Tracks.cpp revision 030033342a6ea17003e6af38a56c7edc6d2ead01
17abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao/* 27abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** 37abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** Copyright 2012, The Android Open Source Project 47abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** 57abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** Licensed under the Apache License, Version 2.0 (the "License"); 67abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** you may not use this file except in compliance with the License. 77abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** You may obtain a copy of the License at 87abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** 97abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** http://www.apache.org/licenses/LICENSE-2.0 107abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** 117abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** Unless required by applicable law or agreed to in writing, software 127abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** distributed under the License is distributed on an "AS IS" BASIS, 137abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 147abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** See the License for the specific language governing permissions and 157abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao** limitations under the License. 167abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao*/ 177abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 187abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 197abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#define LOG_TAG "AudioFlinger" 207abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao//#define LOG_NDEBUG 0 217abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 227abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include "Configuration.h" 237abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include <math.h> 247abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include <utils/Log.h> 257abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 267abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include <private/media/AudioTrackShared.h> 277abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 287abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include <common_time/cc_helper.h> 297abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include <common_time/local_clock.h> 307abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 317abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include "AudioMixer.h" 327abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include "AudioFlinger.h" 337abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include "ServiceUtilities.h" 347abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 357abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include <media/nbaio/Pipe.h> 367abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#include <media/nbaio/PipeReader.h> 377abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 387abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// ---------------------------------------------------------------------------- 397abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 407abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// Note: the following macro is used for extremely verbose logging message. In 417abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 427abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// 0; but one side effect of this is to turn all LOGV's as well. Some messages 437abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// are so verbose that we want to suppress them even when we have ALOG_ASSERT 447abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// turned on. Do not uncomment the #def below unless you really know what you 457abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// are doing and want to see all of the extremely verbose messages. 467abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao//#define VERY_VERY_VERBOSE_LOGGING 477abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#ifdef VERY_VERY_VERBOSE_LOGGING 487abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#define ALOGVV ALOGV 497abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#else 507abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#define ALOGVV(a...) do { } while(0) 517abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao#endif 527abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 537abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liaonamespace android { 547abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 557abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// ---------------------------------------------------------------------------- 567abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// TrackBase 577abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// ---------------------------------------------------------------------------- 587abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 597abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liaostatic volatile int32_t nextTrackId = 55; 607abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 617abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao// TrackBase constructor must be called with AudioFlinger::mLock held 627abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei LiaoAudioFlinger::ThreadBase::TrackBase::TrackBase( 637abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao ThreadBase *thread, 647abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao const sp<Client>& client, 657abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao uint32_t sampleRate, 667abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao audio_format_t format, 677abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao audio_channel_mask_t channelMask, 687abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao size_t frameCount, 697abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao const sp<IMemory>& sharedBuffer, 707abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao int sessionId, 717abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao bool isOut) 727abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao : RefBase(), 737abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mThread(thread), 747abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mClient(client), 757abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mCblk(NULL), 767abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao // mBuffer 777abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mState(IDLE), 787abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mSampleRate(sampleRate), 797abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mFormat(format), 807abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mChannelMask(channelMask), 817abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mChannelCount(popcount(channelMask)), 827abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mFrameSize(audio_is_linear_pcm(format) ? 837abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 847abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mFrameCount(frameCount), 857abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mSessionId(sessionId), 867abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mIsOut(isOut), 877abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mServerProxy(NULL), 887abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mId(android_atomic_inc(&nextTrackId)), 897abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mTerminated(false) 907abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao{ 917abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao // client == 0 implies sharedBuffer == 0 927abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 937abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 947abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 957abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao sharedBuffer->size()); 967abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 977abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 987abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao size_t size = sizeof(audio_track_cblk_t); 997abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 1007abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao if (sharedBuffer == 0) { 1017abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao size += bufferSize; 1027abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao } 1037abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 1047abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao if (client != 0) { 1057abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mCblkMemory = client->heap()->allocate(size); 1067abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao if (mCblkMemory != 0) { 1077abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 1087abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao // can't assume mCblk != NULL 1097abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao } else { 1107abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao ALOGE("not enough memory for AudioTrack size=%u", size); 1117abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao client->heap()->dump("AudioTrack"); 1127abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao return; 1137abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao } 1147abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao } else { 1157abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao // this syntax avoids calling the audio_track_cblk_t constructor twice 1167abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao mCblk = (audio_track_cblk_t *) new uint8_t[size]; 1177abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao // assume mCblk != NULL 1187abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao } 1197abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao 1207abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao // construct the shared structure in-place. 1217abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao if (mCblk != NULL) { 1227abe37e4aee38cc79d91dd069a37d7e91d5bef53Shih-wei Liao new(mCblk) audio_track_cblk_t(); 123 // clear all buffers 124 mCblk->frameCount_ = frameCount; 125 if (sharedBuffer == 0) { 126 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 127 memset(mBuffer, 0, bufferSize); 128 } else { 129 mBuffer = sharedBuffer->pointer(); 130#if 0 131 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 132#endif 133 } 134 135#ifdef TEE_SINK 136 if (mTeeSinkTrackEnabled) { 137 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 138 if (pipeFormat != Format_Invalid) { 139 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 140 size_t numCounterOffers = 0; 141 const NBAIO_Format offers[1] = {pipeFormat}; 142 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 143 ALOG_ASSERT(index == 0); 144 PipeReader *pipeReader = new PipeReader(*pipe); 145 numCounterOffers = 0; 146 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 147 ALOG_ASSERT(index == 0); 148 mTeeSink = pipe; 149 mTeeSource = pipeReader; 150 } 151 } 152#endif 153 154 } 155} 156 157AudioFlinger::ThreadBase::TrackBase::~TrackBase() 158{ 159#ifdef TEE_SINK 160 dumpTee(-1, mTeeSource, mId); 161#endif 162 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 163 delete mServerProxy; 164 if (mCblk != NULL) { 165 if (mClient == 0) { 166 delete mCblk; 167 } else { 168 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 169 } 170 } 171 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 172 if (mClient != 0) { 173 // Client destructor must run with AudioFlinger mutex locked 174 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 175 // If the client's reference count drops to zero, the associated destructor 176 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 177 // relying on the automatic clear() at end of scope. 178 mClient.clear(); 179 } 180} 181 182// AudioBufferProvider interface 183// getNextBuffer() = 0; 184// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 185void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 186{ 187#ifdef TEE_SINK 188 if (mTeeSink != 0) { 189 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 190 } 191#endif 192 193 ServerProxy::Buffer buf; 194 buf.mFrameCount = buffer->frameCount; 195 buf.mRaw = buffer->raw; 196 buffer->frameCount = 0; 197 buffer->raw = NULL; 198 mServerProxy->releaseBuffer(&buf); 199} 200 201status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 202{ 203 mSyncEvents.add(event); 204 return NO_ERROR; 205} 206 207// ---------------------------------------------------------------------------- 208// Playback 209// ---------------------------------------------------------------------------- 210 211AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 212 : BnAudioTrack(), 213 mTrack(track) 214{ 215} 216 217AudioFlinger::TrackHandle::~TrackHandle() { 218 // just stop the track on deletion, associated resources 219 // will be freed from the main thread once all pending buffers have 220 // been played. Unless it's not in the active track list, in which 221 // case we free everything now... 222 mTrack->destroy(); 223} 224 225sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 226 return mTrack->getCblk(); 227} 228 229status_t AudioFlinger::TrackHandle::start() { 230 return mTrack->start(); 231} 232 233void AudioFlinger::TrackHandle::stop() { 234 mTrack->stop(); 235} 236 237void AudioFlinger::TrackHandle::flush() { 238 mTrack->flush(); 239} 240 241void AudioFlinger::TrackHandle::pause() { 242 mTrack->pause(); 243} 244 245status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 246{ 247 return mTrack->attachAuxEffect(EffectId); 248} 249 250status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 251 sp<IMemory>* buffer) { 252 if (!mTrack->isTimedTrack()) 253 return INVALID_OPERATION; 254 255 PlaybackThread::TimedTrack* tt = 256 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 257 return tt->allocateTimedBuffer(size, buffer); 258} 259 260status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 261 int64_t pts) { 262 if (!mTrack->isTimedTrack()) 263 return INVALID_OPERATION; 264 265 PlaybackThread::TimedTrack* tt = 266 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 267 return tt->queueTimedBuffer(buffer, pts); 268} 269 270status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 271 const LinearTransform& xform, int target) { 272 273 if (!mTrack->isTimedTrack()) 274 return INVALID_OPERATION; 275 276 PlaybackThread::TimedTrack* tt = 277 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 278 return tt->setMediaTimeTransform( 279 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 280} 281 282status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 283 return mTrack->setParameters(keyValuePairs); 284} 285 286status_t AudioFlinger::TrackHandle::onTransact( 287 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 288{ 289 return BnAudioTrack::onTransact(code, data, reply, flags); 290} 291 292// ---------------------------------------------------------------------------- 293 294// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 295AudioFlinger::PlaybackThread::Track::Track( 296 PlaybackThread *thread, 297 const sp<Client>& client, 298 audio_stream_type_t streamType, 