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Issue 557333002: Revert of Switch to using media::TimeSource inside media::RendererImpl. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 3 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "media/filters/audio_renderer_impl.h" 5 #include "media/filters/audio_renderer_impl.h"
6 6
7 #include <math.h> 7 #include <math.h>
8 8
9 #include <algorithm> 9 #include <algorithm>
10 10
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56 media_log)), 56 media_log)),
57 hardware_config_(hardware_config), 57 hardware_config_(hardware_config),
58 playback_rate_(0), 58 playback_rate_(0),
59 state_(kUninitialized), 59 state_(kUninitialized),
60 buffering_state_(BUFFERING_HAVE_NOTHING), 60 buffering_state_(BUFFERING_HAVE_NOTHING),
61 rendering_(false), 61 rendering_(false),
62 sink_playing_(false), 62 sink_playing_(false),
63 pending_read_(false), 63 pending_read_(false),
64 received_end_of_stream_(false), 64 received_end_of_stream_(false),
65 rendered_end_of_stream_(false), 65 rendered_end_of_stream_(false),
66 last_timestamp_update_(kNoTimestamp()),
66 weak_factory_(this) { 67 weak_factory_(this) {
67 audio_buffer_stream_->set_splice_observer(base::Bind( 68 audio_buffer_stream_->set_splice_observer(base::Bind(
68 &AudioRendererImpl::OnNewSpliceBuffer, weak_factory_.GetWeakPtr())); 69 &AudioRendererImpl::OnNewSpliceBuffer, weak_factory_.GetWeakPtr()));
69 audio_buffer_stream_->set_config_change_observer(base::Bind( 70 audio_buffer_stream_->set_config_change_observer(base::Bind(
70 &AudioRendererImpl::OnConfigChange, weak_factory_.GetWeakPtr())); 71 &AudioRendererImpl::OnConfigChange, weak_factory_.GetWeakPtr()));
71 } 72 }
72 73
73 AudioRendererImpl::~AudioRendererImpl() { 74 AudioRendererImpl::~AudioRendererImpl() {
74 DVLOG(1) << __FUNCTION__; 75 DVLOG(1) << __FUNCTION__;
75 DCHECK(task_runner_->BelongsToCurrentThread()); 76 DCHECK(task_runner_->BelongsToCurrentThread());
(...skipping 67 matching lines...) Expand 10 before | Expand all | Expand 10 after
143 void AudioRendererImpl::SetMediaTime(base::TimeDelta time) { 144 void AudioRendererImpl::SetMediaTime(base::TimeDelta time) {
144 DVLOG(1) << __FUNCTION__ << "(" << time.InMicroseconds() << ")"; 145 DVLOG(1) << __FUNCTION__ << "(" << time.InMicroseconds() << ")";
145 DCHECK(task_runner_->BelongsToCurrentThread()); 146 DCHECK(task_runner_->BelongsToCurrentThread());
146 147
147 base::AutoLock auto_lock(lock_); 148 base::AutoLock auto_lock(lock_);
148 DCHECK(!rendering_); 149 DCHECK(!rendering_);
149 DCHECK_EQ(state_, kFlushed); 150 DCHECK_EQ(state_, kFlushed);
150 151
151 start_timestamp_ = time; 152 start_timestamp_ = time;
152 ended_timestamp_ = kInfiniteDuration(); 153 ended_timestamp_ = kInfiniteDuration();
153 last_render_ticks_ = base::TimeTicks();
154 audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate())); 154 audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate()));
155 } 155 }
156 156
157 base::TimeDelta AudioRendererImpl::CurrentMediaTime() { 157 base::TimeDelta AudioRendererImpl::CurrentMediaTime() {
158 DVLOG(2) << __FUNCTION__; 158 DVLOG(2) << __FUNCTION__;
159 DCHECK(task_runner_->BelongsToCurrentThread());
159 160
160 // In practice the Render() method is called with a high enough frequency 161 // TODO(scherkus): Finish implementing when ready to switch Pipeline to using
