blob: 315cbbcd6164d8d755c7afd7c6ba0ffc934e04cd [file] [log] [blame]
Eric Laurent81784c32012-11-19 14:55:58 -08001/*
2**
3** Copyright 2012, 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_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <math.h>
23#include <cutils/compiler.h>
24#include <utils/Log.h>
25
26#include <private/media/AudioTrackShared.h>
27
28#include <common_time/cc_helper.h>
29#include <common_time/local_clock.h>
30
31#include "AudioMixer.h"
32#include "AudioFlinger.h"
33#include "ServiceUtilities.h"
34
35// ----------------------------------------------------------------------------
36
37// Note: the following macro is used for extremely verbose logging message. In
38// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
39// 0; but one side effect of this is to turn all LOGV's as well. Some messages
40// are so verbose that we want to suppress them even when we have ALOG_ASSERT
41// turned on. Do not uncomment the #def below unless you really know what you
42// are doing and want to see all of the extremely verbose messages.
43//#define VERY_VERY_VERBOSE_LOGGING
44#ifdef VERY_VERY_VERBOSE_LOGGING
45#define ALOGVV ALOGV
46#else
47#define ALOGVV(a...) do { } while(0)
48#endif
49
50namespace android {
51
52// ----------------------------------------------------------------------------
53// TrackBase
54// ----------------------------------------------------------------------------
55
56// TrackBase constructor must be called with AudioFlinger::mLock held
57AudioFlinger::ThreadBase::TrackBase::TrackBase(
58 ThreadBase *thread,
59 const sp<Client>& client,
60 uint32_t sampleRate,
61 audio_format_t format,
62 audio_channel_mask_t channelMask,
63 size_t frameCount,
64 const sp<IMemory>& sharedBuffer,
Glenn Kastene3aa6592012-12-04 12:22:46 -080065 int sessionId,
66 bool isOut)
Eric Laurent81784c32012-11-19 14:55:58 -080067 : RefBase(),
68 mThread(thread),
69 mClient(client),
70 mCblk(NULL),
71 // mBuffer
72 // mBufferEnd
73 mStepCount(0),
74 mState(IDLE),
75 mSampleRate(sampleRate),
76 mFormat(format),
77 mChannelMask(channelMask),
78 mChannelCount(popcount(channelMask)),
79 mFrameSize(audio_is_linear_pcm(format) ?
80 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
81 mFrameCount(frameCount),
82 mStepServerFailed(false),
Glenn Kastene3aa6592012-12-04 12:22:46 -080083 mSessionId(sessionId),
84 mIsOut(isOut),
85 mServerProxy(NULL)
Eric Laurent81784c32012-11-19 14:55:58 -080086{
87 // client == 0 implies sharedBuffer == 0
88 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
89
90 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
91 sharedBuffer->size());
92
93 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
94 size_t size = sizeof(audio_track_cblk_t);
95 size_t bufferSize = frameCount * mFrameSize;
96 if (sharedBuffer == 0) {
97 size += bufferSize;
98 }
99
100 if (client != 0) {
101 mCblkMemory = client->heap()->allocate(size);
102 if (mCblkMemory != 0) {
103 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
104 // can't assume mCblk != NULL
105 } else {
106 ALOGE("not enough memory for AudioTrack size=%u", size);
107 client->heap()->dump("AudioTrack");
108 return;
109 }
110 } else {
Glenn Kastene3aa6592012-12-04 12:22:46 -0800111 // this syntax avoids calling the audio_track_cblk_t constructor twice
112 mCblk = (audio_track_cblk_t *) new uint8_t[size];
Eric Laurent81784c32012-11-19 14:55:58 -0800113 // assume mCblk != NULL
114 }
115
116 // construct the shared structure in-place.
117 if (mCblk != NULL) {
118 new(mCblk) audio_track_cblk_t();
119 // clear all buffers
120 mCblk->frameCount_ = frameCount;
Eric Laurent81784c32012-11-19 14:55:58 -0800121// uncomment the following lines to quickly test 32-bit wraparound
122// mCblk->user = 0xffff0000;
123// mCblk->server = 0xffff0000;
124// mCblk->userBase = 0xffff0000;
125// mCblk->serverBase = 0xffff0000;
126 if (sharedBuffer == 0) {
127 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
128 memset(mBuffer, 0, bufferSize);
129 // Force underrun condition to avoid false underrun callback until first data is
130 // written to buffer (other flags are cleared)
131 mCblk->flags = CBLK_UNDERRUN;
132 } else {
133 mBuffer = sharedBuffer->pointer();
134 }
135 mBufferEnd = (uint8_t *)mBuffer + bufferSize;
Glenn Kastene3aa6592012-12-04 12:22:46 -0800136 mServerProxy = new ServerProxy(mCblk, mBuffer, frameCount, mFrameSize, isOut);
Eric Laurent81784c32012-11-19 14:55:58 -0800137 }
138}
139
140AudioFlinger::ThreadBase::TrackBase::~TrackBase()
141{
Glenn Kastene3aa6592012-12-04 12:22:46 -0800142 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
143 delete mServerProxy;
Eric Laurent81784c32012-11-19 14:55:58 -0800144 if (mCblk != NULL) {
145 if (mClient == 0) {
146 delete mCblk;
147 } else {
148 mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
149 }
150 }
151 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
152 if (mClient != 0) {
153 // Client destructor must run with AudioFlinger mutex locked
154 Mutex::Autolock _l(mClient->audioFlinger()->mLock);
155 // If the client's reference count drops to zero, the associated destructor
156 // must run with AudioFlinger lock held. Thus the explicit clear() rather than
157 // relying on the automatic clear() at end of scope.
158 mClient.clear();
159 }
160}
161
162// AudioBufferProvider interface
163// getNextBuffer() = 0;
164// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
165void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
166{
167 buffer->raw = NULL;
168 mStepCount = buffer->frameCount;
169 // FIXME See note at getNextBuffer()
170 (void) step(); // ignore return value of step()
171 buffer->frameCount = 0;
172}
173
174bool AudioFlinger::ThreadBase::TrackBase::step() {
Glenn Kastene3aa6592012-12-04 12:22:46 -0800175 bool result = mServerProxy->step(mStepCount);
Eric Laurent81784c32012-11-19 14:55:58 -0800176 if (!result) {
177 ALOGV("stepServer failed acquiring cblk mutex");
178 mStepServerFailed = true;
179 }
180 return result;
181}
182
183void AudioFlinger::ThreadBase::TrackBase::reset() {
184 audio_track_cblk_t* cblk = this->cblk();
185
186 cblk->user = 0;
187 cblk->server = 0;
188 cblk->userBase = 0;
189 cblk->serverBase = 0;
190 mStepServerFailed = false;
191 ALOGV("TrackBase::reset");
192}
193
194uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
Glenn Kastene3aa6592012-12-04 12:22:46 -0800195 return mServerProxy->getSampleRate();
Eric Laurent81784c32012-11-19 14:55:58 -0800196}
197
198void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
199 audio_track_cblk_t* cblk = this->cblk();
200 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
201 int8_t *bufferEnd = bufferStart + frames * mFrameSize;
202
203 // Check validity of returned pointer in case the track control block would have been corrupted.
