AudioTrack.cpp revision a42ff007a17d63df22c60dd5e5fd811ee45ca1b3 
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        size_t* frameCount,
54        audio_stream_type_t streamType,
55        uint32_t sampleRate)
56{
57    if (frameCount == NULL) {
58        return BAD_VALUE;
59    }
60
61    // default to 0 in case of error
62    *frameCount = 0;
63
64    // FIXME merge with similar code in createTrack_l(), except we're missing
65    //       some information here that is available in createTrack_l():
66    //          audio_io_handle_t output
67    //          audio_format_t format
68    //          audio_channel_mask_t channelMask
69    //          audio_output_flags_t flags
70    uint32_t afSampleRate;
71    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
72        return NO_INIT;
73    }
74    size_t afFrameCount;
75    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
76        return NO_INIT;
77    }
78    uint32_t afLatency;
79    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
80        return NO_INIT;
81    }
82
83    // Ensure that buffer depth covers at least audio hardware latency
84    uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
85    if (minBufCount < 2) minBufCount = 2;
86
87    *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
88            afFrameCount * minBufCount * sampleRate / afSampleRate;
89    ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
90            *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
91    return NO_ERROR;
92}
93
94// ---------------------------------------------------------------------------
95
96AudioTrack::AudioTrack()
97    : mStatus(NO_INIT),
98      mIsTimed(false),
99      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
100      mPreviousSchedulingGroup(SP_DEFAULT)
101{
102}
103
104AudioTrack::AudioTrack(
105        audio_stream_type_t streamType,
106        uint32_t sampleRate,
107        audio_format_t format,
108        audio_channel_mask_t channelMask,
109        int frameCount,
110        audio_output_flags_t flags,
111        callback_t cbf,
112        void* user,
113        int notificationFrames,
114        int sessionId)
115    : mStatus(NO_INIT),
116      mIsTimed(false),
117      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
118      mPreviousSchedulingGroup(SP_DEFAULT)
119{
120    mStatus = set(streamType, sampleRate, format, channelMask,
121            frameCount, flags, cbf, user, notificationFrames,
122            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
123}
124
125AudioTrack::AudioTrack(
126        audio_stream_type_t streamType,
127        uint32_t sampleRate,
128        audio_format_t format,
129        audio_channel_mask_t channelMask,
130        const sp<IMemory>& sharedBuffer,
131        audio_output_flags_t flags,
132        callback_t cbf,
133        void* user,
134        int notificationFrames,
135        int sessionId)
136    : mStatus(NO_INIT),
137      mIsTimed(false),
138      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
139      mPreviousSchedulingGroup(SP_DEFAULT)
140{
141    mStatus = set(streamType, sampleRate, format, channelMask,
142            0 /*frameCount*/, flags, cbf, user, notificationFrames,
143            sharedBuffer, false /*threadCanCallJava*/, sessionId);
144}
145
146AudioTrack::~AudioTrack()
147{
148    ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
149
150    if (mStatus == NO_ERROR) {
151        // Make sure that callback function exits in the case where
152        // it is looping on buffer full condition in obtainBuffer().
153        // Otherwise the callback thread will never exit.
154        stop();
155        if (mAudioTrackThread != 0) {
156            mAudioTrackThread->requestExit();   // see comment in AudioTrack.h
157            mAudioTrackThread->requestExitAndWait();
158            mAudioTrackThread.clear();
159        }
160        mAudioTrack.clear();
161        IPCThreadState::self()->flushCommands();
162        AudioSystem::releaseAudioSessionId(mSessionId);
163    }
164}
165
166status_t AudioTrack::set(
167        audio_stream_type_t streamType,
168        uint32_t sampleRate,
169        audio_format_t format,
170        audio_channel_mask_t channelMask,
171        int frameCountInt,
172        audio_output_flags_t flags,
173        callback_t cbf,
174        void* user,
175        int notificationFrames,
176        const sp<IMemory>& sharedBuffer,
177        bool threadCanCallJava,
178        int sessionId)
179{
180    // FIXME "int" here is legacy and will be replaced by size_t later
181    if (frameCountInt < 0) {
182        ALOGE("Invalid frame count %d", frameCountInt);
183        return BAD_VALUE;
184    }
185    size_t frameCount = frameCountInt;
186
187    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
188            sharedBuffer->size());
189
190    ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
191
192    AutoMutex lock(mLock);
193    if (mAudioTrack != 0) {
194        ALOGE("Track already in use");
195        return INVALID_OPERATION;
196    }
197
198    // handle default values first.
199    if (streamType == AUDIO_STREAM_DEFAULT) {
200        streamType = AUDIO_STREAM_MUSIC;
201    }
202
203    if (sampleRate == 0) {
204        uint32_t afSampleRate;
205        if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
206            return NO_INIT;
207        }
208        sampleRate = afSampleRate;
209    }
210
211    // these below should probably come from the audioFlinger too...
212    if (format == AUDIO_FORMAT_DEFAULT) {
213        format = AUDIO_FORMAT_PCM_16_BIT;
214    }
215    if (channelMask == 0) {
216        channelMask = AUDIO_CHANNEL_OUT_STEREO;
217    }
218
219    // validate parameters
220    if (!audio_is_valid_format(format)) {
221        ALOGE("Invalid format");
222        return BAD_VALUE;
223    }
224
225    // AudioFlinger does not currently support 8-bit data in shared memory
226    if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
227        ALOGE("8-bit data in shared memory is not supported");
228        return BAD_VALUE;
229    }
230
231    // force direct flag if format is not linear PCM
232    if (!audio_is_linear_pcm(format)) {
233        flags = (audio_output_flags_t)
234                // FIXME why can't we allow direct AND fast?
