代码收藏家 深度解析 Android 10 车载音频架构中的 AudioTrack
xref: /frameworks/base/media/java/android/media/AudioTrack.java
598 private AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
599 int mode, int sessionId, boolean offload)
600 throws IllegalArgumentException {
601 super(attributes, AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
602 // mState already == STATE_UNINITIALIZED
603
604 mConfiguredAudioAttributes = attributes; // object copy not needed, immutable.
605
606 if (format == null) {
607 throw new IllegalArgumentException("Illegal null AudioFormat");
608 }
609
610 // Check if we should enable deep buffer mode
611 if (shouldEnablePowerSaving(mAttributes, format, bufferSizeInBytes, mode)) {
612 mAttributes = new AudioAttributes.Builder(mAttributes)
613 .replaceFlags((mAttributes.getAllFlags()
614 | AudioAttributes.FLAG_DEEP_BUFFER)
615 & ~AudioAttributes.FLAG_LOW_LATENCY)
616 .build();
617 }
618
619 // remember which looper is associated with the AudioTrack instantiation
620 Looper looper;
621 if ((looper = Looper.myLooper()) == null) {
622 looper = Looper.getMainLooper();
623 }
624
625 int rate = format.getSampleRate();
626 if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
627 rate = 0;
628 }
629
630 int channelIndexMask = 0;
631 if ((format.getPropertySetMask()
632 & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {
633 channelIndexMask = format.getChannelIndexMask();
634 }
635 int channelMask = 0;
636 if ((format.getPropertySetMask()
637 & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {
638 channelMask = format.getChannelMask();
639 } else if (channelIndexMask == 0) { // if no masks at all, use stereo
640 channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT
641 | AudioFormat.CHANNEL_OUT_FRONT_RIGHT;
642 }
643 int encoding = AudioFormat.ENCODING_DEFAULT;
644 if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {
645 encoding = format.getEncoding();
646 }
647 audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode);
648 mOffloaded = offload;
649 mStreamType = AudioSystem.STREAM_DEFAULT;
650
651 audioBuffSizeCheck(bufferSizeInBytes);
652
653 mInitializationLooper = looper;
654
655 if (sessionId < 0) {
656 throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
657 }
658
659 int[] sampleRate = new int[] {mSampleRate};
660 int[] session = new int[1];
661 session[0] = sessionId;
662 // native initialization
663 int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes,
664 sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,
665 mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/,
666 offload);
667 if (initResult != SUCCESS) {
668 loge("Error code "+initResult+" when initializing AudioTrack.");
669 return; // with mState == STATE_UNINITIALIZED
670 }
671
672 mSampleRate = sampleRate[0];
673 mSessionId = session[0];
674
675 if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) != 0) {
676 int frameSizeInBytes;
677 if (AudioFormat.isEncodingLinearFrames(mAudioFormat)) {
678 frameSizeInBytes = mChannelCount * AudioFormat.getBytesPerSample(mAudioFormat);
679 } else {
680 frameSizeInBytes = 1;
681 }
682 mOffset = ((int) Math.ceil(HEADER_V2_SIZE_BYTES / frameSizeInBytes)) * frameSizeInBytes;
683 }
684
685 if (mDataLoadMode == MODE_STATIC) {
686 mState = STATE_NO_STATIC_DATA;
687 } else {
688 mState = STATE_INITIALIZED;
689 }
690
691 baseRegisterPlayer();
692 }
693
694 /**
695 * A constructor which explicitly connects a Native (C++) AudioTrack. For use by
696 * the AudioTrackRoutingProxy subclass.
697 * @param nativeTrackInJavaObj a C/C++ pointer to a native AudioTrack
698 * (associated with an OpenSL ES player).
699 * IMPORTANT: For "N", this method is ONLY called to setup a Java routing proxy,
700 * i.e. IAndroidConfiguration::AcquireJavaProxy(). If we call with a 0 in nativeTrackInJavaObj
701 * it means that the OpenSL player interface hasn't been realized, so there is no native
702 * Audiotrack to connect to. In this case wait to call deferred_connect() until the
703 * OpenSLES interface is realized.
