AudioTrack.cpp revision 867d2f6ce668968e463eb86b856d21525f12fd67 
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/Parcel.h>
36#include <binder/IPCThreadState.h>
37#include <utils/Timers.h>
38#include <cutils/atomic.h>
39
40#define LIKELY( exp )       (__builtin_expect( (exp) != 0, true  ))
41#define UNLIKELY( exp )     (__builtin_expect( (exp) != 0, false ))
42
43namespace android {
44
45// ---------------------------------------------------------------------------
46
47AudioTrack::AudioTrack()
48    : mStatus(NO_INIT)
49{
50}
51
52AudioTrack::AudioTrack(
53        int streamType,
54        uint32_t sampleRate,
55        int format,
56        int channels,
57        int frameCount,
58        uint32_t flags,
59        callback_t cbf,
60        void* user,
61        int notificationFrames)
62    : mStatus(NO_INIT)
63{
64    mStatus = set(streamType, sampleRate, format, channels,
65            frameCount, flags, cbf, user, notificationFrames, 0);
66}
67
68AudioTrack::AudioTrack(
69        int streamType,
70        uint32_t sampleRate,
71        int format,
72        int channels,
73        const sp<IMemory>& sharedBuffer,
74        uint32_t flags,
75        callback_t cbf,
76        void* user,
77        int notificationFrames)
78    : mStatus(NO_INIT)
79{
80    mStatus = set(streamType, sampleRate, format, channels,
81            0, flags, cbf, user, notificationFrames, sharedBuffer);
82}
83
84AudioTrack::~AudioTrack()
85{
86    LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
87
88    if (mStatus == NO_ERROR) {
89        // Make sure that callback function exits in the case where
90        // it is looping on buffer full condition in obtainBuffer().
91        // Otherwise the callback thread will never exit.
92        stop();
93        if (mAudioTrackThread != 0) {
94            mAudioTrackThread->requestExitAndWait();
95            mAudioTrackThread.clear();
96        }
97        mAudioTrack.clear();
98        IPCThreadState::self()->flushCommands();
99    }
100}
101
102status_t AudioTrack::set(
103        int streamType,
104        uint32_t sampleRate,
105        int format,
106        int channels,
107        int frameCount,
108        uint32_t flags,
109        callback_t cbf,
110        void* user,
111        int notificationFrames,
112        const sp<IMemory>& sharedBuffer,
113        bool threadCanCallJava)
114{
115
116    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
117
118    if (mAudioTrack != 0) {
119        LOGE("Track already in use");
120        return INVALID_OPERATION;
121    }
122
123    int afSampleRate;
124    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
125        return NO_INIT;
126    }
127    int afFrameCount;
128    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
129        return NO_INIT;
130    }
131    uint32_t afLatency;
132    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
133        return NO_INIT;
134    }
135
136    // handle default values first.
137    if (streamType == AudioSystem::DEFAULT) {
138        streamType = AudioSystem::MUSIC;
139    }
140    if (sampleRate == 0) {
141        sampleRate = afSampleRate;
142    }
143    // these below should probably come from the audioFlinger too...
