AudioTrack.cpp revision 34f1d8ecd23169a5f299937e3aaf1bd7937578a0 
1/* //device/extlibs/pv/android/AudioTrack.cpp
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/MemoryDealer.h>
36#include <binder/Parcel.h>
37#include <binder/IPCThreadState.h>
38#include <utils/Timers.h>
39#include <cutils/atomic.h>
40
41#define LIKELY( exp )       (__builtin_expect( (exp) != 0, true  ))
42#define UNLIKELY( exp )     (__builtin_expect( (exp) != 0, false ))
43
44namespace android {
45
46// ---------------------------------------------------------------------------
47
48AudioTrack::AudioTrack()
49    : mStatus(NO_INIT)
50{
51}
52
53AudioTrack::AudioTrack(
54        int streamType,
55        uint32_t sampleRate,
56        int format,
57        int channels,
58        int frameCount,
59        uint32_t flags,
60        callback_t cbf,
61        void* user,
62        int notificationFrames)
63    : mStatus(NO_INIT)
64{
65    mStatus = set(streamType, sampleRate, format, channels,
66            frameCount, flags, cbf, user, notificationFrames, 0);
67}
68
69AudioTrack::AudioTrack(
70        int streamType,
71        uint32_t sampleRate,
72        int format,
73        int channels,
74        const sp<IMemory>& sharedBuffer,
75        uint32_t flags,
76        callback_t cbf,
77        void* user,
78        int notificationFrames)
79    : mStatus(NO_INIT)
80{
81    mStatus = set(streamType, sampleRate, format, channels,
82            0, flags, cbf, user, notificationFrames, sharedBuffer);
83}
84
85AudioTrack::~AudioTrack()
86{
87    LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
88
89    if (mStatus == NO_ERROR) {
90        // Make sure that callback function exits in the case where
91        // it is looping on buffer full condition in obtainBuffer().
92        // Otherwise the callback thread will never exit.
93        stop();
94        if (mAudioTrackThread != 0) {
95            mAudioTrackThread->requestExitAndWait();
96            mAudioTrackThread.clear();
97        }
98        mAudioTrack.clear();
99        IPCThreadState::self()->flushCommands();
100        AudioSystem::releaseOutput(getOutput());
101    }
102}
103
104status_t AudioTrack::set(
105        int streamType,
106        uint32_t sampleRate,
107        int format,
108        int channels,
109        int frameCount,
110        uint32_t flags,
111        callback_t cbf,
112        void* user,
113        int notificationFrames,
114        const sp<IMemory>& sharedBuffer,
115        bool threadCanCallJava)
116{
117
118    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
119
120    if (mAudioTrack != 0) {
121        LOGE("Track already in use");
122        return INVALID_OPERATION;
123    }
124
125    int afSampleRate;
126    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
127        return NO_INIT;
128    }
129    int afFrameCount;
130    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
131        return NO_INIT;
132    }
133    uint32_t afLatency;
134    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
135        return NO_INIT;
136    }
137
138    // handle default values first.
139    if (streamType == AudioSystem::DEFAULT) {
140        streamType = AudioSystem::MUSIC;
141    }
142    if (sampleRate == 0) {
143        sampleRate = afSampleRate;
144    }
145    // these below should probably come from the audioFlinger too...