704 */
705 /*package*/ AudioTrack(long nativeTrackInJavaObj) {
706 super(new AudioAttributes.Builder().build(),
707 AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
708 // "final"s
709 mNativeTrackInJavaObj = 0;
710 mJniData = 0;
711
712 // remember which looper is associated with the AudioTrack instantiation
713 Looper looper;
714 if ((looper = Looper.myLooper()) == null) {
715 looper = Looper.getMainLooper();
716 }
717 mInitializationLooper = looper;
718
719 // other initialization...
720 if (nativeTrackInJavaObj != 0) {
721 baseRegisterPlayer();
722 deferred_connect(nativeTrackInJavaObj);
723 } else {
724 mState = STATE_UNINITIALIZED;
725 }
726 }xref: /frameworks/base/core/jni/android_media_AudioTrack.cpp
216 static jint
217 android_media_AudioTrack_setup(JNIEnv *env, jobject thiz, jobject weak_this, jobject jaa,
218 jintArray jSampleRate, jint channelPositionMask, jint channelIndexMask,
219 jint audioFormat, jint buffSizeInBytes, jint memoryMode, jintArray jSession,
220 jlong nativeAudioTrack, jboolean offload) {
221
222 ALOGV("sampleRates=%p, channel mask=%x, index mask=%x, audioFormat(Java)=%d, buffSize=%d,"
223 " nativeAudioTrack=0x%" PRIX64 ", offload=%d",
224 jSampleRate, channelPositionMask, channelIndexMask, audioFormat, buffSizeInBytes,
225 nativeAudioTrack, offload);
226
227 sp<AudioTrack> lpTrack = 0;
228
229 if (jSession == NULL) {
230 ALOGE("Error creating AudioTrack: invalid session ID pointer");
231 return (jint) AUDIO_JAVA_ERROR;
232 }
233
234 jint* nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
235 if (nSession == NULL) {
236 ALOGE("Error creating AudioTrack: Error retrieving session id pointer");
237 return (jint) AUDIO_JAVA_ERROR;
238 }
239 audio_session_t sessionId = (audio_session_t) nSession[0];
240 env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
241 nSession = NULL;
242
243 AudioTrackJniStorage* lpJniStorage = NULL;
244
245 jclass clazz = env->GetObjectClass(thiz);
246 if (clazz == NULL) {
247 ALOGE("Can't find %s when setting up callback.", kClassPathName);
248 return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;
249 }
250
251 // if we pass in an existing *Native* AudioTrack, we don't need to create/initialize one.
252 if (nativeAudioTrack == 0) {
253 if (jaa == 0) {
254 ALOGE("Error creating AudioTrack: invalid audio attributes");
255 return (jint) AUDIO_JAVA_ERROR;
256 }
257
258 if (jSampleRate == 0) {
259 ALOGE("Error creating AudioTrack: invalid sample rates");
260 return (jint) AUDIO_JAVA_ERROR;
261 }
262
263 int* sampleRates = env->GetIntArrayElements(jSampleRate, NULL);
264 int sampleRateInHertz = sampleRates[0];
265 env->ReleaseIntArrayElements(jSampleRate, sampleRates, JNI_ABORT);
266
267 // Invalid channel representations are caught by !audio_is_output_channel() below.
268 audio_channel_mask_t nativeChannelMask = nativeChannelMaskFromJavaChannelMasks(
269 channelPositionMask, channelIndexMask);
270 if (!audio_is_output_channel(nativeChannelMask)) {
271 ALOGE("Error creating AudioTrack: invalid native channel mask %#x.", nativeChannelMask);
272 return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK;
273 }
274
275 uint32_t channelCount = audio_channel_count_from_out_mask(nativeChannelMask);
276
277 // check the format.
