AudioTrack.cpp revision 8424361609e0a94b9a240b43920529a84a63ed15 
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, &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, cblk);
382        status_t status = NO_ERROR;
383        if (!(cblk->flags & CBLK_INVALID)) {
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, &cblk->flags);
390            }
391        }
392        if (cblk->flags & CBLK_INVALID) {
393            audio_track_cblk_t* temp = cblk;
394            status = restoreTrack_l(temp, true);
395            cblk = temp;
396        }
397        cblk->lock.unlock();
398        if (status != NO_ERROR) {
399            ALOGV("start() failed");
400            mActive = false;
401            if (t != 0) {
402                t->pause();
403            } else {
404                setpriority(PRIO_PROCESS, 0, mPreviousPriority);
405                set_sched_policy(0, mPreviousSchedulingGroup);
406            }
407        }
408    }
409
410}
411
412void AudioTrack::stop()
413{
414    sp<AudioTrackThread> t = mAudioTrackThread;
415
416    ALOGV("stop %p", this);
417
418    AutoMutex lock(mLock);
419    if (mActive) {
420        mActive = false;
421        mCblk->cv.signal();
422        mAudioTrack->stop();
423        // Cancel loops (If we are in the middle of a loop, playback
424        // would not stop until loopCount reaches 0).
425        setLoop_l(0, 0, 0);
426        // the playback head position will reset to 0, so if a marker is set, we need
427        // to activate it again
428        mMarkerReached = false;
429        // Force flush if a shared buffer is used otherwise audioflinger
430        // will not stop before end of buffer is reached.
431        if (mSharedBuffer != 0) {
432            flush_l();
433        }
434        if (t != 0) {
435            t->pause();
436        } else {
437            setpriority(PRIO_PROCESS, 0, mPreviousPriority);
438            set_sched_policy(0, mPreviousSchedulingGroup);
439        }
440    }
441
442}
443
444bool AudioTrack::stopped() const
445{
446    AutoMutex lock(mLock);
447    return stopped_l();
448}
449
450void AudioTrack::flush()
451{
452    AutoMutex lock(mLock);
453    flush_l();
454}
455
456// must be called with mLock held
457void AudioTrack::flush_l()
458{
459    ALOGV("flush");
460
461    // clear playback marker and periodic update counter
462    mMarkerPosition = 0;
463    mMarkerReached = false;
464    mUpdatePeriod = 0;
465
466    if (!mActive) {
467        mFlushed = true;
468        mAudioTrack->flush();
469        // Release AudioTrack callback thread in case it was waiting for new buffers
470        // in AudioTrack::obtainBuffer()
471        mCblk->cv.signal();
472    }
473}
474
475void AudioTrack::pause()
476{
477    ALOGV("pause");
478    AutoMutex lock(mLock);
479    if (mActive) {
480        mActive = false;
481        mCblk->cv.signal();
482        mAudioTrack->pause();
483    }
484}
485
486void AudioTrack::mute(bool e)
487{
488    mAudioTrack->mute(e);
489    mMuted = e;
490}
491
492bool AudioTrack::muted() const
493{
494    return mMuted;
495}
496
497status_t AudioTrack::setVolume(float left, float right)
498{
499    if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
500        return BAD_VALUE;
501    }
502
503    AutoMutex lock(mLock);
504    mVolume[LEFT] = left;
505    mVolume[RIGHT] = right;
506
507    mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
508
509    return NO_ERROR;
510}
511
512void AudioTrack::getVolume(float* left, float* right) const
513{
514    if (left != NULL) {
515        *left  = mVolume[LEFT];
516    }
517    if (right != NULL) {
518        *right = mVolume[RIGHT];
519    }
520}
521
522status_t AudioTrack::setAuxEffectSendLevel(float level)
523{
524    ALOGV("setAuxEffectSendLevel(%f)", level);
525    if (level < 0.0f || level > 1.0f) {
526        return BAD_VALUE;
527    }
528    AutoMutex lock(mLock);
529
530    mSendLevel = level;
531
532    mCblk->setSendLevel(level);
533
534    return NO_ERROR;
535}
536
537void AudioTrack::getAuxEffectSendLevel(float* level) const
538{
539    if (level != NULL) {
540        *level  = mSendLevel;
541    }
542}
543
544status_t AudioTrack::setSampleRate(int rate)
545{
546    int afSamplingRate;
547
548    if (mIsTimed) {
549        return INVALID_OPERATION;
550    }
551
552    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
553        return NO_INIT;
554    }
555    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
556    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
557
558    AutoMutex lock(mLock);
559    mCblk->sampleRate = rate;
560    return NO_ERROR;
561}
562
563uint32_t AudioTrack::getSampleRate() const
564{
565    if (mIsTimed) {
566        return INVALID_OPERATION;
567    }
568
569    AutoMutex lock(mLock);
570    return mCblk->sampleRate;
571}
572
573status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
