Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1 | /* |
| 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 | |
| 50 | namespace android { |
| 51 | |
| 52 | // ---------------------------------------------------------------------------- |
| 53 | // TrackBase |
| 54 | // ---------------------------------------------------------------------------- |
| 55 | |
| 56 | // TrackBase constructor must be called with AudioFlinger::mLock held |
| 57 | AudioFlinger::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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 65 | int sessionId, |
| 66 | bool isOut) |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 67 | : 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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 83 | mSessionId(sessionId), |
| 84 | mIsOut(isOut), |
| 85 | mServerProxy(NULL) |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 86 | { |
| 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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 111 | // this syntax avoids calling the audio_track_cblk_t constructor twice |
| 112 | mCblk = (audio_track_cblk_t *) new uint8_t[size]; |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 113 | // 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 Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 121 | // 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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 136 | mServerProxy = new ServerProxy(mCblk, mBuffer, frameCount, mFrameSize, isOut); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 137 | } |
| 138 | } |
| 139 | |
| 140 | AudioFlinger::ThreadBase::TrackBase::~TrackBase() |
| 141 | { |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 142 | // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference |
| 143 | delete mServerProxy; |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 144 | 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 |
| 165 | void 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 | |
| 174 | bool AudioFlinger::ThreadBase::TrackBase::step() { |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 175 | bool result = mServerProxy->step(mStepCount); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 176 | if (!result) { |
| 177 | ALOGV("stepServer failed acquiring cblk mutex"); |
| 178 | mStepServerFailed = true; |
| 179 | } |
| 180 | return result; |
| 181 | } |
| 182 | |
| 183 | void 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 | |
| 194 | uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 195 | return mServerProxy->getSampleRate(); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 196 | } |
| 197 | |
| 198 | void* 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 | |
| 214 | status_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 | |
| 224 | AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) |
| 225 | : BnAudioTrack(), |
| 226 | mTrack(track) |
| 227 | { |
| 228 | } |
| 229 | |
| 230 | AudioFlinger::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 | |
| 238 | sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { |
| 239 | return mTrack->getCblk(); |
| 240 | } |
| 241 | |
| 242 | status_t AudioFlinger::TrackHandle::start() { |
| 243 | return mTrack->start(); |
| 244 | } |
| 245 | |
| 246 | void AudioFlinger::TrackHandle::stop() { |
| 247 | mTrack->stop(); |
| 248 | } |
| 249 | |
| 250 | void AudioFlinger::TrackHandle::flush() { |
| 251 | mTrack->flush(); |
| 252 | } |
| 253 | |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 254 | void AudioFlinger::TrackHandle::pause() { |
| 255 | mTrack->pause(); |
| 256 | } |
| 257 | |
| 258 | status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) |
| 259 | { |
| 260 | return mTrack->attachAuxEffect(EffectId); |
| 261 | } |
| 262 | |
| 263 | status_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 | |
| 273 | status_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 | |
| 283 | status_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 | |
| 295 | status_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 |
| 304 | AudioFlinger::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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 316 | sessionId, true /*isOut*/), |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 317 | 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 Kasten | 5736c35 | 2012-12-04 12:12:34 -0800 | [diff] [blame] | 329 | mCachedVolume(1.0), |
| 330 | mIsInvalid(false) |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 331 | { |
| 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 | |
| 360 | AudioFlinger::PlaybackThread::Track::~Track() |
| 361 | { |
| 362 | ALOGV("PlaybackThread::Track destructor"); |
| 363 | } |
| 364 | |
| 365 | void 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 Kasten | e4756fe | 2012-11-29 13:38:14 -0800 | [diff] [blame] | 399 | result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S F SRate " |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 400 | "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); |
| 401 | } |
| 402 | |
| 403 | void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) |
| 404 | { |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 405 | uint32_t vlr = mServerProxy->getVolumeLR(); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 406 | 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 Kasten | e4756fe | 2012-11-29 13:38:14 -0800 | [diff] [blame] | 463 | snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %5u %5.2g %5.2g " |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 464 | "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 Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 473 | mFillingUpStatus, |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 474 | mServerProxy->getSampleRate(), |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 475 | 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 |
