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| 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 |
| 3 // found in the LICENSE file. |
| 4 |
| 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ |
| 7 |
| 8 #include "base/synchronization/lock.h" |
| 9 #include "base/threading/thread_checker.h" |
| 10 #include "content/common/content_export.h" |
| 11 #include "media/base/audio_converter.h" |
| 12 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface
.h" |
| 13 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h
" |
| 14 #include "third_party/webrtc/modules/interface/module_common_types.h" |
| 15 |
| 16 namespace media { |
| 17 class AudioBus; |
| 18 class AudioFifo; |
| 19 class AudioParameters; |
| 20 } // namespace media |
| 21 |
| 22 namespace webrtc { |
| 23 class AudioFrame; |
| 24 } |
| 25 |
| 26 namespace content { |
| 27 |
| 28 // This class owns an object of webrtc::AudioProcessing which contains signal |
| 29 // processing components like AGC, AEC and NS. It enables the components based |
| 30 // on the constraints, processes the data and outputs it in a unit of 10 ms |
| 31 // data chunk. |
| 32 class CONTENT_EXPORT WebRtcAudioProcessor { |
| 33 public: |
| 34 explicit WebRtcAudioProcessor( |
| 35 const webrtc::MediaConstraintsInterface* constraints); |
| 36 ~WebRtcAudioProcessor(); |
| 37 |
| 38 // Pushes capture data in |audio_source| to the internal FIFO. |
| 39 // Called on the capture audio thread. |
| 40 void PushCaptureData(media::AudioBus* audio_source); |
| 41 |
| 42 // Processes a block of 10 ms data from the internal FIFO and outputs it via |
| 43 // |out|. |
| 44 // Returns true if the internal FIFO has at least 10ms data for processing, |
| 45 // otherwise false. |
| 46 // Called on the capture audio thread. |
| 47 bool ProcessAndConsumeData(int capture_audio_delay_ms, |
| 48 int volume, |
| 49 bool key_pressed, |
| 50 int16** out); |
| 51 |
| 52 // Called when the format of the capture data has changed. |
| 53 // Called on the main render thread. |
| 54 void SetCaptureFormat(const media::AudioParameters& source_params); |
| 55 |
| 56 // Push the render audio to WebRtc::AudioProcessing for analysis. This is |
| 57 // needed iff echo processing is enabled. |
| 58 // Called on the render audio thread. |
| 59 void PushRenderData(const int16* render_audio, |
| 60 int sample_rate, |
| 61 int number_of_channels, |
| 62 int number_of_frames, |
| 63 int render_delay_ms); |
| 64 |
| 65 // The audio format of the output from the processor. |
| 66 const media::AudioParameters& OutputFormat() const; |
| 67 |
| 68 // Accessor to check if the audio processing is enabled or not. |
| 69 bool has_audio_processing() const { return audio_processing_.get() != NULL; } |
| 70 |
| 71 private: |
| 72 class WebRtcAudioConverter; |
| 73 |
| 74 // Helper to initialize the WebRtc AudioProcessing. |
| 75 void InitializeAudioProcessingModule( |
| 76 const webrtc::MediaConstraintsInterface* constraints); |
| 77 |
| 78 // Helper to initialize the render converter. |
| 79 void InitializeRenderConverterIfNeeded(int sample_rate, |
| 80 int number_of_channels, |
| 81 int frames_per_buffer); |
| 82 |
| 83 // Called by ProcessAndConsume10MsData(). |
| 84 void ProcessData(int audio_delay_milliseconds, |
| 85 int volume, |
| 86 bool key_pressed); |
| 87 |
| 88 // Called when the processor is going away. |
| 89 void StopAudioProcessing(); |
| 90 |
| 91 // Cached value for the render delay latency. |
| 92 int render_delay_ms_; |
| 93 |
| 94 // Protects |render_delay_ms_|. |
| 95 // TODO(xians): Can we get rid of the lock? |
| 96 mutable base::Lock lock_; |
| 97 |
| 98 // WebRtc AudioProcessing module which does AEC, AGC, NS, HighPass filter, |
| 99 // ..etc. |
| 100 scoped_ptr<webrtc::AudioProcessing> audio_processing_; |
| 101 |
| 102 // Converter used for the down-mixing and resampling of the capture data. |
| 103 scoped_ptr<WebRtcAudioConverter> capture_converter_; |
| 104 |
| 105 // Converter used for the down-mixing and resampling of the render data when |
| 106 // the AEC is enabled. |
| 107 scoped_ptr<WebRtcAudioConverter> render_converter_; |
| 108 |
| 109 // Data bus to help converting interleaved data to an AudioBus. |
| 110 scoped_ptr<media::AudioBus> render_data_bus_; |
| 111 |
| 112 // Used to DCHECK that some methods are called on the correct thread. |
| 113 base::ThreadChecker thread_checker_; |
| 114 }; |
| 115 |
| 116 } // namespace content |
| 117 |
| 118 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ |
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