144    if (format == 0) {
145        format = AudioSystem::PCM_16_BIT;
146    }
147    if (channels == 0) {
148        channels = AudioSystem::CHANNEL_OUT_STEREO;
149    }
150
151    // validate parameters
152    if (!AudioSystem::isValidFormat(format)) {
153        LOGE("Invalid format");
154        return BAD_VALUE;
155    }
156
157    // force direct flag if format is not linear PCM
158    if (!AudioSystem::isLinearPCM(format)) {
159        flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
160    }
161
162    if (!AudioSystem::isOutputChannel(channels)) {
163        LOGE("Invalid channel mask");
164        return BAD_VALUE;
165    }
166    uint32_t channelCount = AudioSystem::popCount(channels);
167
168    audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
169            sampleRate, format, channels, (AudioSystem::output_flags)flags);
170
171    if (output == 0) {
172        LOGE("Could not get audio output for stream type %d", streamType);
173        return BAD_VALUE;
174    }
175
176    if (!AudioSystem::isLinearPCM(format)) {
177        if (sharedBuffer != 0) {
178            frameCount = sharedBuffer->size();
179        }
180    } else {
181        // Ensure that buffer depth covers at least audio hardware latency
182        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
183        if (minBufCount < 2) minBufCount = 2;
184
185        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
186
187        if (sharedBuffer == 0) {
188            if (frameCount == 0) {
189                frameCount = minFrameCount;
190            }
191            if (notificationFrames == 0) {
192                notificationFrames = frameCount/2;
193            }
194            // Make sure that application is notified with sufficient margin
195            // before underrun
196            if (notificationFrames > frameCount/2) {
197                notificationFrames = frameCount/2;
198            }
199            if (frameCount < minFrameCount) {
200              LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
201              return BAD_VALUE;
202            }
203        } else {
204            // Ensure that buffer alignment matches channelcount
205            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
206                LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
207                return BAD_VALUE;
208            }
209            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
210        }
211    }
212
213    mVolume[LEFT] = 1.0f;
214    mVolume[RIGHT] = 1.0f;
215    // create the IAudioTrack
216    status_t status = createTrack(streamType, sampleRate, format, channelCount,
217                                  frameCount, flags, sharedBuffer, output);
218
219    if (status != NO_ERROR) {
220        return status;
221    }
222
223    if (cbf != 0) {
224        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
225        if (mAudioTrackThread == 0) {
226          LOGE("Could not create callback thread");
227          return NO_INIT;
228        }
229    }
230
231    mStatus = NO_ERROR;
232
233    mStreamType = streamType;
234    mFormat = format;
235    mChannels = channels;
236    mChannelCount = channelCount;
237    mSharedBuffer = sharedBuffer;
238    mMuted = false;
239    mActive = 0;
240    mCbf = cbf;
241    mNotificationFrames = notificationFrames;
242    mRemainingFrames = notificationFrames;
243    mUserData = user;
244    mLatency = afLatency + (1000*mFrameCount) / sampleRate;
245    mLoopCount = 0;
246    mMarkerPosition = 0;
247    mMarkerReached = false;
248    mNewPosition = 0;
249    mUpdatePeriod = 0;
250    mFlags = flags;
251
252    return NO_ERROR;
253}
254
255status_t AudioTrack::initCheck() const
256{
257    return mStatus;
258}
259
260// -------------------------------------------------------------------------
261
262uint32_t AudioTrack::latency() const
263{
264    return mLatency;
265}
266
267int AudioTrack::streamType() const
268{
269    return mStreamType;
270}
271
272int AudioTrack::format() const
273{
274    return mFormat;
275}
276
277int AudioTrack::channelCount() const
278{
279    return mChannelCount;
280}
281
282uint32_t AudioTrack::frameCount() const
283{
284    return mFrameCount;
285}
286
287int AudioTrack::frameSize() const
288{
289    if (AudioSystem::isLinearPCM(mFormat)) {
290        return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
291    } else {
292        return sizeof(uint8_t);
293    }
294}
295
296sp<IMemory>& AudioTrack::sharedBuffer()
297{
298    return mSharedBuffer;
299}
300
301// -------------------------------------------------------------------------
302
303void AudioTrack::start()
304{
305    sp<AudioTrackThread> t = mAudioTrackThread;
306
307    LOGV("start %p", this);
308    if (t != 0) {
309        if (t->exitPending()) {
310            if (t->requestExitAndWait() == WOULD_BLOCK) {
311                LOGE("AudioTrack::start called from thread");
312                return;
313            }
314        }
315        t->mLock.lock();
316     }
317
318    if (android_atomic_or(1, &mActive) == 0) {
319        mNewPosition = mCblk->server + mUpdatePeriod;
320        mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
321        mCblk->waitTimeMs = 0;
322        if (t != 0) {
323           t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
324        } else {
325            setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
326        }
327
328        status_t status = mAudioTrack->start();
329        if (status == DEAD_OBJECT) {
330            LOGV("start() dead IAudioTrack: creating a new one");
331            status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
332                                 mFrameCount, mFlags, mSharedBuffer, getOutput());
333            if (status == NO_ERROR) {
334                status = mAudioTrack->start();
335                if (status == NO_ERROR) {
336                    mNewPosition = mCblk->server + mUpdatePeriod;
337                }
338            }
339        }
340        if (status != NO_ERROR) {
341            LOGV("start() failed");
342            android_atomic_and(~1, &mActive);
343            if (t != 0) {
344                t->requestExit();
345            } else {
346                setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
347            }
348        }
349    }
350
351    if (t != 0) {
352        t->mLock.unlock();
353    }
354}
355
356void AudioTrack::stop()
357{
358    sp<AudioTrackThread> t = mAudioTrackThread;
359
360    LOGV("stop %p", this);
361    if (t != 0) {
362        t->mLock.lock();
363    }
364
365    if (android_atomic_and(~1, &mActive) == 1) {
366        mCblk->cv.signal();
367        mAudioTrack->stop();
368        // Cancel loops (If we are in the middle of a loop, playback
369        // would not stop until loopCount reaches 0).