278 // This function was called from Java, so we compare the format against the Java constants
279 audio_format_t format = audioFormatToNative(audioFormat);
280 if (format == AUDIO_FORMAT_INVALID) {
281 ALOGE("Error creating AudioTrack: unsupported audio format %d.", audioFormat);
282 return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT;
283 }
284
285 // compute the frame count
286 size_t frameCount;
287 if (audio_has_proportional_frames(format)) {
288 const size_t bytesPerSample = audio_bytes_per_sample(format);
289 frameCount = buffSizeInBytes / (channelCount * bytesPerSample);
290 } else {
291 frameCount = buffSizeInBytes;
292 }
293
294 // create the native AudioTrack object
295 lpTrack = new AudioTrack();
296
297 // read the AudioAttributes values
298 auto paa = JNIAudioAttributeHelper::makeUnique();
299 jint jStatus = JNIAudioAttributeHelper::nativeFromJava(env, jaa, paa.get());
300 if (jStatus != (jint)AUDIO_JAVA_SUCCESS) {
301 return jStatus;
302 }
303 ALOGV("AudioTrack_setup for usage=%d content=%d flags=0x%#x tags=%s",
304 paa->usage, paa->content_type, paa->flags, paa->tags);
305
306 // initialize the callback information:
307 // this data will be passed with every AudioTrack callback
308 lpJniStorage = new AudioTrackJniStorage();
309 lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);
310 // we use a weak reference so the AudioTrack object can be garbage collected.
311 lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);
312 lpJniStorage->mCallbackData.isOffload = offload;
313 lpJniStorage->mCallbackData.busy = false;
314
315 audio_offload_info_t offloadInfo;
316 if (offload == JNI_TRUE) {
317 offloadInfo = AUDIO_INFO_INITIALIZER;
318 offloadInfo.format = format;
319 offloadInfo.sample_rate = sampleRateInHertz;
320 offloadInfo.channel_mask = nativeChannelMask;
321 offloadInfo.has_video = false;
322 offloadInfo.stream_type = AUDIO_STREAM_MUSIC; //required for offload
323 }
324
325 // initialize the native AudioTrack object
326 status_t status = NO_ERROR;
327 switch (memoryMode) {
328 case MODE_STREAM:
329 status = lpTrack->set(
330 AUDIO_STREAM_DEFAULT,// stream type, but more info conveyed in paa (last argument)
331 sampleRateInHertz,
332 format,// word length, PCM
333 nativeChannelMask,
334 offload ? 0 : frameCount,
335 offload ? AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_NONE,
336 audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user)
337 0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack
338 0,// shared mem
339 true,// thread can call Java
340 sessionId,// audio session ID
341 offload ? AudioTrack::TRANSFER_SYNC_NOTIF_CALLBACK : AudioTrack::TRANSFER_SYNC,
342 offload ? &offloadInfo : NULL,
343 -1, -1, // default uid, pid values
344 paa.get());
345
346 break;
347
348 case MODE_STATIC:
349 // AudioTrack is using shared memory
350
351 if (!lpJniStorage->allocSharedMem(buffSizeInBytes)) {
352 ALOGE("Error creating AudioTrack in static mode: error creating mem heap base");
353 goto native_init_failure;
354 }
355
356 status = lpTrack->set(
357 AUDIO_STREAM_DEFAULT,// stream type, but more info conveyed in paa (last argument)
358 sampleRateInHertz,
359 format,// word length, PCM
360 nativeChannelMask,
361 frameCount,
362 AUDIO_OUTPUT_FLAG_NONE,
363 audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user));
364 0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack
365 lpJniStorage->mMemBase,// shared mem
366 true,// thread can call Java
367 sessionId,// audio session ID
368 AudioTrack::TRANSFER_SHARED,
369 NULL, // default offloadInfo
370 -1, -1, // default uid, pid values
371 paa.get());
372 break;
373
374 default:
375 ALOGE("Unknown mode %d", memoryMode);
376 goto native_init_failure;
377 }
378
379 if (status != NO_ERROR) {
380 ALOGE("Error %d initializing AudioTrack", status);
381 goto native_init_failure;
382 }
383 } else { // end if (nativeAudioTrack == 0)
384 lpTrack = (AudioTrack*)nativeAudioTrack;
385 // TODO: We need to find out which members of the Java AudioTrack might
386 // need to be initialized from the Native AudioTrack
387 // these are directly returned from getters:
388 // mSampleRate
389 // mAudioFormat
390 // mStreamType
391 // mChannelConfiguration
392 // mChannelCount
393 // mState (?)