204 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
205 "TrackBase::getBuffer buffer out of range:\n"
206 " start: %p, end %p , mBuffer %p mBufferEnd %p\n"
207 " server %u, serverBase %u, user %u, userBase %u, frameSize %u",
208 bufferStart, bufferEnd, mBuffer, mBufferEnd,
209 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
210
211 return bufferStart;
212}
213
214status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
215{
216 mSyncEvents.add(event);
217 return NO_ERROR;
218}
219
220// ----------------------------------------------------------------------------
221// Playback
222// ----------------------------------------------------------------------------
223
224AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
225 : BnAudioTrack(),
226 mTrack(track)
227{
228}
229
230AudioFlinger::TrackHandle::~TrackHandle() {
231 // just stop the track on deletion, associated resources
232 // will be freed from the main thread once all pending buffers have
233 // been played. Unless it's not in the active track list, in which
234 // case we free everything now...
235 mTrack->destroy();
236}
237
238sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
239 return mTrack->getCblk();
240}
241
242status_t AudioFlinger::TrackHandle::start() {
243 return mTrack->start();
244}
245
246void AudioFlinger::TrackHandle::stop() {
247 mTrack->stop();
248}
249
250void AudioFlinger::TrackHandle::flush() {
251 mTrack->flush();
252}
253
Eric Laurent81784c32012-11-19 14:55:58 -0800254void AudioFlinger::TrackHandle::pause() {
255 mTrack->pause();
256}
257
258status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
259{
260 return mTrack->attachAuxEffect(EffectId);
261}
262
263status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
264 sp<IMemory>* buffer) {
265 if (!mTrack->isTimedTrack())
266 return INVALID_OPERATION;
267
268 PlaybackThread::TimedTrack* tt =
269 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
270 return tt->allocateTimedBuffer(size, buffer);
271}
272
273status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
274 int64_t pts) {
275 if (!mTrack->isTimedTrack())
276 return INVALID_OPERATION;
277
278 PlaybackThread::TimedTrack* tt =
279 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
280 return tt->queueTimedBuffer(buffer, pts);
281}
282
283status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
284 const LinearTransform& xform, int target) {
285
286 if (!mTrack->isTimedTrack())
287 return INVALID_OPERATION;
288
289 PlaybackThread::TimedTrack* tt =
290 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
291 return tt->setMediaTimeTransform(
292 xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
293}
294
295status_t AudioFlinger::TrackHandle::onTransact(
296 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
297{
298 return BnAudioTrack::onTransact(code, data, reply, flags);
299}
300
301// ----------------------------------------------------------------------------
302
303// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
304AudioFlinger::PlaybackThread::Track::Track(
305 PlaybackThread *thread,
306 const sp<Client>& client,
307 audio_stream_type_t streamType,
308 uint32_t sampleRate,
309 audio_format_t format,
310 audio_channel_mask_t channelMask,
311 size_t frameCount,
312 const sp<IMemory>& sharedBuffer,
313 int sessionId,
314 IAudioFlinger::track_flags_t flags)
315 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
Glenn Kastene3aa6592012-12-04 12:22:46 -0800316 sessionId, true /*isOut*/),
Eric Laurent81784c32012-11-19 14:55:58 -0800317 mFillingUpStatus(FS_INVALID),
318 // mRetryCount initialized later when needed
319 mSharedBuffer(sharedBuffer),
320 mStreamType(streamType),
321 mName(-1), // see note below
322 mMainBuffer(thread->mixBuffer()),
323 mAuxBuffer(NULL),
324 mAuxEffectId(0), mHasVolumeController(false),
325 mPresentationCompleteFrames(0),
326 mFlags(flags),
327 mFastIndex(-1),
328 mUnderrunCount(0),
Glenn Kasten5736c352012-12-04 12:12:34 -0800329 mCachedVolume(1.0),
330 mIsInvalid(false)
Eric Laurent81784c32012-11-19 14:55:58 -0800331{
332 if (mCblk != NULL) {
333 // to avoid leaking a track name, do not allocate one unless there is an mCblk
334 mName = thread->getTrackName_l(channelMask, sessionId);
335 mCblk->mName = mName;
336 if (mName < 0) {
337 ALOGE("no more track names available");
338 return;
339 }
340 // only allocate a fast track index if we were able to allocate a normal track name
341 if (flags & IAudioFlinger::TRACK_FAST) {
342 ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
343 int i = __builtin_ctz(thread->mFastTrackAvailMask);
344 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks);
345 // FIXME This is too eager. We allocate a fast track index before the
346 // fast track becomes active. Since fast tracks are a scarce resource,
347 // this means we are potentially denying other more important fast tracks from
348 // being created. It would be better to allocate the index dynamically.
349 mFastIndex = i;
350 mCblk->mName = i;
351 // Read the initial underruns because this field is never cleared by the fast mixer
352 mObservedUnderruns = thread->getFastTrackUnderruns(i);
353 thread->mFastTrackAvailMask &= ~(1 << i);
354 }
355 }
356 ALOGV("Track constructor name %d, calling pid %d", mName,
357 IPCThreadState::self()->getCallingPid());
358}
359
360AudioFlinger::PlaybackThread::Track::~Track()
361{
362 ALOGV("PlaybackThread::Track destructor");
363}
364
365void AudioFlinger::PlaybackThread::Track::destroy()
366{
367 // NOTE: destroyTrack_l() can remove a strong reference to this Track
368 // by removing it from mTracks vector, so there is a risk that this Tracks's
369 // destructor is called. As the destructor needs to lock mLock,
370 // we must acquire a strong reference on this Track before locking mLock
371 // here so that the destructor is called only when exiting this function.
372 // On the other hand, as long as Track::destroy() is only called by
373 // TrackHandle destructor, the TrackHandle still holds a strong ref on
374 // this Track with its member mTrack.
375 sp<Track> keep(this);
376 { // scope for mLock
377 sp<ThreadBase> thread = mThread.promote();
378 if (thread != 0) {
379 if (!isOutputTrack()) {
380 if (mState == ACTIVE || mState == RESUMING) {
381 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
382
383#ifdef ADD_BATTERY_DATA
384 // to track the speaker usage
385 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
386#endif
387 }
388 AudioSystem::releaseOutput(thread->id());
389 }
390 Mutex::Autolock _l(thread->mLock);
391 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
392 playbackThread->destroyTrack_l(this);
393 }
394 }
395}
396
397/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
398{
Glenn Kastene4756fe2012-11-29 13:38:14 -0800399 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S F SRate "
Eric Laurent81784c32012-11-19 14:55:58 -0800400 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n");
401}
402
403void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
404{
Glenn Kastene3aa6592012-12-04 12:22:46 -0800405 uint32_t vlr = mServerProxy->getVolumeLR();
Eric Laurent81784c32012-11-19 14:55:58 -0800406 if (isFastTrack()) {
407 sprintf(buffer, " F %2d", mFastIndex);
408 } else {
409 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0);
410 }
411 track_state state = mState;
412 char stateChar;
413 switch (state) {
414 case IDLE:
415 stateChar = 'I';
416 break;
417 case TERMINATED:
418 stateChar = 'T';
419 break;
420 case STOPPING_1:
421 stateChar = 's';
422 break;
423 case STOPPING_2:
424 stateChar = '5';
425 break;
426 case STOPPED:
427 stateChar = 'S';
428 break;
429 case RESUMING:
430 stateChar = 'R';
431 break;
432 case ACTIVE:
433 stateChar = 'A';
434 break;
435 case PAUSING:
436 stateChar = 'p';
437 break;
438 case PAUSED:
439 stateChar = 'P';
440 break;
441 case FLUSHED:
442 stateChar = 'F';
443 break;
444 default:
445 stateChar = '?';
446 break;
447 }
448 char nowInUnderrun;
449 switch (mObservedUnderruns.mBitFields.mMostRecent) {
450 case UNDERRUN_FULL:
451 nowInUnderrun = ' ';
452 break;
453 case UNDERRUN_PARTIAL:
454 nowInUnderrun = '<';
455 break;
456 case UNDERRUN_EMPTY:
457 nowInUnderrun = '*';
458 break;
459 default:
460 nowInUnderrun = '?';
461 break;
462 }
Glenn Kastene4756fe2012-11-29 13:38:14 -0800463 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %5u %5.2g %5.2g "
Eric Laurent81784c32012-11-19 14:55:58 -0800464 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n",
465 (mClient == 0) ? getpid_cached : mClient->pid(),
466 mStreamType,
467 mFormat,
468 mChannelMask,
469 mSessionId,
470 mStepCount,
471 mFrameCount,
472 stateChar,
Eric Laurent81784c32012-11-19 14:55:58 -0800473 mFillingUpStatus,
Glenn Kastene3aa6592012-12-04 12:22:46 -0800474 mServerProxy->getSampleRate(),
Eric Laurent81784c32012-11-19 14:55:58 -0800475 20.0 * log10((vlr & 0xFFFF) / 4096.0),
476 20.0 * log10((vlr >> 16) / 4096.0),
477 mCblk->server,
478 mCblk->user,
479 (int)mMainBuffer,
480 (int)mAuxBuffer,
481 mCblk->flags,
482 mUnderrunCount,
483 nowInUnderrun);
484}
485
486// AudioBufferProvider interface
487status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
488 AudioBufferProvider::Buffer* buffer, int64_t pts)
489{
490 audio_track_cblk_t* cblk = this->cblk();
491 uint32_t framesReady;
492 uint32_t framesReq = buffer->frameCount;
493
494 // Check if last stepServer failed, try to step now
495 if (mStepServerFailed) {
496 // FIXME When called by fast mixer, this takes a mutex with tryLock().