370        setLoop(0, 0, 0);
371        // the playback head position will reset to 0, so if a marker is set, we need
372        // to activate it again
373        mMarkerReached = false;
374        // Force flush if a shared buffer is used otherwise audioflinger
375        // will not stop before end of buffer is reached.
376        if (mSharedBuffer != 0) {
377            flush();
378        }
379        if (t != 0) {
380            t->requestExit();
381        } else {
382            setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
383        }
384    }
385
386    if (t != 0) {
387        t->mLock.unlock();
388    }
389}
390
391bool AudioTrack::stopped() const
392{
393    return !mActive;
394}
395
396void AudioTrack::flush()
397{
398    LOGV("flush");
399
400    // clear playback marker and periodic update counter
401    mMarkerPosition = 0;
402    mMarkerReached = false;
403    mUpdatePeriod = 0;
404
405
406    if (!mActive) {
407        mAudioTrack->flush();
408        // Release AudioTrack callback thread in case it was waiting for new buffers
409        // in AudioTrack::obtainBuffer()
410        mCblk->cv.signal();
411    }
412}
413
414void AudioTrack::pause()
415{
416    LOGV("pause");
417    if (android_atomic_and(~1, &mActive) == 1) {
418        mAudioTrack->pause();
419    }
420}
421
422void AudioTrack::mute(bool e)
423{
424    mAudioTrack->mute(e);
425    mMuted = e;
426}
427
428bool AudioTrack::muted() const
429{
430    return mMuted;
431}
432
433void AudioTrack::setVolume(float left, float right)
434{
435    mVolume[LEFT] = left;
436    mVolume[RIGHT] = right;
437
438    // write must be atomic
439    mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000);
440}
441
442void AudioTrack::getVolume(float* left, float* right)
443{
444    *left  = mVolume[LEFT];
445    *right = mVolume[RIGHT];
446}
447
448status_t AudioTrack::setSampleRate(int rate)
449{
450    int afSamplingRate;
451
452    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
453        return NO_INIT;
454    }
455    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
456    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
457
458    mCblk->sampleRate = rate;
459    return NO_ERROR;
460}
461
462uint32_t AudioTrack::getSampleRate()
463{
464    return mCblk->sampleRate;
465}
466
467status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
468{
469    audio_track_cblk_t* cblk = mCblk;
470
471    Mutex::Autolock _l(cblk->lock);
472
473    if (loopCount == 0) {
474        cblk->loopStart = UINT_MAX;
475        cblk->loopEnd = UINT_MAX;
476        cblk->loopCount = 0;
477        mLoopCount = 0;
478        return NO_ERROR;
479    }
480
481    if (loopStart >= loopEnd ||
482        loopEnd - loopStart > mFrameCount) {
483        LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
484        return BAD_VALUE;
485    }
486
487    if ((mSharedBuffer != 0) && (loopEnd   > mFrameCount)) {
488        LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
489            loopStart, loopEnd, mFrameCount);
490        return BAD_VALUE;
491    }
492
493    cblk->loopStart = loopStart;
494    cblk->loopEnd = loopEnd;
495    cblk->loopCount = loopCount;
496    mLoopCount = loopCount;
497
498    return NO_ERROR;
499}
500
501status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
502{
503    if (loopStart != 0) {
504        *loopStart = mCblk->loopStart;
505    }
506    if (loopEnd != 0) {
507        *loopEnd = mCblk->loopEnd;
508    }
509    if (loopCount != 0) {
510        if (mCblk->loopCount < 0) {
511            *loopCount = -1;
512        } else {
513            *loopCount = mCblk->loopCount;
514        }
515    }
516
517    return NO_ERROR;
518}
519
520status_t AudioTrack::setMarkerPosition(uint32_t marker)
521{
522    if (mCbf == 0) return INVALID_OPERATION;
523
524    mMarkerPosition = marker;
525    mMarkerReached = false;
526
527    return NO_ERROR;
528}
529
530status_t AudioTrack::getMarkerPosition(uint32_t *marker)
531{
532    if (marker == 0) return BAD_VALUE;
533
534    *marker = mMarkerPosition;
535
536    return NO_ERROR;
537}
538
539status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
540{
541    if (mCbf == 0) return INVALID_OPERATION;
542
543    uint32_t curPosition;
544    getPosition(&curPosition);
545    mNewPosition = curPosition + updatePeriod;
546    mUpdatePeriod = updatePeriod;
547
548    return NO_ERROR;
549}
550
551status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
552{
553    if (updatePeriod == 0) return BAD_VALUE;
554
555    *updatePeriod = mUpdatePeriod;
556
557    return NO_ERROR;
558}
559