235                ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
236    }
237    // only allow deep buffering for music stream type
238    if (streamType != AUDIO_STREAM_MUSIC) {
239        flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
240    }
241
242    if (!audio_is_output_channel(channelMask)) {
243        ALOGE("Invalid channel mask %#x", channelMask);
244        return BAD_VALUE;
245    }
246    mChannelMask = channelMask;
247    uint32_t channelCount = popcount(channelMask);
248    mChannelCount = channelCount;
249
250    audio_io_handle_t output = AudioSystem::getOutput(
251                                    streamType,
252                                    sampleRate, format, channelMask,
253                                    flags);
254
255    if (output == 0) {
256        ALOGE("Could not get audio output for stream type %d", streamType);
257        return BAD_VALUE;
258    }
259
260    mVolume[LEFT] = 1.0f;
261    mVolume[RIGHT] = 1.0f;
262    mSendLevel = 0.0f;
263    mFrameCount = frameCount;
264    mReqFrameCount = frameCount;
265    mNotificationFramesReq = notificationFrames;
266    mSessionId = sessionId;
267    mAuxEffectId = 0;
268    mFlags = flags;
269    mCbf = cbf;
270
271    if (cbf != NULL) {
272        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
273        mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
274    }
275
276    // create the IAudioTrack
277    status_t status = createTrack_l(streamType,
278                                  sampleRate,
279                                  format,
280                                  frameCount,
281                                  flags,
282                                  sharedBuffer,
283                                  output);
284
285    if (status != NO_ERROR) {
286        if (mAudioTrackThread != 0) {
287            mAudioTrackThread->requestExit();
288            mAudioTrackThread.clear();
289        }
290        return status;
291    }
292
293    mStatus = NO_ERROR;
294
295    mStreamType = streamType;
296    mFormat = format;
297
298    if (audio_is_linear_pcm(format)) {
299        mFrameSize = channelCount * audio_bytes_per_sample(format);
300        mFrameSizeAF = channelCount * sizeof(int16_t);
301    } else {
302        mFrameSize = sizeof(uint8_t);
303        mFrameSizeAF = sizeof(uint8_t);
304    }
305
306    mSharedBuffer = sharedBuffer;
307    mMuted = false;
308    mActive = false;
309    mUserData = user;
310    mLoopCount = 0;
311    mMarkerPosition = 0;
312    mMarkerReached = false;
313    mNewPosition = 0;
314    mUpdatePeriod = 0;
315    mFlushed = false;
316    AudioSystem::acquireAudioSessionId(mSessionId);
317    return NO_ERROR;
318}
319
320status_t AudioTrack::initCheck() const
321{
322    return mStatus;
323}
324
325// -------------------------------------------------------------------------
326
327uint32_t AudioTrack::latency() const
328{
329    return mLatency;
330}
331
332audio_stream_type_t AudioTrack::streamType() const
333{
334    return mStreamType;
335}
336
337audio_format_t AudioTrack::format() const
338{
339    return mFormat;
340}
341
342uint32_t AudioTrack::channelCount() const
343{
344    return mChannelCount;
345}
346
347size_t AudioTrack::frameCount() const
348{
349    return mFrameCount;
350}
351
352sp<IMemory>& AudioTrack::sharedBuffer()
353{
354    return mSharedBuffer;
355}
356
357// -------------------------------------------------------------------------
358
359void AudioTrack::start()
360{
361    sp<AudioTrackThread> t = mAudioTrackThread;
362
363    ALOGV("start %p", this);
364
365    AutoMutex lock(mLock);
366    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
367    // while we are accessing the cblk
368    sp<IAudioTrack> audioTrack = mAudioTrack;
369    sp<IMemory> iMem = mCblkMemory;
370    audio_track_cblk_t* cblk = mCblk;
371
372    if (!mActive) {
373        mFlushed = false;
374        mActive = true;
375        mNewPosition = cblk->server + mUpdatePeriod;
376        cblk->lock.lock();
377        cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
378        cblk->waitTimeMs = 0;
379        android_atomic_and(~CBLK_DISABLED, &cblk->flags);
380        if (t != 0) {
381            t->resume();
382        } else {
383            mPreviousPriority = getpriority(PRIO_PROCESS, 0);
384            get_sched_policy(0, &mPreviousSchedulingGroup);
385            androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
386        }
387
388        ALOGV("start %p before lock cblk %p", this, cblk);
389        status_t status = NO_ERROR;
390        if (!(cblk->flags & CBLK_INVALID)) {
391            cblk->lock.unlock();
392            ALOGV("mAudioTrack->start()");
393            status = mAudioTrack->start();
394            cblk->lock.lock();
395            if (status == DEAD_OBJECT) {
396                android_atomic_or(CBLK_INVALID, &cblk->flags);
397            }
398        }
399        if (cblk->flags & CBLK_INVALID) {
400            audio_track_cblk_t* temp = cblk;
401            status = restoreTrack_l(temp, true /*fromStart*/);
402            cblk = temp;
403        }
404        cblk->lock.unlock();
405        if (status != NO_ERROR) {
406            ALOGV("start() failed");
407            mActive = false;
408            if (t != 0) {
409                t->pause();
410            } else {
411                setpriority(PRIO_PROCESS, 0, mPreviousPriority);
412                set_sched_policy(0, mPreviousSchedulingGroup);
413            }
414        }
415    }
416
417}
418
419void AudioTrack::stop()
420{
421    sp<AudioTrackThread> t = mAudioTrackThread;
422
423    ALOGV("stop %p", this);
424
425    AutoMutex lock(mLock);
426    if (mActive) {
427        mActive = false;
428        mCblk->cv.signal();
429        mAudioTrack->stop();
430        // Cancel loops (If we are in the middle of a loop, playback
431        // would not stop until loopCount reaches 0).
432        setLoop_l(0, 0, 0);
433        // the playback head position will reset to 0, so if a marker is set, we need
434        // to activate it again
435        mMarkerReached = false;
436        // Force flush if a shared buffer is used otherwise audioflinger
437        // will not stop before end of buffer is reached.
