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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 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
| 7 | 7 |
| 8 #include "base/atomicops.h" | 8 #include "base/atomicops.h" |
| 9 #include "base/files/file.h" | 9 #include "base/files/file.h" |
| 10 #include "base/gtest_prod_util.h" | 10 #include "base/gtest_prod_util.h" |
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| 114 | 114 |
| 115 private: | 115 private: |
| 116 friend class MediaStreamAudioProcessorTest; | 116 friend class MediaStreamAudioProcessorTest; |
| 117 | 117 |
| 118 FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest, | 118 FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest, |
| 119 GetAecDumpMessageFilter); | 119 GetAecDumpMessageFilter); |
| 120 | 120 |
| 121 // WebRtcPlayoutDataSource::Sink implementation. | 121 // WebRtcPlayoutDataSource::Sink implementation. |
| 122 void OnPlayoutData(media::AudioBus* audio_bus, | 122 void OnPlayoutData(media::AudioBus* audio_bus, |
| 123 int sample_rate, | 123 int sample_rate, |
| 124 int audio_delay_milliseconds) override; | 124 int audio_delay_milliseconds, |
| 125 uint32_t skipped_frames) override; |
| 125 void OnPlayoutDataSourceChanged() override; | 126 void OnPlayoutDataSourceChanged() override; |
| 126 | 127 |
| 127 // webrtc::AudioProcessorInterface implementation. | 128 // webrtc::AudioProcessorInterface implementation. |
| 128 // This method is called on the libjingle thread. | 129 // This method is called on the libjingle thread. |
| 129 void GetStats(AudioProcessorStats* stats) override; | 130 void GetStats(AudioProcessorStats* stats) override; |
| 130 | 131 |
| 131 // Helper to initialize the WebRtc AudioProcessing. | 132 // Helper to initialize the WebRtc AudioProcessing. |
| 132 void InitializeAudioProcessingModule( | 133 void InitializeAudioProcessingModule( |
| 133 const blink::WebMediaConstraints& constraints, | 134 const blink::WebMediaConstraints& constraints, |
| 134 const MediaStreamDevice::AudioDeviceParameters& input_params); | 135 const MediaStreamDevice::AudioDeviceParameters& input_params); |
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| 200 // Communication with browser for AEC dump. | 201 // Communication with browser for AEC dump. |
| 201 scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_; | 202 scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_; |
| 202 | 203 |
| 203 // Flag to avoid executing Stop() more than once. | 204 // Flag to avoid executing Stop() more than once. |
| 204 bool stopped_; | 205 bool stopped_; |
| 205 | 206 |
| 206 // Object for logging echo information when the AEC is enabled. Accessible by | 207 // Object for logging echo information when the AEC is enabled. Accessible by |
| 207 // the libjingle thread through GetStats(). | 208 // the libjingle thread through GetStats(). |
| 208 scoped_ptr<EchoInformation> echo_information_; | 209 scoped_ptr<EchoInformation> echo_information_; |
| 209 | 210 |
| 211 // Stores the number of skipped output frames. The APM only accepts 10 ms |
| 212 // chunks of data, so we ned to store it until it reaches that amount. |
| 213 uint32_t skipped_output_frames_; |
| 214 |
| 210 DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor); | 215 DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor); |
| 211 }; | 216 }; |
| 212 | 217 |
| 213 } // namespace content | 218 } // namespace content |
| 214 | 219 |
| 215 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ | 220 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
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