394 // mPlayState (?)
395 // these may be used internally (Java AudioTrack.audioParamCheck():
396 // mChannelMask
397 // mChannelIndexMask
398 // mDataLoadMode
399
400 // initialize the callback information:
401 // this data will be passed with every AudioTrack callback
402 lpJniStorage = new AudioTrackJniStorage();
403 lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);
404 // we use a weak reference so the AudioTrack object can be garbage collected.
405 lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);
406 lpJniStorage->mCallbackData.busy = false;
407 }
408
409 nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
410 if (nSession == NULL) {
411 ALOGE("Error creating AudioTrack: Error retrieving session id pointer");
412 goto native_init_failure;
413 }
414 // read the audio session ID back from AudioTrack in case we create a new session
415 nSession[0] = lpTrack->getSessionId();
416 env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
417 nSession = NULL;
418
419 {
420 const jint elements[1] = { (jint) lpTrack->getSampleRate() };
421 env->SetIntArrayRegion(jSampleRate, 0, 1, elements);
422 }
423
424 { // scope for the lock
425 Mutex::Autolock l(sLock);
426 sAudioTrackCallBackCookies.add(&lpJniStorage->mCallbackData);
427 }
428 // save our newly created C++ AudioTrack in the "nativeTrackInJavaObj" field
429 // of the Java object (in mNativeTrackInJavaObj)
430 setAudioTrack(env, thiz, lpTrack);
431
432 // save the JNI resources so we can free them later
433 //ALOGV("storing lpJniStorage: %x\n", (long)lpJniStorage);
434 env->SetLongField(thiz, javaAudioTrackFields.jniData, (jlong)lpJniStorage);
435
436 // since we had audio attributes, the stream type was derived from them during the
437 // creation of the native AudioTrack: push the same value to the Java object
438 env->SetIntField(thiz, javaAudioTrackFields.fieldStreamType, (jint) lpTrack->streamType());
439
440 return (jint) AUDIO_JAVA_SUCCESS;
441
442 // failures:
443 native_init_failure:
444 if (nSession != NULL) {
445 env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
446 }
447 env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_class);
448 env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_ref);
449 delete lpJniStorage;
450 env->SetLongField(thiz, javaAudioTrackFields.jniData, 0);
451
452 // lpTrack goes out of scope, so reference count drops to zero
453 return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;
454 }1.mode的取值：
构造方法中会传人一个mode参数，如果是MODE_STATIC则代表数据由app一次性传人,MODE_STREAM则代表数据不是一次性传人。android_media_AudioTrack.cpp的android_media_AudioTrack_setup方法中会创建一个不带参的lpTrack = new AudioTrack();
对于MODE_STATIC会通过allocSharedMem分配共享内存，对于MODE_STREAM不会分配共享内存。
xref: /frameworks/av/services/audioflinger/Tracks.cpp
496 // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
497 AudioFlinger::PlaybackThread::Track::Track(
498 PlaybackThread *thread,
499 const sp<Client>& client,
500 audio_stream_type_t streamType,
501 const audio_attributes_t& attr,
502 uint32_t sampleRate,
503 audio_format_t format,
504 audio_channel_mask_t channelMask,
505 size_t frameCount,
506 void *buffer,
507 size_t bufferSize,
508 const sp<IMemory>& sharedBuffer,
509 audio_session_t sessionId,
510 pid_t creatorPid,
511 uid_t uid,
512 audio_output_flags_t flags,
513 track_type type,
514 audio_port_handle_t portId)
515 : TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
516 (sharedBuffer != 0) ? sharedBuffer->pointer() : buffer,
517 (sharedBuffer != 0) ? sharedBuffer->size() : bufferSize,
518 sessionId, creatorPid, uid, true /*isOut*/,
519 (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
520 type, portId),
521 mFillingUpStatus(FS_INVALID),
522 // mRetryCount initialized later when needed
523 mSharedBuffer(sharedBuffer),
524 mStreamType(streamType),
525 mMainBuffer(thread->sinkBuffer()),
526 mAuxBuffer(NULL),
527 mAuxEffectId(0), mHasVolumeController(false),
528 mPresentationCompleteFrames(0),
529 mFrameMap(16 /* sink-frame-to-track-frame map memory */),
530 mVolumeHandler(new media::VolumeHandler(sampleRate)),
531 mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(uid, attr, id(), streamType)),
532 // mSinkTimestamp
533 mFastIndex(-1),
534 mCachedVolume(1.0),
535 /* The track might not play immediately after being active, similarly as if its volume was 0.
