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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 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 "content/renderer/media/webrtc_local_audio_source_provider.h" | 5 #include "content/renderer/media/webrtc_local_audio_source_provider.h" |
6 | 6 |
7 #include "base/logging.h" | 7 #include "base/logging.h" |
8 #include "content/renderer/render_thread_impl.h" | 8 #include "content/renderer/render_thread_impl.h" |
9 #include "media/audio/audio_parameters.h" | 9 #include "media/audio/audio_parameters.h" |
10 #include "media/base/audio_fifo.h" | 10 #include "media/base/audio_fifo.h" |
11 #include "media/base/audio_hardware_config.h" | 11 #include "media/base/audio_hardware_config.h" |
12 #include "third_party/WebKit/public/web/WebAudioSourceProviderClient.h" | 12 #include "third_party/WebKit/public/web/WebAudioSourceProviderClient.h" |
13 | 13 |
14 using WebKit::WebVector; | 14 using blink::WebVector; |
15 | 15 |
16 namespace content { | 16 namespace content { |
17 | 17 |
18 static const size_t kMaxNumberOfBuffers = 10; | 18 static const size_t kMaxNumberOfBuffers = 10; |
19 | 19 |
20 // Size of the buffer that WebAudio processes each time, it is the same value | 20 // Size of the buffer that WebAudio processes each time, it is the same value |
21 // as AudioNode::ProcessingSizeInFrames in WebKit. | 21 // as AudioNode::ProcessingSizeInFrames in WebKit. |
22 // static | 22 // static |
23 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128; | 23 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128; |
24 | 24 |
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100 elapsed_ms = static_cast<int>( | 100 elapsed_ms = static_cast<int>( |
101 (base::TimeTicks::Now() - last_fill_).InMilliseconds()); | 101 (base::TimeTicks::Now() - last_fill_).InMilliseconds()); |
102 } | 102 } |
103 *delay_ms = audio_delay_ms_ + elapsed_ms + static_cast<int>( | 103 *delay_ms = audio_delay_ms_ + elapsed_ms + static_cast<int>( |
104 1000 * fifo_->frames() / source_params_.sample_rate() + 0.5); | 104 1000 * fifo_->frames() / source_params_.sample_rate() + 0.5); |
105 *volume = volume_; | 105 *volume = volume_; |
106 *key_pressed = key_pressed_; | 106 *key_pressed = key_pressed_; |
107 } | 107 } |
108 | 108 |
109 void WebRtcLocalAudioSourceProvider::setClient( | 109 void WebRtcLocalAudioSourceProvider::setClient( |
110 WebKit::WebAudioSourceProviderClient* client) { | 110 blink::WebAudioSourceProviderClient* client) { |
111 NOTREACHED(); | 111 NOTREACHED(); |
112 } | 112 } |
113 | 113 |
114 void WebRtcLocalAudioSourceProvider::provideInput( | 114 void WebRtcLocalAudioSourceProvider::provideInput( |
115 const WebVector<float*>& audio_data, size_t number_of_frames) { | 115 const WebVector<float*>& audio_data, size_t number_of_frames) { |
116 DCHECK_EQ(number_of_frames, kWebAudioRenderBufferSize); | 116 DCHECK_EQ(number_of_frames, kWebAudioRenderBufferSize); |
117 if (!bus_wrapper_ || | 117 if (!bus_wrapper_ || |
118 static_cast<size_t>(bus_wrapper_->channels()) != audio_data.size()) { | 118 static_cast<size_t>(bus_wrapper_->channels()) != audio_data.size()) { |
119 bus_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size()); | 119 bus_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size()); |
120 } | 120 } |
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146 return 1.0; | 146 return 1.0; |
147 } | 147 } |
148 | 148 |
149 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting( | 149 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting( |
150 const media::AudioParameters& sink_params) { | 150 const media::AudioParameters& sink_params) { |
151 DCHECK(thread_checker_.CalledOnValidThread()); | 151 DCHECK(thread_checker_.CalledOnValidThread()); |
152 sink_params_ = sink_params; | 152 sink_params_ = sink_params; |
153 } | 153 } |
154 | 154 |
155 } // namespace content | 155 } // namespace content |
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