146    if (format == 0) {
147        format = AudioSystem::PCM_16_BIT;
148    }
149    if (channels == 0) {
150        channels = AudioSystem::CHANNEL_OUT_STEREO;
151    }
152
153    // validate parameters
154    if (!AudioSystem::isValidFormat(format)) {
155        LOGE("Invalid format");
156        return BAD_VALUE;
157    }
158
159    // force direct flag if format is not linear PCM
160    if (!AudioSystem::isLinearPCM(format)) {
161        flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
162    }
163
164    if (!AudioSystem::isOutputChannel(channels)) {
165        LOGE("Invalid channel mask");
166        return BAD_VALUE;
167    }
168    uint32_t channelCount = AudioSystem::popCount(channels);
169
170    audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
171            sampleRate, format, channels, (AudioSystem::output_flags)flags);
172
173    if (output == 0) {
174        LOGE("Could not get audio output for stream type %d", streamType);
175        return BAD_VALUE;
176    }
177
178    if (!AudioSystem::isLinearPCM(format)) {
179        if (sharedBuffer != 0) {
180            frameCount = sharedBuffer->size();
181        }
182    } else {
183        // Ensure that buffer depth covers at least audio hardware latency
184        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
185        if (minBufCount < 2) minBufCount = 2;
186
187        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
188
189        if (sharedBuffer == 0) {
190            if (frameCount == 0) {
191                frameCount = minFrameCount;
192            }
193            if (notificationFrames == 0) {
194                notificationFrames = frameCount/2;
195            }
196            // Make sure that application is notified with sufficient margin
197            // before underrun
198            if (notificationFrames > frameCount/2) {
199                notificationFrames = frameCount/2;
200            }
201            if (frameCount < minFrameCount) {
202              LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
203              return BAD_VALUE;
204            }
205        } else {
206            // Ensure that buffer alignment matches channelcount
207            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
208                LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
209                return BAD_VALUE;
210            }
211            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
212        }
213    }
214
215    mVolume[LEFT] = 1.0f;
216    mVolume[RIGHT] = 1.0f;
217    // create the IAudioTrack
218    status_t status = createTrack(streamType, sampleRate, format, channelCount,
219                                  frameCount, flags, sharedBuffer, output);
220
221    if (status != NO_ERROR) {
222        return status;
223    }
224
225    if (cbf != 0) {
226        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
227        if (mAudioTrackThread == 0) {
228          LOGE("Could not create callback thread");
229          return NO_INIT;
230        }
231    }
232
233    mStatus = NO_ERROR;
234
235    mStreamType = streamType;
236    mFormat = format;
237    mChannels = channels;
238    mChannelCount = channelCount;
239    mSharedBuffer = sharedBuffer;
240    mMuted = false;
241    mActive = 0;
242    mCbf = cbf;
243    mNotificationFrames = notificationFrames;
244    mRemainingFrames = notificationFrames;
245    mUserData = user;
246    mLatency = afLatency + (1000*mFrameCount) / sampleRate;
247    mLoopCount = 0;
248    mMarkerPosition = 0;
249    mMarkerReached = false;
250    mNewPosition = 0;
251    mUpdatePeriod = 0;
252    mFlags = flags;
253
254    return NO_ERROR;
255}
256
257status_t AudioTrack::initCheck() const
258{
259    return mStatus;
260}
261
262// -------------------------------------------------------------------------
263
264uint32_t AudioTrack::latency() const
265{
266    return mLatency;
267}
268
269int AudioTrack::streamType() const
270{
271    return mStreamType;
272}
273
274int AudioTrack::format() const
275{
276    return mFormat;
277}
278
279int AudioTrack::channelCount() const
280{
281    return mChannelCount;
282}
283
284uint32_t AudioTrack::frameCount() const
285{
286    return mFrameCount;
287}
288
289int AudioTrack::frameSize() const
290{
291    if (AudioSystem::isLinearPCM(mFormat)) {
292        return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
293    } else {
294        return sizeof(uint8_t);
295    }
296}
297
298sp<IMemory>& AudioTrack::sharedBuffer()
299{
300    return mSharedBuffer;
301}
302
303// -------------------------------------------------------------------------
304
305void AudioTrack::start()
306{
307    sp<AudioTrackThread> t = mAudioTrackThread;
308
309    LOGV("start %p", this);
310    if (t != 0) {
311        if (t->exitPending()) {
312            if (t->requestExitAndWait() == WOULD_BLOCK) {
313                LOGE("AudioTrack::start called from thread");
314                return;
315            }
316        }
317        t->mLock.lock();
318     }
319
320    if (android_atomic_or(1, &mActive) == 0) {
321        audio_io_handle_t output = AudioTrack::getOutput();
322        status_t status = mAudioTrack->start();
323        if (status == DEAD_OBJECT) {
324            LOGV("start() dead IAudioTrack: creating a new one");
325            status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
326                                 mFrameCount, mFlags, mSharedBuffer, output);
327        }
328        if (status == NO_ERROR) {
329            AudioSystem::startOutput(output, (AudioSystem::stream_type)mStreamType);
330            mNewPosition = mCblk->server + mUpdatePeriod;
331            mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
332            mCblk->waitTimeMs = 0;
333            if (t != 0) {
334               t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
335            } else {
336                setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
337            }
338        } else {
339            LOGV("start() failed");
340            android_atomic_and(~1, &mActive);
341        }
342    }
343
344    if (t != 0) {
345        t->mLock.unlock();
346    }
347}
348
349void AudioTrack::stop()
350{
351    sp<AudioTrackThread> t = mAudioTrackThread;
352
353    LOGV("stop %p", this);
354    if (t != 0) {
355        t->mLock.lock();
356    }
357
358    if (android_atomic_and(~1, &mActive) == 1) {
359        mCblk->cv.signal();
360        mAudioTrack->stop();
361        // Cancel loops (If we are in the middle of a loop, playback
362        // would not stop until loopCount reaches 0).