161 // that returning only the front timestamp is good enough and also prevents 162 // TimeSource http://crbug.com/370634
162 // returning values that go backwards in time. 163 NOTIMPLEMENTED();
163 base::AutoLock auto_lock(lock_);
164 return audio_clock_->front_timestamp();
165 }
166 164
167 base::TimeDelta AudioRendererImpl::CurrentMediaTimeForSyncingVideo() { 165 return base::TimeDelta();
168 DVLOG(2) << __FUNCTION__;
169
170 base::AutoLock auto_lock(lock_);
171 if (last_render_ticks_.is_null())
172 return audio_clock_->front_timestamp();
173
174 return audio_clock_->TimestampSinceWriting(base::TimeTicks::Now() -
175 last_render_ticks_);
176 } 166 }
177 167
178 TimeSource* AudioRendererImpl::GetTimeSource() { 168 TimeSource* AudioRendererImpl::GetTimeSource() {
179 return this; 169 return this;
180 } 170 }
181 171
182 void AudioRendererImpl::Flush(const base::Closure& callback) { 172 void AudioRendererImpl::Flush(const base::Closure& callback) {
183 DVLOG(1) << __FUNCTION__; 173 DVLOG(1) << __FUNCTION__;
184 DCHECK(task_runner_->BelongsToCurrentThread()); 174 DCHECK(task_runner_->BelongsToCurrentThread());
185 175
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209 } 199 }
210 200
211 void AudioRendererImpl::ResetDecoderDone() { 201 void AudioRendererImpl::ResetDecoderDone() {
212 DCHECK(task_runner_->BelongsToCurrentThread()); 202 DCHECK(task_runner_->BelongsToCurrentThread());
213 { 203 {
214 base::AutoLock auto_lock(lock_); 204 base::AutoLock auto_lock(lock_);
215 205
216 DCHECK_EQ(state_, kFlushed); 206 DCHECK_EQ(state_, kFlushed);
217 DCHECK(!flush_cb_.is_null()); 207 DCHECK(!flush_cb_.is_null());
218 208
209 audio_clock_.reset();
219 received_end_of_stream_ = false; 210 received_end_of_stream_ = false;
220 rendered_end_of_stream_ = false; 211 rendered_end_of_stream_ = false;
212 last_timestamp_update_ = kNoTimestamp();
221 213
222 // Flush() may have been called while underflowed/not fully buffered. 214 // Flush() may have been called while underflowed/not fully buffered.
223 if (buffering_state_ != BUFFERING_HAVE_NOTHING) 215 if (buffering_state_ != BUFFERING_HAVE_NOTHING)
224 SetBufferingState_Locked(BUFFERING_HAVE_NOTHING); 216 SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
225 217
226 splicer_->Reset(); 218 splicer_->Reset();
227 if (buffer_converter_) 219 if (buffer_converter_)
228 buffer_converter_->Reset(); 220 buffer_converter_->Reset();
229 algorithm_->FlushBuffers(); 221 algorithm_->FlushBuffers();
230 } 222 }
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244 DCHECK_EQ(buffering_state_, BUFFERING_HAVE_NOTHING); 236 DCHECK_EQ(buffering_state_, BUFFERING_HAVE_NOTHING);
245 DCHECK(!pending_read_) << "Pending read must complete before seeking"; 237 DCHECK(!pending_read_) << "Pending read must complete before seeking";
246 238
247 ChangeState_Locked(kPlaying); 239 ChangeState_Locked(kPlaying);
248 AttemptRead_Locked(); 240 AttemptRead_Locked();
249 } 241 }
250 242
251 void AudioRendererImpl::Initialize(DemuxerStream* stream, 243 void AudioRendererImpl::Initialize(DemuxerStream* stream,
252 const PipelineStatusCB& init_cb, 244 const PipelineStatusCB& init_cb,
253 const StatisticsCB& statistics_cb, 245 const StatisticsCB& statistics_cb,
246 const TimeCB& time_cb,
254 const BufferingStateCB& buffering_state_cb, 247 const BufferingStateCB& buffering_state_cb,