536 * When the track starts playing, its volume will be computed. */
537 mFinalVolume(0.f),
538 mResumeToStopping(false),
539 mFlushHwPending(false),
540 mFlags(flags)
541 {
542 // client == 0 implies sharedBuffer == 0
543 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
544
545 ALOGV_IF(sharedBuffer != 0, "%s(%d): sharedBuffer: %p, size: %zu",
546 __func__, mId, sharedBuffer->pointer(), sharedBuffer->size());
547
548 if (mCblk == NULL) {
549 return;
550 }
551
552 if (sharedBuffer == 0) {
553 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
554 mFrameSize, !isExternalTrack(), sampleRate);
555 } else {
556 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
557 mFrameSize);
558 }
559 mServerProxy = mAudioTrackServerProxy;
560
561 if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
562 ALOGE("%s(%d): no more tracks available", __func__, mId);
563 return;
564 }
565 // only allocate a fast track index if we were able to allocate a normal track name
566 if (flags & AUDIO_OUTPUT_FLAG_FAST) {
567 // FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential
568 // race with setSyncEvent(). However, if we call it, we cannot properly start
569 // static fast tracks (SoundPool) immediately after stopping.
570 //mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
571 ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
572 int i = __builtin_ctz(thread->mFastTrackAvailMask);
573 ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
574 // FIXME This is too eager. We allocate a fast track index before the
575 // fast track becomes active. Since fast tracks are a scarce resource,
576 // this means we are potentially denying other more important fast tracks from
577 // being created. It would be better to allocate the index dynamically.
578 mFastIndex = i;
579 thread->mFastTrackAvailMask &= ~(1 << i);
580 }
581
582 mServerLatencySupported = thread->type() == ThreadBase::MIXER
583 || thread->type() == ThreadBase::DUPLICATING;
584 #ifdef TEE_SINK
585 mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
586 + "_" + std::to_string(mId) + "_T");
587 #endif
588
589 if (channelMask & AUDIO_CHANNEL_HAPTIC_ALL) {
590 mAudioVibrationController = new AudioVibrationController(this);
591 mExternalVibration = new os::ExternalVibration(
592 mUid, "" /* pkg */, mAttr, mAudioVibrationController);
593 }
594 }xref: /frameworks/av/services/audioflinger/Tracks.cpp
64 // TrackBase constructor must be called with AudioFlinger::mLock held
65 AudioFlinger::ThreadBase::TrackBase::TrackBase(
66 ThreadBase *thread,
67 const sp<Client>& client,
68 const audio_attributes_t& attr,
69 uint32_t sampleRate,
70 audio_format_t format,
71 audio_channel_mask_t channelMask,
72 size_t frameCount,
73 void *buffer,
74 size_t bufferSize,
75 audio_session_t sessionId,
76 pid_t creatorPid,
77 uid_t clientUid,
78 bool isOut,
79 alloc_type alloc,
80 track_type type,
81 audio_port_handle_t portId)
82 : RefBase(),
83 mThread(thread),
84 mClient(client),
85 mCblk(NULL),
86 // mBuffer, mBufferSize
87 mState(IDLE),
88 mAttr(attr),
89 mSampleRate(sampleRate),
90 mFormat(format),
91 mChannelMask(channelMask),
92 mChannelCount(isOut ?
93 audio_channel_count_from_out_mask(channelMask) :
94 audio_channel_count_from_in_mask(channelMask)),
95 mFrameSize(audio_has_proportional_frames(format) ?