560status_t AudioTrack::setPosition(uint32_t position)
561{
562    Mutex::Autolock _l(mCblk->lock);
563
564    if (!stopped()) return INVALID_OPERATION;
565
566    if (position > mCblk->user) return BAD_VALUE;
567
568    mCblk->server = position;
569    mCblk->forceReady = 1;
570
571    return NO_ERROR;
572}
573
574status_t AudioTrack::getPosition(uint32_t *position)
575{
576    if (position == 0) return BAD_VALUE;
577
578    *position = mCblk->server;
579
580    return NO_ERROR;
581}
582
583status_t AudioTrack::reload()
584{
585    if (!stopped()) return INVALID_OPERATION;
586
587    flush();
588
589    mCblk->stepUser(mFrameCount);
590
591    return NO_ERROR;
592}
593
594audio_io_handle_t AudioTrack::getOutput()
595{
596    return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
597            mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
598}
599
600// -------------------------------------------------------------------------
601
602status_t AudioTrack::createTrack(
603        int streamType,
604        uint32_t sampleRate,
605        int format,
606        int channelCount,
607        int frameCount,
608        uint32_t flags,
609        const sp<IMemory>& sharedBuffer,
610        audio_io_handle_t output)
611{
612    status_t status;
613    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
614    if (audioFlinger == 0) {
615       LOGE("Could not get audioflinger");
616       return NO_INIT;
617    }
618
619    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
620                                                      streamType,
621                                                      sampleRate,
622                                                      format,
623                                                      channelCount,
624                                                      frameCount,
625                                                      ((uint16_t)flags) << 16,
626                                                      sharedBuffer,
627                                                      output,
628                                                      &status);
629
630    if (track == 0) {
631        LOGE("AudioFlinger could not create track, status: %d", status);
632        return status;
633    }
634    sp<IMemory> cblk = track->getCblk();
635    if (cblk == 0) {
636        LOGE("Could not get control block");
637        return NO_INIT;
638    }
639    mAudioTrack.clear();
640    mAudioTrack = track;
641    mCblkMemory.clear();
642    mCblkMemory = cblk;
643    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
644    mCblk->out = 1;
645    // Update buffer size in case it has been limited by AudioFlinger during track creation
646    mFrameCount = mCblk->frameCount;
647    if (sharedBuffer == 0) {
648        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
649    } else {
650        mCblk->buffers = sharedBuffer->pointer();
651         // Force buffer full condition as data is already present in shared memory
652        mCblk->stepUser(mFrameCount);
653    }
654
655    mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
656    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
657    mCblk->waitTimeMs = 0;
658    return NO_ERROR;
659}
660
661status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
662{
663    int active;
664    status_t result;
665    audio_track_cblk_t* cblk = mCblk;
666    uint32_t framesReq = audioBuffer->frameCount;
667    uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
668
669    audioBuffer->frameCount  = 0;
670    audioBuffer->size = 0;
671
672    uint32_t framesAvail = cblk->framesAvailable();
673
674    if (framesAvail == 0) {
675        cblk->lock.lock();
676        goto start_loop_here;
677        while (framesAvail == 0) {
678            active = mActive;
679            if (UNLIKELY(!active)) {
680                LOGV("Not active and NO_MORE_BUFFERS");
681                cblk->lock.unlock();
682                return NO_MORE_BUFFERS;
683            }
684            if (UNLIKELY(!waitCount)) {
685                cblk->lock.unlock();
686                return WOULD_BLOCK;
687            }
688
689            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
690            if (__builtin_expect(result!=NO_ERROR, false)) {
691                cblk->waitTimeMs += waitTimeMs;
692                if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
693                    // timing out when a loop has been set and we have already written upto loop end
694                    // is a normal condition: no need to wake AudioFlinger up.