497 // Since the fast mixer is higher priority than client callback thread,
498 // it does not result in priority inversion for client.
499 // But a non-blocking solution would be preferable to avoid
500 // fast mixer being unable to tryLock(), and
501 // to avoid the extra context switches if the client wakes up,
502 // discovers the mutex is locked, then has to wait for fast mixer to unlock.
503 if (!step()) goto getNextBuffer_exit;
504 ALOGV("stepServer recovered");
505 mStepServerFailed = false;
506 }
507
508 // FIXME Same as above
Glenn Kastene3aa6592012-12-04 12:22:46 -0800509 framesReady = mServerProxy->framesReady();
Eric Laurent81784c32012-11-19 14:55:58 -0800510
511 if (CC_LIKELY(framesReady)) {
512 uint32_t s = cblk->server;
513 uint32_t bufferEnd = cblk->serverBase + mFrameCount;
514
515 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
516 if (framesReq > framesReady) {
517 framesReq = framesReady;
518 }
519 if (framesReq > bufferEnd - s) {
520 framesReq = bufferEnd - s;
521 }
522
523 buffer->raw = getBuffer(s, framesReq);
524 buffer->frameCount = framesReq;
525 return NO_ERROR;
526 }
527
528getNextBuffer_exit:
529 buffer->raw = NULL;
530 buffer->frameCount = 0;
531 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
532 return NOT_ENOUGH_DATA;
533}
534
535// Note that framesReady() takes a mutex on the control block using tryLock().
536// This could result in priority inversion if framesReady() is called by the normal mixer,
537// as the normal mixer thread runs at lower
538// priority than the client's callback thread: there is a short window within framesReady()
539// during which the normal mixer could be preempted, and the client callback would block.
540// Another problem can occur if framesReady() is called by the fast mixer:
541// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer.
542// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue.
543size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
Glenn Kastene3aa6592012-12-04 12:22:46 -0800544 return mServerProxy->framesReady();
Eric Laurent81784c32012-11-19 14:55:58 -0800545}
546
547// Don't call for fast tracks; the framesReady() could result in priority inversion
548bool AudioFlinger::PlaybackThread::Track::isReady() const {
549 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
550 return true;
551 }
552
553 if (framesReady() >= mFrameCount ||
554 (mCblk->flags & CBLK_FORCEREADY)) {
555 mFillingUpStatus = FS_FILLED;
556 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
557 return true;
558 }
559 return false;
560}
561
562status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
563 int triggerSession)
564{
565 status_t status = NO_ERROR;
566 ALOGV("start(%d), calling pid %d session %d",
567 mName, IPCThreadState::self()->getCallingPid(), mSessionId);
568
569 sp<ThreadBase> thread = mThread.promote();
570 if (thread != 0) {
571 Mutex::Autolock _l(thread->mLock);
Glenn Kasten7f5d3352013-02-15 23:55:04 +0000572 thread->mNBLogWriter->logf("start mName=%d", mName);
Eric Laurent81784c32012-11-19 14:55:58 -0800573 track_state state = mState;
574 // here the track could be either new, or restarted
575 // in both cases "unstop" the track
576 if (mState == PAUSED) {
577 mState = TrackBase::RESUMING;
578 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
579 } else {
580 mState = TrackBase::ACTIVE;
581 ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
582 }
583
584 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
585 thread->mLock.unlock();
586 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
587 thread->mLock.lock();
588
589#ifdef ADD_BATTERY_DATA
590 // to track the speaker usage
591 if (status == NO_ERROR) {
592 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
593 }
594#endif
595 }
596 if (status == NO_ERROR) {
597 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
598 playbackThread->addTrack_l(this);
599 } else {
600 mState = state;
601 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
602 }
603 } else {
604 status = BAD_VALUE;
605 }
606 return status;
607}
608
609void AudioFlinger::PlaybackThread::Track::stop()
610{
611 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
612 sp<ThreadBase> thread = mThread.promote();
613 if (thread != 0) {
614 Mutex::Autolock _l(thread->mLock);
Glenn Kasten7f5d3352013-02-15 23:55:04 +0000615 thread->mNBLogWriter->logf("stop mName=%d", mName);
Eric Laurent81784c32012-11-19 14:55:58 -0800616 track_state state = mState;
617 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
618 // If the track is not active (PAUSED and buffers full), flush buffers
619 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
620 if (playbackThread->mActiveTracks.indexOf(this) < 0) {
621 reset();
622 mState = STOPPED;
623 } else if (!isFastTrack()) {
624 mState = STOPPED;
625 } else {
626 // prepareTracks_l() will set state to STOPPING_2 after next underrun,
627 // and then to STOPPED and reset() when presentation is complete
628 mState = STOPPING_1;
629 }
630 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
631 playbackThread);
632 }
633 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
634 thread->mLock.unlock();
635 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
636 thread->mLock.lock();
637
638#ifdef ADD_BATTERY_DATA
639 // to track the speaker usage
640 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
641#endif
642 }
643 }
644}
645
646void AudioFlinger::PlaybackThread::Track::pause()
647{
648 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
649 sp<ThreadBase> thread = mThread.promote();
650 if (thread != 0) {
651 Mutex::Autolock _l(thread->mLock);
Glenn Kasten7f5d3352013-02-15 23:55:04 +0000652 thread->mNBLogWriter->logf("pause mName=%d", mName);
Eric Laurent81784c32012-11-19 14:55:58 -0800653 if (mState == ACTIVE || mState == RESUMING) {
654 mState = PAUSING;
655 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
656 if (!isOutputTrack()) {
657 thread->mLock.unlock();
658 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
659 thread->mLock.lock();
660
661#ifdef ADD_BATTERY_DATA
662 // to track the speaker usage
663 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
664#endif
665 }
666 }
667 }
668}
669
670void AudioFlinger::PlaybackThread::Track::flush()
671{
672 ALOGV("flush(%d)", mName);
673 sp<ThreadBase> thread = mThread.promote();
674 if (thread != 0) {
675 Mutex::Autolock _l(thread->mLock);
Glenn Kasten7f5d3352013-02-15 23:55:04 +0000676 thread->mNBLogWriter->logf("flush mName=%d", mName);
Eric Laurent81784c32012-11-19 14:55:58 -0800677 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED &&
678 mState != PAUSING && mState != IDLE && mState != FLUSHED) {
679 return;
680 }
681 // No point remaining in PAUSED state after a flush => go to
682 // FLUSHED state
683 mState = FLUSHED;
684 // do not reset the track if it is still in the process of being stopped or paused.