96 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
97 mFrameCount(frameCount),
98 mSessionId(sessionId),
99 mIsOut(isOut),
100 mId(android_atomic_inc(&nextTrackId)),
101 mTerminated(false),
102 mType(type),
103 mThreadIoHandle(thread ? thread->id() : AUDIO_IO_HANDLE_NONE),
104 mPortId(portId),
105 mIsInvalid(false),
106 mCreatorPid(creatorPid)
107 {
108 const uid_t callingUid = IPCThreadState::self()->getCallingUid();
109 if (!isAudioServerOrMediaServerUid(callingUid) || clientUid == AUDIO_UID_INVALID) {
110 ALOGW_IF(clientUid != AUDIO_UID_INVALID && clientUid != callingUid,
111 "%s(%d): uid %d tried to pass itself off as %d",
112 __func__, mId, callingUid, clientUid);
113 clientUid = callingUid;
114 }
115 // clientUid contains the uid of the app that is responsible for this track, so we can blame
116 // battery usage on it.
117 mUid = clientUid;
118
119 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
120
121 size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount;
122 // check overflow when computing bufferSize due to multiplication by mFrameSize.
123 if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2
124 || mFrameSize == 0 // format needs to be correct
125 || minBufferSize > SIZE_MAX / mFrameSize) {
126 android_errorWriteLog(0x534e4554, "34749571");
127 return;
128 }
129 minBufferSize *= mFrameSize;
130
131 if (buffer == nullptr) {
132 bufferSize = minBufferSize; // allocated here.
133 } else if (minBufferSize > bufferSize) {
134 android_errorWriteLog(0x534e4554, "38340117");
135 return;
136 }
137
138 size_t size = sizeof(audio_track_cblk_t);
139 if (buffer == NULL && alloc == ALLOC_CBLK) {
140 // check overflow when computing allocation size for streaming tracks.
141 if (size > SIZE_MAX - bufferSize) {
142 android_errorWriteLog(0x534e4554, "34749571");
143 return;
144 }
145 size += bufferSize;
146 }
147
148 if (client != 0) {
149 mCblkMemory = client->heap()->allocate(size);
150 if (mCblkMemory == 0 ||
151 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
152 ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
153 client->heap()->dump("AudioTrack");
154 mCblkMemory.clear();
155 return;
156 }
157 } else {
158 mCblk = (audio_track_cblk_t *) malloc(size);
159 if (mCblk == NULL) {
160 ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
161 return;
162 }
163 }
164
165 // construct the shared structure in-place.
166 if (mCblk != NULL) {
167 new(mCblk) audio_track_cblk_t();
168 switch (alloc) {
169 case ALLOC_READONLY: {
170 const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
171 if (roHeap == 0 ||
172 (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
173 (mBuffer = mBufferMemory->pointer()) == NULL) {
174 ALOGE("%s(%d): not enough memory for read-only buffer size=%zu",
175 __func__, mId, bufferSize);
176 if (roHeap != 0) {
177 roHeap->dump("buffer");
178 }
179 mCblkMemory.clear();
180 mBufferMemory.clear();
181 return;
182 }
183 memset(mBuffer, 0, bufferSize);
184 } break;
185 case ALLOC_PIPE:
186 mBufferMemory = thread->pipeMemory();
187 // mBuffer is the virtual address as seen from current process (mediaserver),
188 // and should normally be coming from mBufferMemory->pointer().
189 // However in this case the TrackBase does not reference the buffer directly.
190 // It should references the buffer via the pipe.
