently supported by AudioFlinger
787        size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
788        if (channelCount > 1) {
789            // More than 2 channels does not require stronger alignment than stereo
790            alignment <<= 1;
791        }
792        if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
793            ALOGE("Invalid buffer alignment: address %p, channelCount %d",
794                    sharedBuffer->pointer(), channelCount);
795            return BAD_VALUE;
796        }
797
798        // When initializing a shared buffer AudioTrack via constructors,
799        // there's no frameCount parameter.
800        // But when initializing a shared buffer AudioTrack via set(),
801        // there _is_ a frameCount parameter.  We silently ignore it.
802        frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
803
804    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
805
806        // FIXME move these calculations and associated checks to server
807        int afSampleRate;
808        if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
809            return NO_INIT;
810        }
811        int afFrameCount;
812        if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
813            return NO_INIT;
814        }
815
816        // Ensure that buffer depth covers at least audio hardware latency
817        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
818        if (minBufCount < 2) minBufCount = 2;
819
820        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
821        ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
822                ", afLatency=%d",
823                minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
824
825        if (frameCount == 0) {
826            frameCount = minFrameCount;
827        }
828        if (mNotificationFramesAct == 0) {
829            mNotificationFramesAct = frameCount/2;
830        }
831        // Make sure that application is notified with sufficient margin
832        // before underrun
833        if (mNotificationFramesAct > (uint32_t)frameCount/2) {
834            mNotificationFramesAct = frameCount/2;
835        }
836        if (frameCount < minFrameCount) {
837            // not ALOGW because it happens all the time when playing key clicks over A2DP
838            ALOGV("Minimum buffer size corrected from %d to %d",
839                     frameCount, minFrameCount);
840            frameCount = minFrameCount;
841        }
842
843    } else {
844        // For fast tracks, the frame count calculations and checks are done by server
845    }
846
847    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
848    if (mIsTimed) {
849        trackFlags |= IAudioFlinger::TRACK_TIMED;
850    }
851
852    pid_t tid = -1;
853    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
854        trackFlags |= IAudioFlinger::TRACK_FAST;
855        if (mAudioTrackThread != 0) {
856            tid = mAudioTrackThread->getTid();
857        }
858    }
859
860    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
861                                                      streamType,
862                                                      sampleRate,
863                                                      format,
864                                                      channelMask,
865                                                      frameCount,
866                                                      trackFlags,
867                                                      sharedBuffer,
868                                                      output,
869                                                      tid,
870                                                      &mSessionId,
871                                                      &status);
872
873    if (track == 0) {
874        ALOGE("AudioFlinger could not create track, status: %d", status);
875        return status;
876    }
877    sp<IMemory> cblk = track->getCblk();
878    if (cblk == 0) {
879        ALOGE("Could not get control block");
880        return NO_INIT;
881    }
882    mAudioTrack = track;
883    mCblkMemory = cblk;
884    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
885    // old has the previous value of mCblk->flags before the "or" operation
886    int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
887    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
888        if (old & CBLK_FAST) {
889            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount);
890        } else {
891            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount);
892            // once denied, do not request again if IAudioTrack is re-created
893            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
894            mFlags = flags;
895        }
896        if (sharedBuffer == 0) {
897            mNotificationFramesAct = mCblk->frameCount/2;
898        }
899    }
900    if (sharedBuffer == 0) {
901        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
902    } else {
903        mCblk->buffers = sharedBuffer->pointer();
904        // Force buffer full condition as data is already present in shared memory
905        mCblk->stepUser(mCblk->frameCount);
906    }
907
908    mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
909            uint16_t(mVolume[LEFT] * 0x1000));
910    mCblk->setSendLevel(mSendLevel);
911    mAudioTrack->attachAuxEffect(mAuxEffectId);
912    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
913    mCblk->waitTimeMs = 0;
914    mRemainingFrames = mNotificationFramesAct;
915    // FIXME don't believe this lie
916    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
917    // If IAudioTrack is re-created, don't let the requested frameCount
918    // decrease.  This can confuse clients that cache frameCount().