685 // this will be done by prepareTracks_l() when the track is stopped.
686 // prepareTracks_l() will see mState == FLUSHED, then
687 // remove from active track list, reset(), and trigger presentation complete
688 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
689 if (playbackThread->mActiveTracks.indexOf(this) < 0) {
690 reset();
691 }
692 }
693}
694
695void AudioFlinger::PlaybackThread::Track::reset()
696{
697 // Do not reset twice to avoid discarding data written just after a flush and before
698 // the audioflinger thread detects the track is stopped.
699 if (!mResetDone) {
700 TrackBase::reset();
701 // Force underrun condition to avoid false underrun callback until first data is
702 // written to buffer
703 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags);
704 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
705 mFillingUpStatus = FS_FILLING;
706 mResetDone = true;
707 if (mState == FLUSHED) {
708 mState = IDLE;
709 }
710 }
711}
712
Eric Laurent81784c32012-11-19 14:55:58 -0800713status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
714{
715 status_t status = DEAD_OBJECT;
716 sp<ThreadBase> thread = mThread.promote();
717 if (thread != 0) {
718 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
719 sp<AudioFlinger> af = mClient->audioFlinger();
720
721 Mutex::Autolock _l(af->mLock);
722
723 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
724
725 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) {
726 Mutex::Autolock _dl(playbackThread->mLock);
727 Mutex::Autolock _sl(srcThread->mLock);
728 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
729 if (chain == 0) {
730 return INVALID_OPERATION;
731 }
732
733 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId);
734 if (effect == 0) {
735 return INVALID_OPERATION;
736 }
737 srcThread->removeEffect_l(effect);
738 playbackThread->addEffect_l(effect);
739 // removeEffect_l() has stopped the effect if it was active so it must be restarted
740 if (effect->state() == EffectModule::ACTIVE ||
741 effect->state() == EffectModule::STOPPING) {
742 effect->start();
743 }
744
745 sp<EffectChain> dstChain = effect->chain().promote();
746 if (dstChain == 0) {
747 srcThread->addEffect_l(effect);
748 return INVALID_OPERATION;
749 }
750 AudioSystem::unregisterEffect(effect->id());
751 AudioSystem::registerEffect(&effect->desc(),
752 srcThread->id(),
753 dstChain->strategy(),
754 AUDIO_SESSION_OUTPUT_MIX,
755 effect->id());
756 }
757 status = playbackThread->attachAuxEffect(this, EffectId);
758 }
759 return status;
760}
761
762void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
763{
764 mAuxEffectId = EffectId;
765 mAuxBuffer = buffer;
766}
767
768bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten,
769 size_t audioHalFrames)
770{
771 // a track is considered presented when the total number of frames written to audio HAL
772 // corresponds to the number of frames written when presentationComplete() is called for the
773 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
774 if (mPresentationCompleteFrames == 0) {
775 mPresentationCompleteFrames = framesWritten + audioHalFrames;
776 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d",
777 mPresentationCompleteFrames, audioHalFrames);
778 }
779 if (framesWritten >= mPresentationCompleteFrames) {
780 ALOGV("presentationComplete() session %d complete: framesWritten %d",
781 mSessionId, framesWritten);
782 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
783 return true;
784 }
785 return false;
786}
787
788void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
789{
790 for (int i = 0; i < (int)mSyncEvents.size(); i++) {
791 if (mSyncEvents[i]->type() == type) {
792 mSyncEvents[i]->trigger();
793 mSyncEvents.removeAt(i);
794 i--;
795 }
796 }
797}
798
799// implement VolumeBufferProvider interface
800
801uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
802{
803 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
804 ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
Glenn Kastene3aa6592012-12-04 12:22:46 -0800805 uint32_t vlr = mServerProxy->getVolumeLR();
Eric Laurent81784c32012-11-19 14:55:58 -0800806 uint32_t vl = vlr & 0xFFFF;
807 uint32_t vr = vlr >> 16;
808 // track volumes come from shared memory, so can't be trusted and must be clamped
809 if (vl > MAX_GAIN_INT) {
810 vl = MAX_GAIN_INT;
811 }
812 if (vr > MAX_GAIN_INT) {
813 vr = MAX_GAIN_INT;
814 }
815 // now apply the cached master volume and stream type volume;
816 // this is trusted but lacks any synchronization or barrier so may be stale
817 float v = mCachedVolume;
818 vl *= v;
819 vr *= v;
820 // re-combine into U4.16
821 vlr = (vr << 16) | (vl & 0xFFFF);
822 // FIXME look at mute, pause, and stop flags
823 return vlr;
824}
825
826status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
827{
828 if (mState == TERMINATED || mState == PAUSED ||
829 ((framesReady() == 0) && ((mSharedBuffer != 0) ||
830 (mState == STOPPED)))) {
831 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ",
832 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
833 event->cancel();
834 return INVALID_OPERATION;
835 }
836 (void) TrackBase::setSyncEvent(event);
837 return NO_ERROR;
838}
839
Glenn Kasten5736c352012-12-04 12:12:34 -0800840void AudioFlinger::PlaybackThread::Track::invalidate()
841{
842 // FIXME should use proxy
843 android_atomic_or(CBLK_INVALID, &mCblk->flags);
844 mCblk->cv.signal();
845 mIsInvalid = true;
846}
847
Eric Laurent81784c32012-11-19 14:55:58 -0800848// ----------------------------------------------------------------------------
849
850sp<AudioFlinger::PlaybackThread::TimedTrack>
851AudioFlinger::PlaybackThread::TimedTrack::create(
852 PlaybackThread *thread,
853 const sp<Client>& client,
854 audio_stream_type_t streamType,
855 uint32_t sampleRate,
856 audio_format_t format,
857 audio_channel_mask_t channelMask,
858 size_t frameCount,
859 const sp<IMemory>& sharedBuffer,
860 int sessionId) {
861 if (!client->reserveTimedTrack())
862 return 0;
863
864 return new TimedTrack(
865 thread, client, streamType, sampleRate, format, channelMask, frameCount,
866 sharedBuffer, sessionId);
867}
868
869AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
870 PlaybackThread *thread,
871 const sp<Client>& client,
872 audio_stream_type_t streamType,
873 uint32_t sampleRate,
874 audio_format_t format,
875 audio_channel_mask_t channelMask,
876 size_t frameCount,
877 const sp<IMemory>& sharedBuffer,
878 int sessionId)
879 : Track(thread, client, streamType, sampleRate, format, channelMask,
880 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
881 mQueueHeadInFlight(false),
882 mTrimQueueHeadOnRelease(false),
883 mFramesPendingInQueue(0),
884 mTimedSilenceBuffer(NULL),
885 mTimedSilenceBufferSize(0),
886 mTimedAudioOutputOnTime(false),
887 mMediaTimeTransformValid(false)
888{
889 LocalClock lc;
890 mLocalTimeFreq = lc.getLocalFreq();
891
892 mLocalTimeToSampleTransform.a_zero = 0;
893 mLocalTimeToSampleTransform.b_zero = 0;
894 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
895 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
896 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
897 &mLocalTimeToSampleTransform.a_to_b_denom);
898
899 mMediaTimeToSampleTransform.a_zero = 0;
900 mMediaTimeToSampleTransform.b_zero = 0;
901 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate;
902 mMediaTimeToSampleTransform.a_to_b_denom = 1000000;
903 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer,
904 &mMediaTimeToSampleTransform.a_to_b_denom);
905}
906
907AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
908 mClient->releaseTimedTrack();
909 delete [] mTimedSilenceBuffer;
910}
911
912status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
913 size_t size, sp<IMemory>* buffer) {
914
915 Mutex::Autolock _l(mTimedBufferQueueLock);
916
917 trimTimedBufferQueue_l();
918
919 // lazily initialize the shared memory heap for timed buffers
920 if (mTimedMemoryDealer == NULL) {