299 uint32_t sampleRate, 300 audio_format_t format, 301 audio_channel_mask_t channelMask, 302 size_t frameCount, 303 const sp<IMemory>& sharedBuffer, 304 int sessionId, 305 IAudioFlinger::track_flags_t flags) 306 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 307 sessionId, true /*isOut*/), 308 mFillingUpStatus(FS_INVALID), 309 // mRetryCount initialized later when needed 310 mSharedBuffer(sharedBuffer), 311 mStreamType(streamType), 312 mName(-1), // see note below 313 mMainBuffer(thread->mixBuffer()), 314 mAuxBuffer(NULL), 315 mAuxEffectId(0), mHasVolumeController(false), 316 mPresentationCompleteFrames(0), 317 mFlags(flags), 318 mFastIndex(-1), 319 mCachedVolume(1.0), 320 mIsInvalid(false), 321 mAudioTrackServerProxy(NULL), 322 mResumeToStopping(false) 323{ 324 if (mCblk != NULL) { 325 if (sharedBuffer == 0) { 326 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 327 mFrameSize); 328 } else { 329 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 330 mFrameSize); 331 } 332 mServerProxy = mAudioTrackServerProxy; 333 // to avoid leaking a track name, do not allocate one unless there is an mCblk 334 mName = thread->getTrackName_l(channelMask, sessionId); 335 if (mName < 0) { 336 ALOGE("no more track names available"); 337 return; 338 } 339 // only allocate a fast track index if we were able to allocate a normal track name 340 if (flags & IAudioFlinger::TRACK_FAST) { 341 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 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 // Read the initial underruns because this field is never cleared by the fast mixer 351 mObservedUnderruns = thread->getFastTrackUnderruns(i); 352 thread->mFastTrackAvailMask &= ~(1 << i); 353 } 354 } 355 ALOGV("Track constructor name %d, calling pid %d", mName, 356 IPCThreadState::self()->getCallingPid()); 357} 358 359AudioFlinger::PlaybackThread::Track::~Track() 360{ 361 ALOGV("PlaybackThread::Track destructor"); 362} 363 364status_t AudioFlinger::PlaybackThread::Track::initCheck() const 365{ 366 status_t status = TrackBase::initCheck(); 367 if (status == NO_ERROR && mName < 0) { 368 status = NO_MEMORY; 369 } 370 return status; 371} 372 373void AudioFlinger::PlaybackThread::Track::destroy() 374{ 375 // NOTE: destroyTrack_l() can remove a strong reference to this Track 376 // by removing it from mTracks vector, so there is a risk that this Tracks's 377 // destructor is called. As the destructor needs to lock mLock, 378 // we must acquire a strong reference on this Track before locking mLock 379 // here so that the destructor is called only when exiting this function. 380 // On the other hand, as long as Track::destroy() is only called by 381 // TrackHandle destructor, the TrackHandle still holds a strong ref on 382 // this Track with its member mTrack. 383 sp<Track> keep(this); 384 { // scope for mLock 385 sp<ThreadBase> thread = mThread.promote(); 386 if (thread != 0) { 387 Mutex::Autolock _l(thread->mLock); 388 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 389 bool wasActive = playbackThread->destroyTrack_l(this); 390 if (!isOutputTrack() && !wasActive) { 391 AudioSystem::releaseOutput(thread->id()); 392 } 393 } 394 } 395} 396 397/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 398{ 399 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 400 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 401} 402 403void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 404{ 405 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 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 if (isTerminated()) { 414 stateChar = 'T'; 415 } else { 416 switch (state) { 417 case IDLE: 418 stateChar = 'I'; 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 } 449 char nowInUnderrun; 450 switch (mObservedUnderruns.mBitFields.mMostRecent) { 451 case UNDERRUN_FULL: 452 nowInUnderrun = ' '; 453 break; 454 case UNDERRUN_PARTIAL: 455 nowInUnderrun = '<'; 456 break; 457 case UNDERRUN_EMPTY: 458 nowInUnderrun = '*'; 459 break; 460 default: 461 nowInUnderrun = '?'; 462 break; 463 } 464 snprintf(&buffer[7], size-7, " %6u %4u %3u %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 465 "%08X %08X %08X 0x%03X %9u%c\n", 466 (mClient == 0) ? getpid_cached : mClient->pid(), 467 mStreamType, 468 mFormat, 469 mChannelMask, 470 mSessionId, 471 mFrameCount, 472 stateChar, 473 mFillingUpStatus, 474 mAudioTrackServerProxy->getSampleRate(), 475 20.0 * log10((vlr & 0xFFFF) / 4096.0), 476 20.0 * log10((vlr >> 16) / 4096.0), 477 mCblk->mServer, 478 (int)mMainBuffer, 479 (int)mAuxBuffer, 480 mCblk->mFlags, 481 mAudioTrackServerProxy->getUnderrunFrames(), 482 nowInUnderrun); 483} 484 485uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 486 return mAudioTrackServerProxy->getSampleRate(); 487} 488 489// AudioBufferProvider interface 490status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 491 AudioBufferProvider::Buffer* buffer, int64_t pts) 492{ 493 ServerProxy::Buffer buf; 494 size_t desiredFrames = buffer->frameCount; 495 buf.mFrameCount = desiredFrames; 496 status_t status = mServerProxy->obtainBuffer(&buf); 497 buffer->frameCount = buf.mFrameCount; 498 buffer->raw = buf.mRaw; 499 if (buf.mFrameCount == 0) { 500 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 501 } 502 return status; 503} 504 505// Note that framesReady() takes a mutex on the control block using tryLock(). 506// This could result in priority inversion if framesReady() is called by the normal mixer, 507// as the normal mixer thread runs at lower 508// priority than the client's callback thread: there is a short window within framesReady() 509// during which the normal mixer could be preempted, and the client callback would block. 510// Another problem can occur if framesReady() is called by the fast mixer: 511// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 512// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 513size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 514 return mAudioTrackServerProxy->framesReady(); 515} 516 517// Don't call for fast tracks; the framesReady() could result in priority inversion 518bool AudioFlinger::PlaybackThread::Track::isReady() const { 519 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 520 return true; 521 } 522 523 if (framesReady() >= mFrameCount || 524 (mCblk->mFlags & CBLK_FORCEREADY)) { 525 mFillingUpStatus = FS_FILLED; 526 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 527 return true; 528 } 529 return false; 530} 531 532status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 533 int triggerSession) 534{ 535 status_t status = NO_ERROR; 536 ALOGV("start(%d), calling pid %d session %d", 537 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 538 539 sp<ThreadBase> thread = mThread.promote(); 540 if (thread != 0) { 541 Mutex::Autolock _l(thread->mLock); 542 track_state state = mState; 543 // here the track could be either new, or restarted 544 // in both cases "unstop" the track 545 546 if (state == PAUSED) { 547 if (mResumeToStopping) { 548 // happened we need to resume to STOPPING_1 549 mState = TrackBase::STOPPING_1; 550 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 551 } else { 552 mState = TrackBase::RESUMING; 553 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 554 } 555 } else { 556 mState = TrackBase::ACTIVE; 557 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 558 } 559 560 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 561 status = playbackThread->addTrack_l(this); 562 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 563 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 564 // restore previous state if start was rejected by policy manager 565 if (status == PERMISSION_DENIED) { 566 mState = state; 567 } 568 } 569 // track was already in the