574{
575    AutoMutex lock(mLock);
576    return setLoop_l(loopStart, loopEnd, loopCount);
577}
578
579// must be called with mLock held
580status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
581{
582    audio_track_cblk_t* cblk = mCblk;
583
584    Mutex::Autolock _l(cblk->lock);
585
586    if (loopCount == 0) {
587        cblk->loopStart = UINT_MAX;
588        cblk->loopEnd = UINT_MAX;
589        cblk->loopCount = 0;
590        mLoopCount = 0;
591        return NO_ERROR;
592    }
593
594    if (mIsTimed) {
595        return INVALID_OPERATION;
596    }
597
598    if (loopStart >= loopEnd ||
599        loopEnd - loopStart > cblk->frameCount ||
600        cblk->server > loopStart) {
601        ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, "
602              "user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
603        return BAD_VALUE;
604    }
605
606    if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
607        ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, "
608            "framecount %d",
609            loopStart, loopEnd, cblk->frameCount);
610        return BAD_VALUE;
611    }
612
613    cblk->loopStart = loopStart;
614    cblk->loopEnd = loopEnd;
615    cblk->loopCount = loopCount;
616    mLoopCount = loopCount;
617
618    return NO_ERROR;
619}
620
621status_t AudioTrack::setMarkerPosition(uint32_t marker)
622{
623    if (mCbf == NULL) return INVALID_OPERATION;
624
625    mMarkerPosition = marker;
626    mMarkerReached = false;
627
628    return NO_ERROR;
629}
630
631status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
632{
633    if (marker == NULL) return BAD_VALUE;
634
635    *marker = mMarkerPosition;
636
637    return NO_ERROR;
638}
639
640status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
641{
642    if (mCbf == NULL) return INVALID_OPERATION;
643
644    uint32_t curPosition;
645    getPosition(&curPosition);
646    mNewPosition = curPosition + updatePeriod;
647    mUpdatePeriod = updatePeriod;
648
649    return NO_ERROR;
650}
651
652status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
653{
654    if (updatePeriod == NULL) return BAD_VALUE;
655
656    *updatePeriod = mUpdatePeriod;
657
658    return NO_ERROR;
659}
660
661status_t AudioTrack::setPosition(uint32_t position)
662{
663    if (mIsTimed) return INVALID_OPERATION;
664
665    AutoMutex lock(mLock);
666
667    if (!stopped_l()) return INVALID_OPERATION;
668
669    audio_track_cblk_t* cblk = mCblk;
670    Mutex::Autolock _l(cblk->lock);
671
672    if (position > cblk->user) return BAD_VALUE;
673
674    cblk->server = position;
675    android_atomic_or(CBLK_FORCEREADY, &cblk->flags);
676
677    return NO_ERROR;
678}
679
680status_t AudioTrack::getPosition(uint32_t *position)
681{
682    if (position == NULL) return BAD_VALUE;
683    AutoMutex lock(mLock);
684    *position = mFlushed ? 0 : mCblk->server;
685
686    return NO_ERROR;
687}
688
689status_t AudioTrack::reload()
690{
691    AutoMutex lock(mLock);
692
693    if (!stopped_l()) return INVALID_OPERATION;
694
695    flush_l();
696
697    audio_track_cblk_t* cblk = mCblk;
698    cblk->stepUser(cblk->frameCount);
699
700    return NO_ERROR;
701}
702
703audio_io_handle_t AudioTrack::getOutput()
704{
705    AutoMutex lock(mLock);
706    return getOutput_l();
707}
708
709// must be called with mLock held
710audio_io_handle_t AudioTrack::getOutput_l()
711{
712    return AudioSystem::getOutput(mStreamType,
713            mCblk->sampleRate, mFormat, mChannelMask, mFlags);
714}
715
716int AudioTrack::getSessionId() const
717{
718    return mSessionId;
719}
720
721status_t AudioTrack::attachAuxEffect(int effectId)
722{
723    ALOGV("attachAuxEffect(%d)", effectId);
724    status_t status = mAudioTrack->attachAuxEffect(effectId);
725    if (status == NO_ERROR) {
726        mAuxEffectId = effectId;
727    }
728    return status;
729}
730
731// -------------------------------------------------------------------------
732
733// must be called with mLock held
734status_t AudioTrack::createTrack_l(
735        audio_stream_type_t streamType,
736        uint32_t sampleRate,
737        audio_format_t format,
738        audio_channel_mask_t channelMask,
739        int frameCount,
740        audio_output_flags_t flags,
741        const sp<IMemory>& sharedBuffer,
742        audio_io_handle_t output)
743{
744    status_t status;
745    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
746    if (audioFlinger == 0) {
747        ALOGE("Could not get audioflinger");
748        return NO_INIT;
749    }
750
751    uint32_t afLatency;
752    if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) {
753        return NO_INIT;
754    }
755
756    // Client decides whether the track is TIMED (see below), but can only express a preference
757    // for FAST.  Server will perform additional tests.