438        if (mSharedBuffer != 0) {
439            flush_l();
440        }
441        if (t != 0) {
442            t->pause();
443        } else {
444            setpriority(PRIO_PROCESS, 0, mPreviousPriority);
445            set_sched_policy(0, mPreviousSchedulingGroup);
446        }
447    }
448
449}
450
451bool AudioTrack::stopped() const
452{
453    AutoMutex lock(mLock);
454    return stopped_l();
455}
456
457void AudioTrack::flush()
458{
459    AutoMutex lock(mLock);
460    flush_l();
461}
462
463// must be called with mLock held
464void AudioTrack::flush_l()
465{
466    ALOGV("flush");
467
468    // clear playback marker and periodic update counter
469    mMarkerPosition = 0;
470    mMarkerReached = false;
471    mUpdatePeriod = 0;
472
473    if (!mActive) {
474        mFlushed = true;
475        mAudioTrack->flush();
476        // Release AudioTrack callback thread in case it was waiting for new buffers
477        // in AudioTrack::obtainBuffer()
478        mCblk->cv.signal();
479    }
480}
481
482void AudioTrack::pause()
483{
484    ALOGV("pause");
485    AutoMutex lock(mLock);
486    if (mActive) {
487        mActive = false;
488        mCblk->cv.signal();
489        mAudioTrack->pause();
490    }
491}
492
493void AudioTrack::mute(bool e)
494{
495    mAudioTrack->mute(e);
496    mMuted = e;
497}
498
499bool AudioTrack::muted() const
500{
501    return mMuted;
502}
503
504status_t AudioTrack::setVolume(float left, float right)
505{
506    if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
507        return BAD_VALUE;
508    }
509
510    AutoMutex lock(mLock);
511    mVolume[LEFT] = left;
512    mVolume[RIGHT] = right;
513
514    mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
515
516    return NO_ERROR;
517}
518
519status_t AudioTrack::setVolume(float volume)
520{
521    return setVolume(volume, volume);
522}
523
524status_t AudioTrack::setAuxEffectSendLevel(float level)
525{
526    ALOGV("setAuxEffectSendLevel(%f)", level);
527    if (level < 0.0f || level > 1.0f) {
528        return BAD_VALUE;
529    }
530    AutoMutex lock(mLock);
531
532    mSendLevel = level;
533
534    mCblk->setSendLevel(level);
535
536    return NO_ERROR;
537}
538
539void AudioTrack::getAuxEffectSendLevel(float* level) const
540{
541    if (level != NULL) {
542        *level  = mSendLevel;
543    }
544}
545
546status_t AudioTrack::setSampleRate(uint32_t rate)
547{
548    uint32_t afSamplingRate;
549
550    if (mIsTimed) {
551        return INVALID_OPERATION;
552    }
553
554    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
555        return NO_INIT;
556    }
557    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
558    if (rate == 0 || rate > afSamplingRate*2 ) {
559        return BAD_VALUE;
560    }
561
562    AutoMutex lock(mLock);
563    mCblk->sampleRate = rate;
564    return NO_ERROR;
565}
566
567uint32_t AudioTrack::getSampleRate() const
568{
569    if (mIsTimed) {
570        return 0;
571    }
572
573    AutoMutex lock(mLock);
574    return mCblk->sampleRate;
575}
576
577status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
578{
579    AutoMutex lock(mLock);
580    return setLoop_l(loopStart, loopEnd, loopCount);
581}
582
583// must be called with mLock held
584status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
585{
586    audio_track_cblk_t* cblk = mCblk;
587
588    Mutex::Autolock _l(cblk->lock);
589
590    if (loopCount == 0) {
591        cblk->loopStart = UINT_MAX;
592        cblk->loopEnd = UINT_MAX;
593        cblk->loopCount = 0;
594        mLoopCount = 0;
595        return NO_ERROR;
596    }
597
598    if (mIsTimed) {
599        return INVALID_OPERATION;
600    }
601
602    if (loopStart >= loopEnd ||
603        loopEnd - loopStart > mFrameCount ||
604        cblk->server > loopStart) {
605        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, "
606              "user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
607        return BAD_VALUE;
608    }
609
610    if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) {
611        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, "
612            "framecount %d",
613            loopStart, loopEnd, mFrameCount);
614        return BAD_VALUE;
615    }
616
617    cblk->loopStart = loopStart;
618    cblk->loopEnd = loopEnd;
619    cblk->loopCount = loopCount;
620    mLoopCount = loopCount;
621
622    return NO_ERROR;
623}
624
625status_t AudioTrack::setMarkerPosition(uint32_t marker)
626{
627    if (mCbf == NULL) {
628        return INVALID_OPERATION;
629    }
630
631    mMarkerPosition = marker;
632    mMarkerReached = false;
633
634    return NO_ERROR;
635}
636
637status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
638{
639    if (marker == NULL) {
640        return BAD_VALUE;
641    }
642
643    *marker = mMarkerPosition;
644
645    return NO_ERROR;
646}
647
648status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
649{
650    if (mCbf == NULL) {
651        return INVALID_OPERATION;
652    }
653
654    uint32_t curPosition;
655    getPosition(&curPosition);
656    mNewPosition = curPosition + updatePeriod;
657    mUpdatePeriod = updatePeriod;
658
659    return NO_ERROR;
660}
661
662status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
663{
664    if (updatePeriod == NULL) {
665        return BAD_VALUE;
666    }
667
668    *updatePeriod = mUpdatePeriod;
669
670    return NO_ERROR;
671}
672
673status_t AudioTrack::setPosition(uint32_t position)
674{
675    if (mIsTimed) {
676        return INVALID_OPERATION;
677    }
678
679    AutoMutex lock(mLock);
680
681    if (!stopped_l()) {
682        return INVALID_OPERATION;
683    }
684
685    audio_track_cblk_t* cblk = mCblk;
686    Mutex::Autolock _l(cblk->lock);
687
688    if (position > cblk->user) {
689        return BAD_VALUE;
690    }
691
692    cblk->server = position;
693    android_atomic_or(CBLK_FORCEREADY, &cblk->flags);
694
695    return NO_ERROR;
696}
697
698status_t AudioTrack::getPosition(uint32_t *position)
699{
700    if (position == NULL) {
701        return BAD_VALUE;
702    }
703    AutoMutex lock(mLock);
704    *position = mFlushed ? 0 : mCblk->server;
705
706    return NO_ERROR;
707}
708
709status_t AudioTrack::reload()
710{
711    AutoMutex lock(mLock);
712
713    if (!stopped_l()) {
714        return INVALID_OPERATION;
715    }
716
717    flush_l();
718
719    audio_track_cblk_t* cblk = mCblk;
720    cblk->stepUserOut(mFrameCount, mFrameCount);
721
722    return NO_ERROR;
723}
724
725audio_io_handle_t AudioTrack::getOutput()
726{
727    AutoMutex lock(mLock);
728    return getOutput_l();
729}
730
731// must be called with mLock held
732audio_io_handle_t AudioTrack::getOutput_l()
733{
734    return AudioSystem::getOutput(mStreamType,
735            mCblk->sampleRate, mFormat, mChannelMask, mFlags);
736}
737
738int AudioTrack::getSessionId() const
739{
740    return mSessionId;
741}
742
743status_t AudioTrack::attachAuxEffect(int effectId)
744{
745    ALOGV("attachAuxEffect(%d)", effectId);
746    status_t status = mAudioTrack->attachAuxEffect(effectId);
747    if (status == NO_ERROR) {
748        mAuxEffectId = effectId;
749    }
750    return status;
751}
752
753// -------------------------------------------------------------------------
754
755// must be called with mLock held
756status_t AudioTrack::createTrack_l(
757        audio_stream_type_t streamType,
758        uint32_t sampleRate,
759        audio_format_t format,
760        size_t frameCount,
761        audio_output_flags_t flags,
762        const sp<IMemory>& sharedBuffer,
763        audio_io_handle_t output)
764{
765    status_t status;
766    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
767    if (audioFlinger == 0) {
768        ALOGE("Could not get audioflinger");
769        return NO_INIT;
770    }
771
772    uint32_t afLatency;
773    if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
774        return NO_INIT;
775    }
776
777    // Client decides whether the track is TIMED (see below), but can only express a preference
778    // for FAST.  Server will perform additional tests.
