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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> |
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| 29 #include "media/base/timestamp_constants.h" | 29 #include "media/base/timestamp_constants.h" |
| 30 #include "media/filters/audio_clock.h" | 30 #include "media/filters/audio_clock.h" |
| 31 #include "media/filters/decrypting_demuxer_stream.h" | 31 #include "media/filters/decrypting_demuxer_stream.h" |
| 32 | 32 |
| 33 namespace media { | 33 namespace media { |
| 34 | 34 |
| 35 AudioRendererImpl::AudioRendererImpl( | 35 AudioRendererImpl::AudioRendererImpl( |
| 36 const scoped_refptr<base::SingleThreadTaskRunner>& task_runner, | 36 const scoped_refptr<base::SingleThreadTaskRunner>& task_runner, |
| 37 media::AudioRendererSink* sink, | 37 media::AudioRendererSink* sink, |
| 38 ScopedVector<AudioDecoder> decoders, | 38 ScopedVector<AudioDecoder> decoders, |
| 39 const AudioHardwareConfig& hardware_config, | |
| 40 const scoped_refptr<MediaLog>& media_log) | 39 const scoped_refptr<MediaLog>& media_log) |
| 41 : task_runner_(task_runner), | 40 : task_runner_(task_runner), |
| 42 expecting_config_changes_(false), | 41 expecting_config_changes_(false), |
| 43 sink_(sink), | 42 sink_(sink), |
| 44 audio_buffer_stream_( | 43 audio_buffer_stream_( |
| 45 new AudioBufferStream(task_runner, std::move(decoders), media_log)), | 44 new AudioBufferStream(task_runner, std::move(decoders), media_log)), |
| 46 hardware_config_(hardware_config), | |
| 47 media_log_(media_log), | 45 media_log_(media_log), |
| 48 client_(nullptr), | 46 client_(nullptr), |
| 49 tick_clock_(new base::DefaultTickClock()), | 47 tick_clock_(new base::DefaultTickClock()), |
| 50 last_audio_memory_usage_(0), | 48 last_audio_memory_usage_(0), |
| 51 last_decoded_sample_rate_(0), | 49 last_decoded_sample_rate_(0), |
| 52 playback_rate_(0.0), | 50 playback_rate_(0.0), |
| 53 state_(kUninitialized), | 51 state_(kUninitialized), |
| 54 buffering_state_(BUFFERING_HAVE_NOTHING), | 52 buffering_state_(BUFFERING_HAVE_NOTHING), |
| 55 rendering_(false), | 53 rendering_(false), |
| 56 sink_playing_(false), | 54 sink_playing_(false), |
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| 326 DCHECK_EQ(kUninitialized, state_); | 324 DCHECK_EQ(kUninitialized, state_); |
| 327 DCHECK(sink_.get()); | 325 DCHECK(sink_.get()); |
| 328 | 326 |
| 329 state_ = kInitializing; | 327 state_ = kInitializing; |
| 330 client_ = client; | 328 client_ = client; |
| 331 | 329 |
| 332 // Always post |init_cb_| because |this| could be destroyed if initialization | 330 // Always post |init_cb_| because |this| could be destroyed if initialization |
| 333 // failed. | 331 // failed. |
| 334 init_cb_ = BindToCurrentLoop(init_cb); | 332 init_cb_ = BindToCurrentLoop(init_cb); |
| 335 | 333 |
| 336 const AudioParameters& hw_params = hardware_config_.GetOutputConfig(); | 334 const AudioParameters& hw_params = |
|
DaleCurtis
2016/05/26 19:18:09
Do you know what this returns when the element is
o1ka
2016/05/27 13:48:52
This is a good question.
Here
https://code.google
| |
| 335 sink_->GetOutputDeviceInfo().output_params(); | |
| 337 expecting_config_changes_ = stream->SupportsConfigChanges(); | 336 expecting_config_changes_ = stream->SupportsConfigChanges(); |
| 338 if (!expecting_config_changes_ || !hw_params.IsValid() || | 337 if (!expecting_config_changes_ || !hw_params.IsValid() || |
| 339 hw_params.format() == AudioParameters::AUDIO_FAKE) { | 338 hw_params.format() == AudioParameters::AUDIO_FAKE) { |
| 340 // The actual buffer size is controlled via the size of the AudioBus | 339 // The actual buffer size is controlled via the size of the AudioBus |
| 341 // provided to Render(), so just choose something reasonable here for looks. | 340 // provided to Render(), so just choose something reasonable here for looks. |
| 342 int buffer_size = stream->audio_decoder_config().samples_per_second() / 100; | 341 int buffer_size = stream->audio_decoder_config().samples_per_second() / 100; |
| 343 audio_parameters_.Reset( | 342 audio_parameters_.Reset( |
| 344 AudioParameters::AUDIO_PCM_LOW_LATENCY, | 343 AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| 345 stream->audio_decoder_config().channel_layout(), | 344 stream->audio_decoder_config().channel_layout(), |
| 346 stream->audio_decoder_config().samples_per_second(), | 345 stream->audio_decoder_config().samples_per_second(), |
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| 926 DCHECK_NE(buffering_state_, buffering_state); | 925 DCHECK_NE(buffering_state_, buffering_state); |
| 927 lock_.AssertAcquired(); | 926 lock_.AssertAcquired(); |
| 928 buffering_state_ = buffering_state; | 927 buffering_state_ = buffering_state; |
| 929 | 928 |
| 930 task_runner_->PostTask( | 929 task_runner_->PostTask( |
| 931 FROM_HERE, base::Bind(&AudioRendererImpl::OnBufferingStateChange, | 930 FROM_HERE, base::Bind(&AudioRendererImpl::OnBufferingStateChange, |
| 932 weak_factory_.GetWeakPtr(), buffering_state_)); | 931 weak_factory_.GetWeakPtr(), buffering_state_)); |
| 933 } | 932 } |
| 934 | 933 |
| 935 } // namespace media | 934 } // namespace media |
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