active list, not a problem 570 if (status == ALREADY_EXISTS) { 571 status = NO_ERROR; 572 } 573 } else { 574 status = BAD_VALUE; 575 } 576 return status; 577} 578 579void AudioFlinger::PlaybackThread::Track::stop() 580{ 581 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 582 sp<ThreadBase> thread = mThread.promote(); 583 if (thread != 0) { 584 Mutex::Autolock _l(thread->mLock); 585 track_state state = mState; 586 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 587 // If the track is not active (PAUSED and buffers full), flush buffers 588 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 589 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 590 reset(); 591 mState = STOPPED; 592 } else if (!isFastTrack() && !isOffloaded()) { 593 mState = STOPPED; 594 } else { 595 // For fast tracks prepareTracks_l() will set state to STOPPING_2 596 // presentation is complete 597 // For an offloaded track this starts a drain and state will 598 // move to STOPPING_2 when drain completes and then STOPPED 599 mState = STOPPING_1; 600 } 601 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 602 playbackThread); 603 } 604 } 605} 606 607void AudioFlinger::PlaybackThread::Track::pause() 608{ 609 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 610 sp<ThreadBase> thread = mThread.promote(); 611 if (thread != 0) { 612 Mutex::Autolock _l(thread->mLock); 613 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 614 switch (mState) { 615 case STOPPING_1: 616 case STOPPING_2: 617 if (!isOffloaded()) { 618 /* nothing to do if track is not offloaded */ 619 break; 620 } 621 622 // Offloaded track was draining, we need to carry on draining when resumed 623 mResumeToStopping = true; 624 // fall through... 625 case ACTIVE: 626 case RESUMING: 627 mState = PAUSING; 628 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 629 playbackThread->signal_l(); 630 break; 631 632 default: 633 break; 634 } 635 } 636} 637 638void AudioFlinger::PlaybackThread::Track::flush() 639{ 640 ALOGV("flush(%d)", mName); 641 sp<ThreadBase> thread = mThread.promote(); 642 if (thread != 0) { 643 Mutex::Autolock _l(thread->mLock); 644 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 645 646 if (isOffloaded()) { 647 // If offloaded we allow flush during any state except terminated 648 // and keep the track active to avoid problems if user is seeking 649 // rapidly and underlying hardware has a significant delay handling 650 // a pause 651 if (isTerminated()) { 652 return; 653 } 654 655 ALOGV("flush: offload flush"); 656 reset(); 657 658 if (mState == STOPPING_1 || mState == STOPPING_2) { 659 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 660 mState = ACTIVE; 661 } 662 663 if (mState == ACTIVE) { 664 ALOGV("flush called in active state, resetting buffer time out retry count"); 665 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 666 } 667 668 mResumeToStopping = false; 669 } else { 670 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 671 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 672 return; 673 } 674 // No point remaining in PAUSED state after a flush => go to 675 // FLUSHED state 676 mState = FLUSHED; 677 // do not reset the track if it is still in the process of being stopped or paused. 678 // this will be done by prepareTracks_l() when the track is stopped. 679 // prepareTracks_l() will see mState == FLUSHED, then 680 // remove from active track list, reset(), and trigger presentation complete 681 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 682 reset(); 683 } 684 } 685 // Prevent flush being lost if the track is flushed and then resumed 686 // before mixer thread can run. This is important when offloading 687 // because the hardware buffer could hold a large amount of audio 688 playbackThread->flushOutput_l(); 689 playbackThread->signal_l(); 690 } 691} 692 693void AudioFlinger::PlaybackThread::Track::reset() 694{ 695 // Do not reset twice to avoid discarding data written just after a flush and before 696 // the audioflinger thread detects the track is stopped. 697 if (!mResetDone) { 698 // Force underrun condition to avoid false underrun callback until first data is 699 // written to buffer 700 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 701 mFillingUpStatus = FS_FILLING; 702 mResetDone = true; 703 if (mState == FLUSHED) { 704 mState = IDLE; 705 } 706 } 707} 708 709status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 710{ 711 sp<ThreadBase> thread = mThread.promote(); 712 if (thread == 0) { 713 ALOGE("thread is dead"); 714 return FAILED_TRANSACTION; 715 } else if ((thread->type() == ThreadBase::DIRECT) || 716 (thread->type() == ThreadBase::OFFLOAD)) { 717 return thread->setParameters(keyValuePairs); 718 } else { 719 return PERMISSION_DENIED; 720 } 721} 722 723status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 724{ 725 status_t status = DEAD_OBJECT; 726 sp<ThreadBase> thread = mThread.promote(); 727 if (thread != 0) { 728 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 729 sp<AudioFlinger> af = mClient->audioFlinger(); 730 731 Mutex::Autolock _l(af->mLock); 732 733 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 734 735 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 736 Mutex::Autolock _dl(playbackThread->mLock); 737 Mutex::Autolock _sl(srcThread->mLock); 738 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 739 if (chain == 0) { 740 return INVALID_OPERATION; 741 } 742 743 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 744 if (effect == 0) { 745 return INVALID_OPERATION; 746 } 747 srcThread->removeEffect_l(effect); 748 playbackThread->addEffect_l(effect); 749 // removeEffect_l() has stopped the effect if it was active so it must be restarted 750 if (effect->state() == EffectModule::ACTIVE || 751 effect->state() == EffectModule::STOPPING) { 752 effect->start(); 753 } 754 755 sp<EffectChain> dstChain = effect->chain().promote(); 756 if (dstChain == 0) { 757 srcThread->addEffect_l(effect); 758 return INVALID_OPERATION; 759 } 760 AudioSystem::unregisterEffect(effect->id()); 761 AudioSystem::registerEffect(&effect->desc(), 762 srcThread->id(), 763 dstChain->strategy(), 764 AUDIO_SESSION_OUTPUT_MIX, 765 effect->id()); 766 } 767 status = playbackThread->attachAuxEffect(this, EffectId); 768 } 769 return status; 770} 771 772void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 773{ 774 mAuxEffectId = EffectId; 775 mAuxBuffer = buffer; 776} 777 778bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 779 size_t audioHalFrames) 780{ 781 // a track is considered presented when the total number of frames written to audio HAL 782 // corresponds to the number of frames written when presentationComplete() is called for the 783 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 784 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 785 // to detect when all frames have been played. In this case framesWritten isn't 786 // useful because it doesn't always reflect whether there is data in the h/w 787 // buffers, particularly if a track has been paused and resumed during draining 788 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 789 mPresentationCompleteFrames, framesWritten); 790 if (mPresentationCompleteFrames == 0) { 791 mPresentationCompleteFrames = framesWritten + audioHalFrames; 792 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 793 mPresentationCompleteFrames, audioHalFrames); 794 } 795 796 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 797 ALOGV("presentationComplete() session %d complete: framesWritten %d", 798 mSessionId, framesWritten); 799 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 800 mAudioTrackServerProxy->setStreamEndDone(); 801 return true; 802 } 803 return false; 804} 805 