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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/renderers/audio_renderer_impl.h" | 5 #include "media/renderers/audio_renderer_impl.h" |
6 | 6 |
7 #include <math.h> | 7 #include <math.h> |
8 #include <stddef.h> | 8 #include <stddef.h> |
9 #include <algorithm> | 9 #include <algorithm> |
10 #include <utility> | 10 #include <utility> |
11 | 11 |
12 #include "base/bind.h" | 12 #include "base/bind.h" |
13 #include "base/callback.h" | 13 #include "base/callback.h" |
14 #include "base/callback_helpers.h" | 14 #include "base/callback_helpers.h" |
15 #include "base/command_line.h" | |
15 #include "base/logging.h" | 16 #include "base/logging.h" |
16 #include "base/single_thread_task_runner.h" | 17 #include "base/single_thread_task_runner.h" |
17 #include "base/time/default_tick_clock.h" | 18 #include "base/time/default_tick_clock.h" |
18 #include "build/build_config.h" | 19 #include "build/build_config.h" |
19 #include "media/base/audio_buffer.h" | 20 #include "media/base/audio_buffer.h" |
20 #include "media/base/audio_buffer_converter.h" | 21 #include "media/base/audio_buffer_converter.h" |
21 #include "media/base/audio_hardware_config.h" | 22 #include "media/base/audio_hardware_config.h" |
22 #include "media/base/audio_splicer.h" | 23 #include "media/base/audio_splicer.h" |
23 #include "media/base/bind_to_current_loop.h" | 24 #include "media/base/bind_to_current_loop.h" |
24 #include "media/base/demuxer_stream.h" | 25 #include "media/base/demuxer_stream.h" |
25 #include "media/base/media_log.h" | 26 #include "media/base/media_log.h" |
27 #include "media/base/media_switches.h" | |
26 #include "media/base/timestamp_constants.h" | 28 #include "media/base/timestamp_constants.h" |
27 #include "media/filters/audio_clock.h" | 29 #include "media/filters/audio_clock.h" |
28 #include "media/filters/decrypting_demuxer_stream.h" | 30 #include "media/filters/decrypting_demuxer_stream.h" |
29 | 31 |
30 namespace media { | 32 namespace media { |
31 | 33 |
32 AudioRendererImpl::AudioRendererImpl( | 34 AudioRendererImpl::AudioRendererImpl( |
33 const scoped_refptr<base::SingleThreadTaskRunner>& task_runner, | 35 const scoped_refptr<base::SingleThreadTaskRunner>& task_runner, |
34 media::AudioRendererSink* sink, | 36 media::AudioRendererSink* sink, |
35 ScopedVector<AudioDecoder> decoders, | 37 ScopedVector<AudioDecoder> decoders, |
36 const AudioHardwareConfig& hardware_config, | 38 const AudioHardwareConfig& hardware_config, |
37 const scoped_refptr<MediaLog>& media_log) | 39 const scoped_refptr<MediaLog>& media_log) |
38 : task_runner_(task_runner), | 40 : task_runner_(task_runner), |
39 expecting_config_changes_(false), | 41 expecting_config_changes_(false), |
40 sink_(sink), | 42 sink_(sink), |
41 audio_buffer_stream_( | 43 audio_buffer_stream_( |
42 new AudioBufferStream(task_runner, std::move(decoders), media_log)), | 44 new AudioBufferStream(task_runner, std::move(decoders), media_log)), |
43 hardware_config_(hardware_config), | 45 hardware_config_(hardware_config), |
44 media_log_(media_log), | 46 media_log_(media_log), |
45 tick_clock_(new base::DefaultTickClock()), | 47 tick_clock_(new base::DefaultTickClock()), |
46 last_audio_memory_usage_(0), | 48 last_audio_memory_usage_(0), |
49 last_decoded_sample_rate_(0), | |
47 playback_rate_(0.0), | 50 playback_rate_(0.0), |
48 state_(kUninitialized), | 51 state_(kUninitialized), |
49 buffering_state_(BUFFERING_HAVE_NOTHING), | 52 buffering_state_(BUFFERING_HAVE_NOTHING), |
50 rendering_(false), | 53 rendering_(false), |
51 sink_playing_(false), | 54 sink_playing_(false), |
52 pending_read_(false), | 55 pending_read_(false), |
53 received_end_of_stream_(false), | 56 received_end_of_stream_(false), |
54 rendered_end_of_stream_(false), | 57 rendered_end_of_stream_(false), |
55 weak_factory_(this) { | 58 weak_factory_(this) { |
56 audio_buffer_stream_->set_splice_observer(base::Bind( | 59 audio_buffer_stream_->set_splice_observer(base::Bind( |
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362 #if defined(OS_CHROMEOS) | 365 #if defined(OS_CHROMEOS) |