758    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
759            // either of these use cases:
760            // use case 1: shared buffer
761            (sharedBuffer != 0) ||
762            // use case 2: callback handler
763            (mCbf != NULL))) {
764        ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
765        // once denied, do not request again if IAudioTrack is re-created
766        flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
767        mFlags = flags;
768    }
769    ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
770
771    mNotificationFramesAct = mNotificationFramesReq;
772
773    if (!audio_is_linear_pcm(format)) {
774
775        if (sharedBuffer != 0) {
776            // Same comment as below about ignoring frameCount parameter for set()
777            frameCount = sharedBuffer->size();
778        } else if (frameCount == 0) {
779            int afFrameCount;
780            if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
781                return NO_INIT;
782            }
783            frameCount = afFrameCount;
784        }
785
786    } else if (sharedBuffer != 0) {
787
788        // Ensure that buffer alignment matches channelCount
789        int channelCount = popcount(channelMask);
790        // 8-bit data in shared memory is not currently supported by AudioFlinger
791        size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
792        if (channelCount > 1) {
793            // More than 2 channels does not require stronger alignment than stereo
794            alignment <<= 1;
795        }
796        if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
797            ALOGE("Invalid buffer alignment: address %p, channelCount %d",
798                    sharedBuffer->pointer(), channelCount);
799            return BAD_VALUE;
800        }
801
802        // When initializing a shared buffer AudioTrack via constructors,
803        // there's no frameCount parameter.
804        // But when initializing a shared buffer AudioTrack via set(),
805        // there _is_ a frameCount parameter.  We silently ignore it.
806        frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
807
808    } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
809
810        // FIXME move these calculations and associated checks to server
811        int afSampleRate;
812        if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
813            return NO_INIT;
814        }
815        int afFrameCount;
816        if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
817            return NO_INIT;
818        }
819
820        // Ensure that buffer depth covers at least audio hardware latency
821        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
822        if (minBufCount < 2) minBufCount = 2;
823
824        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
825        ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
826                ", afLatency=%d",
827                minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
828
829        if (frameCount == 0) {
830            frameCount = minFrameCount;
831        }
832        if (mNotificationFramesAct == 0) {
833            mNotificationFramesAct = frameCount/2;
834        }
835        // Make sure that application is notified with sufficient margin
836        // before underrun
837        if (mNotificationFramesAct > (uint32_t)frameCount/2) {
838            mNotificationFramesAct = frameCount/2;
839        }
840        if (frameCount < minFrameCount) {
841            // not ALOGW because it happens all the time when playing key clicks over A2DP
842            ALOGV("Minimum buffer size corrected from %d to %d",
843                     frameCount, minFrameCount);
844            frameCount = minFrameCount;
845        }
846
847    } else {
848        // For fast tracks, the frame count calculations and checks are done by server
849    }
850
851    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
852    if (mIsTimed) {
853        trackFlags |= IAudioFlinger::TRACK_TIMED;
854    }
855
856    pid_t tid = -1;
857    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
858        trackFlags |= IAudioFlinger::TRACK_FAST;
859        if (mAudioTrackThread != 0) {
860            tid = mAudioTrackThread->getTid();
861        }
862    }
863
864    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
865                                                      streamType,
866                                                      sampleRate,
867                                                      format,
868                                                      channelMask,
869                                                      frameCount,
870                                                      trackFlags,
871                                                      sharedBuffer,
872                                                      output,
873                                                      tid,
874                                                      &mSessionId,
875                                                      &status);
876
877    if (track == 0) {
878        ALOGE("AudioFlinger could not create track, status: %d", status);
879        return status;
880    }
881    sp<IMemory> iMem = track->getCblk();
882    if (iMem == 0) {
883        ALOGE("Could not get control block");
884        return NO_INIT;
885    }
886    mAudioTrack = track;
887    mCblkMemory = iMem;
888    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
889    mCblk = cblk;
890    // old has the previous value of cblk->flags before the "or" operation
891    int32_t old = android_atomic_or(CBLK_DIRECTION, &cblk->flags);
892    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
893        if (old & CBLK_FAST) {
894            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", cblk->frameCount);
895        } else {
896            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", cblk->frameCount);
897            // once denied, do not request again if IAudioTrack is re-created
898            flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
899            mFlags = flags;
900        }
901        if (sharedBuffer == 0) {
902            mNotificationFramesAct = cblk->frameCount/2;
903        }
904    }
905    if (sharedBuffer == 0) {
906        cblk->buffers = (char*)cblk + sizeof(audio_track_cblk_t);
907    } else {
908        cblk->buffers = sharedBuffer->pointer();
909        // Force buffer full condition as data is already present in shared memory
910        cblk->stepUser(cblk->frameCount);
911    }
912
913    cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) |
914            uint16_t(mVolume[LEFT] * 0x1000));
915    cblk->setSendLevel(mSendLevel);
916    mAudioTrack->attachAuxEffect(mAuxEffectId);
917    cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
918    cblk->waitTimeMs = 0;
919    mRemainingFrames = mNotificationFramesAct;
920    // FIXME don't believe this lie
921    mLatency = afLatency + (1000*cblk->frameCount) / sampleRate;
922    // If IAudioTrack is re-created, don't let the requested frameCount
923    // decrease.  This can confuse clients that cache frameCount().