695                    if (cblk->user < cblk->loopEnd) {
696                        LOGW(   "obtainBuffer timed out (is the CPU pegged?) %p "
697                                "user=%08x, server=%08x", this, cblk->user, cblk->server);
698                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
699                        cblk->lock.unlock();
700                        result = mAudioTrack->start();
701                        if (result == DEAD_OBJECT) {
702                            LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
703                            result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
704                                                 mFrameCount, mFlags, mSharedBuffer, getOutput());
705                            if (result == NO_ERROR) {
706                                cblk = mCblk;
707                                cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
708                                mAudioTrack->start();
709                            }
710                        }
711                        cblk->lock.lock();
712                    }
713                    cblk->waitTimeMs = 0;
714                }
715
716                if (--waitCount == 0) {
717                    cblk->lock.unlock();
718                    return TIMED_OUT;
719                }
720            }
721            // read the server count again
722        start_loop_here:
723            framesAvail = cblk->framesAvailable_l();
724        }
725        cblk->lock.unlock();
726    }
727
728    cblk->waitTimeMs = 0;
729
730    if (framesReq > framesAvail) {
731        framesReq = framesAvail;
732    }
733
734    uint32_t u = cblk->user;
735    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
736
737    if (u + framesReq > bufferEnd) {
738        framesReq = bufferEnd - u;
739    }
740
741    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
742    audioBuffer->channelCount = mChannelCount;
743    audioBuffer->frameCount = framesReq;
744    audioBuffer->size = framesReq * cblk->frameSize;
745    if (AudioSystem::isLinearPCM(mFormat)) {
746        audioBuffer->format = AudioSystem::PCM_16_BIT;
747    } else {
748        audioBuffer->format = mFormat;
749    }
750    audioBuffer->raw = (int8_t *)cblk->buffer(u);
751    active = mActive;
752    return active ? status_t(NO_ERROR) : status_t(STOPPED);
753}
754
755void AudioTrack::releaseBuffer(Buffer* audioBuffer)
756{
757    audio_track_cblk_t* cblk = mCblk;
758    cblk->stepUser(audioBuffer->frameCount);
759}
760
761// -------------------------------------------------------------------------
762
763ssize_t AudioTrack::write(const void* buffer, size_t userSize)
764{
765
766    if (mSharedBuffer != 0) return INVALID_OPERATION;
767
768    if (ssize_t(userSize) < 0) {
769        // sanity-check. user is most-likely passing an error code.
770        LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
771                buffer, userSize, userSize);
772        return BAD_VALUE;
773    }
774
775    LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
776
777    ssize_t written = 0;
778    const int8_t *src = (const int8_t *)buffer;
779    Buffer audioBuffer;
780
781    do {
782        audioBuffer.frameCount = userSize/frameSize();
783
784        // Calling obtainBuffer() with a negative wait count causes
785        // an (almost) infinite wait time.