363 // On ChromeOS let the OS level resampler handle resampling unless the | 366 // On ChromeOS let the OS level resampler handle resampling unless the |
364 // initial sample rate is too low; this allows support for sample rate | 367 // initial sample rate is too low; this allows support for sample rate |
365 // adaptations where necessary. | 368 // adaptations where necessary. |
366 if (stream->audio_decoder_config().samples_per_second() >= 44100) { | 369 if (stream->audio_decoder_config().samples_per_second() >= 44100) { |
367 sample_rate = stream->audio_decoder_config().samples_per_second(); | 370 sample_rate = stream->audio_decoder_config().samples_per_second(); |
368 preferred_buffer_size = 0; // No preference. | 371 preferred_buffer_size = 0; // No preference. |
369 } | 372 } |
370 #endif | 373 #endif |
371 | 374 |
372 audio_parameters_.Reset( | 375 int stream_channel_count = ChannelLayoutToChannelCount( |
373 hw_params.format(), | 376 stream->audio_decoder_config().channel_layout()); |
374 // Always use the source's channel layout to avoid premature downmixing | 377 |
375 // (http://crbug.com/379288), platform specific issues around channel | 378 bool try_supported_channel_layouts_win = false; |
DaleCurtis
2016/03/25 23:57:51
Remove win suffix.
chcunningham
2016/03/26 00:31:36
Done.
| |
376 // layouts (http://crbug.com/266674), and unnecessary upmixing overhead. | 379 #if defined(OS_WIN) |
377 stream->audio_decoder_config().channel_layout(), sample_rate, | 380 const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
378 hw_params.bits_per_sample(), | 381 try_supported_channel_layouts = |
379 AudioHardwareConfig::GetHighLatencyBufferSize(sample_rate, | 382 cmd_line->HasSwitch(switches::kTrySupportedChannelLayouts); |
DaleCurtis
2016/03/25 23:57:51
inline cmd_line?
chcunningham
2016/03/26 00:31:36
Done.
| |
380 preferred_buffer_size)); | 383 #endif |
384 | |
385 // We don't know how to up-mix for DISCRETE layouts (fancy multichannel | |
386 // hardware with non-standard speaker arrangement). Instead, pretend the | |
387 // hardware layout is stereo and let the OS take care of further up-mixing | |
388 // to the discrete layout (http://crbug.com/266674). Additionally, pretend | |
389 // hardware is stereo whenever kTrySupportedChannelLayouts is set. This flag | |
390 // is for savvy users who want stereo content to output in all surround | |
391 // speakers. Using the actual layout (likely 5.1 or higher) will mean our | |
392 // mixer will attempt to up-mix stereo source streams to just the left/right | |
393 // speaker of the 5.1 setup, nulling out the other channels | |
394 // (http://crbug.com/177872). | |
395 ChannelLayout hw_channel_layout = | |
396 hw_params.channel_layout() == CHANNEL_LAYOUT_DISCRETE || | |
397 try_supported_channel_layouts_win | |
398 ? CHANNEL_LAYOUT_STEREO | |
399 : hw_params.channel_layout(); | |
400 int hw_channel_count = ChannelLayoutToChannelCount(hw_channel_layout); | |
401 | |
402 // The layout we pass to |audio_parameters_| will be used for the lifetime | |
403 // of this audio renderer, regardless of changes to hardware and/or stream | |
404 // properties. Below we choose the max of stream layout vs. hardware layout | |
405 // to leave room for changes to the hardware and/or stream (i.e. avoid | |
406 // premature down-mixing - http://crbug.com/379288). | |
407 // If stream_channels < hw_channels: | |
408 // Taking max means we up-mix to hardware layout. If stream later changes | |
409 // to have more channels, we aren't locked into down-mixing to the | |
410 // initial stream layout. | |
411 // If stream_channels > hw_channels: | |
412 // We choose to output stream's layout, meaning mixing is a no-op for the | |
413 // renderer. Browser-side will down-mix to the hardware config. If the | |
414 // hardware later changes to equal stream channels, browser-side will stop | |
415 // down-mixing and use the data from all stream channels. | |
416 ChannelLayout renderer_channel_layout = | |
417 (hw_channel_count > stream_channel_count) | |
DaleCurtis
2016/03/25 23:57:51
Remove extra parens.