924    if (cblk->frameCount > mFrameCount) {
925        mFrameCount = cblk->frameCount;
926    }
927    return NO_ERROR;
928}
929
930status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
931{
932    AutoMutex lock(mLock);
933    bool active;
934    status_t result = NO_ERROR;
935    audio_track_cblk_t* cblk = mCblk;
936    uint32_t framesReq = audioBuffer->frameCount;
937    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
938
939    audioBuffer->frameCount  = 0;
940    audioBuffer->size = 0;
941
942    uint32_t framesAvail = cblk->framesAvailable();
943
944    cblk->lock.lock();
945    if (cblk->flags & CBLK_INVALID) {
946        goto create_new_track;
947    }
948    cblk->lock.unlock();
949
950    if (framesAvail == 0) {
951        cblk->lock.lock();
952        goto start_loop_here;
953        while (framesAvail == 0) {
954            active = mActive;
955            if (CC_UNLIKELY(!active)) {
956                ALOGV("Not active and NO_MORE_BUFFERS");
957                cblk->lock.unlock();
958                return NO_MORE_BUFFERS;
959            }
960            if (CC_UNLIKELY(!waitCount)) {
961                cblk->lock.unlock();
962                return WOULD_BLOCK;
963            }
964            if (!(cblk->flags & CBLK_INVALID)) {
965                mLock.unlock();
966                result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
967                cblk->lock.unlock();
968                mLock.lock();
969                if (!mActive) {
970                    return status_t(STOPPED);
971                }
972                cblk->lock.lock();
973            }
974
975            if (cblk->flags & CBLK_INVALID) {
976                goto create_new_track;
977            }
978            if (CC_UNLIKELY(result != NO_ERROR)) {
979                cblk->waitTimeMs += waitTimeMs;
980                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
981                    // timing out when a loop has been set and we have already written upto loop end
982                    // is a normal condition: no need to wake AudioFlinger up.
983                    if (cblk->user < cblk->loopEnd) {
984                        ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, "
985                              "server=%08x", this, cblk->mName, cblk->user, cblk->server);
986                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
987                        cblk->lock.unlock();
988                        result = mAudioTrack->start();
989                        cblk->lock.lock();
990                        if (result == DEAD_OBJECT) {
991                            android_atomic_or(CBLK_INVALID, &cblk->flags);
992create_new_track:
993                            audio_track_cblk_t* temp = cblk;
994                            result = restoreTrack_l(temp, false);
995                            cblk = temp;
996                        }
997                        if (result != NO_ERROR) {
998                            ALOGW("obtainBuffer create Track error %d", result);
999                            cblk->lock.unlock();
1000                            return result;
1001                        }
1002                    }
1003                    cblk->waitTimeMs = 0;
1004                }
1005
1006                if (--waitCount == 0) {
1007                    cblk->lock.unlock();
1008                    return TIMED_OUT;
1009                }
1010            }
1011            // read the server count again
1012        start_loop_here:
1013            framesAvail = cblk->framesAvailable_l();
1014        }
1015        cblk->lock.unlock();
1016    }
1017
1018    cblk->waitTimeMs = 0;
1019
1020    if (framesReq > framesAvail) {
1021        framesReq = framesAvail;
1022    }
1023
1024    uint32_t u = cblk->user;
1025    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
1026
1027    if (framesReq > bufferEnd - u) {
1028        framesReq = bufferEnd - u;
1029    }
1030
1031    audioBuffer->frameCount = framesReq;
1032    audioBuffer->size = framesReq * cblk->frameSize;
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    audio_track_cblk_t* cblk = mCblk;
1042    cblk->stepUser(audioBuffer->frameCount);
1043    if (audioBuffer->frameCount > 0) {
1044        // restart track if it was disabled by audioflinger due to previous underrun
1045        if (mActive && (cblk->flags & CBLK_DISABLED)) {
1046            android_atomic_and(~CBLK_DISABLED, &cblk->flags);
1047            ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName);
1048            mAudioTrack->start();
1049        }
1050    }
1051}
1052
1053// -------------------------------------------------------------------------
1054
1055ssize_t AudioTrack::write(const void* buffer, size_t userSize)
1056{
1057
1058    if (mSharedBuffer != 0) return INVALID_OPERATION;
1059    if (mIsTimed) return INVALID_OPERATION;
1060
1061    if (ssize_t(userSize) < 0) {
1062        // Sanity-check: user is most-likely passing an error code, and it would
1063        // make the return value ambiguous (actualSize vs error).