363        setLoop(0, 0, 0);
364        // the playback head position will reset to 0, so if a marker is set, we need
365        // to activate it again
366        mMarkerReached = false;
367        // Force flush if a shared buffer is used otherwise audioflinger
368        // will not stop before end of buffer is reached.
369        if (mSharedBuffer != 0) {
370            flush();
371        }
372        if (t != 0) {
373            t->requestExit();
374        } else {
375            setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
376        }
377        AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
378    }
379
380    if (t != 0) {
381        t->mLock.unlock();
382    }
383}
384
385bool AudioTrack::stopped() const
386{
387    return !mActive;
388}
389
390void AudioTrack::flush()
391{
392    LOGV("flush");
393
394    // clear playback marker and periodic update counter
395    mMarkerPosition = 0;
396    mMarkerReached = false;
397    mUpdatePeriod = 0;
398
399
400    if (!mActive) {
401        mAudioTrack->flush();
402        // Release AudioTrack callback thread in case it was waiting for new buffers
403        // in AudioTrack::obtainBuffer()
404        mCblk->cv.signal();
405    }
406}
407
408void AudioTrack::pause()
409{
410    LOGV("pause");
411    if (android_atomic_and(~1, &mActive) == 1) {
412        mActive = 0;
413        mAudioTrack->pause();
414        AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
415    }
416}
417
418void AudioTrack::mute(bool e)
419{
420    mAudioTrack->mute(e);
421    mMuted = e;
422}
423
424bool AudioTrack::muted() const
425{
426    return mMuted;
427}
428
429void AudioTrack::setVolume(float left, float right)
430{
431    mVolume[LEFT] = left;
432    mVolume[RIGHT] = right;
433
434    // write must be atomic
435    mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000);
436}
437
438void AudioTrack::getVolume(float* left, float* right)
439{
440    *left  = mVolume[LEFT];
441    *right = mVolume[RIGHT];
442}
443
444status_t AudioTrack::setSampleRate(int rate)
445{
446    int afSamplingRate;
447
448    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
449        return NO_INIT;
450    }
451    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
452    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
453
454    mCblk->sampleRate = rate;
455    return NO_ERROR;
456}
457
458uint32_t AudioTrack::getSampleRate()
459{
460    return mCblk->sampleRate;
461}
462
463status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
464{
465    audio_track_cblk_t* cblk = mCblk;
466
467    Mutex::Autolock _l(cblk->lock);
468
469    if (loopCount == 0) {
470        cblk->loopStart = UINT_MAX;
471        cblk->loopEnd = UINT_MAX;
472        cblk->loopCount = 0;
473        mLoopCount = 0;
474        return NO_ERROR;
475    }
476
477    if (loopStart >= loopEnd ||
478        loopEnd - loopStart > mFrameCount) {
479        LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
480        return BAD_VALUE;
481    }
482
483    if ((mSharedBuffer != 0) && (loopEnd   > mFrameCount)) {
484        LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
485            loopStart, loopEnd, mFrameCount);
486        return BAD_VALUE;
487    }
488
489    cblk->loopStart = loopStart;
490    cblk->loopEnd = loopEnd;
491    cblk->loopCount = loopCount;
492    mLoopCount = loopCount;
493
494    return NO_ERROR;
495}
496
497status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
498{
499    if (loopStart != 0) {
500        *loopStart = mCblk->loopStart;
501    }
502    if (loopEnd != 0) {
503        *loopEnd = mCblk->loopEnd;
504    }
505    if (loopCount != 0) {
506        if (mCblk->loopCount < 0) {
507            *loopCount = -1;
508        } else {
509            *loopCount = mCblk->loopCount;
510        }
511    }
512
513    return NO_ERROR;
514}
515
516status_t AudioTrack::setMarkerPosition(uint32_t marker)
517{
518    if (mCbf == 0) return INVALID_OPERATION;
519
520    mMarkerPosition = marker;
521    mMarkerReached = false;
522
523    return NO_ERROR;
524}
525
526status_t AudioTrack::getMarkerPosition(uint32_t *marker)
527{
528    if (marker == 0) return BAD_VALUE;
529
530    *marker = mMarkerPosition;
531
532    return NO_ERROR;
533}
534
535status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
536{
537    if (mCbf == 0) return INVALID_OPERATION;
538
539    uint32_t curPosition;
540    getPosition(&curPosition);
541    mNewPosition = curPosition + updatePeriod;
542    mUpdatePeriod = updatePeriod;
543
544    return NO_ERROR;
545}
546
547status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
548{