191 // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
192 mBuffer = NULL;
193 bufferSize = 0;
194 break;
195 case ALLOC_CBLK:
196 // clear all buffers
197 if (buffer == NULL) {
198 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
199 memset(mBuffer, 0, bufferSize);
200 } else {
201 mBuffer = buffer;
202 #if 0
203 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic
204 #endif
205 }
206 break;
207 case ALLOC_LOCAL:
208 mBuffer = calloc(1, bufferSize);
209 break;
210 case ALLOC_NONE:
211 mBuffer = buffer;
212 break;
213 default:
214 LOG_ALWAYS_FATAL("%s(%d): invalid allocation type: %d", __func__, mId, (int)alloc);
215 }
216 mBufferSize = bufferSize;
217
218 #ifdef TEE_SINK
219 mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK);
220 #endif
221
222 }
223 }
2.共享内存size的分配
每一个应用程序的AudioTrack都对应PlaybackThread中mTracks中的一个track，sharedBuffer是应用程序传过来的的共享内存。如果sharedBuffer等于null，则track的基类TrackBase中会创建一个共享内存。 如果应用程序提供Buffer,只分配mCblk。如果应用程序没有提供Buffer，则分配mCblk+ sizeof(audio_track_cblk_t)。
552 if (sharedBuffer == 0) {
553 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
554 mFrameSize, !isExternalTrack(), sampleRate);
555 } else {
556 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
557 mFrameSize);
558 }
559 mServerProxy = mAudioTrackServerProxy;3.buffer的管理 如果应用程序没有创建sharedBuffer，则通过AudioTrackServerProxy来管理buffer。如果应用程序创建了sharedBuffer，则通过StaticAudioTrackServerProxy来管理buffer。
634 template <typename T>
635 static jint writeToTrack(const sp<AudioTrack>& track, jint audioFormat, const T *data,
636 jint offsetInSamples, jint sizeInSamples, bool blocking) {
637 // give the data to the native AudioTrack object (the data starts at the offset)
638 ssize_t written = 0;
639 // regular write() or copy the data to the AudioTrack's shared memory?
640 size_t sizeInBytes = sizeInSamples * sizeof(T);
641 if (track->sharedBuffer() == 0) {
642 written = track->write(data + offsetInSamples, sizeInBytes, blocking);
643 // for compatibility with earlier behavior of write(), return 0 in this case
644 if (written == (ssize_t) WOULD_BLOCK) {
645 written = 0;
646 }
647 } else {
648 // writing to shared memory, check for capacity
649 if ((size_t)sizeInBytes > track->sharedBuffer()->size()) {
650 sizeInBytes = track->sharedBuffer()->size();
651 }
652 memcpy(track->sharedBuffer()->pointer(), data + offsetInSamples, sizeInBytes);
653 written = sizeInBytes;
654 }
655 if (written >= 0) {
656 return written / sizeof(T);
657 }
658 return interpretWriteSizeError(written);
659 }xref: /frameworks/av/services/audioflinger/Tracks.cpp
786 // AudioBufferProvider interface
787 status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
788 {
789 ServerProxy::Buffer buf;
790 size_t desiredFrames = buffer->frameCount;
791 buf.mFrameCount = desiredFrames;
792 status_t status = mServerProxy->obtainBuffer(&buf);
793 buffer->frameCount = buf.mFrameCount;
794 buffer->raw = buf.mRaw;
795 if (buf.mFrameCount == 0 && !isStopping() && !isStopped() && !isPaused()) {
796 ALOGV("%s(%d): underrun, framesReady(%zu) < framesDesired(%zd), state: %d",
797 __func__, mId, buf.mFrameCount, desiredFrames, mState);
798 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
799 } else {
800 mAudioTrackServerProxy->tallyUnderrunFrames(0);
801 }
802 return status;
803 }4.buffer的传递 通过writeToTrack写数据给PlaybackThread,在AudioFlinger::PlaybackThread::Track::getNextBuffer中通过mServerProxy->obtainBuffer获取buffer。
xref: /frameworks/av/media/libaudioclient/AudioTrackShared.cpp
1109 __attribute__((no_sanitize("integer")))
1110 status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
1111 {
1112 if (mIsShutdown) {
1113 buffer->mFrameCount = 0;
1114 buffer->mRaw = NULL;
1115 buffer->mNonContig = 0;
1116 mUnreleased = 0;
1117 return NO_INIT;
1118 }
1119 ssize_t positionOrStatus = pollPosition();
1120 if (positionOrStatus < 0) {
1121 buffer->mFrameCount = 0;
1122 buffer->mRaw = NULL;
1123 buffer->mNonContig = 0;
1124 mUnreleased = 0;
1125 return (status_t) positionOrStatus;
1126 }
1127 size_t position = (size_t) positionOrStatus;
1128 size_t end = mState.mLoopCount != 0 ? mState.mLoopEnd : mFrameCount;
1129 size_t avail;
1130 if (position < end) {
1131 avail = end - position;
1132 size_t wanted = buffer->mFrameCount;
1133 if (avail < wanted) {
1134 buffer->mFrameCount = avail;
1135 } else {
1136 avail = wanted;
1137 }
1138 buffer->mRaw = &((char *) mBuffers)[position * mFrameSize];
1139 } else {
1140 avail = 0;
1141 buffer->mFrameCount = 0;
1142 buffer->mRaw = NULL;
1143 }
1144 // As mFramesReady is the total remaining frames in the static audio track,
1145 // it is always larger or equal to avail.