1064        ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
1065                buffer, userSize, userSize);
1066        return BAD_VALUE;
1067    }
1068
1069    ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
1070
1071    if (userSize == 0) {
1072        return 0;
1073    }
1074
1075    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1076    // while we are accessing the cblk
1077    mLock.lock();
1078    sp<IAudioTrack> audioTrack = mAudioTrack;
1079    sp<IMemory> iMem = mCblkMemory;
1080    mLock.unlock();
1081
1082    ssize_t written = 0;
1083    const int8_t *src = (const int8_t *)buffer;
1084    Buffer audioBuffer;
1085    size_t frameSz = frameSize();
1086
1087    do {
1088        audioBuffer.frameCount = userSize/frameSz;
1089
1090        status_t err = obtainBuffer(&audioBuffer, -1);
1091        if (err < 0) {
1092            // out of buffers, return #bytes written
1093            if (err == status_t(NO_MORE_BUFFERS))
1094                break;
1095            return ssize_t(err);
1096        }
1097
1098        size_t toWrite;
1099
1100        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1101            // Divide capacity by 2 to take expansion into account
1102            toWrite = audioBuffer.size>>1;
1103            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
1104        } else {
1105            toWrite = audioBuffer.size;
1106            memcpy(audioBuffer.i8, src, toWrite);
1107            src += toWrite;
1108        }
1109        userSize -= toWrite;
1110        written += toWrite;
1111
1112        releaseBuffer(&audioBuffer);
1113    } while (userSize >= frameSz);
1114
1115    return written;
1116}
1117
1118// -------------------------------------------------------------------------
1119
1120TimedAudioTrack::TimedAudioTrack() {
1121    mIsTimed = true;
1122}
1123
1124status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
1125{
1126    status_t result = UNKNOWN_ERROR;
1127
1128    // If the track is not invalid already, try to allocate a buffer.  alloc
1129    // fails indicating that the server is dead, flag the track as invalid so
1130    // we can attempt to restore in just a bit.
1131    audio_track_cblk_t* cblk = mCblk;
1132    if (!(cblk->flags & CBLK_INVALID)) {
1133        result = mAudioTrack->allocateTimedBuffer(size, buffer);
1134        if (result == DEAD_OBJECT) {
1135            android_atomic_or(CBLK_INVALID, &cblk->flags);
1136        }
1137    }
1138
1139    // If the track is invalid at this point, attempt to restore it. and try the
1140    // allocation one more time.
1141    if (cblk->flags & CBLK_INVALID) {
1142        cblk->lock.lock();
1143        audio_track_cblk_t* temp = cblk;
1144        result = restoreTrack_l(temp, false);
1145        cblk = temp;
1146        cblk->lock.unlock();
1147
1148        if (result == OK)
1149            result = mAudioTrack->allocateTimedBuffer(size, buffer);
1150    }
1151
1152    return result;
1153}
1154
1155status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
1156                                           int64_t pts)
1157{
1158    status_t status = mAudioTrack->queueTimedBuffer(buffer, pts);
1159    {
1160        AutoMutex lock(mLock);
1161        audio_track_cblk_t* cblk = mCblk;
1162        // restart track if it was disabled by audioflinger due to previous underrun
1163        if (buffer->size() != 0 && status == NO_ERROR &&
1164                mActive && (cblk->flags & CBLK_DISABLED)) {
1165            android_atomic_and(~CBLK_DISABLED, &cblk->flags);
1166            ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
1167            mAudioTrack->start();
1168        }
1169    }
1170    return status;
1171}
1172
1173status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
1174                                                TargetTimeline target)
1175{
1176    return mAudioTrack->setMediaTimeTransform(xform, target);
1177}
1178
1179// -------------------------------------------------------------------------
1180
1181bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
1182{
1183    Buffer audioBuffer;
1184    uint32_t frames;
1185    size_t writtenSize;
1186
1187    mLock.lock();
1188    // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
1189    // while we are accessing the cblk
1190    sp<IAudioTrack> audioTrack = mAudioTrack;
1191    sp<IMemory> iMem = mCblkMemory;
1192    audio_track_cblk_t* cblk = mCblk;
1193    bool active = mActive;
1194    mLock.unlock();
1195
1196    // Manage underrun callback
1197    if (active && (cblk->framesAvailable() == cblk->frameCount)) {
1198        ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
1199        if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) {
1200            mCbf(EVENT_UNDERRUN, mUserData, 0);
1201            if (cblk->server == cblk->frameCount) {
1202                mCbf(EVENT_BUFFER_END, mUserData, 0);
1203            }
1204            if (mSharedBuffer != 0) return false;
1205        }
1206    }
1207
1208    // Manage loop end callback
1209    while (mLoopCount > cblk->loopCount) {
1210        int loopCount = -1;
1211        mLoopCount--;
1212        if (mLoopCount >= 0) loopCount = mLoopCount;
1213
1214        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
1215    }
1216
1217    // Manage marker callback
1218    if (!mMarkerReached && (mMarkerPosition > 0)) {
1219        if (cblk->server >= mMarkerPosition) {
1220            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
1221            mMarkerReached = true;
1222        }
1223    }
1224
1225    // Manage new position callback
1226    if (mUpdatePeriod > 0) {
1227        while (cblk->server >= mNewPosition) {
1228            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
1229            mNewPosition += mUpdatePeriod;
1230        }
1231    }
1232
1233    // If Shared buffer is used, no data is requested from client.