549    if (updatePeriod == 0) return BAD_VALUE;
550
551    *updatePeriod = mUpdatePeriod;
552
553    return NO_ERROR;
554}
555
556status_t AudioTrack::setPosition(uint32_t position)
557{
558    Mutex::Autolock _l(mCblk->lock);
559
560    if (!stopped()) return INVALID_OPERATION;
561
562    if (position > mCblk->user) return BAD_VALUE;
563
564    mCblk->server = position;
565    mCblk->forceReady = 1;
566
567    return NO_ERROR;
568}
569
570status_t AudioTrack::getPosition(uint32_t *position)
571{
572    if (position == 0) return BAD_VALUE;
573
574    *position = mCblk->server;
575
576    return NO_ERROR;
577}
578
579status_t AudioTrack::reload()
580{
581    if (!stopped()) return INVALID_OPERATION;
582
583    flush();
584
585    mCblk->stepUser(mFrameCount);
586
587    return NO_ERROR;
588}
589
590audio_io_handle_t AudioTrack::getOutput()
591{
592    return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
593            mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
594}
595
596// -------------------------------------------------------------------------
597
598status_t AudioTrack::createTrack(
599        int streamType,
600        uint32_t sampleRate,
601        int format,
602        int channelCount,
603        int frameCount,
604        uint32_t flags,
605        const sp<IMemory>& sharedBuffer,
606        audio_io_handle_t output)
607{
608    status_t status;
609    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
610    if (audioFlinger == 0) {
611       LOGE("Could not get audioflinger");
612       return NO_INIT;
613    }
614
615    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
616                                                      streamType,
617                                                      sampleRate,
618                                                      format,
619                                                      channelCount,
620                                                      frameCount,
621                                                      ((uint16_t)flags) << 16,
622                                                      sharedBuffer,
623                                                      output,
624                                                      &status);
625
626    if (track == 0) {
627        LOGE("AudioFlinger could not create track, status: %d", status);
628        return status;
629    }
630    sp<IMemory> cblk = track->getCblk();
631    if (cblk == 0) {
632        LOGE("Could not get control block");
633        return NO_INIT;
634    }
635    mAudioTrack.clear();
636    mAudioTrack = track;
637    mCblkMemory.clear();
638    mCblkMemory = cblk;
639    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
640    mCblk->out = 1;
641    // Update buffer size in case it has been limited by AudioFlinger during track creation
642    mFrameCount = mCblk->frameCount;
643    if (sharedBuffer == 0) {
644        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
645    } else {
646        mCblk->buffers = sharedBuffer->pointer();
647         // Force buffer full condition as data is already present in shared memory
648        mCblk->stepUser(mFrameCount);
649    }
650
651    mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
652
653    return NO_ERROR;
654}
655
656status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
657{
658    int active;
659    status_t result;
660    audio_track_cblk_t* cblk = mCblk;
661    uint32_t framesReq = audioBuffer->frameCount;
662    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
663
664    audioBuffer->frameCount  = 0;
665    audioBuffer->size = 0;
666
667    uint32_t framesAvail = cblk->framesAvailable();
668
669    if (framesAvail == 0) {
670        cblk->lock.lock();
671        goto start_loop_here;
672        while (framesAvail == 0) {
673            active = mActive;
674            if (UNLIKELY(!active)) {
675                LOGV("Not active and NO_MORE_BUFFERS");
676                cblk->lock.unlock();
677                return NO_MORE_BUFFERS;
678            }
679            if (UNLIKELY(!waitCount)) {
680                cblk->lock.unlock();
681                return WOULD_BLOCK;
682            }
683
684            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
685            if (__builtin_expect(result!=NO_ERROR, false)) {
686                cblk->waitTimeMs += waitTimeMs;
687                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
688                    // timing out when a loop has been set and we have already written upto loop end
689                    // is a normal condition: no need to wake AudioFlinger up.