779    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
780            // either of these use cases:
781            // use case 1: shared buffer
782            (sharedBuffer != 0) ||
783            // use case 2: callback handler
784            (mCbf != NULL))) {
785        ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
786        // once denied, do not request again if IAudioTrack is re-created
787        flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
788        mFlags = flags;
789    }
790    ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
791
792    mNotificationFramesAct = mNotificationFramesReq;
793
794    if (!audio_is_linear_pcm(format)) {
795
796        if (sharedBuffer != 0) {
797            // Same comment as below about ignoring frameCount parameter for set()
798            frameCount = sharedBuffer->size();
799        } else if (frameCount == 0) {
800            size_t afFrameCount;
801            if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
802                return NO_INIT;
803            }
804            frameCount = afFrameCount;
805        }
806
807    } else if (sharedBuffer != 0) {
808
809        // Ensure that buffer alignment matches channel count
810        // 8-bit data in shared memory is not currently supported by AudioFlinger
811        size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
812        if (mChannelCount > 1) {
813            // More than 2 channels does not require stronger alignment than stereo
814            alignment <<= 1;
815        }
816        if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
817            ALOGE("Invalid buffer alignment: address %p, channel count %u",
818                    sharedBuffer->pointer(), mChannelCount);
819            return BAD_VALUE;
820        }
821
822        // When initializing a shared buffer AudioTrack via constructors,
823        // there's no frameCount parameter.
824        // But when initializing a shared buffer AudioTrack via set(),
825        // there _is_ a frameCount parameter.  We silently ignore it.
826        frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
827
828    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
829
830        // FIXME move these calculations and associated checks to server
831        uint32_t afSampleRate;
832        if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
833            return NO_INIT;
834        }
835        size_t afFrameCount;
836        if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
837            return NO_INIT;
838        }
839
840        // Ensure that buffer depth covers at least audio hardware latency
841        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
842        if (minBufCount < 2) minBufCount = 2;
843
844        size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
845        ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
846                ", afLatency=%d",
847                minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
848
849        if (frameCount == 0) {
850            frameCount = minFrameCount;
851        }
852        if (mNotificationFramesAct == 0) {
853            mNotificationFramesAct = frameCount/2;
854        }
855        // Make sure that application is notified with sufficient margin
856        // before underrun
857        if (mNotificationFramesAct > frameCount/2) {
858            mNotificationFramesAct = frameCount/2;
859        }
860        if (frameCount < minFrameCount) {
861            // not ALOGW because it happens all the time when playing key clicks over A2DP
862            ALOGV("Minimum buffer size corrected from %d to %d",
863                     frameCount, minFrameCount);
864            frameCount = minFrameCount;
865        }
866
867    } else {
868        // For fast tracks, the frame count calculations and checks are done by server
869    }
870
871    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
872    if (mIsTimed) {
873        trackFlags |= IAudioFlinger::TRACK_TIMED;
874    }
875
876    pid_t tid = -1;
877    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
878        trackFlags |= IAudioFlinger::TRACK_FAST;
879        if (mAudioTrackThread != 0) {
880            tid = mAudioTrackThread->getTid();
881        }
882    }
883
884    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
885                                                      streamType,
886                                                      sampleRate,
887                                                      // AudioFlinger only sees 16-bit PCM
888                                                      format == AUDIO_FORMAT_PCM_8_BIT ?
889                                                              AUDIO_FORMAT_PCM_16_BIT : format,
890                                                      mChannelMask,
891                                                      frameCount,
892                                                      &trackFlags,
893                                                      sharedBuffer,
894                                                      output,
895                                                      tid,
896                                                      &mSessionId,
897                                                      &status);
898
899    if (track == 0) {
900        ALOGE("AudioFlinger could not create track, status: %d", status);
901        return status;
902    }
903    sp<IMemory> iMem = track->getCblk();
904    if (iMem == 0) {
905        ALOGE("Could not get control block");
906        return NO_INIT;
907    }
908    mAudioTrack = track;
909    mCblkMemory = iMem;
910    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
911    mCblk = cblk;
912    size_t temp = cblk->frameCount_;
913    if (temp < frameCount || (frameCount == 0 && temp == 0)) {
914        // In current design, AudioTrack client checks and ensures frame count validity before
915        // passing it to AudioFlinger so AudioFlinger should not return a different value except
916        // for fast track as it uses a special method of assigning frame count.