255 const base::Closure& ended_cb, 248 const base::Closure& ended_cb,
256 const PipelineStatusCB& error_cb) { 249 const PipelineStatusCB& error_cb) {
257 DCHECK(task_runner_->BelongsToCurrentThread()); 250 DCHECK(task_runner_->BelongsToCurrentThread());
258 DCHECK(stream); 251 DCHECK(stream);
259 DCHECK_EQ(stream->type(), DemuxerStream::AUDIO); 252 DCHECK_EQ(stream->type(), DemuxerStream::AUDIO);
260 DCHECK(!init_cb.is_null()); 253 DCHECK(!init_cb.is_null());
261 DCHECK(!statistics_cb.is_null()); 254 DCHECK(!statistics_cb.is_null());
255 DCHECK(!time_cb.is_null());
262 DCHECK(!buffering_state_cb.is_null()); 256 DCHECK(!buffering_state_cb.is_null());
263 DCHECK(!ended_cb.is_null()); 257 DCHECK(!ended_cb.is_null());
264 DCHECK(!error_cb.is_null()); 258 DCHECK(!error_cb.is_null());
265 DCHECK_EQ(kUninitialized, state_); 259 DCHECK_EQ(kUninitialized, state_);
266 DCHECK(sink_.get()); 260 DCHECK(sink_.get());
267 261
268 state_ = kInitializing; 262 state_ = kInitializing;
269 263
270 // Always post |init_cb_| because |this| could be destroyed if initialization 264 // Always post |init_cb_| because |this| could be destroyed if initialization
271 // failed. 265 // failed.
272 init_cb_ = BindToCurrentLoop(init_cb); 266 init_cb_ = BindToCurrentLoop(init_cb);
273 267
268 time_cb_ = time_cb;
274 buffering_state_cb_ = buffering_state_cb; 269 buffering_state_cb_ = buffering_state_cb;
275 ended_cb_ = ended_cb; 270 ended_cb_ = ended_cb;
276 error_cb_ = error_cb; 271 error_cb_ = error_cb;
277 272
278 expecting_config_changes_ = stream->SupportsConfigChanges(); 273 expecting_config_changes_ = stream->SupportsConfigChanges();
279 if (!expecting_config_changes_) { 274 if (!expecting_config_changes_) {
280 // The actual buffer size is controlled via the size of the AudioBus 275 // The actual buffer size is controlled via the size of the AudioBus
281 // provided to Render(), so just choose something reasonable here for looks. 276 // provided to Render(), so just choose something reasonable here for looks.
282 int buffer_size = stream->audio_decoder_config().samples_per_second() / 100; 277 int buffer_size = stream->audio_decoder_config().samples_per_second() / 100;
283 audio_parameters_.Reset( 278 audio_parameters_.Reset(
(...skipping 266 matching lines...) Expand 10 before | Expand all | Expand 10 after
550 int AudioRendererImpl::Render(AudioBus* audio_bus, 545 int AudioRendererImpl::Render(AudioBus* audio_bus,
551 int audio_delay_milliseconds) { 546 int audio_delay_milliseconds) {
552 const int requested_frames = audio_bus->frames(); 547 const int requested_frames = audio_bus->frames();
553 base::TimeDelta playback_delay = base::TimeDelta::FromMilliseconds( 548 base::TimeDelta playback_delay = base::TimeDelta::FromMilliseconds(
554 audio_delay_milliseconds); 549 audio_delay_milliseconds);
555 const int delay_frames = static_cast<int>(playback_delay.InSecondsF() * 550 const int delay_frames = static_cast<int>(playback_delay.InSecondsF() *
556 audio_parameters_.sample_rate()); 551 audio_parameters_.sample_rate());
557 int frames_written = 0; 552 int frames_written = 0;
558 { 553 {
559 base::AutoLock auto_lock(lock_); 554 base::AutoLock auto_lock(lock_);
560 last_render_ticks_ = base::TimeTicks::Now();
561 555
562 // Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread. 556 // Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread.