690                    if (cblk->user < cblk->loopEnd) {
691                        LOGW(   "obtainBuffer timed out (is the CPU pegged?) %p "
692                                "user=%08x, server=%08x", this, cblk->user, cblk->server);
693                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
694                        cblk->lock.unlock();
695                        result = mAudioTrack->start();
696                        if (result == DEAD_OBJECT) {
697                            LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
698                            result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
699                                                 mFrameCount, mFlags, mSharedBuffer, getOutput());
700                            if (result == NO_ERROR) {
701                                cblk = mCblk;
702                                cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
703                            }
704                        }
705                        cblk->lock.lock();
706                    }
707                    cblk->waitTimeMs = 0;
708                }
709
710                if (--waitCount == 0) {
711                    cblk->lock.unlock();
712                    return TIMED_OUT;
713                }
714            }
715            // read the server count again
716        start_loop_here:
717            framesAvail = cblk->framesAvailable_l();
718        }
719        cblk->lock.unlock();
720    }
721
722    cblk->waitTimeMs = 0;
723
724    if (framesReq > framesAvail) {
725        framesReq = framesAvail;
726    }
727
728    uint32_t u = cblk->user;
729    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
730
731    if (u + framesReq > bufferEnd) {
732        framesReq = bufferEnd - u;
733    }
734
735    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
736    audioBuffer->channelCount = mChannelCount;
737    audioBuffer->frameCount = framesReq;
738    audioBuffer->size = framesReq * cblk->frameSize;
739    if (AudioSystem::isLinearPCM(mFormat)) {
740        audioBuffer->format = AudioSystem::PCM_16_BIT;
741    } else {
742        audioBuffer->format = mFormat;
743    }
744    audioBuffer->raw = (int8_t *)cblk->buffer(u);
745    active = mActive;
746    return active ? status_t(NO_ERROR) : status_t(STOPPED);
747}
748
749void AudioTrack::releaseBuffer(Buffer* audioBuffer)
750{
751    audio_track_cblk_t* cblk = mCblk;
752    cblk->stepUser(audioBuffer->frameCount);
753}
754
755// -------------------------------------------------------------------------
756
757ssize_t AudioTrack::write(const void* buffer, size_t userSize)
758{
759
760    if (mSharedBuffer != 0) return INVALID_OPERATION;
761
762    if (ssize_t(userSize) < 0) {
763        // sanity-check. user is most-likely passing an error code.
764        LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
765                buffer, userSize, userSize);
766        return BAD_VALUE;
767    }
768
769    LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
770
771    ssize_t written = 0;
772    const int8_t *src = (const int8_t *)buffer;
773    Buffer audioBuffer;
774
775    do {
776        audioBuffer.frameCount = userSize/frameSize();
777
778        // Calling obtainBuffer() with a negative wait count causes
779        // an (almost) infinite wait time.