1146 LOG_ALWAYS_FATAL_IF(mFramesReady < (int64_t) avail,
1147 "%s: mFramesReady out of range, mFramesReady:%lld < avail:%zu",
1148 __func__, (long long)mFramesReady, avail);
1149 buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail);
1150 if (!ackFlush) {
1151 mUnreleased = avail;
1152 }
1153 return NO_ERROR;
1154 }5.buffer的获取 根据mSharedBuffer创建不同的AudioTrack，我们这里以StaticAudioTrackClientProxy的创建为例。StaticAudioTrackServerProxy::obtainBuffer用来获取buffer。具体计算省略…
xref: /frameworks/av/media/libaudioclient/AudioTrackShared.cpp
1156 __attribute__((no_sanitize("integer")))
1157 void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
1158 {
1159 size_t stepCount = buffer->mFrameCount;
1160 LOG_ALWAYS_FATAL_IF(!((int64_t) stepCount <= mFramesReady),
1161 "%s: stepCount out of range, "
1162 "!(stepCount:%zu <= mFramesReady:%lld)",
1163 __func__, stepCount, (long long)mFramesReady);
1164 LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased),
1165 "%s: stepCount out of range, "
1166 "!(stepCount:%zu <= mUnreleased:%zu)",
1167 __func__, stepCount, mUnreleased);
1168 if (stepCount == 0) {
1169 // prevent accidental re-use of buffer
1170 buffer->mRaw = NULL;
1171 buffer->mNonContig = 0;
1172 return;
1173 }
1174 mUnreleased -= stepCount;
1175 audio_track_cblk_t* cblk = mCblk;
1176 size_t position = mState.mPosition;
1177 size_t newPosition = position + stepCount;
1178 int32_t setFlags = 0;
1179 if (!(position <= newPosition && newPosition <= mFrameCount)) {
1180 ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position,
1181 mFrameCount);
1182 newPosition = mFrameCount;
1183 } else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
1184 newPosition = mState.mLoopStart;
1185 if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) {
1186 setFlags = CBLK_LOOP_CYCLE;
1187 } else {
1188 setFlags = CBLK_LOOP_FINAL;
1189 }
1190 }
1191 if (newPosition == mFrameCount) {
1192 setFlags |= CBLK_BUFFER_END;
1193 }
1194 mState.mPosition = newPosition;
1195 if (mFramesReady != INT64_MAX) {
1196 mFramesReady -= stepCount;
1197 }
1198 mFramesReadySafe = clampToSize(mFramesReady);
1199
1200 cblk->mServer += stepCount;
1201 mReleased += stepCount;
1202
1203 // This may overflow, but client is not supposed to rely on it
1204 StaticAudioTrackPosLoop posLoop;
1205 posLoop.mBufferPosition = mState.mPosition;
1206 posLoop.mLoopCount = mState.mLoopCount;
1207 mPosLoopMutator.push(posLoop);
1208 if (setFlags != 0) {
1209 (void) android_atomic_or(setFlags, &cblk->mFlags);
1210 // this would be a good place to wake a futex
1211 }
1212
1213 buffer->mFrameCount = 0;
1214 buffer->mRaw = NULL;
1215 buffer->mNonContig = 0;
1216 }6.释放buffer
7.数据的同步
MODE_STREAM模式时AudioTrack和PlaybackThread之间数据的传递采用环形缓冲区。
作者：@OuYang