921 const int kTimedBufferHeapSize = 512 << 10;
922
923 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
924 "AudioFlingerTimed");
925 if (mTimedMemoryDealer == NULL)
926 return NO_MEMORY;
927 }
928
929 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
930 if (newBuffer == NULL) {
931 newBuffer = mTimedMemoryDealer->allocate(size);
932 if (newBuffer == NULL)
933 return NO_MEMORY;
934 }
935
936 *buffer = newBuffer;
937 return NO_ERROR;
938}
939
940// caller must hold mTimedBufferQueueLock
941void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
942 int64_t mediaTimeNow;
943 {
944 Mutex::Autolock mttLock(mMediaTimeTransformLock);
945 if (!mMediaTimeTransformValid)
946 return;
947
948 int64_t targetTimeNow;
949 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
950 ? mCCHelper.getCommonTime(&targetTimeNow)
951 : mCCHelper.getLocalTime(&targetTimeNow);
952
953 if (OK != res)
954 return;
955
956 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
957 &mediaTimeNow)) {
958 return;
959 }
960 }
961
962 size_t trimEnd;
963 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
964 int64_t bufEnd;
965
966 if ((trimEnd + 1) < mTimedBufferQueue.size()) {
967 // We have a next buffer. Just use its PTS as the PTS of the frame
968 // following the last frame in this buffer. If the stream is sparse
969 // (ie, there are deliberate gaps left in the stream which should be
970 // filled with silence by the TimedAudioTrack), then this can result
971 // in one extra buffer being left un-trimmed when it could have
972 // been. In general, this is not typical, and we would rather
973 // optimized away the TS calculation below for the more common case
974 // where PTSes are contiguous.
975 bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
976 } else {
977 // We have no next buffer. Compute the PTS of the frame following
978 // the last frame in this buffer by computing the duration of of
979 // this frame in media time units and adding it to the PTS of the
980 // buffer.
981 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
982 / mFrameSize;
983
984 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
985 &bufEnd)) {
986 ALOGE("Failed to convert frame count of %lld to media time"
987 " duration" " (scale factor %d/%u) in %s",
988 frameCount,
989 mMediaTimeToSampleTransform.a_to_b_numer,
990 mMediaTimeToSampleTransform.a_to_b_denom,
991 __PRETTY_FUNCTION__);
992 break;
993 }
994 bufEnd += mTimedBufferQueue[trimEnd].pts();
995 }
996
997 if (bufEnd > mediaTimeNow)
998 break;
999
1000 // Is the buffer we want to use in the middle of a mix operation right
1001 // now? If so, don't actually trim it. Just wait for the releaseBuffer
1002 // from the mixer which should be coming back shortly.
1003 if (!trimEnd && mQueueHeadInFlight) {
1004 mTrimQueueHeadOnRelease = true;
1005 }
1006 }
1007
1008 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
1009 if (trimStart < trimEnd) {
1010 // Update the bookkeeping for framesReady()
1011 for (size_t i = trimStart; i < trimEnd; ++i) {
1012 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
1013 }
1014
1015 // Now actually remove the buffers from the queue.
1016 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
1017 }
1018}
1019
1020void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
1021 const char* logTag) {
1022 ALOG_ASSERT(mTimedBufferQueue.size() > 0,
1023 "%s called (reason \"%s\"), but timed buffer queue has no"
1024 " elements to trim.", __FUNCTION__, logTag);
1025
1026 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
1027 mTimedBufferQueue.removeAt(0);
1028}
1029
1030void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
1031 const TimedBuffer& buf,
1032 const char* logTag) {
1033 uint32_t bufBytes = buf.buffer()->size();
1034 uint32_t consumedAlready = buf.position();
1035
1036 ALOG_ASSERT(consumedAlready <= bufBytes,
1037 "Bad bookkeeping while updating frames pending. Timed buffer is"
1038 " only %u bytes long, but claims to have consumed %u"
1039 " bytes. (update reason: \"%s\")",
1040 bufBytes, consumedAlready, logTag);
1041
1042 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
1043 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
1044 "Bad bookkeeping while updating frames pending. Should have at"
1045 " least %u queued frames, but we think we have only %u. (update"
1046 " reason: \"%s\")",
1047 bufFrames, mFramesPendingInQueue, logTag);
1048
1049 mFramesPendingInQueue -= bufFrames;
1050}
1051
1052status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
1053 const sp<IMemory>& buffer, int64_t pts) {
1054
1055 {
1056 Mutex::Autolock mttLock(mMediaTimeTransformLock);
1057 if (!mMediaTimeTransformValid)
1058 return INVALID_OPERATION;
1059 }
1060
1061 Mutex::Autolock _l(mTimedBufferQueueLock);
1062
1063 uint32_t bufFrames = buffer->size() / mFrameSize;
1064 mFramesPendingInQueue += bufFrames;
1065 mTimedBufferQueue.add(TimedBuffer(buffer, pts));
1066
1067 return NO_ERROR;
1068}
1069
1070status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
1071 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
1072
1073 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
1074 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
1075 target);
1076
1077 if (!(target == TimedAudioTrack::LOCAL_TIME ||
1078 target == TimedAudioTrack::COMMON_TIME)) {
1079 return BAD_VALUE;
1080 }
1081
1082 Mutex::Autolock lock(mMediaTimeTransformLock);
1083 mMediaTimeTransform = xform;
1084 mMediaTimeTransformTarget = target;
1085 mMediaTimeTransformValid = true;
1086
1087 return NO_ERROR;
1088}
1089
1090#define min(a, b) ((a) < (b) ? (a) : (b))
1091
1092// implementation of getNextBuffer for tracks whose buffers have timestamps
1093status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
1094 AudioBufferProvider::Buffer* buffer, int64_t pts)
1095{
1096 if (pts == AudioBufferProvider::kInvalidPTS) {
1097 buffer->raw = NULL;
1098 buffer->frameCount = 0;
1099 mTimedAudioOutputOnTime = false;
1100 return INVALID_OPERATION;
1101 }
1102
1103 Mutex::Autolock _l(mTimedBufferQueueLock);
1104
1105 ALOG_ASSERT(!mQueueHeadInFlight,
1106 "getNextBuffer called without releaseBuffer!");
1107
1108 while (true) {
1109
1110 // if we have no timed buffers, then fail
1111 if (mTimedBufferQueue.isEmpty()) {
1112 buffer->raw = NULL;
1113 buffer->frameCount = 0;
1114 return NOT_ENOUGH_DATA;
1115 }
1116
1117 TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1118
1119 // calculate the PTS of the head of the timed buffer queue expressed in
1120 // local time
1121 int64_t headLocalPTS;
1122 {
1123 Mutex::Autolock mttLock(mMediaTimeTransformLock);
1124
1125 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
1126
1127 if (mMediaTimeTransform.a_to_b_denom == 0) {
1128 // the transform represents a pause, so yield silence
1129 timedYieldSilence_l(buffer->frameCount, buffer);
1130 return NO_ERROR;
1131 }
1132
1133 int64_t transformedPTS;
1134 if (!mMediaTimeTransform.doForwardTransform(head.pts(),
1135 &transformedPTS)) {
1136 // the transform failed. this shouldn't happen, but if it does
1137 // then just drop this buffer
1138 ALOGW("timedGetNextBuffer transform failed");
1139 buffer->raw = NULL;
1140 buffer->frameCount = 0;
1141 trimTimedBufferQueueHead_l("getNextBuffer; no transform");
1142 return NO_ERROR;
1143 }
1144
1145 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
1146 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
1147 &headLocalPTS)) {
1148 buffer->raw = NULL;
1149 buffer->frameCount = 0;
1150 return INVALID_OPERATION;
1151 }
1152 } else {
1153 headLocalPTS = transformedPTS;
1154 }
1155 }
1156
1157 // adjust the head buffer's PTS to reflect the portion of the head buffer
1158 // that has already been consumed
1159 int64_t effectivePTS = headLocalPTS +
1160 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
1161
1162 // Calculate the delta in samples between the head of the input buffer
1163 // queue and the start of the next output buffer that will be written.