806void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 807{ 808 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 809 if (mSyncEvents[i]->type() == type) { 810 mSyncEvents[i]->trigger(); 811 mSyncEvents.removeAt(i); 812 i--; 813 } 814 } 815} 816 817// implement VolumeBufferProvider interface 818 819uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 820{ 821 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 822 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 823 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 824 uint32_t vl = vlr & 0xFFFF; 825 uint32_t vr = vlr >> 16; 826 // track volumes come from shared memory, so can't be trusted and must be clamped 827 if (vl > MAX_GAIN_INT) { 828 vl = MAX_GAIN_INT; 829 } 830 if (vr > MAX_GAIN_INT) { 831 vr = MAX_GAIN_INT; 832 } 833 // now apply the cached master volume and stream type volume; 834 // this is trusted but lacks any synchronization or barrier so may be stale 835 float v = mCachedVolume; 836 vl *= v; 837 vr *= v; 838 // re-combine into U4.16 839 vlr = (vr << 16) | (vl & 0xFFFF); 840 // FIXME look at mute, pause, and stop flags 841 return vlr; 842} 843 844status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 845{ 846 if (isTerminated() || mState == PAUSED || 847 ((framesReady() == 0) && ((mSharedBuffer != 0) || 848 (mState == STOPPED)))) { 849 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 850 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 851 event->cancel(); 852 return INVALID_OPERATION; 853 } 854 (void) TrackBase::setSyncEvent(event); 855 return NO_ERROR; 856} 857 858void AudioFlinger::PlaybackThread::Track::invalidate() 859{ 860 // FIXME should use proxy, and needs work 861 audio_track_cblk_t* cblk = mCblk; 862 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 863 android_atomic_release_store(0x40000000, &cblk->mFutex); 864 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 865 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 866 mIsInvalid = true; 867} 868 869// ---------------------------------------------------------------------------- 870 871sp<AudioFlinger::PlaybackThread::TimedTrack> 872AudioFlinger::PlaybackThread::TimedTrack::create( 873 PlaybackThread *thread, 874 const sp<Client>& client, 875 audio_stream_type_t streamType, 876 uint32_t sampleRate, 877 audio_format_t format, 878 audio_channel_mask_t channelMask, 879 size_t frameCount, 880 const sp<IMemory>& sharedBuffer, 881 int sessionId) { 882 if (!client->reserveTimedTrack()) 883 return 0; 884 885 return new TimedTrack( 886 thread, client, streamType, sampleRate, format, channelMask, frameCount, 887 sharedBuffer, sessionId); 888} 889 890AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 891 PlaybackThread *thread, 892 const sp<Client>& client, 893 audio_stream_type_t streamType, 894 uint32_t sampleRate, 895 audio_format_t format, 896 audio_channel_mask_t channelMask, 897 size_t frameCount, 898 const sp<IMemory>& sharedBuffer, 899 int sessionId) 900 : Track(thread, client, streamType, sampleRate, format, channelMask, 901 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 902 mQueueHeadInFlight(false), 903 mTrimQueueHeadOnRelease(false), 904 mFramesPendingInQueue(0), 905 mTimedSilenceBuffer(NULL), 906 mTimedSilenceBufferSize(0), 907 mTimedAudioOutputOnTime(false), 908 mMediaTimeTransformValid(false) 909{ 910 LocalClock lc; 911 mLocalTimeFreq = lc.getLocalFreq(); 912 913 mLocalTimeToSampleTransform.a_zero = 0; 914 mLocalTimeToSampleTransform.b_zero = 0; 915 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 916 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 917 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 918 &mLocalTimeToSampleTransform.a_to_b_denom); 919 920 mMediaTimeToSampleTransform.a_zero = 0; 921 mMediaTimeToSampleTransform.b_zero = 0; 922 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 923 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 924 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 925 &mMediaTimeToSampleTransform.a_to_b_denom); 926} 927 928AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 929 mClient->releaseTimedTrack(); 930 delete [] mTimedSilenceBuffer; 931} 932 933status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 934 size_t size, sp<IMemory>* buffer) { 935 936 Mutex::Autolock _l(mTimedBufferQueueLock); 937 938 trimTimedBufferQueue_l(); 939 940 // lazily initialize the shared memory heap for timed buffers 941 if (mTimedMemoryDealer == NULL) { 942 const int kTimedBufferHeapSize = 512 << 10; 943 944 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 945 "AudioFlingerTimed"); 946 if (mTimedMemoryDealer == NULL) 947 return NO_MEMORY; 948 } 949 950 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 951 if (newBuffer == NULL) { 952 newBuffer = mTimedMemoryDealer->allocate(size); 953 if (newBuffer == NULL) 954 return NO_MEMORY; 955 } 956 957 *buffer = newBuffer; 958 return NO_ERROR; 959} 960 961// caller must hold mTimedBufferQueueLock 962void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 963 int64_t mediaTimeNow; 964 { 965 Mutex::Autolock mttLock(mMediaTimeTransformLock); 966 if (!mMediaTimeTransformValid) 967 return; 968 969 int64_t targetTimeNow; 970 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 971 ? mCCHelper.getCommonTime(&targetTimeNow) 972 : mCCHelper.getLocalTime(&targetTimeNow); 973 974 if (OK != res) 975 return; 976 977 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 978 &mediaTimeNow)) { 979 return; 980 } 981 } 982 983 size_t trimEnd; 984 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 985 int64_t bufEnd; 986 987 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 988 // We have a next buffer. Just use its PTS as the PTS of the frame 989 // following the last frame in this buffer. If the stream is sparse 990 // (ie, there are deliberate gaps left in the stream which should be 991 // filled with silence by the TimedAudioTrack), then this can result 992 // in one extra buffer being left un-trimmed when it could have 993 // been. In general, this is not typical, and we would rather 994 // optimized away the TS calculation below for the more common case 995 // where PTSes are contiguous. 996 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 997 } else { 998 // We have no next buffer. Compute the PTS of the frame following 999 // the last frame in this buffer by computing the duration of of 1000 // this frame in media time units and adding it to the PTS of the 1001 // buffer. 1002 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1003 / mFrameSize; 1004 1005 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1006 &bufEnd)) { 1007 ALOGE("Failed to convert frame count of %lld to media time" 1008 " duration" " (scale factor %d/%u) in %s", 1009 frameCount, 1010 mMediaTimeToSampleTransform.a_to_b_numer, 1011 mMediaTimeToSampleTransform.a_to_b_denom, 1012 __PRETTY_FUNCTION__); 1013 break; 1014 } 1015 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1016 } 1017 1018 if (bufEnd > mediaTimeNow) 1019 break; 1020 1021 // Is the buffer we want to use in the middle of a mix operation right 1022 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1023 // from the mixer which should be coming back shortly. 1024 if (!trimEnd && mQueueHeadInFlight) { 1025 mTrimQueueHeadOnRelease = true; 1026 } 1027 } 1028 1029 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1030 if (trimStart < trimEnd) { 1031 // Update the bookkeeping for framesReady() 1032 for (size_t i = trimStart; i < trimEnd; ++i) { 1033 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1034 } 1035 1036 // Now actually remove the buffers from the queue. 