919    if (mCblk->frameCount > mFrameCount) {
920        mFrameCount = mCblk->frameCount;
921    }
922    return NO_ERROR;
923}
924
925status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
926{
927    AutoMutex lock(mLock);
928    bool active;
929    status_t result = NO_ERROR;
930    audio_track_cblk_t* cblk = mCblk;
931    uint32_t framesReq = audioBuffer->frameCount;
932    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
933
934    audioBuffer->frameCount  = 0;
935    audioBuffer->size = 0;
936
937    uint32_t framesAvail = cblk->framesAvailable();
938
939    cblk->lock.lock();
940    if (cblk->flags & CBLK_INVALID_MSK) {
941        goto create_new_track;
942    }
943    cblk->lock.unlock();
944
945    if (framesAvail == 0) {
946        cblk->lock.lock();
947        goto start_loop_here;
948        while (framesAvail == 0) {
949            active = mActive;
950            if (CC_UNLIKELY(!active)) {
951                ALOGV("Not active and NO_MORE_BUFFERS");
952                cblk->lock.unlock();
953                return NO_MORE_BUFFERS;
954            }
955            if (CC_UNLIKELY(!waitCount)) {
956                cblk->lock.unlock();
957                return WOULD_BLOCK;
958            }
959            if (!(cblk->flags & CBLK_INVALID_MSK)) {
960                mLock.unlock();
961                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
962                cblk->lock.unlock();
963                mLock.lock();
964                if (!mActive) {
965                    return status_t(STOPPED);
966                }
967                cblk->lock.lock();
968            }
969
970            if (cblk->flags & CBLK_INVALID_MSK) {
971                goto create_new_track;
972            }
973            if (CC_UNLIKELY(result != NO_ERROR)) {
974                cblk->waitTimeMs += waitTimeMs;
975                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
976                    // timing out when a loop has been set and we have already written upto loop end
977                    // is a normal condition: no need to wake AudioFlinger up.
978                    if (cblk->user < cblk->loopEnd) {
979                        ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, "
980                              "server=%08x", this, cblk->mName, cblk->user, cblk->server);
981                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
982                        cblk->lock.unlock();
983                        result = mAudioTrack->start();
984                        cblk->lock.lock();
985                        if (result == DEAD_OBJECT) {
986                            android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
987create_new_track:
988                            result = restoreTrack_l(cblk, false);
989                        }
990                        if (result != NO_ERROR) {
991                            ALOGW("obtainBuffer create Track error %d", result);
992                            cblk->lock.unlock();
993                            return result;
994                        }
995                    }
996                    cblk->waitTimeMs = 0;
997                }
998
999                if (--waitCount == 0) {
1000                    cblk->lock.unlock();
1001                    return TIMED_OUT;
1002                }
1003            }
1004            // read the server count again
1005        start_loop_here:
1006            framesAvail = cblk->framesAvailable_l();
1007        }
1008        cblk->lock.unlock();
1009    }
1010
1011    cblk->waitTimeMs = 0;
1012
1013    if (framesReq > framesAvail) {
1014        framesReq = framesAvail;
1015    }
1016
1017    uint32_t u = cblk->user;
1018    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
1019
1020    if (framesReq > bufferEnd - u) {
1021        framesReq = bufferEnd - u;
1022    }
1023
1024    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
1025    audioBuffer->channelCount = mChannelCount;
1026    audioBuffer->frameCount = framesReq;
1027    audioBuffer->size = framesReq * cblk->frameSize;
1028    if (audio_is_linear_pcm(mFormat)) {
1029        audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
1030    } else {
1031        audioBuffer->format = mFormat;
1032    }
1033    audioBuffer->raw = (int8_t *)cblk->buffer(u);
1034    active = mActive;
1035    return active ? status_t(NO_ERROR) : status_t(STOPPED);
1036}
1037
1038void AudioTrack::releaseBuffer(Buffer* audioBuffer)
1039{
1040    AutoMutex lock(mLock);
1041    mCblk->stepUser(audioBuffer->frameCount);
1042    if (audioBuffer->frameCount > 0) {
1043        // restart track if it was disabled by audioflinger due to previous underrun
1044        if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
1045            android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
1046            ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName);
1047            mAudioTrack->start();
1048        }
1049    }
1050}
1051
1052// -------------------------------------------------------------------------
1053
1054ssize_t AudioTrack::write(const void* buffer, size_t userSize)
1055{
1056
1057    if (mSharedBuffer != 0) return INVALID_OPERATION;
1058    if (mIsTimed) return INVALID_OPERATION;
1059
1060    if (ssize_t(userSize) < 0) {
1061        // Sanity-check: user is most-likely passing an error code, and it would
1062        // make the return value ambiguous (actualSize vs error).