786        status_t err = obtainBuffer(&audioBuffer, -1);
787        if (err < 0) {
788            // out of buffers, return #bytes written
789            if (err == status_t(NO_MORE_BUFFERS))
790                break;
791            return ssize_t(err);
792        }
793
794        size_t toWrite;
795
796        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
797            // Divide capacity by 2 to take expansion into account
798            toWrite = audioBuffer.size>>1;
799            // 8 to 16 bit conversion
800            int count = toWrite;
801            int16_t *dst = (int16_t *)(audioBuffer.i8);
802            while(count--) {
803                *dst++ = (int16_t)(*src++^0x80) << 8;
804            }
805        } else {
806            toWrite = audioBuffer.size;
807            memcpy(audioBuffer.i8, src, toWrite);
808            src += toWrite;
809        }
810        userSize -= toWrite;
811        written += toWrite;
812
813        releaseBuffer(&audioBuffer);
814    } while (userSize);
815
816    return written;
817}
818
819// -------------------------------------------------------------------------
820
821bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
822{
823    Buffer audioBuffer;
824    uint32_t frames;
825    size_t writtenSize;
826
827    // Manage underrun callback
828    if (mActive && (mCblk->framesReady() == 0)) {
829        LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
830        if (mCblk->flowControlFlag == 0) {
831            mCbf(EVENT_UNDERRUN, mUserData, 0);
832            if (mCblk->server == mCblk->frameCount) {
833                mCbf(EVENT_BUFFER_END, mUserData, 0);
834            }
835            mCblk->flowControlFlag = 1;
836            if (mSharedBuffer != 0) return false;
837        }
838    }
839
840    // Manage loop end callback
841    while (mLoopCount > mCblk->loopCount) {
842        int loopCount = -1;
843        mLoopCount--;
844        if (mLoopCount >= 0) loopCount = mLoopCount;
845
846        mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
847    }
848
849    // Manage marker callback
850    if (!mMarkerReached && (mMarkerPosition > 0)) {
851        if (mCblk->server >= mMarkerPosition) {
852            mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
853            mMarkerReached = true;
854        }
855    }
856
857    // Manage new position callback
858    if (mUpdatePeriod > 0) {
859        while (mCblk->server >= mNewPosition) {
860            mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
861            mNewPosition += mUpdatePeriod;
862        }
863    }
864
865    // If Shared buffer is used, no data is requested from client.
866    if (mSharedBuffer != 0) {
867        frames = 0;
868    } else {
869        frames = mRemainingFrames;
870    }
871
872    do {
873
874        audioBuffer.frameCount = frames;
875
876        // Calling obtainBuffer() with a wait count of 1
877        // limits wait time to WAIT_PERIOD_MS. This prevents from being
878        // stuck here not being able to handle timed events (position, markers, loops).
879        status_t err = obtainBuffer(&audioBuffer, 1);
880        if (err < NO_ERROR) {
881            if (err != TIMED_OUT) {
882                LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
883                return false;
884            }
885            break;
886        }
887        if (err == status_t(STOPPED)) return false;
888
889        // Divide buffer size by 2 to take into account the expansion
890        // due to 8 to 16 bit conversion: the callback must fill only half
891        // of the destination buffer
892        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
893            audioBuffer.size >>= 1;
894        }
895
896        size_t reqSize = audioBuffer.size;
897        mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
898        writtenSize = audioBuffer.size;
899
900        // Sanity check on returned size
901        if (ssize_t(writtenSize) <= 0) {
902            // The callback is done filling buffers
903            // Keep this thread going to handle timed events and
904            // still try to get more data in intervals of WAIT_PERIOD_MS
905            // but don't just loop and block the CPU, so wait
906            usleep(WAIT_PERIOD_MS*1000);
907            break;
908        }
909        if (writtenSize > reqSize) writtenSize = reqSize;
910
911        if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
912            // 8 to 16 bit conversion
913            const int8_t *src = audioBuffer.i8 + writtenSize-1;
914            int count = writtenSize;
915            int16_t *dst = audioBuffer.i16 + writtenSize-1;
916            while(count--) {
917                *dst-- = (int16_t)(*src--^0x80) << 8;
918            }
919            writtenSize <<= 1;
920        }
921
922        audioBuffer.size = writtenSize;
923        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
924        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sampel size of
925        // 16 bit.