1037 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1038 } 1039} 1040 1041void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1042 const char* logTag) { 1043 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1044 "%s called (reason \"%s\"), but timed buffer queue has no" 1045 " elements to trim.", __FUNCTION__, logTag); 1046 1047 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1048 mTimedBufferQueue.removeAt(0); 1049} 1050 1051void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1052 const TimedBuffer& buf, 1053 const char* logTag) { 1054 uint32_t bufBytes = buf.buffer()->size(); 1055 uint32_t consumedAlready = buf.position(); 1056 1057 ALOG_ASSERT(consumedAlready <= bufBytes, 1058 "Bad bookkeeping while updating frames pending. Timed buffer is" 1059 " only %u bytes long, but claims to have consumed %u" 1060 " bytes. (update reason: \"%s\")", 1061 bufBytes, consumedAlready, logTag); 1062 1063 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1064 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1065 "Bad bookkeeping while updating frames pending. Should have at" 1066 " least %u queued frames, but we think we have only %u. (update" 1067 " reason: \"%s\")", 1068 bufFrames, mFramesPendingInQueue, logTag); 1069 1070 mFramesPendingInQueue -= bufFrames; 1071} 1072 1073status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1074 const sp<IMemory>& buffer, int64_t pts) { 1075 1076 { 1077 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1078 if (!mMediaTimeTransformValid) 1079 return INVALID_OPERATION; 1080 } 1081 1082 Mutex::Autolock _l(mTimedBufferQueueLock); 1083 1084 uint32_t bufFrames = buffer->size() / mFrameSize; 1085 mFramesPendingInQueue += bufFrames; 1086 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1087 1088 return NO_ERROR; 1089} 1090 1091status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1092 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1093 1094 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1095 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1096 target); 1097 1098 if (!(target == TimedAudioTrack::LOCAL_TIME || 1099 target == TimedAudioTrack::COMMON_TIME)) { 1100 return BAD_VALUE; 1101 } 1102 1103 Mutex::Autolock lock(mMediaTimeTransformLock); 1104 mMediaTimeTransform = xform; 1105 mMediaTimeTransformTarget = target; 1106 mMediaTimeTransformValid = true; 1107 1108 return NO_ERROR; 1109} 1110 1111#define min(a, b) ((a) < (b) ? (a) : (b)) 1112 1113// implementation of getNextBuffer for tracks whose buffers have timestamps 1114status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1115 AudioBufferProvider::Buffer* buffer, int64_t pts) 1116{ 1117 if (pts == AudioBufferProvider::kInvalidPTS) { 1118 buffer->raw = NULL; 1119 buffer->frameCount = 0; 1120 mTimedAudioOutputOnTime = false; 1121 return INVALID_OPERATION; 1122 } 1123 1124 Mutex::Autolock _l(mTimedBufferQueueLock); 1125 1126 ALOG_ASSERT(!mQueueHeadInFlight, 1127 "getNextBuffer called without releaseBuffer!"); 1128 1129 while (true) { 1130 1131 // if we have no timed buffers, then fail 1132 if (mTimedBufferQueue.isEmpty()) { 1133 buffer->raw = NULL; 1134 buffer->frameCount = 0; 1135 return NOT_ENOUGH_DATA; 1136 } 1137 1138 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1139 1140 // calculate the PTS of the head of the timed buffer queue expressed in 1141 // local time 1142 int64_t headLocalPTS; 1143 { 1144 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1145 1146 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1147 1148 if (mMediaTimeTransform.a_to_b_denom == 0) { 1149 // the transform represents a pause, so yield silence 1150 timedYieldSilence_l(buffer->frameCount, buffer); 1151 return NO_ERROR; 1152 } 1153 1154 int64_t transformedPTS; 1155 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1156 &transformedPTS)) { 1157 // the transform failed. this shouldn't happen, but if it does 1158 // then just drop this buffer 1159 ALOGW("timedGetNextBuffer transform failed"); 1160 buffer->raw = NULL; 1161 buffer->frameCount = 0; 1162 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1163 return NO_ERROR; 1164 } 1165 1166 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1167 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1168 &headLocalPTS)) { 1169 buffer->raw = NULL; 1170 buffer->frameCount = 0; 1171 return INVALID_OPERATION; 1172 } 1173 } else { 1174 headLocalPTS = transformedPTS; 1175 } 1176 } 1177 1178 uint32_t sr = sampleRate(); 1179 1180 // adjust the head buffer's PTS to reflect the portion of the head buffer 1181 // that has already been consumed 1182 int64_t effectivePTS = headLocalPTS + 1183 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1184 1185 // Calculate the delta in samples between the head of the input buffer 1186 // queue and the start of the next output buffer that will be written. 1187 // If the transformation fails because of over or underflow, it means 1188 // that the sample's position in the output stream is so far out of 1189 // whack that it should just be dropped. 1190 int64_t sampleDelta; 1191 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1192 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1193 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1194 " mix"); 1195 continue; 1196 } 1197 if (!mLocalTimeToSampleTransform.doForwardTransform( 1198 (effectivePTS - pts) << 32, &sampleDelta)) { 1199 ALOGV("*** too late during sample rate transform: dropped buffer"); 1200 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1201 continue; 1202 } 1203 1204 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1205 " sampleDelta=[%d.%08x]", 1206 head.pts(), head.position(), pts, 1207 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1208 + (sampleDelta >> 32)), 1209 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1210 1211 // if the delta between the ideal placement for the next input sample and 1212 // the current output position is within this threshold, then we will 1213 // concatenate the next input samples to the previous output 1214 const int64_t kSampleContinuityThreshold = 1215 (static_cast<int64_t>(sr) << 32) / 250; 1216 1217 // if this is the first buffer of audio that we're emitting from this track 1218 // then it should be almost exactly on time. 1219 const int64_t kSampleStartupThreshold = 1LL << 32; 1220 1221 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1222 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1223 // the next input is close enough to being on time, so concatenate it 1224 // with the last output 1225 timedYieldSamples_l(buffer); 1226 1227 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1228 head.position(), buffer->frameCount); 1229 return NO_ERROR; 1230 } 1231 1232 // Looks like our output is not on time. Reset our on timed status. 1233 // Next time we mix samples from our input queue, then should be within 1234 // the StartupThreshold. 1235 mTimedAudioOutputOnTime = false; 1236 if (sampleDelta > 0) { 1237 // the gap between the current output position and the proper start of 1238 // the next input sample is too big, so fill it with silence 1239 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1240 1241 timedYieldSilence_l(framesUntilNextInput, buffer); 1242 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1243 return NO_ERROR; 1244 } else { 1245 // the next input sample is late 1246 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1247 size_t onTimeSamplePosition = 1248 head.position() + lateFrames * mFrameSize; 1249 1250 if (onTimeSamplePosition > head.buffer()->size()) { 1251 // all the remaining samples in the head are too late, so 1252 // drop it and move on 1253 ALOGV("*** too late: dropped buffer"); 1254 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1255 continue; 1256 } else { 1257 // skip over the late samples 1258 head.setPosition(onTimeSamplePosition); 1259 1260 // yield the available samples 1261 timedYieldSamples_l(buffer); 1262 1263 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1264 return NO_ERROR; 1265 } 1266 } 1267 } 1268} 1269 1270// Yield samples from the timed buffer queue head up to the given output 1271// buffer's capacity. 