917        ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
918    }
919    frameCount = temp;
920    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
921        if (trackFlags & IAudioFlinger::TRACK_FAST) {
922            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
923        } else {
924            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
925            // once denied, do not request again if IAudioTrack is re-created
926            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
927            mFlags = flags;
928        }
929        if (sharedBuffer == 0) {
930            mNotificationFramesAct = frameCount/2;
931        }
932    }
933    if (sharedBuffer == 0) {
934        mBuffers = (char*)cblk + sizeof(audio_track_cblk_t);
935    } else {
936        mBuffers = sharedBuffer->pointer();
937        // Force buffer full condition as data is already present in shared memory
938        cblk->stepUserOut(frameCount, frameCount);
939    }
940
941    cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
942            uint16_t(mVolume[LEFT] * 0x1000));
943    cblk->setSendLevel(mSendLevel);
944    mAudioTrack->attachAuxEffect(mAuxEffectId);
945    cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
946    cblk->waitTimeMs = 0;
947    mRemainingFrames = mNotificationFramesAct;
948    // FIXME don't believe this lie
949    mLatency = afLatency + (1000*frameCount) / sampleRate;
950    mFrameCount = frameCount;
951    // If IAudioTrack is re-created, don't let the requested frameCount
952    // decrease.  This can confuse clients that cache frameCount().
953    if (frameCount > mReqFrameCount) {
954        mReqFrameCount = frameCount;
955    }
956    return NO_ERROR;
957}
958
959status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
960{
961    AutoMutex lock(mLock);
962    bool active;
963    status_t result = NO_ERROR;
964    audio_track_cblk_t* cblk = mCblk;
965    uint32_t framesReq = audioBuffer->frameCount;
966    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
967
968    audioBuffer->frameCount  = 0;
969    audioBuffer->size = 0;
970
971    uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount);
972
973    cblk->lock.lock();
974    if (cblk->flags & CBLK_INVALID) {
975        goto create_new_track;
976    }
977    cblk->lock.unlock();
978
979    if (framesAvail == 0) {
980        cblk->lock.lock();
981        goto start_loop_here;
982        while (framesAvail == 0) {
983            active = mActive;
984            if (CC_UNLIKELY(!active)) {
985                ALOGV("Not active and NO_MORE_BUFFERS");
986                cblk->lock.unlock();
987                return NO_MORE_BUFFERS;
988            }
989            if (CC_UNLIKELY(!waitCount)) {
990                cblk->lock.unlock();
991                return WOULD_BLOCK;
992            }
993            if (!(cblk->flags & CBLK_INVALID)) {
994                mLock.unlock();
995                // this condition is in shared memory, so if IAudioTrack and control block
996                // are replaced due to mediaserver death or IAudioTrack invalidation then
997                // cv won't be signalled, but fortunately the timeout will limit the wait
998                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
999                cblk->lock.unlock();
1000                mLock.lock();
1001                if (!mActive) {
1002                    return status_t(STOPPED);
1003                }
1004                // IAudioTrack may have been re-created while mLock was unlocked
1005                cblk = mCblk;
1006                cblk->lock.lock();
1007            }
1008
1009            if (cblk->flags & CBLK_INVALID) {
1010                goto create_new_track;
1011            }
1012            if (CC_UNLIKELY(result != NO_ERROR)) {
1013                cblk->waitTimeMs += waitTimeMs;
1014                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
1015                    // timing out when a loop has been set and we have already written upto loop end
1016                    // is a normal condition: no need to wake AudioFlinger up.
1017                    if (cblk->user < cblk->loopEnd) {
1018                        ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, "
1019                              "server=%08x", this, cblk->mName, cblk->user, cblk->server);
1020                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
1021                        cblk->lock.unlock();
1022                        result = mAudioTrack->start();
1023                        cblk->lock.lock();
1024                        if (result == DEAD_OBJECT) {
1025                            android_atomic_or(CBLK_INVALID, &cblk->flags);
1026create_new_track:
1027                            audio_track_cblk_t* temp = cblk;
1028                            result = restoreTrack_l(temp, false /*fromStart*/);
1029                            cblk = temp;
1030                        }
1031                        if (result != NO_ERROR) {
1032                            ALOGW("obtainBuffer create Track error %d", result);
1033                            cblk->lock.unlock();
1034                            return result;
1035                        }
1036                    }
1037                    cblk->waitTimeMs = 0;
1038                }
1039
1040                if (--waitCount == 0) {
1041                    cblk->lock.unlock();
1042                    return TIMED_OUT;
1043                }
1044            }
1045            // read the server count again
1046        start_loop_here:
1047            framesAvail = cblk->framesAvailableOut_l(mFrameCount);
1048        }
1049        cblk->lock.unlock();
1050    }
1051
1052    cblk->waitTimeMs = 0;
1053
1054    if (framesReq > framesAvail) {
1055        framesReq = framesAvail;
1056    }
1057
1058    uint32_t u = cblk->user;
1059    uint32_t bufferEnd = cblk->userBase + mFrameCount;
1060
1061    if (framesReq > bufferEnd - u) {
1062        framesReq = bufferEnd - u;
1063    }
1064
1065    audioBuffer->frameCount = framesReq;
1066    audioBuffer->size = framesReq * mFrameSizeAF;
1067    audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u);
1068    active = mActive;
1069    return active ? status_t(NO_ERROR) : status_t(STOPPED);
1070}
1071
1072void AudioTrack::releaseBuffer(Buffer* audioBuffer)
1073{
1074    AutoMutex lock(mLock);
1075    audio_track_cblk_t* cblk = mCblk;
1076    cblk->stepUserOut(audioBuffer->frameCount, mFrameCount);
1077    if (audioBuffer->frameCount > 0) {
1078        // restart track if it was disabled by audioflinger due to previous underrun
1079        if (mActive && (cblk->flags & CBLK_DISABLED)) {
1080            android_atomic_and(~CBLK_DISABLED, &cblk->flags);
1081            ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName);
1082            mAudioTrack->start();
1083        }
1084    }
1085}
1086
1087// -------------------------------------------------------------------------
1088
1089ssize_t AudioTrack::write(const void* buffer, size_t userSize)
1090{
1091
1092    if (mSharedBuffer != 0) {
1093        return INVALID_OPERATION;
1094    }
1095    if (mIsTimed) {
1096        return INVALID_OPERATION;
1097    }
1098
1099    if (ssize_t(userSize) < 0) {
1100        // Sanity-check: user is most-likely passing an error code, and it would
1101        // make the return value ambiguous (actualSize vs error).