1234    if (mSharedBuffer != 0) {
1235        frames = 0;
1236    } else {
1237        frames = mRemainingFrames;
1238    }
1239
1240    // See description of waitCount parameter at declaration of obtainBuffer().
1241    // The logic below prevents us from being stuck below at obtainBuffer()
1242    // not being able to handle timed events (position, markers, loops).
1243    int32_t waitCount = -1;
1244    if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
1245        waitCount = 1;
1246    }
1247
1248    do {
1249
1250        audioBuffer.frameCount = frames;
1251
1252        status_t err = obtainBuffer(&audioBuffer, waitCount);
1253        if (err < NO_ERROR) {
1254            if (err != TIMED_OUT) {
1255                ALOGE_IF(err != status_t(NO_MORE_BUFFERS),
1256                        "Error obtaining an audio buffer, giving up.");
1257                return false;
1258            }
1259            break;
1260        }
1261        if (err == status_t(STOPPED)) return false;
1262
1263        // Divide buffer size by 2 to take into account the expansion
1264        // due to 8 to 16 bit conversion: the callback must fill only half
1265        // of the destination buffer
1266        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1267            audioBuffer.size >>= 1;
1268        }
1269
1270        size_t reqSize = audioBuffer.size;
1271        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
1272        writtenSize = audioBuffer.size;
1273
1274        // Sanity check on returned size
1275        if (ssize_t(writtenSize) <= 0) {
1276            // The callback is done filling buffers
1277            // Keep this thread going to handle timed events and
1278            // still try to get more data in intervals of WAIT_PERIOD_MS
1279            // but don't just loop and block the CPU, so wait
1280            usleep(WAIT_PERIOD_MS*1000);
1281            break;
1282        }
1283
1284        if (writtenSize > reqSize) writtenSize = reqSize;
1285
1286        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
1287            // 8 to 16 bit conversion, note that source and destination are the same address
1288            memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
1289            writtenSize <<= 1;
1290        }
1291
1292        audioBuffer.size = writtenSize;
1293        // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for
1294        // 8 bit PCM data: in this case,  cblk->frameSize is based on a sample size of
1295        // 16 bit.
1296        audioBuffer.frameCount = writtenSize/cblk->frameSize;
1297
1298        frames -= audioBuffer.frameCount;
1299
1300        releaseBuffer(&audioBuffer);
1301    }
1302    while (frames);
1303
1304    if (frames == 0) {
1305        mRemainingFrames = mNotificationFramesAct;
1306    } else {
1307        mRemainingFrames = frames;
1308    }
1309    return true;
1310}
1311
1312// must be called with mLock and refCblk.lock held. Callers must also hold strong references on
1313// the IAudioTrack and IMemory in case they are recreated here.
1314// If the IAudioTrack is successfully restored, the refCblk pointer is updated
1315// FIXME Don't depend on caller to hold strong references.
1316status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart)
1317{
1318    status_t result;
1319
1320    audio_track_cblk_t* cblk = refCblk;
1321    audio_track_cblk_t* newCblk = cblk;
1322    if (!(android_atomic_or(CBLK_RESTORING, &cblk->flags) & CBLK_RESTORING)) {
1323        ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
1324            fromStart ? "start()" : "obtainBuffer()", gettid());
1325
1326        // signal old cblk condition so that other threads waiting for available buffers stop
1327        // waiting now
1328        cblk->cv.broadcast();
1329        cblk->lock.unlock();
1330
1331        // refresh the audio configuration cache in this process to make sure we get new
1332        // output parameters in getOutput_l() and createTrack_l()
1333        AudioSystem::clearAudioConfigCache();
1334
1335        // if the new IAudioTrack is created, createTrack_l() will modify the
1336        // following member variables: mAudioTrack, mCblkMemory and mCblk.
1337        // It will also delete the strong references on previous IAudioTrack and IMemory
1338        result = createTrack_l(mStreamType,
1339                               cblk->sampleRate,
1340                               mFormat,
1341                               mChannelMask,
1342                               mFrameCount,
1343                               mFlags,
1344                               mSharedBuffer,
1345                               getOutput_l());
1346
1347        if (result == NO_ERROR) {
1348            uint32_t user = cblk->user;
1349            uint32_t server = cblk->server;
1350            // restore write index and set other indexes to reflect empty buffer status
1351            newCblk = mCblk;
1352            newCblk->user = user;
1353            newCblk->server = user;
1354            newCblk->userBase = user;
1355            newCblk->serverBase = user;
1356            // restore loop: this is not guaranteed to succeed if new frame count is not
1357            // compatible with loop length
1358            setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount);
1359            if (!fromStart) {
1360                newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1361                // Make sure that a client relying on callback events indicating underrun or
1362                // the actual amount of audio frames played (e.g SoundPool) receives them.