563 if (!algorithm_) { 557 if (!algorithm_) {
564 audio_clock_->WroteAudio( 558 audio_clock_->WroteAudio(
565 0, requested_frames, delay_frames, playback_rate_); 559 0, requested_frames, delay_frames, playback_rate_);
566 return 0; 560 return 0;
567 } 561 }
568 562
569 if (playback_rate_ == 0) { 563 if (playback_rate_ == 0) {
570 audio_clock_->WroteAudio( 564 audio_clock_->WroteAudio(
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621 requested_frames, 615 requested_frames,
622 delay_frames, 616 delay_frames,
623 playback_rate_); 617 playback_rate_);
624 618
625 if (CanRead_Locked()) { 619 if (CanRead_Locked()) {
626 task_runner_->PostTask(FROM_HERE, 620 task_runner_->PostTask(FROM_HERE,
627 base::Bind(&AudioRendererImpl::AttemptRead, 621 base::Bind(&AudioRendererImpl::AttemptRead,
628 weak_factory_.GetWeakPtr())); 622 weak_factory_.GetWeakPtr()));
629 } 623 }
630 624
631 if (audio_clock_->front_timestamp() >= ended_timestamp_ && 625 if (last_timestamp_update_ != audio_clock_->front_timestamp()) {
632 !rendered_end_of_stream_) { 626 // Since |max_time| uses linear interpolation, only provide an upper bound
633 rendered_end_of_stream_ = true; 627 // that is for audio data at the same playback rate. Failing to do so can
634 task_runner_->PostTask(FROM_HERE, ended_cb_); 628 // make time jump backwards when the linear interpolated time advances
629 // past buffered regions of audio at different rates.
630 last_timestamp_update_ = audio_clock_->front_timestamp();
631 base::TimeDelta max_time =
632 last_timestamp_update_ +
633 audio_clock_->contiguous_audio_data_buffered_at_same_rate();
634 task_runner_->PostTask(
635 FROM_HERE, base::Bind(time_cb_, last_timestamp_update_, max_time));
636
637 if (last_timestamp_update_ >= ended_timestamp_ &&
638 !rendered_end_of_stream_) {
639 rendered_end_of_stream_ = true;
640 task_runner_->PostTask(FROM_HERE, ended_cb_);
641 }
635 } 642 }
636 } 643 }
637 644
638 DCHECK_LE(frames_written, requested_frames); 645 DCHECK_LE(frames_written, requested_frames);
639 return frames_written; 646 return frames_written;
640 } 647 }
641 648
642 void AudioRendererImpl::OnRenderError() { 649 void AudioRendererImpl::OnRenderError() {
643 // UMA data tells us this happens ~0.01% of the time. Trigger an error instead 650 // UMA data tells us this happens ~0.01% of the time. Trigger an error instead
644 // of trying to gracefully fall back to a fake sink. It's very likely 651 // of trying to gracefully fall back to a fake sink. It's very likely
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707 << buffering_state; 714 << buffering_state;
708 DCHECK_NE(buffering_state_, buffering_state); 715 DCHECK_NE(buffering_state_, buffering_state);
709 lock_.AssertAcquired(); 716 lock_.AssertAcquired();
710 buffering_state_ = buffering_state; 717 buffering_state_ = buffering_state;
711 718
712 task_runner_->PostTask(FROM_HERE, 719 task_runner_->PostTask(FROM_HERE,
713 base::Bind(buffering_state_cb_, buffering_state_)); 720 base::Bind(buffering_state_cb_, buffering_state_));
714 } 721 }
715 722
716 } // namespace media 723 } // namespace media
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