| 487 | status_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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 509 | framesReady = mServerProxy->framesReady(); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 510 | |
| 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 | |
| 528 | getNextBuffer_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. |
| 543 | size_t AudioFlinger::PlaybackThread::Track::framesReady() const { |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 544 | return mServerProxy->framesReady(); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 545 | } |
| 546 | |
| 547 | // Don't call for fast tracks; the framesReady() could result in priority inversion |
| 548 | bool 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 | |
| 562 | status_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 Kasten | 7f5d335 | 2013-02-15 23:55:04 +0000 | [diff] [blame] | 572 | thread->mNBLogWriter->logf("start mName=%d", mName); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 573 | 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 | |
| 609 | void 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 Kasten | 7f5d335 | 2013-02-15 23:55:04 +0000 | [diff] [blame] | 615 | thread->mNBLogWriter->logf("stop mName=%d", mName); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 616 | 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 | |
| 646 | void 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 Kasten | 7f5d335 | 2013-02-15 23:55:04 +0000 | [diff] [blame] | 652 | thread->mNBLogWriter->logf("pause mName=%d", mName); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 653 | 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 | |
| 670 | void 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 Kasten | 7f5d335 | 2013-02-15 23:55:04 +0000 | [diff] [blame] | 676 | thread->mNBLogWriter->logf("flush mName=%d", mName); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 677 | 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 | |
| 695 | void 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 Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 713 | status_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 | |
| 762 | void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) |
| 763 | { |
| 764 | mAuxEffectId = EffectId; |
| 765 | mAuxBuffer = buffer; |
| 766 | } |
| 767 | |
| 768 | bool 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 | |
| 788 | void 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 | |
| 801 | uint32_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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 805 | uint32_t vlr = mServerProxy->getVolumeLR(); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 806 | 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 | |
| 826 | status_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 Kasten | 5736c35 | 2012-12-04 12:12:34 -0800 | [diff] [blame] | 840 | void 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 Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 848 | // ---------------------------------------------------------------------------- |
| 849 | |
| 850 | sp<AudioFlinger::PlaybackThread::TimedTrack> |
| 851 | AudioFlinger::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 | |
| 869 | AudioFlinger::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 | |
| 907 | AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { |
| 908 | mClient->releaseTimedTrack(); |
| 909 | delete [] mTimedSilenceBuffer; |
| 910 | } |
| 911 | |
| 912 | status_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 |
| 941 | void 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 | |
| 1020 | void 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 | |
| 1030 | void 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 | |
| 1052 | status_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 | |
| 1070 | status_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 |
| 1093 | status_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 |
| 1251 | void 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 |
| 1271 | void 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 |
| 1290 | void 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 | |
| 1345 | done: |
| 1346 | buffer->raw = 0; |
| 1347 | buffer->frameCount = 0; |
| 1348 | } |
| 1349 | |
| 1350 | size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { |
| 1351 | Mutex::Autolock _l(mTimedBufferQueueLock); |
| 1352 | return mFramesPendingInQueue; |
| 1353 | } |
| 1354 | |
| 1355 | AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() |
| 1356 | : mPTS(0), mPosition(0) {} |
| 1357 | |
| 1358 | AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( |
| 1359 | const sp<IMemory>& buffer, int64_t pts) |
| 1360 | : mBuffer(buffer), mPTS(pts), mPosition(0) {} |
| 1361 | |
| 1362 | |
| 1363 | // ---------------------------------------------------------------------------- |
| 1364 | |
| 1365 | AudioFlinger::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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1374 | mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1375 | { |
| 1376 | |
| 1377 | if (mCblk != NULL) { |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1378 | mOutBuffer.frameCount = 0; |
| 1379 | playbackThread->mTracks.add(this); |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1380 | 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 Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1387 | } else { |