1063        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
1064                buffer, userSize, userSize);
1065        return BAD_VALUE;
1066    }
1067
1068    ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
1069
1070    if (userSize == 0) {
1071        return 0;
1072    }
1073
1074    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1075    // while we are accessing the cblk
1076    mLock.lock();
1077    sp<IAudioTrack> audioTrack = mAudioTrack;
1078    sp<IMemory> iMem = mCblkMemory;
1079    mLock.unlock();
1080
1081    ssize_t written = 0;
1082    const int8_t *src = (const int8_t *)buffer;
1083    Buffer audioBuffer;
1084    size_t frameSz = frameSize();
1085
1086    do {
1087        audioBuffer.frameCount = userSize/frameSz;
1088
1089        status_t err = obtainBuffer(&audioBuffer, -1);
1090        if (err < 0) {
1091            // out of buffers, return #bytes written
1092            if (err == status_t(NO_MORE_BUFFERS))
1093                break;
1094            return ssize_t(err);
1095        }
1096
1097        size_t toWrite;
1098
1099        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1100            // Divide capacity by 2 to take expansion into account
1101            toWrite = audioBuffer.size>>1;
1102            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
1103        } else {
1104            toWrite = audioBuffer.size;
1105            memcpy(audioBuffer.i8, src, toWrite);
1106            src += toWrite;
1107        }
1108        userSize -= toWrite;
1109        written += toWrite;
1110
1111        releaseBuffer(&audioBuffer);
1112    } while (userSize >= frameSz);
1113
1114    return written;
1115}
1116
1117// -------------------------------------------------------------------------
1118
1119TimedAudioTrack::TimedAudioTrack() {
1120    mIsTimed = true;
1121}
1122
1123status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
1124{
1125    status_t result = UNKNOWN_ERROR;
1126
1127    // If the track is not invalid already, try to allocate a buffer.  alloc
1128    // fails indicating that the server is dead, flag the track as invalid so
1129    // we can attempt to restore in just a bit.
1130    if (!(mCblk->flags & CBLK_INVALID_MSK)) {
1131        result = mAudioTrack->allocateTimedBuffer(size, buffer);
1132        if (result == DEAD_OBJECT) {
1133            android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
1134        }
1135    }
1136
1137    // If the track is invalid at this point, attempt to restore it. and try the
1138    // allocation one more time.