1102        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
1103                buffer, userSize, userSize);
1104        return BAD_VALUE;
1105    }
1106
1107    ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
1108
1109    if (userSize == 0) {
1110        return 0;
1111    }
1112
1113    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1114    // while we are accessing the cblk
1115    mLock.lock();
1116    sp<IAudioTrack> audioTrack = mAudioTrack;
1117    sp<IMemory> iMem = mCblkMemory;
1118    mLock.unlock();
1119
1120    // since mLock is unlocked the IAudioTrack and shared memory may be re-created,
1121    // so all cblk references might still refer to old shared memory, but that should be benign
1122
1123    ssize_t written = 0;
1124    const int8_t *src = (const int8_t *)buffer;
1125    Buffer audioBuffer;
1126    size_t frameSz = frameSize();
1127
1128    do {
1129        audioBuffer.frameCount = userSize/frameSz;
1130
1131        status_t err = obtainBuffer(&audioBuffer, -1);
1132        if (err < 0) {
1133            // out of buffers, return #bytes written
1134            if (err == status_t(NO_MORE_BUFFERS)) {
1135                break;
1136            }
1137            return ssize_t(err);
1138        }
1139
1140        size_t toWrite;
1141
1142        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1143            // Divide capacity by 2 to take expansion into account
1144            toWrite = audioBuffer.size>>1;
1145            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
1146        } else {
1147            toWrite = audioBuffer.size;
1148            memcpy(audioBuffer.i8, src, toWrite);
1149        }
1150        src += toWrite;
1151        userSize -= toWrite;
1152        written += toWrite;
1153
1154        releaseBuffer(&audioBuffer);
1155    } while (userSize >= frameSz);
1156
1157    return written;
1158}
1159
1160// -------------------------------------------------------------------------
1161
1162TimedAudioTrack::TimedAudioTrack() {
1163    mIsTimed = true;
1164}
1165
1166status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
1167{
1168    AutoMutex lock(mLock);
1169    status_t result = UNKNOWN_ERROR;
1170
1171    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1172    // while we are accessing the cblk
1173    sp<IAudioTrack> audioTrack = mAudioTrack;
1174    sp<IMemory> iMem = mCblkMemory;
1175
1176    // If the track is not invalid already, try to allocate a buffer.  alloc
1177    // fails indicating that the server is dead, flag the track as invalid so
1178    // we can attempt to restore in just a bit.
1179    audio_track_cblk_t* cblk = mCblk;
1180    if (!(cblk->flags & CBLK_INVALID)) {
1181        result = mAudioTrack->allocateTimedBuffer(size, buffer);
1182        if (result == DEAD_OBJECT) {
1183            android_atomic_or(CBLK_INVALID, &cblk->flags);
1184        }
1185    }
1186
1187    // If the track is invalid at this point, attempt to restore it. and try the
1188    // allocation one more time.
1189    if (cblk->flags & CBLK_INVALID) {
1190        cblk->lock.lock();
1191        audio_track_cblk_t* temp = cblk;
1192        result = restoreTrack_l(temp, false /*fromStart*/);
1193        cblk = temp;
1194        cblk->lock.unlock();
1195
1196        if (result == OK) {
1197            result = mAudioTrack->allocateTimedBuffer(size, buffer);
1198        }
1199    }
1200
1201    return result;
1202}
1203
1204status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
1205                                           int64_t pts)
1206{
1207    status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
1208    {
1209        AutoMutex lock(mLock);
1210        audio_track_cblk_t* cblk = mCblk;
1211        // restart track if it was disabled by audioflinger due to previous underrun
1212        if (buffer->size() != 0 && status == NO_ERROR &&
1213                mActive && (cblk->flags & CBLK_DISABLED)) {
1214            android_atomic_and(~CBLK_DISABLED, &cblk->flags);
1215            ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
1216            mAudioTrack->start();
1217        }
1218    }
1219    return status;
1220}
1221
1222status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
1223                                                TargetTimeline target)
1224{
1225    return mAudioTrack->setMediaTimeTransform(xform, target);
1226}
1227
1228// -------------------------------------------------------------------------
1229
1230bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
1231{
1232    Buffer audioBuffer;
1233    uint32_t frames;
1234    size_t writtenSize;
1235
1236    mLock.lock();
1237    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1238    // while we are accessing the cblk
1239    sp<IAudioTrack> audioTrack = mAudioTrack;
1240    sp<IMemory> iMem = mCblkMemory;
1241    audio_track_cblk_t* cblk = mCblk;
1242    bool active = mActive;
1243    mLock.unlock();
1244
1245    // since mLock is unlocked the IAudioTrack and shared memory may be re-created,
1246    // so all cblk references might still refer to old shared memory, but that should be benign
1247
1248    // Manage underrun callback
1249    if (active && (cblk->framesAvailableOut(mFrameCount) == mFrameCount)) {
1250        ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
1251        if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) {
1252            mCbf(EVENT_UNDERRUN, mUserData, 0);
1253            if (cblk->server == mFrameCount) {
1254                mCbf(EVENT_BUFFER_END, mUserData, 0);
1255            }
1256            if (mSharedBuffer != 0) {
1257                return false;
1258            }
1259        }
1260    }
1261
1262    // Manage loop end callback
1263    while (mLoopCount > cblk->loopCount) {
1264        int loopCount = -1;
1265        mLoopCount--;
1266        if (mLoopCount >= 0) loopCount = mLoopCount;
1267
1268        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
1269    }
1270
1271    // Manage marker callback
1272    if (!mMarkerReached && (mMarkerPosition > 0)) {
1273        if (cblk->server >= mMarkerPosition) {
1274            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
1275            mMarkerReached = true;
1276        }
1277    }
1278
1279    // Manage new position callback
1280    if (mUpdatePeriod > 0) {
1281        while (cblk->server >= mNewPosition) {
1282            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
1283            mNewPosition += mUpdatePeriod;
1284        }
1285    }
1286
1287    // If Shared buffer is used, no data is requested from client.
1288    if (mSharedBuffer != 0) {
1289        frames = 0;
1290    } else {
1291        frames = mRemainingFrames;
1292    }
1293
1294    // See description of waitCount parameter at declaration of obtainBuffer().
1295    // The logic below prevents us from being stuck below at obtainBuffer()
1296    // not being able to handle timed events (position, markers, loops).