1272// 1273// Caller must hold mTimedBufferQueueLock 1274void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1275 AudioBufferProvider::Buffer* buffer) { 1276 1277 const TimedBuffer& head = mTimedBufferQueue[0]; 1278 1279 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1280 head.position()); 1281 1282 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1283 mFrameSize); 1284 size_t framesRequested = buffer->frameCount; 1285 buffer->frameCount = min(framesLeftInHead, framesRequested); 1286 1287 mQueueHeadInFlight = true; 1288 mTimedAudioOutputOnTime = true; 1289} 1290 1291// Yield samples of silence up to the given output buffer's capacity 1292// 1293// Caller must hold mTimedBufferQueueLock 1294void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1295 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1296 1297 // lazily allocate a buffer filled with silence 1298 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1299 delete [] mTimedSilenceBuffer; 1300 mTimedSilenceBufferSize = numFrames * mFrameSize; 1301 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1302 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1303 } 1304 1305 buffer->raw = mTimedSilenceBuffer; 1306 size_t framesRequested = buffer->frameCount; 1307 buffer->frameCount = min(numFrames, framesRequested); 1308 1309 mTimedAudioOutputOnTime = false; 1310} 1311 1312// AudioBufferProvider interface 1313void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1314 AudioBufferProvider::Buffer* buffer) { 1315 1316 Mutex::Autolock _l(mTimedBufferQueueLock); 1317 1318 // If the buffer which was just released is part of the buffer at the head 1319 // of the queue, be sure to update the amt of the buffer which has been 1320 // consumed. If the buffer being returned is not part of the head of the 1321 // queue, its either because the buffer is part of the silence buffer, or 1322 // because the head of the timed queue was trimmed after the mixer called 1323 // getNextBuffer but before the mixer called releaseBuffer. 1324 if (buffer->raw == mTimedSilenceBuffer) { 1325 ALOG_ASSERT(!mQueueHeadInFlight, 1326 "Queue head in flight during release of silence buffer!"); 1327 goto done; 1328 } 1329 1330 ALOG_ASSERT(mQueueHeadInFlight, 1331 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1332 " head in flight."); 1333 1334 if (mTimedBufferQueue.size()) { 1335 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1336 1337 void* start = head.buffer()->pointer(); 1338 void* end = reinterpret_cast<void*>( 1339 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1340 + head.buffer()->size()); 1341 1342 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1343 "released buffer not within the head of the timed buffer" 1344 " queue; qHead = [%p, %p], released buffer = %p", 1345 start, end, buffer->raw); 1346 1347 head.setPosition(head.position() + 1348 (buffer->frameCount * mFrameSize)); 1349 mQueueHeadInFlight = false; 1350 1351 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1352 "Bad bookkeeping during releaseBuffer! Should have at" 1353 " least %u queued frames, but we think we have only %u", 1354 buffer->frameCount, mFramesPendingInQueue); 1355 1356 mFramesPendingInQueue -= buffer->frameCount; 1357 1358 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1359 || mTrimQueueHeadOnRelease) { 1360 trimTimedBufferQueueHead_l("releaseBuffer"); 1361 mTrimQueueHeadOnRelease = false; 1362 } 1363 } else { 1364 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1365 " buffers in the timed buffer queue"); 1366 } 1367 1368done: 1369 buffer->raw = 0; 1370 buffer->frameCount = 0; 1371} 1372 1373size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1374 Mutex::Autolock _l(mTimedBufferQueueLock); 1375 return mFramesPendingInQueue; 1376} 1377 1378AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1379 : mPTS(0), mPosition(0) {} 1380 1381AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1382 const sp<IMemory>& buffer, int64_t pts) 1383 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1384 1385 1386// ---------------------------------------------------------------------------- 1387 1388AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1389 PlaybackThread *playbackThread, 1390 DuplicatingThread *sourceThread, 1391 uint32_t sampleRate, 1392 audio_format_t format, 1393 audio_channel_mask_t channelMask, 1394 size_t frameCount) 1395 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1396 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1397 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1398{ 1399 1400 if (mCblk != NULL) { 1401 mOutBuffer.frameCount = 0; 1402 playbackThread->mTracks.add(this); 1403 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1404 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1405 mCblk, mBuffer, 1406 mCblk->frameCount_, mChannelMask); 1407 // since client and server are in the same process, 1408 // the buffer has the same virtual address on both sides 1409 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1410 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1411 mClientProxy->setSendLevel(0.0); 1412 mClientProxy->setSampleRate(sampleRate); 1413 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1414 true /*clientInServer*/); 1415 } else { 1416 ALOGW("Error creating output track on thread %p", playbackThread); 1417 } 1418} 1419 1420AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1421{ 1422 clearBufferQueue(); 1423 delete mClientProxy; 1424 // superclass destructor will now delete the server proxy and shared memory both refer to 1425} 1426 1427status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1428 int triggerSession) 1429{ 1430 status_t status = Track::start(event, triggerSession); 1431 if (status != NO_ERROR) { 1432 return status; 1433 } 1434 1435 mActive = true; 1436 mRetryCount = 127; 1437 return status; 1438} 1439 1440void AudioFlinger::PlaybackThread::OutputTrack::stop() 1441{ 1442 Track::stop(); 1443 clearBufferQueue(); 1444 mOutBuffer.frameCount = 0; 1445 mActive = false; 1446} 1447 1448bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1449{ 1450 Buffer *pInBuffer; 1451 Buffer inBuffer; 1452 uint32_t channelCount = mChannelCount; 1453 bool outputBufferFull = false; 1454 inBuffer.frameCount = frames; 1455 inBuffer.i16 = data; 1456 1457 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1458 1459 if (!mActive && frames != 0) { 1460 start(); 1461 sp<ThreadBase> thread = mThread.promote(); 1462 if (thread != 0) { 1463 MixerThread *mixerThread = (MixerThread *)thread.get(); 1464 if (mFrameCount > frames) { 1465 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1466 uint32_t startFrames = (mFrameCount - frames); 1467 pInBuffer = new Buffer; 1468 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1469 pInBuffer->frameCount = startFrames; 1470 pInBuffer->i16 = pInBuffer->mBuffer; 1471 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1472 mBufferQueue.add(pInBuffer); 1473 } else { 1474 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1475 } 1476 } 1477 } 1478 } 1479 1480 while (waitTimeLeftMs) { 1481 // First write pending buffers, then new data 1482 if (mBufferQueue.size()) { 1483 pInBuffer = mBufferQueue.itemAt(0); 1484 } else { 1485 pInBuffer = &inBuffer; 1486 } 1487 1488 if (pInBuffer->frameCount == 0) { 1489 break; 1490 } 1491 1492 if (mOutBuffer.frameCount == 0) { 1493 mOutBuffer.frameCount = pInBuffer->frameCount; 1494 nsecs_t startTime = systemTime(); 1495 