1139    if (mCblk->flags & CBLK_INVALID_MSK) {
1140        mCblk->lock.lock();
1141        result = restoreTrack_l(mCblk, false);
1142        mCblk->lock.unlock();
1143
1144        if (result == OK)
1145            result = mAudioTrack->allocateTimedBuffer(size, buffer);
1146    }
1147
1148    return result;
1149}
1150
1151status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
1152                                           int64_t pts)
1153{
1154    status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
1155    {
1156        AutoMutex lock(mLock);
1157        // restart track if it was disabled by audioflinger due to previous underrun
1158        if (buffer->size() != 0 && status == NO_ERROR &&
1159                mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
1160            android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
1161            ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
1162            mAudioTrack->start();
1163        }
1164    }
1165    return status;
1166}
1167
1168status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
1169                                                TargetTimeline target)
1170{
1171    return mAudioTrack->setMediaTimeTransform(xform, target);
1172}
1173
1174// -------------------------------------------------------------------------
1175
1176bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
1177{
1178    Buffer audioBuffer;
1179    uint32_t frames;
1180    size_t writtenSize;
1181
1182    mLock.lock();
1183    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1184    // while we are accessing the cblk
1185    sp<IAudioTrack> audioTrack = mAudioTrack;
1186    sp<IMemory> iMem = mCblkMemory;
1187    audio_track_cblk_t* cblk = mCblk;
1188    bool active = mActive;
1189    mLock.unlock();
1190
1191    // Manage underrun callback
1192    if (active && (cblk->framesAvailable() == cblk->frameCount)) {
1193        ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
1194        if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
1195            mCbf(EVENT_UNDERRUN, mUserData, 0);
1196            if (cblk->server == cblk->frameCount) {
1197                mCbf(EVENT_BUFFER_END, mUserData, 0);
1198            }
1199            if (mSharedBuffer != 0) return false;
1200        }
1201    }
1202
1203    // Manage loop end callback
1204    while (mLoopCount > cblk->loopCount) {
1205        int loopCount = -1;
1206        mLoopCount--;
1207        if (mLoopCount >= 0) loopCount = mLoopCount;
1208
1209        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
1210    }
1211
1212    // Manage marker callback
1213    if (!mMarkerReached && (mMarkerPosition > 0)) {
1214        if (cblk->server >= mMarkerPosition) {
1215            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
1216            mMarkerReached = true;
1217        }
1218    }
1219
1220    // Manage new position callback
1221    if (mUpdatePeriod > 0) {
1222        while (cblk->server >= mNewPosition) {
1223            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
1224            mNewPosition += mUpdatePeriod;
1225        }
1226    }
1227
1228    // If Shared buffer is used, no data is requested from client.
1229    if (mSharedBuffer != 0) {
1230        frames = 0;
1231    } else {
1232        frames = mRemainingFrames;
1233    }
1234
1235    // See description of waitCount parameter at declaration of obtainBuffer().
1236    // The logic below prevents us from being stuck below at obtainBuffer()
1237    // not being able to handle timed events (position, markers, loops).
1238    int32_t waitCount = -1;
1239    if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
1240        waitCount = 1;
1241    }
1242
1243    do {
1244
1245        audioBuffer.frameCount = frames;
1246
1247        status_t err = obtainBuffer(&audioBuffer, waitCount);
1248        if (err < NO_ERROR) {
1249            if (err != TIMED_OUT) {
1250                ALOGE_IF(err != status_t(NO_MORE_BUFFERS),
1251                        "Error obtaining an audio buffer, giving up.");
1252                return false;
1253            }
1254            break;
1255        }
1256        if (err == status_t(STOPPED)) return false;
1257
1258        // Divide buffer size by 2 to take into account the expansion
1259        // due to 8 to 16 bit conversion: the callback must fill only half
1260        // of the destination buffer
1261        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1262            audioBuffer.size >>= 1;
1263        }
1264
1265        size_t reqSize = audioBuffer.size;
1266        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1267        writtenSize = audioBuffer.size;
1268
1269        // Sanity check on returned size
1270        if (ssize_t(writtenSize) <= 0) {
1271            // The callback is done filling buffers
1272            // Keep this thread going to handle timed events and
1273            // still try to get more data in intervals of WAIT_PERIOD_MS
1274            // but don't just loop and block the CPU, so wait
1275            usleep(WAIT_PERIOD_MS*1000);
1276            break;
1277        }
1278
1279        if (writtenSize > reqSize) writtenSize = reqSize;
1280
1281        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1282            // 8 to 16 bit conversion, note that source and destination are the same address
1283            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
1284            writtenSize <<= 1;
1285        }
1286
1287        audioBuffer.size = writtenSize;
1288        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
1289        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
1290        // 16 bit.