1164 // If the transformation fails because of over or underflow, it means
1165 // that the sample's position in the output stream is so far out of
1166 // whack that it should just be dropped.
1167 int64_t sampleDelta;
1168 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
1169 ALOGV("*** head buffer is too far from PTS: dropped buffer");
1170 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from"
1171 " mix");
1172 continue;
1173 }
1174 if (!mLocalTimeToSampleTransform.doForwardTransform(
1175 (effectivePTS - pts) << 32, &sampleDelta)) {
1176 ALOGV("*** too late during sample rate transform: dropped buffer");
1177 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
1178 continue;
1179 }
1180
1181 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld"
1182 " sampleDelta=[%d.%08x]",
1183 head.pts(), head.position(), pts,
1184 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1)
1185 + (sampleDelta >> 32)),
1186 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
1187
1188 // if the delta between the ideal placement for the next input sample and
1189 // the current output position is within this threshold, then we will
1190 // concatenate the next input samples to the previous output
1191 const int64_t kSampleContinuityThreshold =
1192 (static_cast<int64_t>(sampleRate()) << 32) / 250;
1193
1194 // if this is the first buffer of audio that we're emitting from this track
1195 // then it should be almost exactly on time.
1196 const int64_t kSampleStartupThreshold = 1LL << 32;
1197
1198 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
1199 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
1200 // the next input is close enough to being on time, so concatenate it
1201 // with the last output
1202 timedYieldSamples_l(buffer);
1203
1204 ALOGVV("*** on time: head.pos=%d frameCount=%u",
1205 head.position(), buffer->frameCount);
1206 return NO_ERROR;
1207 }
1208
1209 // Looks like our output is not on time. Reset our on timed status.
1210 // Next time we mix samples from our input queue, then should be within
1211 // the StartupThreshold.
1212 mTimedAudioOutputOnTime = false;
1213 if (sampleDelta > 0) {
1214 // the gap between the current output position and the proper start of
1215 // the next input sample is too big, so fill it with silence
1216 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
1217
1218 timedYieldSilence_l(framesUntilNextInput, buffer);
1219 ALOGV("*** silence: frameCount=%u", buffer->frameCount);
1220 return NO_ERROR;
1221 } else {
1222 // the next input sample is late
1223 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
1224 size_t onTimeSamplePosition =
1225 head.position() + lateFrames * mFrameSize;
1226
1227 if (onTimeSamplePosition > head.buffer()->size()) {
1228 // all the remaining samples in the head are too late, so
1229 // drop it and move on
1230 ALOGV("*** too late: dropped buffer");
1231 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
1232 continue;
1233 } else {
1234 // skip over the late samples
1235 head.setPosition(onTimeSamplePosition);
1236
1237 // yield the available samples
1238 timedYieldSamples_l(buffer);
1239
1240 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
1241 return NO_ERROR;
1242 }
1243 }
1244 }
1245}
1246
1247// Yield samples from the timed buffer queue head up to the given output
1248// buffer's capacity.
1249//
1250// Caller must hold mTimedBufferQueueLock
1251void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l(
1252 AudioBufferProvider::Buffer* buffer) {
1253
1254 const TimedBuffer& head = mTimedBufferQueue[0];
1255
1256 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
1257 head.position());
1258
1259 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
1260 mFrameSize);
1261 size_t framesRequested = buffer->frameCount;
1262 buffer->frameCount = min(framesLeftInHead, framesRequested);
1263
1264 mQueueHeadInFlight = true;
1265 mTimedAudioOutputOnTime = true;
1266}
1267
1268// Yield samples of silence up to the given output buffer's capacity
1269//
1270// Caller must hold mTimedBufferQueueLock
1271void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l(
1272 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
1273
1274 // lazily allocate a buffer filled with silence
1275 if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
1276 delete [] mTimedSilenceBuffer;
1277 mTimedSilenceBufferSize = numFrames * mFrameSize;
1278 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
1279 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
1280 }
1281
1282 buffer->raw = mTimedSilenceBuffer;
1283 size_t framesRequested = buffer->frameCount;
1284 buffer->frameCount = min(numFrames, framesRequested);
1285
1286 mTimedAudioOutputOnTime = false;
1287}
1288
1289// AudioBufferProvider interface
1290void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
1291 AudioBufferProvider::Buffer* buffer) {
1292
1293 Mutex::Autolock _l(mTimedBufferQueueLock);
1294
1295 // If the buffer which was just released is part of the buffer at the head
1296 // of the queue, be sure to update the amt of the buffer which has been
1297 // consumed. If the buffer being returned is not part of the head of the
1298 // queue, its either because the buffer is part of the silence buffer, or
1299 // because the head of the timed queue was trimmed after the mixer called
1300 // getNextBuffer but before the mixer called releaseBuffer.