780        status_t err = obtainBuffer(&audioBuffer, -1);
781        if (err < 0) {
782            // out of buffers, return #bytes written
783            if (err == status_t(NO_MORE_BUFFERS))
784                break;
785            return ssize_t(err);
786        }
787
788        size_t toWrite;
789
790        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
791            // Divide capacity by 2 to take expansion into account
792            toWrite = audioBuffer.size>>1;
793            // 8 to 16 bit conversion
794            int count = toWrite;
795            int16_t *dst = (int16_t *)(audioBuffer.i8);
796            while(count--) {
797                *dst++ = (int16_t)(*src++^0x80) << 8;
798            }
799        } else {
800            toWrite = audioBuffer.size;
801            memcpy(audioBuffer.i8, src, toWrite);
802            src += toWrite;
803        }
804        userSize -= toWrite;
805        written += toWrite;
806
807        releaseBuffer(&audioBuffer);
808    } while (userSize);
809
810    return written;
811}
812
813// -------------------------------------------------------------------------
814
815bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
816{
817    Buffer audioBuffer;
818    uint32_t frames;
819    size_t writtenSize;
820
821    // Manage underrun callback
822    if (mActive && (mCblk->framesReady() == 0)) {
823        LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
824        if (mCblk->flowControlFlag == 0) {
825            mCbf(EVENT_UNDERRUN, mUserData, 0);
826            if (mCblk->server == mCblk->frameCount) {
827                mCbf(EVENT_BUFFER_END, mUserData, 0);
828            }
829            mCblk->flowControlFlag = 1;
830            if (mSharedBuffer != 0) return false;
831        }
832    }
833
834    // Manage loop end callback
835    while (mLoopCount > mCblk->loopCount) {
836        int loopCount = -1;
837        mLoopCount--;
838        if (mLoopCount >= 0) loopCount = mLoopCount;
839
840        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
841    }
842
843    // Manage marker callback
844    if (!mMarkerReached && (mMarkerPosition > 0)) {
845        if (mCblk->server >= mMarkerPosition) {
846            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
847            mMarkerReached = true;
848        }
849    }
850
851    // Manage new position callback
852    if (mUpdatePeriod > 0) {
853        while (mCblk->server >= mNewPosition) {
854            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
855            mNewPosition += mUpdatePeriod;
856        }
857    }
858
859    // If Shared buffer is used, no data is requested from client.
860    if (mSharedBuffer != 0) {
861        frames = 0;
862    } else {
863        frames = mRemainingFrames;
864    }
865
866    do {
867
868        audioBuffer.frameCount = frames;
869
870        // Calling obtainBuffer() with a wait count of 1
871        // limits wait time to WAIT_PERIOD_MS. This prevents from being
872        // stuck here not being able to handle timed events (position, markers, loops).
873        status_t err = obtainBuffer(&audioBuffer, 1);
874        if (err < NO_ERROR) {
875            if (err != TIMED_OUT) {
876                LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
877                return false;
878            }
879            break;
880        }
881        if (err == status_t(STOPPED)) return false;
882
883        // Divide buffer size by 2 to take into account the expansion
884        // due to 8 to 16 bit conversion: the callback must fill only half
885        // of the destination buffer
886        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
887            audioBuffer.size >>= 1;
888        }
889
890        size_t reqSize = audioBuffer.size;
891        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
892        writtenSize = audioBuffer.size;
893
894        // Sanity check on returned size
895        if (ssize_t(writtenSize) <= 0) {
896            // The callback is done filling buffers
897            // Keep this thread going to handle timed events and
898            // still try to get more data in intervals of WAIT_PERIOD_MS
899            // but don't just loop and block the CPU, so wait
900            usleep(WAIT_PERIOD_MS*1000);
901            break;
902        }
903        if (writtenSize > reqSize) writtenSize = reqSize;
904
905        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
906            // 8 to 16 bit conversion
907            const int8_t *src = audioBuffer.i8 + writtenSize-1;
908            int count = writtenSize;
909            int16_t *dst = audioBuffer.i16 + writtenSize-1;
910            while(count--) {
911                *dst-- = (int16_t)(*src--^0x80) << 8;
912            }
913            writtenSize <<= 1;
914        }
915
916        audioBuffer.size = writtenSize;
917        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
918        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sampel size of
919        // 16 bit.