1291        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
1292
1293        frames -= audioBuffer.frameCount;
1294
1295        releaseBuffer(&audioBuffer);
1296    }
1297    while (frames);
1298
1299    if (frames == 0) {
1300        mRemainingFrames = mNotificationFramesAct;
1301    } else {
1302        mRemainingFrames = frames;
1303    }
1304    return true;
1305}
1306
1307// must be called with mLock and cblk.lock held. Callers must also hold strong references on
1308// the IAudioTrack and IMemory in case they are recreated here.
1309// If the IAudioTrack is successfully restored, the cblk pointer is updated
1310status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart)
1311{
1312    status_t result;
1313
1314    if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
1315        ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
1316            fromStart ? "start()" : "obtainBuffer()", gettid());
1317
1318        // signal old cblk condition so that other threads waiting for available buffers stop
1319        // waiting now
1320        cblk->cv.broadcast();
1321        cblk->lock.unlock();
1322
1323        // refresh the audio configuration cache in this process to make sure we get new
1324        // output parameters in getOutput_l() and createTrack_l()
1325        AudioSystem::clearAudioConfigCache();
1326
1327        // if the new IAudioTrack is created, createTrack_l() will modify the
1328        // following member variables: mAudioTrack, mCblkMemory and mCblk.
1329        // It will also delete the strong references on previous IAudioTrack and IMemory
1330        result = createTrack_l(mStreamType,
1331                               cblk->sampleRate,
1332                               mFormat,
1333                               mChannelMask,
1334                               mFrameCount,
1335                               mFlags,
1336                               mSharedBuffer,
1337                               getOutput_l());
1338
1339        if (result == NO_ERROR) {
1340            uint32_t user = cblk->user;
1341            uint32_t server = cblk->server;
1342            // restore write index and set other indexes to reflect empty buffer status
1343            mCblk->user = user;
1344            mCblk->server = user;
1345            mCblk->userBase = user;
1346            mCblk->serverBase = user;
1347            // restore loop: this is not guaranteed to succeed if new frame count is not
1348            // compatible with loop length
1349            setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
1350            if (!fromStart) {
1351                mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1352                // Make sure that a client relying on callback events indicating underrun or
1353                // the actual amount of audio frames played (e.g SoundPool) receives them.
1354                if (mSharedBuffer == 0) {
1355                    uint32_t frames = 0;
1356                    if (user > server) {
1357                        frames = ((user - server) > mCblk->frameCount) ?
1358                                mCblk->frameCount : (user - server);
1359                        memset(mCblk->buffers, 0, frames * mCblk->frameSize);
1360                    }
1361                    // restart playback even if buffer is not completely filled.