1297    int32_t waitCount = -1;
1298    if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
1299        waitCount = 1;
1300    }
1301
1302    do {
1303
1304        audioBuffer.frameCount = frames;
1305
1306        status_t err = obtainBuffer(&audioBuffer, waitCount);
1307        if (err < NO_ERROR) {
1308            if (err != TIMED_OUT) {
1309                ALOGE_IF(err != status_t(NO_MORE_BUFFERS),
1310                        "Error obtaining an audio buffer, giving up.");
1311                return false;
1312            }
1313            break;
1314        }
1315        if (err == status_t(STOPPED)) {
1316            return false;
1317        }
1318
1319        // Divide buffer size by 2 to take into account the expansion
1320        // due to 8 to 16 bit conversion: the callback must fill only half
1321        // of the destination buffer
1322        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1323            audioBuffer.size >>= 1;
1324        }
1325
1326        size_t reqSize = audioBuffer.size;
1327        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1328        writtenSize = audioBuffer.size;
1329
1330        // Sanity check on returned size
1331        if (ssize_t(writtenSize) <= 0) {
1332            // The callback is done filling buffers
1333            // Keep this thread going to handle timed events and
1334            // still try to get more data in intervals of WAIT_PERIOD_MS
1335            // but don't just loop and block the CPU, so wait
1336            usleep(WAIT_PERIOD_MS*1000);
1337            break;
1338        }
1339
1340        if (writtenSize > reqSize) {
1341            writtenSize = reqSize;
1342        }
1343
1344        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1345            // 8 to 16 bit conversion, note that source and destination are the same address
1346            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
1347            writtenSize <<= 1;
1348        }
1349
1350        audioBuffer.size = writtenSize;
1351        // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for
1352        // 8 bit PCM data: in this case,  cblk->frameSize is based on a sample size of
1353        // 16 bit.
1354        audioBuffer.frameCount = writtenSize / mFrameSizeAF;
1355
1356        frames -= audioBuffer.frameCount;
1357
1358        releaseBuffer(&audioBuffer);
1359    }
1360    while (frames);
1361
1362    if (frames == 0) {
1363        mRemainingFrames = mNotificationFramesAct;
1364    } else {
1365        mRemainingFrames = frames;
1366    }
1367    return true;
1368}
1369
1370// must be called with mLock and refCblk.lock held. Callers must also hold strong references on
1371// the IAudioTrack and IMemory in case they are recreated here.
1372// If the IAudioTrack is successfully restored, the refCblk pointer is updated
1373// FIXME Don't depend on caller to hold strong references.
1374status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart)
1375{
1376    status_t result;
1377
1378    audio_track_cblk_t* cblk = refCblk;
1379    audio_track_cblk_t* newCblk = cblk;
1380    ALOGW("dead IAudioTrack, creating a new one from %s",
1381        fromStart ? "start()" : "obtainBuffer()");
1382
1383    // signal old cblk condition so that other threads waiting for available buffers stop
1384    // waiting now
1385    cblk->cv.broadcast();
1386    cblk->lock.unlock();
1387
1388    // refresh the audio configuration cache in this process to make sure we get new
1389    // output parameters in getOutput_l() and createTrack_l()
1390    AudioSystem::clearAudioConfigCache();
1391
1392    // if the new IAudioTrack is created, createTrack_l() will modify the
1393    // following member variables: mAudioTrack, mCblkMemory and mCblk.
1394    // It will also delete the strong references on previous IAudioTrack and IMemory
1395    result = createTrack_l(mStreamType,
1396                           cblk->sampleRate,
1397                           mFormat,
1398                           mReqFrameCount,  // so that frame count never goes down
1399                           mFlags,
1400                           mSharedBuffer,
1401                           getOutput_l());
1402
1403    if (result == NO_ERROR) {
1404        uint32_t user = cblk->user;
1405        uint32_t server = cblk->server;
1406        // restore write index and set other indexes to reflect empty buffer status
1407        newCblk = mCblk;
1408        newCblk->user = user;
1409        newCblk->server = user;
1410        newCblk->userBase = user;
1411        newCblk->serverBase = user;
1412        // restore loop: this is not guaranteed to succeed if new frame count is not
1413        // compatible with loop length
1414        setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
1415        if (!fromStart) {
1416            newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1417            // Make sure that a client relying on callback events indicating underrun or
1418            // the actual amount of audio frames played (e.g SoundPool) receives them.
1419            if (mSharedBuffer == 0) {
1420                uint32_t frames = 0;
1421                if (user > server) {
1422                    frames = ((user - server) > mFrameCount) ?
1423                            mFrameCount : (user - server);
1424                    memset(mBuffers, 0, frames * mFrameSizeAF);
1425                }
1426                // restart playback even if buffer is not completely filled.