1363                if (mSharedBuffer == 0) {
1364                    uint32_t frames = 0;
1365                    if (user > server) {
1366                        frames = ((user - server) > newCblk->frameCount) ?
1367                                newCblk->frameCount : (user - server);
1368                        memset(newCblk->buffers, 0, frames * newCblk->frameSize);
1369                    }
1370                    // restart playback even if buffer is not completely filled.
1371                    android_atomic_or(CBLK_FORCEREADY, &newCblk->flags);
1372                    // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to
1373                    // the client
1374                    newCblk->stepUser(frames);
1375                }
1376            }
1377            if (mSharedBuffer != 0) {
1378                newCblk->stepUser(newCblk->frameCount);
1379            }
1380            if (mActive) {
1381                result = mAudioTrack->start();
1382                ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
1383            }
1384            if (fromStart && result == NO_ERROR) {
1385                mNewPosition = newCblk->server + mUpdatePeriod;
1386            }
1387        }
1388        if (result != NO_ERROR) {
1389            android_atomic_and(~CBLK_RESTORING, &cblk->flags);
1390            ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result);
1391        }
1392        mRestoreStatus = result;
1393        // signal old cblk condition for other threads waiting for restore completion
1394        android_atomic_or(CBLK_RESTORED, &cblk->flags);
1395        cblk->cv.broadcast();
1396    } else {
1397        bool haveLogged = false;
1398        for (;;) {
1399            if (cblk->flags & CBLK_RESTORED) {
1400                ALOGW("dead IAudioTrack restored");
1401                result = mRestoreStatus;
1402                cblk->lock.unlock();
1403                break;
1404            }
1405            if (!haveLogged) {
1406                ALOGW("dead IAudioTrack, waiting for a new one");
1407                haveLogged = true;
1408            }
1409            mLock.unlock();
1410            result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS));
1411            cblk->lock.unlock();
1412            mLock.lock();
1413            if (result != NO_ERROR) {
1414                ALOGW("timed out");
1415                break;
1416            }
1417            cblk->lock.lock();
1418        }
1419    }
1420    ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
1421        result, mActive, newCblk, cblk, newCblk->flags, cblk->flags);
1422
1423    if (result == NO_ERROR) {
1424        // from now on we switch to the newly created cblk
1425        refCblk = newCblk;
1426    }
1427    newCblk->lock.lock();
1428
1429    ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid());
1430
1431    return result;
1432}
1433
1434status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
1435{
1436
1437    const size_t SIZE = 256;
1438    char buffer[SIZE];
1439    String8 result;
1440
1441    audio_track_cblk_t* cblk = mCblk;
1442    result.append(" AudioTrack::dump\n");
1443    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
1444            mVolume[0], mVolume[1]);
1445    result.append(buffer);
1446    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
1447            mChannelCount, cblk->frameCount);
1448    result.append(buffer);
1449    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n",
1450            (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted);
1451    result.append(buffer);
1452    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
1453    result.append(buffer);
1454    ::write(fd, result.string(), result.size());
1455    return NO_ERROR;
1456}
1457
1458// =========================================================================
1459
1460AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
1461    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true)
1462{
1463}
1464
1465AudioTrack::AudioTrackThread::~AudioTrackThread()
1466{
1467}
1468
1469bool AudioTrack::AudioTrackThread::threadLoop()
1470{
1471    {
1472        AutoMutex _l(mMyLock);
1473        if (mPaused) {
1474            mMyCond.wait(mMyLock);
1475            // caller will check for exitPending()
1476            return true;
1477        }
1478    }
1479    if (!mReceiver.processAudioBuffer(this)) {
1480        pause();
1481    }
1482    return true;
1483}
1484
1485void AudioTrack::AudioTrackThread::requestExit()
1486{
1487    // must be in this order to avoid a race condition
1488    Thread::requestExit();
1489    resume();
1490}
1491
1492void AudioTrack::AudioTrackThread::pause()
1493{
1494    AutoMutex _l(mMyLock);
1495    mPaused = true;
1496}
1497
1498void AudioTrack::AudioTrackThread::resume()
1499{
1500    AutoMutex _l(mMyLock);
1501    if (mPaused) {
1502        mPaused = false;
1503        mMyCond.signal();
1504    }
1505}
1506
1507// =========================================================================
1508
1509
1510audio_track_cblk_t::audio_track_cblk_t()
1511    : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
1512    userBase(0), serverBase(0), buffers(NULL), frameCount(0),
1513    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
1514    mSendLevel(0), flags(0)
1515{
1516}
1517
1518uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
1519{
1520    ALOGV("stepuser %08x %08x %d", user, server, frameCount);
1521
1522    uint32_t u = user;
1523    u += frameCount;
1524    // Ensure that user is never ahead of server for AudioRecord
1525    if (flags & CBLK_DIRECTION) {
1526        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
1527        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1528            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1529        }
1530    } else if (u > server) {
1531        ALOGW("stepUser occurred after track reset");
1532        u = server;
1533    }
1534
1535    uint32_t fc = this->frameCount;
1536    if (u >= fc) {
1537        // common case, user didn't just wrap
1538        if (u - fc >= userBase ) {
1539            userBase += fc;
1540        }
1541    } else if (u >= userBase + fc) {
1542        // user just wrapped
1543        userBase += fc;
1544    }
1545
1546    user = u;
1547
1548    // Clear flow control error condition as new data has been written/read to/from buffer.