1301 if (buffer->raw == mTimedSilenceBuffer) {
1302 ALOG_ASSERT(!mQueueHeadInFlight,
1303 "Queue head in flight during release of silence buffer!");
1304 goto done;
1305 }
1306
1307 ALOG_ASSERT(mQueueHeadInFlight,
1308 "TimedTrack::releaseBuffer of non-silence buffer, but no queue"
1309 " head in flight.");
1310
1311 if (mTimedBufferQueue.size()) {
1312 TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
1313
1314 void* start = head.buffer()->pointer();
1315 void* end = reinterpret_cast<void*>(
1316 reinterpret_cast<uint8_t*>(head.buffer()->pointer())
1317 + head.buffer()->size());
1318
1319 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end),
1320 "released buffer not within the head of the timed buffer"
1321 " queue; qHead = [%p, %p], released buffer = %p",
1322 start, end, buffer->raw);
1323
1324 head.setPosition(head.position() +
1325 (buffer->frameCount * mFrameSize));
1326 mQueueHeadInFlight = false;
1327
1328 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
1329 "Bad bookkeeping during releaseBuffer! Should have at"
1330 " least %u queued frames, but we think we have only %u",
1331 buffer->frameCount, mFramesPendingInQueue);
1332
1333 mFramesPendingInQueue -= buffer->frameCount;
1334
1335 if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
1336 || mTrimQueueHeadOnRelease) {
1337 trimTimedBufferQueueHead_l("releaseBuffer");
1338 mTrimQueueHeadOnRelease = false;
1339 }
1340 } else {
1341 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no"
1342 " buffers in the timed buffer queue");
1343 }
1344
1345done:
1346 buffer->raw = 0;
1347 buffer->frameCount = 0;
1348}
1349
1350size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
1351 Mutex::Autolock _l(mTimedBufferQueueLock);
1352 return mFramesPendingInQueue;
1353}
1354
1355AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
1356 : mPTS(0), mPosition(0) {}
1357
1358AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
1359 const sp<IMemory>& buffer, int64_t pts)
1360 : mBuffer(buffer), mPTS(pts), mPosition(0) {}
1361
1362
1363// ----------------------------------------------------------------------------
1364
1365AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
1366 PlaybackThread *playbackThread,
1367 DuplicatingThread *sourceThread,
1368 uint32_t sampleRate,
1369 audio_format_t format,
1370 audio_channel_mask_t channelMask,
1371 size_t frameCount)
1372 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
1373 NULL, 0, IAudioFlinger::TRACK_DEFAULT),
Glenn Kastene3aa6592012-12-04 12:22:46 -08001374 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
Eric Laurent81784c32012-11-19 14:55:58 -08001375{
1376
1377 if (mCblk != NULL) {
Eric Laurent81784c32012-11-19 14:55:58 -08001378 mOutBuffer.frameCount = 0;
1379 playbackThread->mTracks.add(this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001380 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
1381 "mCblk->frameCount_ %u, mChannelMask 0x%08x mBufferEnd %p",
1382 mCblk, mBuffer,
1383 mCblk->frameCount_, mChannelMask, mBufferEnd);
1384 // since client and server are in the same process,
1385 // the buffer has the same virtual address on both sides
1386 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
Eric Laurent81784c32012-11-19 14:55:58 -08001387 } else {
1388 ALOGW("Error creating output track on thread %p", playbackThread);
1389 }
1390}
1391
1392AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
1393{
1394 clearBufferQueue();
Glenn Kastene3aa6592012-12-04 12:22:46 -08001395 delete mClientProxy;
1396 // superclass destructor will now delete the server proxy and shared memory both refer to
Eric Laurent81784c32012-11-19 14:55:58 -08001397}
1398
1399status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
1400 int triggerSession)
1401{
1402 status_t status = Track::start(event, triggerSession);
1403 if (status != NO_ERROR) {
1404 return status;
1405 }
1406
1407 mActive = true;
1408 mRetryCount = 127;
1409 return status;
1410}
1411
1412void AudioFlinger::PlaybackThread::OutputTrack::stop()
1413{
1414 Track::stop();
1415 clearBufferQueue();
1416 mOutBuffer.frameCount = 0;
1417 mActive = false;
1418}
1419
1420bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
1421{
1422 Buffer *pInBuffer;
1423 Buffer inBuffer;
1424 uint32_t channelCount = mChannelCount;
1425 bool outputBufferFull = false;
1426 inBuffer.frameCount = frames;
1427 inBuffer.i16 = data;
1428
1429 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
1430
1431 if (!mActive && frames != 0) {
1432 start();
1433 sp<ThreadBase> thread = mThread.promote();
1434 if (thread != 0) {
1435 MixerThread *mixerThread = (MixerThread *)thread.get();
1436 if (mFrameCount > frames) {
1437 if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1438 uint32_t startFrames = (mFrameCount - frames);
1439 pInBuffer = new Buffer;
1440 pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
1441 pInBuffer->frameCount = startFrames;
1442 pInBuffer->i16 = pInBuffer->mBuffer;
1443 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
1444 mBufferQueue.add(pInBuffer);
1445 } else {
1446 ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
1447 }
1448 }
1449 }
1450 }
1451
1452 while (waitTimeLeftMs) {
1453 // First write pending buffers, then new data
1454 if (mBufferQueue.size()) {
1455 pInBuffer = mBufferQueue.itemAt(0);
1456 } else {
1457 pInBuffer = &inBuffer;
1458 }
1459
1460 if (pInBuffer->frameCount == 0) {
1461 break;
1462 }
1463
1464 if (mOutBuffer.frameCount == 0) {
1465 mOutBuffer.frameCount = pInBuffer->frameCount;
1466 nsecs_t startTime = systemTime();
1467 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
1468 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
1469 mThread.unsafe_get());
1470 outputBufferFull = true;
1471 break;
1472 }
1473 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
1474 if (waitTimeLeftMs >= waitTimeMs) {
1475 waitTimeLeftMs -= waitTimeMs;
1476 } else {
1477 waitTimeLeftMs = 0;
1478 }
1479 }
1480
1481 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
1482 pInBuffer->frameCount;
1483 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
Glenn Kastene3aa6592012-12-04 12:22:46 -08001484 mClientProxy->stepUser(outFrames);
Eric Laurent81784c32012-11-19 14:55:58 -08001485 pInBuffer->frameCount -= outFrames;
1486 pInBuffer->i16 += outFrames * channelCount;
1487 mOutBuffer.frameCount -= outFrames;
1488 mOutBuffer.i16 += outFrames * channelCount;
1489
1490 if (pInBuffer->frameCount == 0) {
1491 if (mBufferQueue.size()) {
1492 mBufferQueue.removeAt(0);
1493 delete [] pInBuffer->mBuffer;
1494 delete pInBuffer;
1495 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
1496 mThread.unsafe_get(), mBufferQueue.size());
1497 } else {
1498 break;
1499 }
1500 }
1501 }
1502
1503 // If we could not write all frames, allocate a buffer and queue it for next time.
1504 if (inBuffer.frameCount) {
1505 sp<ThreadBase> thread = mThread.promote();
1506 if (thread != 0 && !thread->standby()) {
1507 if (mBufferQueue.size() < kMaxOverFlowBuffers) {
1508 pInBuffer = new Buffer;
1509 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
1510 pInBuffer->frameCount = inBuffer.frameCount;
1511 pInBuffer->i16 = pInBuffer->mBuffer;
1512 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
1513 sizeof(int16_t));
1514 mBufferQueue.add(pInBuffer);
1515 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
1516 mThread.unsafe_get(), mBufferQueue.size());
1517 } else {
1518 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
1519 mThread.unsafe_get(), this);
1520 }
1521 }
1522 }
1523
1524 // Calling write() with a 0 length buffer, means that no more data will be written:
1525 // If no more buffers are pending, fill output track buffer to make sure it is started
1526 // by output mixer.