chcunningham
2016/03/26 00:31:36
Done.
| |
418 ? hw_channel_layout | |
419 : stream->audio_decoder_config().channel_layout(); | |
420 | |
421 audio_parameters_.Reset(hw_params.format(), renderer_channel_layout, | |
DaleCurtis
2016/03/25 23:57:51
Do you want to initialize last_decoded_sample_rate
chcunningham
2016/03/26 00:31:36
Done. I added a check that its been set below befo
| |
422 sample_rate, hw_params.bits_per_sample(), | |
423 AudioHardwareConfig::GetHighLatencyBufferSize( | |
424 sample_rate, preferred_buffer_size)); | |
381 } | 425 } |
382 | 426 |
383 audio_clock_.reset( | 427 audio_clock_.reset( |
384 new AudioClock(base::TimeDelta(), audio_parameters_.sample_rate())); | 428 new AudioClock(base::TimeDelta(), audio_parameters_.sample_rate())); |
385 | 429 |
386 audio_buffer_stream_->Initialize( | 430 audio_buffer_stream_->Initialize( |
387 stream, base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized, | 431 stream, base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized, |
388 weak_factory_.GetWeakPtr()), | 432 weak_factory_.GetWeakPtr()), |
389 cdm_context, statistics_cb, waiting_for_decryption_key_cb); | 433 cdm_context, statistics_cb, waiting_for_decryption_key_cb); |
390 } | 434 } |
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465 DCHECK_EQ(status, AudioBufferStream::OK); | 509 DCHECK_EQ(status, AudioBufferStream::OK); |
466 DCHECK(buffer.get()); | 510 DCHECK(buffer.get()); |
467 | 511 |
468 if (state_ == kFlushing) { | 512 if (state_ == kFlushing) { |
469 ChangeState_Locked(kFlushed); | 513 ChangeState_Locked(kFlushed); |
470 DoFlush_Locked(); | 514 DoFlush_Locked(); |
471 return; | 515 return; |
472 } | 516 } |
473 | 517 |
474 if (expecting_config_changes_) { | 518 if (expecting_config_changes_) { |
519 if (buffer->sample_rate() != last_decoded_sample_rate_) { | |
520 DVLOG(1) << __FUNCTION__ << " Updating audio sample_rate." | |
521 << " ts:" << buffer->timestamp().InMicroseconds() | |
522 << " old:" << last_decoded_sample_rate_ | |
523 << " new:" << buffer->sample_rate(); | |
524 OnConfigChange(); | |
525 } | |
526 last_decoded_sample_rate_ = buffer->sample_rate(); | |
527 | |
475 DCHECK(buffer_converter_); | 528 DCHECK(buffer_converter_); |
476 buffer_converter_->AddInput(buffer); | 529 buffer_converter_->AddInput(buffer); |
477 while (buffer_converter_->HasNextBuffer()) { | 530 while (buffer_converter_->HasNextBuffer()) { |
478 if (!splicer_->AddInput(buffer_converter_->GetNextBuffer())) { | 531 if (!splicer_->AddInput(buffer_converter_->GetNextBuffer())) { |
479 HandleAbortedReadOrDecodeError(AUDIO_RENDERER_ERROR_SPLICE_FAILED); | 532 HandleAbortedReadOrDecodeError(AUDIO_RENDERER_ERROR_SPLICE_FAILED); |
480 return; | 533 return; |
481 } | 534 } |
482 } | 535 } |
483 } else { | 536 } else { |
484 if (!splicer_->AddInput(buffer)) { | 537 if (!splicer_->AddInput(buffer)) { |
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827 << buffering_state; | 880 << buffering_state; |
828 DCHECK_NE(buffering_state_, buffering_state); | 881 DCHECK_NE(buffering_state_, buffering_state); |
829 lock_.AssertAcquired(); | 882 lock_.AssertAcquired(); |
830 buffering_state_ = buffering_state; | 883 buffering_state_ = buffering_state; |
831 | 884 |
832 task_runner_->PostTask(FROM_HERE, | 885 task_runner_->PostTask(FROM_HERE, |
833 base::Bind(buffering_state_cb_, buffering_state_)); | 886 base::Bind(buffering_state_cb_, buffering_state_)); |
834 } | 887 } |
835 | 888 |
836 } // namespace media | 889 } // namespace media |
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