1362                    android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags);
1363                    // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to
1364                    // the client
1365                    mCblk->stepUser(frames);
1366                }
1367            }
1368            if (mSharedBuffer != 0) {
1369                mCblk->stepUser(mCblk->frameCount);
1370            }
1371            if (mActive) {
1372                result = mAudioTrack->start();
1373                ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
1374            }
1375            if (fromStart && result == NO_ERROR) {
1376                mNewPosition = mCblk->server + mUpdatePeriod;
1377            }
1378        }
1379        if (result != NO_ERROR) {
1380            android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags);
1381            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
1382        }
1383        mRestoreStatus = result;
1384        // signal old cblk condition for other threads waiting for restore completion
1385        android_atomic_or(CBLK_RESTORED_ON, &cblk->flags);
1386        cblk->cv.broadcast();
1387    } else {
1388        bool haveLogged = false;
1389        for (;;) {
1390            if (cblk->flags & CBLK_RESTORED_MSK) {
1391                ALOGW("dead IAudioTrack restored");
1392                result = mRestoreStatus;
1393                cblk->lock.unlock();
1394                break;
1395            }
1396            if (!haveLogged) {
1397                ALOGW("dead IAudioTrack, waiting for a new one");
1398                haveLogged = true;
1399            }
1400            mLock.unlock();
1401            result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
1402            cblk->lock.unlock();
1403            mLock.lock();
1404            if (result != NO_ERROR) {
1405                ALOGW("timed out");
1406                break;
1407            }
1408            cblk->lock.lock();
1409        }
1410    }
1411    ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
1412        result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
1413
1414    if (result == NO_ERROR) {
1415        // from now on we switch to the newly created cblk
1416        cblk = mCblk;
1417    }
1418    cblk->lock.lock();
1419
1420    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
1421
1422    return result;
1423}
1424
1425status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1426{
1427
1428    const size_t SIZE = 256;
1429    char buffer[SIZE];
1430    String8 result;
1431
1432    result.append(" AudioTrack::dump\n");
1433    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
1434            mVolume[0], mVolume[1]);
1435    result.append(buffer);
1436    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
1437            mChannelCount, mCblk->frameCount);
1438    result.append(buffer);
1439    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n",
1440            (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
1441    result.append(buffer);
1442    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1443    result.append(buffer);
1444    ::write(fd, result.string(), result.size());
1445    return NO_ERROR;
1446}
1447
1448// =========================================================================
1449
1450AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1451    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
1452{
1453}
1454
1455AudioTrack::AudioTrackThread::~AudioTrackThread()
1456{
1457}
1458
1459bool AudioTrack::AudioTrackThread::threadLoop()
1460{
1461    {
1462        AutoMutex _l(mMyLock);
1463        if (mPaused) {
1464            mMyCond.wait(mMyLock);
1465            // caller will check for exitPending()
1466            return true;
1467        }
1468    }
1469    if (!mReceiver.processAudioBuffer(this)) {
1470        pause();
1471    }
1472    return true;
1473}
1474
1475void AudioTrack::AudioTrackThread::requestExit()
1476{
1477    // must be in this order to avoid a race condition
1478    Thread::requestExit();
1479    resume();
1480}
1481
1482void AudioTrack::AudioTrackThread::pause()
1483{
1484    AutoMutex _l(mMyLock);
1485    mPaused = true;
1486}
1487
1488void AudioTrack::AudioTrackThread::resume()
1489{
1490    AutoMutex _l(mMyLock);
1491    if (mPaused) {
1492        mPaused = false;
1493        mMyCond.signal();
1494    }
1495}
1496
1497// =========================================================================
1498
1499
1500audio_track_cblk_t::audio_track_cblk_t()
1501    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1502    userBase(0), serverBase(0), buffers(NULL), frameCount(0),
1503    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
1504    mSendLevel(0), flags(0)
1505{
1506}
1507
1508uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1509{
1510    ALOGV("stepuser %08x %08x %d", user, server, frameCount);
1511
1512    uint32_t u = user;
1513    u += frameCount;
1514    // Ensure that user is never ahead of server for AudioRecord
1515    if (flags & CBLK_DIRECTION_MSK) {
1516        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1517        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1518            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1519        }
1520    } else if (u > server) {
1521        ALOGW("stepUser occurred after track reset");
1522        u = server;
1523    }
1524
1525    uint32_t fc = this->frameCount;
1526    if (u >= fc) {
1527        // common case, user didn't just wrap
1528        if (u - fc >= userBase ) {
1529            userBase += fc;
1530        }
1531    } else if (u >= userBase + fc) {
1532        // user just wrapped
1533        userBase += fc;
1534    }
1535
1536    user = u;
1537
1538    // Clear flow control error condition as new data has been written/read to/from buffer.