1527 if (frames == 0 && mBufferQueue.size() == 0) {
1528 if (mCblk->user < mFrameCount) {
1529 frames = mFrameCount - mCblk->user;
1530 pInBuffer = new Buffer;
1531 pInBuffer->mBuffer = new int16_t[frames * channelCount];
1532 pInBuffer->frameCount = frames;
1533 pInBuffer->i16 = pInBuffer->mBuffer;
1534 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
1535 mBufferQueue.add(pInBuffer);
1536 } else if (mActive) {
1537 stop();
1538 }
1539 }
1540
1541 return outputBufferFull;
1542}
1543
1544status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
1545 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
1546{
Eric Laurent81784c32012-11-19 14:55:58 -08001547 audio_track_cblk_t* cblk = mCblk;
1548 uint32_t framesReq = buffer->frameCount;
1549
1550 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
1551 buffer->frameCount = 0;
1552
Glenn Kastene3aa6592012-12-04 12:22:46 -08001553 size_t framesAvail;
1554 {
Eric Laurent81784c32012-11-19 14:55:58 -08001555 Mutex::Autolock _l(cblk->lock);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001556
1557 // read the server count again
1558 while (!(framesAvail = mClientProxy->framesAvailable_l())) {
1559 if (CC_UNLIKELY(!mActive)) {
Eric Laurent81784c32012-11-19 14:55:58 -08001560 ALOGV("Not active and NO_MORE_BUFFERS");
1561 return NO_MORE_BUFFERS;
1562 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001563 status_t result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
Eric Laurent81784c32012-11-19 14:55:58 -08001564 if (result != NO_ERROR) {
1565 return NO_MORE_BUFFERS;
1566 }
Eric Laurent81784c32012-11-19 14:55:58 -08001567 }
1568 }
1569
Eric Laurent81784c32012-11-19 14:55:58 -08001570 if (framesReq > framesAvail) {
1571 framesReq = framesAvail;
1572 }
1573
1574 uint32_t u = cblk->user;
1575 uint32_t bufferEnd = cblk->userBase + mFrameCount;
1576
1577 if (framesReq > bufferEnd - u) {
1578 framesReq = bufferEnd - u;
1579 }
1580
1581 buffer->frameCount = framesReq;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001582 buffer->raw = mClientProxy->buffer(u);
Eric Laurent81784c32012-11-19 14:55:58 -08001583 return NO_ERROR;
1584}
1585
1586
1587void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
1588{
1589 size_t size = mBufferQueue.size();
1590
1591 for (size_t i = 0; i < size; i++) {
1592 Buffer *pBuffer = mBufferQueue.itemAt(i);
1593 delete [] pBuffer->mBuffer;
1594 delete pBuffer;
1595 }
1596 mBufferQueue.clear();
1597}
1598
1599
1600// ----------------------------------------------------------------------------
1601// Record
1602// ----------------------------------------------------------------------------
1603
1604AudioFlinger::RecordHandle::RecordHandle(
1605 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
1606 : BnAudioRecord(),
1607 mRecordTrack(recordTrack)
1608{
1609}
1610
1611AudioFlinger::RecordHandle::~RecordHandle() {
1612 stop_nonvirtual();
1613 mRecordTrack->destroy();
1614}
1615
1616sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
1617 return mRecordTrack->getCblk();
1618}
1619
1620status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
1621 int triggerSession) {
1622 ALOGV("RecordHandle::start()");
1623 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
1624}
1625
1626void AudioFlinger::RecordHandle::stop() {
1627 stop_nonvirtual();
1628}
1629
1630void AudioFlinger::RecordHandle::stop_nonvirtual() {
1631 ALOGV("RecordHandle::stop()");
1632 mRecordTrack->stop();
1633}
1634
1635status_t AudioFlinger::RecordHandle::onTransact(
1636 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
1637{
1638 return BnAudioRecord::onTransact(code, data, reply, flags);
1639}
1640
1641// ----------------------------------------------------------------------------
1642
1643// RecordTrack constructor must be called with AudioFlinger::mLock held
1644AudioFlinger::RecordThread::RecordTrack::RecordTrack(
1645 RecordThread *thread,
1646 const sp<Client>& client,
1647 uint32_t sampleRate,
1648 audio_format_t format,
1649 audio_channel_mask_t channelMask,
1650 size_t frameCount,
1651 int sessionId)
1652 : TrackBase(thread, client, sampleRate, format,
Glenn Kastene3aa6592012-12-04 12:22:46 -08001653 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
Eric Laurent81784c32012-11-19 14:55:58 -08001654 mOverflow(false)
1655{
1656 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
1657}
1658
1659AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
1660{
1661 ALOGV("%s", __func__);
1662}
1663
1664// AudioBufferProvider interface
1665status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
1666 int64_t pts)
1667{
1668 audio_track_cblk_t* cblk = this->cblk();
1669 uint32_t framesAvail;
1670 uint32_t framesReq = buffer->frameCount;
1671
1672 // Check if last stepServer failed, try to step now
1673 if (mStepServerFailed) {
1674 if (!step()) {
1675 goto getNextBuffer_exit;
1676 }
1677 ALOGV("stepServer recovered");
1678 mStepServerFailed = false;
1679 }
1680
1681 // FIXME lock is not actually held, so overrun is possible
Glenn Kastene3aa6592012-12-04 12:22:46 -08001682 framesAvail = mServerProxy->framesAvailableIn_l();
Eric Laurent81784c32012-11-19 14:55:58 -08001683
1684 if (CC_LIKELY(framesAvail)) {
1685 uint32_t s = cblk->server;
1686 uint32_t bufferEnd = cblk->serverBase + mFrameCount;
1687
1688 if (framesReq > framesAvail) {
1689 framesReq = framesAvail;
1690 }
1691 if (framesReq > bufferEnd - s) {
1692 framesReq = bufferEnd - s;
1693 }
1694
1695 buffer->raw = getBuffer(s, framesReq);
1696 buffer->frameCount = framesReq;
1697 return NO_ERROR;
1698 }
1699
1700getNextBuffer_exit:
1701 buffer->raw = NULL;
1702 buffer->frameCount = 0;
1703 return NOT_ENOUGH_DATA;
1704}
1705
1706status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
1707 int triggerSession)
1708{
1709 sp<ThreadBase> thread = mThread.promote();
1710 if (thread != 0) {
1711 RecordThread *recordThread = (RecordThread *)thread.get();
1712 return recordThread->start(this, event, triggerSession);
1713 } else {
1714 return BAD_VALUE;
1715 }
1716}
1717
1718void AudioFlinger::RecordThread::RecordTrack::stop()
1719{
1720 sp<ThreadBase> thread = mThread.promote();
1721 if (thread != 0) {
1722 RecordThread *recordThread = (RecordThread *)thread.get();
1723 recordThread->mLock.lock();
1724 bool doStop = recordThread->stop_l(this);
1725 if (doStop) {
1726 TrackBase::reset();
1727 // Force overrun condition to avoid false overrun callback until first data is
1728 // read from buffer
1729 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags);
1730 }
1731 recordThread->mLock.unlock();
1732 if (doStop) {
1733 AudioSystem::stopInput(recordThread->id());
1734 }
1735 }
1736}
1737
1738void AudioFlinger::RecordThread::RecordTrack::destroy()
1739{
1740 // see comments at AudioFlinger::PlaybackThread::Track::destroy()
1741 sp<RecordTrack> keep(this);
1742 {
1743 sp<ThreadBase> thread = mThread.promote();
1744 if (thread != 0) {
1745 if (mState == ACTIVE || mState == RESUMING) {
1746 AudioSystem::stopInput(thread->id());
1747 }
1748 AudioSystem::releaseInput(thread->id());
1749 Mutex::Autolock _l(thread->mLock);
1750 RecordThread *recordThread = (RecordThread *) thread.get();
1751 recordThread->destroyTrack_l(this);
1752 }
1753 }
1754}
1755
1756
1757/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
1758{
Glenn Kastene3aa6592012-12-04 12:22:46 -08001759 result.append(" Clien Fmt Chn mask Session Step S Serv User FrameCount\n");
Eric Laurent81784c32012-11-19 14:55:58 -08001760}
1761
1762void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
1763{
Glenn Kastene3aa6592012-12-04 12:22:46 -08001764 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %08x %05d\n",
Eric Laurent81784c32012-11-19 14:55:58 -08001765 (mClient == 0) ? getpid_cached : mClient->pid(),
1766 mFormat,
1767 mChannelMask,
1768 mSessionId,
1769 mStepCount,
1770 mState,
Eric Laurent81784c32012-11-19 14:55:58 -08001771 mCblk->server,
1772 mCblk->user,
1773 mFrameCount);
1774}
1775
Eric Laurent81784c32012-11-19 14:55:58 -08001776}; // namespace android