1549    if (flags & CBLK_UNDERRUN) {
1550        android_atomic_and(~CBLK_UNDERRUN, &flags);
1551    }
1552
1553    return u;
1554}
1555
1556bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1557{
1558    ALOGV("stepserver %08x %08x %d", user, server, frameCount);
1559
1560    if (!tryLock()) {
1561        ALOGW("stepServer() could not lock cblk");
1562        return false;
1563    }
1564
1565    uint32_t s = server;
1566    bool flushed = (s == user);
1567
1568    s += frameCount;
1569    if (flags & CBLK_DIRECTION) {
1570        // Mark that we have read the first buffer so that next time stepUser() is called
1571        // we switch to normal obtainBuffer() timeout period
1572        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1573            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1574        }
1575        // It is possible that we receive a flush()
1576        // while the mixer is processing a block: in this case,
1577        // stepServer() is called After the flush() has reset u & s and
1578        // we have s > u
1579        if (flushed) {
1580            ALOGW("stepServer occurred after track reset");
1581            s = user;
1582        }
1583    }
1584
1585    if (s >= loopEnd) {
1586        ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1587        s = loopStart;
1588        if (--loopCount == 0) {
1589            loopEnd = UINT_MAX;
1590            loopStart = UINT_MAX;
1591        }
1592    }
1593
1594    uint32_t fc = this->frameCount;
1595    if (s >= fc) {
1596        // common case, server didn't just wrap
1597        if (s - fc >= serverBase ) {
1598            serverBase += fc;
1599        }
1600    } else if (s >= serverBase + fc) {
1601        // server just wrapped
1602        serverBase += fc;
1603    }
1604
1605    server = s;
1606
1607    if (!(flags & CBLK_INVALID)) {
1608        cv.signal();
1609    }
1610    lock.unlock();
1611    return true;
1612}
1613
1614void* audio_track_cblk_t::buffer(uint32_t offset) const
1615{
1616    return (int8_t *)buffers + (offset - userBase) * frameSize;
1617}
1618
1619uint32_t audio_track_cblk_t::framesAvailable()
1620{
1621    Mutex::Autolock _l(lock);
1622    return framesAvailable_l();
1623}
1624
1625uint32_t audio_track_cblk_t::framesAvailable_l()
1626{
1627    uint32_t u = user;
1628    uint32_t s = server;
1629
1630    if (flags & CBLK_DIRECTION) {
1631        uint32_t limit = (s < loopStart) ? s : loopStart;
1632        return limit + frameCount - u;
1633    } else {
1634        return frameCount + u - s;
1635    }
1636}
1637
1638uint32_t audio_track_cblk_t::framesReady()
1639{
1640    uint32_t u = user;
1641    uint32_t s = server;
1642
1643    if (flags & CBLK_DIRECTION) {
1644        if (u < loopEnd) {
1645            return u - s;
1646        } else {
1647            // do not block on mutex shared with client on AudioFlinger side
1648            if (!tryLock()) {
1649                ALOGW("framesReady() could not lock cblk");
1650                return 0;
1651            }
1652            uint32_t frames = UINT_MAX;
1653            if (loopCount >= 0) {
1654                frames = (loopEnd - loopStart)*loopCount + u - s;
1655            }
1656            lock.unlock();
1657            return frames;
1658        }
1659    } else {
1660        return s - u;
1661    }
1662}
1663
1664bool audio_track_cblk_t::tryLock()
1665{
1666    // the code below simulates lock-with-timeout
1667    // we MUST do this to protect the AudioFlinger server
1668    // as this lock is shared with the client.
1669    status_t err;
1670
1671    err = lock.tryLock();
1672    if (err == -EBUSY) { // just wait a bit
1673        usleep(1000);
1674        err = lock.tryLock();
1675    }
1676    if (err != NO_ERROR) {
1677        // probably, the client just died.
1678        return false;
1679    }
1680    return true;
1681}
1682
1683// -------------------------------------------------------------------------
1684
1685}; // namespace android
1686