1539    if (flags & CBLK_UNDERRUN_MSK) {
1540        android_atomic_and(~CBLK_UNDERRUN_MSK, &flags);
1541    }
1542
1543    return u;
1544}
1545
1546bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1547{
1548    ALOGV("stepserver %08x %08x %d", user, server, frameCount);
1549
1550    if (!tryLock()) {
1551        ALOGW("stepServer() could not lock cblk");
1552        return false;
1553    }
1554
1555    uint32_t s = server;
1556    bool flushed = (s == user);
1557
1558    s += frameCount;
1559    if (flags & CBLK_DIRECTION_MSK) {
1560        // Mark that we have read the first buffer so that next time stepUser() is called
1561        // we switch to normal obtainBuffer() timeout period
1562        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1563            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1564        }
1565        // It is possible that we receive a flush()
1566        // while the mixer is processing a block: in this case,
1567        // stepServer() is called After the flush() has reset u & s and
1568        // we have s > u
1569        if (flushed) {
1570            ALOGW("stepServer occurred after track reset");
1571            s = user;
1572        }
1573    }
1574
1575    if (s >= loopEnd) {
1576        ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1577        s = loopStart;
1578        if (--loopCount == 0) {
1579            loopEnd = UINT_MAX;
1580            loopStart = UINT_MAX;
1581        }
1582    }
1583
1584    uint32_t fc = this->frameCount;
1585    if (s >= fc) {
1586        // common case, server didn't just wrap
1587        if (s - fc >= serverBase ) {
1588            serverBase += fc;
1589        }
1590    } else if (s >= serverBase + fc) {
1591        // server just wrapped
1592        serverBase += fc;
1593    }
1594
1595    server = s;
1596
1597    if (!(flags & CBLK_INVALID_MSK)) {
1598        cv.signal();
1599    }
1600    lock.unlock();
1601    return true;
1602}
1603
1604void* audio_track_cblk_t::buffer(uint32_t offset) const
1605{
1606    return (int8_t *)buffers + (offset - userBase) * frameSize;
1607}
1608
1609uint32_t audio_track_cblk_t::framesAvailable()
1610{
1611    Mutex::Autolock _l(lock);
1612    return framesAvailable_l();
1613}
1614
1615uint32_t audio_track_cblk_t::framesAvailable_l()
1616{
1617    uint32_t u = user;
1618    uint32_t s = server;
1619
1620    if (flags & CBLK_DIRECTION_MSK) {
1621        uint32_t limit = (s < loopStart) ? s : loopStart;
1622        return limit + frameCount - u;
1623    } else {
1624        return frameCount + u - s;
1625    }
1626}
1627
1628uint32_t audio_track_cblk_t::framesReady()
1629{
1630    uint32_t u = user;
1631    uint32_t s = server;
1632
1633    if (flags & CBLK_DIRECTION_MSK) {
1634        if (u < loopEnd) {
1635            return u - s;
1636        } else {
1637            // do not block on mutex shared with client on AudioFlinger side
1638            if (!tryLock()) {
1639                ALOGW("framesReady() could not lock cblk");
1640                return 0;
1641            }
1642            uint32_t frames = UINT_MAX;
1643            if (loopCount >= 0) {
1644                frames = (loopEnd - loopStart)*loopCount + u - s;
1645            }
1646            lock.unlock();
1647            return frames;
1648        }
1649    } else {
1650        return s - u;
1651    }
1652}
1653
1654bool audio_track_cblk_t::tryLock()
1655{
1656    // the code below simulates lock-with-timeout
1657    // we MUST do this to protect the AudioFlinger server
1658    // as this lock is shared with the client.
1659    status_t err;
1660
1661    err = lock.tryLock();
1662    if (err == -EBUSY) { // just wait a bit
1663        usleep(1000);
1664        err = lock.tryLock();
1665    }
1666    if (err != NO_ERROR) {
1667        // probably, the client just died.
1668        return false;
1669    }
1670    return true;
1671}
1672
1673// -------------------------------------------------------------------------
1674
1675}; // namespace android
1676