| 1388 | ALOGW("Error creating output track on thread %p", playbackThread); |
| 1389 | } |
| 1390 | } |
| 1391 | |
| 1392 | AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() |
| 1393 | { |
| 1394 | clearBufferQueue(); |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1395 | delete mClientProxy; |
| 1396 | // superclass destructor will now delete the server proxy and shared memory both refer to |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1397 | } |
| 1398 | |
| 1399 | status_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 | |
| 1412 | void AudioFlinger::PlaybackThread::OutputTrack::stop() |
| 1413 | { |
| 1414 | Track::stop(); |
| 1415 | clearBufferQueue(); |
| 1416 | mOutBuffer.frameCount = 0; |
| 1417 | mActive = false; |
| 1418 | } |
| 1419 | |
| 1420 | bool 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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1484 | mClientProxy->stepUser(outFrames); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1485 | 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 | |
| 1544 | status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( |
| 1545 | AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) |
| 1546 | { |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1547 | 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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1553 | size_t framesAvail; |
| 1554 | { |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1555 | Mutex::Autolock _l(cblk->lock); |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1556 | |
| 1557 | // read the server count again |
| 1558 | while (!(framesAvail = mClientProxy->framesAvailable_l())) { |
| 1559 | if (CC_UNLIKELY(!mActive)) { |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1560 | ALOGV("Not active and NO_MORE_BUFFERS"); |
| 1561 | return NO_MORE_BUFFERS; |
| 1562 | } |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1563 | status_t result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1564 | if (result != NO_ERROR) { |
| 1565 | return NO_MORE_BUFFERS; |
| 1566 | } |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1567 | } |
| 1568 | } |
| 1569 | |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1570 | 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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1582 | buffer->raw = mClientProxy->buffer(u); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1583 | return NO_ERROR; |
| 1584 | } |
| 1585 | |
| 1586 | |
| 1587 | void 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 | |
| 1604 | AudioFlinger::RecordHandle::RecordHandle( |
| 1605 | const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) |
| 1606 | : BnAudioRecord(), |
| 1607 | mRecordTrack(recordTrack) |
| 1608 | { |
| 1609 | } |
| 1610 | |
| 1611 | AudioFlinger::RecordHandle::~RecordHandle() { |
| 1612 | stop_nonvirtual(); |
| 1613 | mRecordTrack->destroy(); |
| 1614 | } |
| 1615 | |
| 1616 | sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { |
| 1617 | return mRecordTrack->getCblk(); |
| 1618 | } |
| 1619 | |
| 1620 | status_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 | |
| 1626 | void AudioFlinger::RecordHandle::stop() { |
| 1627 | stop_nonvirtual(); |
| 1628 | } |
| 1629 | |
| 1630 | void AudioFlinger::RecordHandle::stop_nonvirtual() { |
| 1631 | ALOGV("RecordHandle::stop()"); |
| 1632 | mRecordTrack->stop(); |
| 1633 | } |
| 1634 | |
| 1635 | status_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 |
| 1644 | AudioFlinger::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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1653 | channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1654 | mOverflow(false) |
| 1655 | { |
| 1656 | ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); |
| 1657 | } |
| 1658 | |
| 1659 | AudioFlinger::RecordThread::RecordTrack::~RecordTrack() |
| 1660 | { |
| 1661 | ALOGV("%s", __func__); |
| 1662 | } |
| 1663 | |
| 1664 | // AudioBufferProvider interface |
| 1665 | status_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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1682 | framesAvail = mServerProxy->framesAvailableIn_l(); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1683 | |
| 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 | |
| 1700 | getNextBuffer_exit: |
| 1701 | buffer->raw = NULL; |
| 1702 | buffer->frameCount = 0; |
| 1703 | return NOT_ENOUGH_DATA; |
| 1704 | } |
| 1705 | |
| 1706 | status_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 | |
| 1718 | void 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 | |
| 1738 | void 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 Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1759 | result.append(" Clien Fmt Chn mask Session Step S Serv User FrameCount\n"); |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1760 | } |
| 1761 | |
| 1762 | void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) |
| 1763 | { |
Glenn Kasten | e3aa659 | 2012-12-04 12:22:46 -0800 | [diff] [blame] | 1764 | snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %08x %08x %05d\n", |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1765 | (mClient == 0) ? getpid_cached : mClient->pid(), |
| 1766 | mFormat, |
| 1767 | mChannelMask, |
| 1768 | mSessionId, |
| 1769 | mStepCount, |
| 1770 | mState, |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1771 | mCblk->server, |
| 1772 | mCblk->user, |
| 1773 | mFrameCount); |
| 1774 | } |
| 1775 | |
Eric Laurent | 81784c3 | 2012-11-19 14:55:58 -0800 | [diff] [blame] | 1776 | }; // namespace android |