1427                android_atomic_or(CBLK_FORCEREADY, &newCblk->flags);
1428                // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to
1429                // the client
1430                newCblk->stepUserOut(frames, mFrameCount);
1431            }
1432        }
1433        if (mSharedBuffer != 0) {
1434            newCblk->stepUserOut(mFrameCount, mFrameCount);
1435        }
1436        if (mActive) {
1437            result = mAudioTrack->start();
1438            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
1439        }
1440        if (fromStart && result == NO_ERROR) {
1441            mNewPosition = newCblk->server + mUpdatePeriod;
1442        }
1443    }
1444    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
1445    ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
1446        result, mActive, newCblk, cblk, newCblk->flags, cblk->flags);
1447
1448    if (result == NO_ERROR) {
1449        // from now on we switch to the newly created cblk
1450        refCblk = newCblk;
1451    }
1452    newCblk->lock.lock();
1453
1454    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result);
1455
1456    return result;
1457}
1458
1459status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1460{
1461
1462    const size_t SIZE = 256;
1463    char buffer[SIZE];
1464    String8 result;
1465
1466    audio_track_cblk_t* cblk = mCblk;
1467    result.append(" AudioTrack::dump\n");
1468    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
1469            mVolume[0], mVolume[1]);
1470    result.append(buffer);
1471    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
1472            mChannelCount, mFrameCount);
1473    result.append(buffer);
1474    snprintf(buffer, 255, "  sample rate(%u), status(%d), muted(%d)\n",
1475            (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted);
1476    result.append(buffer);
1477    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1478    result.append(buffer);
1479    ::write(fd, result.string(), result.size());
1480    return NO_ERROR;
1481}
1482
1483// =========================================================================
1484
1485AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1486    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
1487{
1488}
1489
1490AudioTrack::AudioTrackThread::~AudioTrackThread()
1491{
1492}
1493
1494bool AudioTrack::AudioTrackThread::threadLoop()
1495{
1496    {
1497        AutoMutex _l(mMyLock);
1498        if (mPaused) {
1499            mMyCond.wait(mMyLock);
1500            // caller will check for exitPending()
1501            return true;
1502        }
1503    }
1504    if (!mReceiver.processAudioBuffer(this)) {
1505        pause();
1506    }
1507    return true;
1508}
1509
1510void AudioTrack::AudioTrackThread::requestExit()
1511{
1512    // must be in this order to avoid a race condition
1513    Thread::requestExit();
1514    resume();
1515}
1516
1517void AudioTrack::AudioTrackThread::pause()
1518{
1519    AutoMutex _l(mMyLock);
1520    mPaused = true;
1521}
1522
1523void AudioTrack::AudioTrackThread::resume()
1524{
1525    AutoMutex _l(mMyLock);
1526    if (mPaused) {
1527        mPaused = false;
1528        mMyCond.signal();
1529    }
1530}
1531
1532// =========================================================================
1533
1534
1535audio_track_cblk_t::audio_track_cblk_t()
1536    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1537    userBase(0), serverBase(0), frameCount_(0),
1538    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
1539    mSendLevel(0), flags(0)
1540{
1541}
1542
1543uint32_t audio_track_cblk_t::stepUser(size_t stepCount, size_t frameCount, bool isOut)
1544{
1545    ALOGV("stepuser %08x %08x %d", user, server, stepCount);
1546
1547    uint32_t u = user;
1548    u += stepCount;
1549    // Ensure that user is never ahead of server for AudioRecord
1550    if (isOut) {
1551        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1552        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1553            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1554        }
1555    } else if (u > server) {
1556        ALOGW("stepUser occurred after track reset");
1557        u = server;
1558    }
1559
1560    if (u >= frameCount) {
1561        // common case, user didn't just wrap
1562        if (u - frameCount >= userBase ) {
1563            userBase += frameCount;
1564        }
1565    } else if (u >= userBase + frameCount) {
1566        // user just wrapped
1567        userBase += frameCount;
1568    }
1569
1570    user = u;
1571
1572    // Clear flow control error condition as new data has been written/read to/from buffer.
1573    if (flags & CBLK_UNDERRUN) {
1574        android_atomic_and(~CBLK_UNDERRUN, &flags);
1575    }
1576
1577    return u;
1578}
1579
1580bool audio_track_cblk_t::stepServer(size_t stepCount, size_t frameCount, bool isOut)
1581{
1582    ALOGV("stepserver %08x %08x %d", user, server, stepCount);
1583
1584    if (!tryLock()) {
1585        ALOGW("stepServer() could not lock cblk");
1586        return false;
1587    }
1588
1589    uint32_t s = server;
1590    bool flushed = (s == user);
1591
1592    s += stepCount;
1593    if (isOut) {
1594        // Mark that we have read the first buffer so that next time stepUser() is called
1595        // we switch to normal obtainBuffer() timeout period
1596        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1597            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1598        }
1599        // It is possible that we receive a flush()
1600        // while the mixer is processing a block: in this case,
1601        // stepServer() is called After the flush() has reset u & s and
1602        // we have s > u
1603        if (flushed) {
1604            ALOGW("stepServer occurred after track reset");
1605            s = user;
1606        }
1607    }
1608
1609    if (s >= loopEnd) {
1610        ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1611        s = loopStart;
1612        if (--loopCount == 0) {
1613            loopEnd = UINT_MAX;
1614            loopStart = UINT_MAX;
1615        }
1616    }
1617
1618    if (s >= frameCount) {
1619        // common case, server didn't just wrap
1620        if (s - frameCount >= serverBase ) {
1621            serverBase += frameCount;
1622        }
1623    } else if (s >= serverBase + frameCount) {
1624        // server just wrapped
1625        serverBase += frameCount;
1626    }
1627
1628    server = s;
1629
1630    if (!(flags & CBLK_INVALID)) {
1631        cv.signal();
1632    }
1633    lock.unlock();
1634    return true;
1635}
1636
1637void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const
1638{
1639    return (int8_t *)buffers + (offset - userBase) * frameSize;
1640}
1641
1642uint32_t audio_track_cblk_t::framesAvailable(size_t frameCount, bool isOut)
1643{
1644    Mutex::Autolock _l(lock);
1645    return framesAvailable_l(frameCount, isOut);
1646}
1647
1648uint32_t audio_track_cblk_t::framesAvailable_l(size_t frameCount, bool isOut)
1649{
1650    uint32_t u = user;
1651    uint32_t s = server;
1652
1653    if (isOut) {
1654        uint32_t limit = (s < loopStart) ? s : loopStart;
1655        return limit + frameCount - u;
1656    } else {
1657        return frameCount + u - s;
1658    }
1659}
1660
1661uint32_t audio_track_cblk_t::framesReady(bool isOut)
1662{
1663    uint32_t u = user;
1664    uint32_t s = server;
1665
1666    if (isOut) {
1667        if (u < loopEnd) {
1668            return u - s;
1669        } else {
1670            // do not block on mutex shared with client on AudioFlinger side
1671            if (!tryLock()) {
1672                ALOGW("framesReady() could not lock cblk");
1673                return 0;
1674            }
1675            uint32_t frames = UINT_MAX;
1676            if (loopCount >= 0) {
1677                frames = (loopEnd - loopStart)*loopCount + u - s;
1678            }
1679            lock.unlock();
1680            return frames;
1681        }
1682    } else {
1683        return s - u;
1684    }
1685}
1686
1687bool audio_track_cblk_t::tryLock()
1688{
1689    // the code below simulates lock-with-timeout
1690    // we MUST do this to protect the AudioFlinger server
1691    // as this lock is shared with the client.
1692    status_t err;
1693
1694    err = lock.tryLock();
1695    if (err == -EBUSY) { // just wait a bit
1696        usleep(1000);
1697        err = lock.tryLock();
1698    }
1699    if (err != NO_ERROR) {
1700        // probably, the client just died.
1701        return false;
1702    }
1703    return true;
1704}
1705
1706// -------------------------------------------------------------------------
1707
1708}; // namespace android
1709