920        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
921
922        frames -= audioBuffer.frameCount;
923
924        releaseBuffer(&audioBuffer);
925    }
926    while (frames);
927
928    if (frames == 0) {
929        mRemainingFrames = mNotificationFrames;
930    } else {
931        mRemainingFrames = frames;
932    }
933    return true;
934}
935
936status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
937{
938
939    const size_t SIZE = 256;
940    char buffer[SIZE];
941    String8 result;
942
943    result.append(" AudioTrack::dump\n");
944    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
945    result.append(buffer);
946    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
947    result.append(buffer);
948    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
949    result.append(buffer);
950    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
951    result.append(buffer);
952    ::write(fd, result.string(), result.size());
953    return NO_ERROR;
954}
955
956// =========================================================================
957
958AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
959    : Thread(bCanCallJava), mReceiver(receiver)
960{
961}
962
963bool AudioTrack::AudioTrackThread::threadLoop()
964{
965    return mReceiver.processAudioBuffer(this);
966}
967
968status_t AudioTrack::AudioTrackThread::readyToRun()
969{
970    return NO_ERROR;
971}
972
973void AudioTrack::AudioTrackThread::onFirstRef()
974{
975}
976
977// =========================================================================
978
979audio_track_cblk_t::audio_track_cblk_t()
980    : lock(Mutex::SHARED), user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0),
981    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0)
982{
983}
984
985uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
986{
987    uint32_t u = this->user;
988
989    u += frameCount;
990    // Ensure that user is never ahead of server for AudioRecord
991    if (out) {
992        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
993        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
994            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
995        }
996    } else if (u > this->server) {
997        LOGW("stepServer occured after track reset");
998        u = this->server;
999    }
1000
1001    if (u >= userBase + this->frameCount) {
1002        userBase += this->frameCount;
1003    }
1004
1005    this->user = u;
1006
1007    // Clear flow control error condition as new data has been written/read to/from buffer.
1008    flowControlFlag = 0;
1009
1010    return u;
1011}
1012
1013bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1014{
1015    // the code below simulates lock-with-timeout
1016    // we MUST do this to protect the AudioFlinger server
1017    // as this lock is shared with the client.
1018    status_t err;
1019
1020    err = lock.tryLock();
1021    if (err == -EBUSY) { // just wait a bit
1022        usleep(1000);
1023        err = lock.tryLock();
1024    }
1025    if (err != NO_ERROR) {
1026        // probably, the client just died.
1027        return false;
1028    }
1029
1030    uint32_t s = this->server;
1031
1032    s += frameCount;
1033    if (out) {
1034        // Mark that we have read the first buffer so that next time stepUser() is called
1035        // we switch to normal obtainBuffer() timeout period
1036        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1037            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1038        }
1039        // It is possible that we receive a flush()
1040        // while the mixer is processing a block: in this case,
1041        // stepServer() is called After the flush() has reset u & s and
1042        // we have s > u
1043        if (s > this->user) {
1044            LOGW("stepServer occured after track reset");
1045            s = this->user;
1046        }
1047    }
1048
1049    if (s >= loopEnd) {
1050        LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1051        s = loopStart;
1052        if (--loopCount == 0) {
1053            loopEnd = UINT_MAX;
1054            loopStart = UINT_MAX;
1055        }
1056    }
1057    if (s >= serverBase + this->frameCount) {
1058        serverBase += this->frameCount;
1059    }
1060
1061    this->server = s;
1062
1063    cv.signal();
1064    lock.unlock();
1065    return true;
1066}
1067
1068void* audio_track_cblk_t::buffer(uint32_t offset) const
1069{
1070    return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
1071}
1072
1073uint32_t audio_track_cblk_t::framesAvailable()
1074{
1075    Mutex::Autolock _l(lock);
1076    return framesAvailable_l();
1077}
1078
1079uint32_t audio_track_cblk_t::framesAvailable_l()
1080{
1081    uint32_t u = this->user;
1082    uint32_t s = this->server;
1083
1084    if (out) {
1085        uint32_t limit = (s < loopStart) ? s : loopStart;
1086        return limit + frameCount - u;
1087    } else {
1088        return frameCount + u - s;
1089    }
1090}
1091
1092uint32_t audio_track_cblk_t::framesReady()
1093{
1094    uint32_t u = this->user;
1095    uint32_t s = this->server;
1096
1097    if (out) {
1098        if (u < loopEnd) {
1099            return u - s;
1100        } else {
1101            Mutex::Autolock _l(lock);
1102            if (loopCount >= 0) {
1103                return (loopEnd - loopStart)*loopCount + u - s;
1104            } else {
1105                return UINT_MAX;
1106            }
1107        }
1108    } else {
1109        return s - u;
1110    }
1111}
1112
1113// -------------------------------------------------------------------------
1114
1115}; // namespace android
1116
1117