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1496 if (status != NO_ERROR) { 1497 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1498 mThread.unsafe_get(), status); 1499 outputBufferFull = true; 1500 break; 1501 } 1502 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1503 if (waitTimeLeftMs >= waitTimeMs) { 1504 waitTimeLeftMs -= waitTimeMs; 1505 } else { 1506 waitTimeLeftMs = 0; 1507 } 1508 } 1509 1510 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1511 pInBuffer->frameCount; 1512 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1513 Proxy::Buffer buf; 1514 buf.mFrameCount = outFrames; 1515 buf.mRaw = NULL; 1516 mClientProxy->releaseBuffer(&buf); 1517 pInBuffer->frameCount -= outFrames; 1518 pInBuffer->i16 += outFrames * channelCount; 1519 mOutBuffer.frameCount -= outFrames; 1520 mOutBuffer.i16 += outFrames * channelCount; 1521 1522 if (pInBuffer->frameCount == 0) { 1523 if (mBufferQueue.size()) { 1524 mBufferQueue.removeAt(0); 1525 delete [] pInBuffer->mBuffer; 1526 delete pInBuffer; 1527 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1528 mThread.unsafe_get(), mBufferQueue.size()); 1529 } else { 1530 break; 1531 } 1532 } 1533 } 1534 1535 // If we could not write all frames, allocate a buffer and queue it for next time. 1536 if (inBuffer.frameCount) { 1537 sp<ThreadBase> thread = mThread.promote(); 1538 if (thread != 0 && !thread->standby()) { 1539 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1540 pInBuffer = new Buffer; 1541 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1542 pInBuffer->frameCount = inBuffer.frameCount; 1543 pInBuffer->i16 = pInBuffer->mBuffer; 1544 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1545 sizeof(int16_t)); 1546 mBufferQueue.add(pInBuffer); 1547 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1548 mThread.unsafe_get(), mBufferQueue.size()); 1549 } else { 1550 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1551 mThread.unsafe_get(), this); 1552 } 1553 } 1554 } 1555 1556 // Calling write() with a 0 length buffer, means that no more data will be written: 1557 // If no more buffers are pending, fill output track buffer to make sure it is started 1558 // by output mixer. 1559 if (frames == 0 && mBufferQueue.size() == 0) { 1560 // FIXME borken, replace by getting framesReady() from proxy 1561 size_t user = 0; // was mCblk->user 1562 if (user < mFrameCount) { 1563 frames = mFrameCount - user; 1564 pInBuffer = new Buffer; 1565 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1566 pInBuffer->frameCount = frames; 1567 pInBuffer->i16 = pInBuffer->mBuffer; 1568 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1569 mBufferQueue.add(pInBuffer); 1570 } else if (mActive) { 1571 stop(); 1572 } 1573 } 1574 1575 return outputBufferFull; 1576} 1577 1578status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1579 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1580{ 1581 ClientProxy::Buffer buf; 1582 buf.mFrameCount = buffer->frameCount; 1583 struct timespec timeout; 1584 timeout.tv_sec = waitTimeMs / 1000; 1585 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1586 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1587 buffer->frameCount = buf.mFrameCount; 1588 buffer->raw = buf.mRaw; 1589 return status; 1590} 1591 1592void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1593{ 1594 size_t size = mBufferQueue.size(); 1595 1596 for (size_t i = 0; i < size; i++) { 1597 Buffer *pBuffer = mBufferQueue.itemAt(i); 1598 delete [] pBuffer->mBuffer; 1599 delete pBuffer; 1600 } 1601 mBufferQueue.clear(); 1602} 1603 1604 1605// ---------------------------------------------------------------------------- 1606// Record 1607// ---------------------------------------------------------------------------- 1608 1609AudioFlinger::RecordHandle::RecordHandle( 1610 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1611 : BnAudioRecord(), 1612 mRecordTrack(recordTrack) 1613{ 1614} 1615 1616AudioFlinger::RecordHandle::~RecordHandle() { 1617 stop_nonvirtual(); 1618 mRecordTrack->destroy(); 1619} 1620 1621sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1622 return mRecordTrack->getCblk(); 1623} 1624 1625status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1626 int triggerSession) { 1627 ALOGV("RecordHandle::start()"); 1628 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1629} 1630 1631void AudioFlinger::RecordHandle::stop() { 1632 stop_nonvirtual(); 1633} 1634 1635void AudioFlinger::RecordHandle::stop_nonvirtual() { 1636 ALOGV("RecordHandle::stop()"); 1637 mRecordTrack->stop(); 1638} 1639 1640status_t AudioFlinger::RecordHandle::onTransact( 1641 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1642{ 1643 return BnAudioRecord::onTransact(code, data, reply, flags); 1644} 1645 1646// ---------------------------------------------------------------------------- 1647 1648// RecordTrack constructor must be called with AudioFlinger::mLock held 1649AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1650 RecordThread *thread, 1651 const sp<Client>& client, 1652 uint32_t sampleRate, 1653 audio_format_t format, 1654 audio_channel_mask_t channelMask, 1655 size_t frameCount, 1656 int sessionId) 1657 : TrackBase(thread, client, sampleRate, format, 1658 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1659 mOverflow(false) 1660{ 1661 ALOGV("RecordTrack constructor"); 1662 if (mCblk != NULL) { 1663 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); 1664 } 1665} 1666 1667AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1668{ 1669 ALOGV("%s", __func__); 1670} 1671 1672// AudioBufferProvider interface 1673status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1674 int64_t pts) 1675{ 1676 ServerProxy::Buffer buf; 1677 buf.mFrameCount = buffer->frameCount; 1678 status_t status = mServerProxy->obtainBuffer(&buf); 1679 buffer->frameCount = buf.mFrameCount; 1680 buffer->raw = buf.mRaw; 1681 if (buf.mFrameCount == 0) { 1682 // FIXME also wake futex so that overrun is noticed more quickly 1683 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1684 } 1685 return status; 1686} 1687 1688status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1689 int triggerSession) 1690{ 1691 sp<ThreadBase> thread = mThread.promote(); 1692 if (thread != 0) { 1693 RecordThread *recordThread = (RecordThread *)thread.get(); 1694 return recordThread->start(this, event, triggerSession); 1695 } else { 1696 return BAD_VALUE; 1697 } 1698} 1699 1700void AudioFlinger::RecordThread::RecordTrack::stop() 1701{ 1702 sp<ThreadBase> thread = mThread.promote(); 1703 if (thread != 0) { 1704 RecordThread *recordThread = (RecordThread *)thread.get(); 1705 if (recordThread->stop(this)) { 1706 AudioSystem::stopInput(recordThread->id()); 1707 } 1708 } 1709} 1710 1711void AudioFlinger::RecordThread::RecordTrack::destroy() 1712{ 1713 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1714 sp<RecordTrack> keep(this); 1715 { 1716 sp<ThreadBase> thread = mThread.promote(); 1717 if (thread != 0) { 1718 if (mState == ACTIVE || mState == RESUMING) { 1719 AudioSystem::stopInput(thread->id()); 1720 } 1721 AudioSystem::releaseInput(thread->id()); 1722 Mutex::Autolock _l(thread->mLock); 1723 RecordThread *recordThread = (RecordThread *) thread.get(); 1724 recordThread->destroyTrack_l(this); 1725 } 1726 } 1727} 1728 1729 1730/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1731{ 1732 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1733} 1734 1735void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1736{ 1737 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1738 (mClient == 0) ? getpid_cached : mClient->pid(), 1739 mFormat, 1740 mChannelMask, 1741 mSessionId, 1742 mState, 1743 mCblk->mServer, 1744 mFrameCount); 1745} 1746 1747}; // namespace android 1748