926        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
927
928        frames -= audioBuffer.frameCount;
929
930        releaseBuffer(&audioBuffer);
931    }
932    while (frames);
933
934    if (frames == 0) {
935        mRemainingFrames = mNotificationFrames;
936    } else {
937        mRemainingFrames = frames;
938    }
939    return true;
940}
941
942status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
943{
944
945    const size_t SIZE = 256;
946    char buffer[SIZE];
947    String8 result;
948
949    result.append(" AudioTrack::dump\n");
950    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
951    result.append(buffer);
952    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
953    result.append(buffer);
954    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
955    result.append(buffer);
956    snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
957    result.append(buffer);
958    ::write(fd, result.string(), result.size());
959    return NO_ERROR;
960}
961
962// =========================================================================
963
964AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
965    : Thread(bCanCallJava), mReceiver(receiver)
966{
967}
968
969bool AudioTrack::AudioTrackThread::threadLoop()
970{
971    return mReceiver.processAudioBuffer(this);
972}
973
974status_t AudioTrack::AudioTrackThread::readyToRun()
975{
976    return NO_ERROR;
977}
978
979void AudioTrack::AudioTrackThread::onFirstRef()
980{
981}
982
983// =========================================================================
984
985audio_track_cblk_t::audio_track_cblk_t()
986    : lock(Mutex::SHARED), user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0),
987    loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0)
988{
989}
990
991uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
992{
993    uint32_t u = this->user;
994
995    u += frameCount;
996    // Ensure that user is never ahead of server for AudioRecord
997    if (out) {
998        // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
999        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
1000            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
1001        }
1002    } else if (u > this->server) {
1003        LOGW("stepServer occured after track reset");
1004        u = this->server;
1005    }
1006
1007    if (u >= userBase + this->frameCount) {
1008        userBase += this->frameCount;
1009    }
1010
1011    this->user = u;
1012
1013    // Clear flow control error condition as new data has been written/read to/from buffer.
1014    flowControlFlag = 0;
1015
1016    return u;
1017}
1018
1019bool audio_track_cblk_t::stepServer(uint32_t frameCount)
1020{
1021    // the code below simulates lock-with-timeout
1022    // we MUST do this to protect the AudioFlinger server
1023    // as this lock is shared with the client.
1024    status_t err;
1025
1026    err = lock.tryLock();
1027    if (err == -EBUSY) { // just wait a bit
1028        usleep(1000);
1029        err = lock.tryLock();
1030    }
1031    if (err != NO_ERROR) {
1032        // probably, the client just died.
1033        return false;
1034    }
1035
1036    uint32_t s = this->server;
1037
1038    s += frameCount;
1039    if (out) {
1040        // Mark that we have read the first buffer so that next time stepUser() is called
1041        // we switch to normal obtainBuffer() timeout period
1042        if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
1043            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
1044        }
1045        // It is possible that we receive a flush()
1046        // while the mixer is processing a block: in this case,
1047        // stepServer() is called After the flush() has reset u & s and
1048        // we have s > u
1049        if (s > this->user) {
1050            LOGW("stepServer occured after track reset");
1051            s = this->user;
1052        }
1053    }
1054
1055    if (s >= loopEnd) {
1056        LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
1057        s = loopStart;
1058        if (--loopCount == 0) {
1059            loopEnd = UINT_MAX;
1060            loopStart = UINT_MAX;
1061        }
1062    }
1063    if (s >= serverBase + this->frameCount) {
1064        serverBase += this->frameCount;
1065    }
1066
1067    this->server = s;
1068
1069    cv.signal();
1070    lock.unlock();
1071    return true;
1072}
1073
1074void* audio_track_cblk_t::buffer(uint32_t offset) const
1075{
1076    return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
1077}
1078
1079uint32_t audio_track_cblk_t::framesAvailable()
1080{
1081    Mutex::Autolock _l(lock);
1082    return framesAvailable_l();
1083}
1084
1085uint32_t audio_track_cblk_t::framesAvailable_l()
1086{
1087    uint32_t u = this->user;
1088    uint32_t s = this->server;
1089
1090    if (out) {
1091        uint32_t limit = (s < loopStart) ? s : loopStart;
1092        return limit + frameCount - u;
1093    } else {
1094        return frameCount + u - s;
1095    }
1096}
1097
1098uint32_t audio_track_cblk_t::framesReady()
1099{
1100    uint32_t u = this->user;
1101    uint32_t s = this->server;
1102
1103    if (out) {
1104        if (u < loopEnd) {
1105            return u - s;
1106        } else {
1107            Mutex::Autolock _l(lock);
1108            if (loopCount >= 0) {
1109                return (loopEnd - loopStart)*loopCount + u - s;
1110            } else {
1111                return UINT_MAX;
1112            }
1113        }
1114    } else {
1115        return s - u;
1116    }
1117}
1118
1119// -------------------------------------------------------------------------
1120
1121}; // namespace android
1122
1123