| // Copyright 2013 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
| #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
| |
| #include "base/atomicops.h" |
| #include "base/files/file.h" |
| #include "base/synchronization/lock.h" |
| #include "base/threading/thread_checker.h" |
| #include "base/time/time.h" |
| #include "content/common/content_export.h" |
| #include "content/renderer/media/aec_dump_message_filter.h" |
| #include "content/renderer/media/webrtc_audio_device_impl.h" |
| #include "media/base/audio_converter.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
| #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "third_party/webrtc/modules/interface/module_common_types.h" |
| |
| namespace blink { |
| class WebMediaConstraints; |
| } |
| |
| namespace media { |
| class AudioBus; |
| class AudioFifo; |
| class AudioParameters; |
| } // namespace media |
| |
| namespace webrtc { |
| class AudioFrame; |
| class TypingDetection; |
| } |
| |
| namespace content { |
| |
| class RTCMediaConstraints; |
| |
| using webrtc::AudioProcessorInterface; |
| |
| // This class owns an object of webrtc::AudioProcessing which contains signal |
| // processing components like AGC, AEC and NS. It enables the components based |
| // on the getUserMedia constraints, processes the data and outputs it in a unit |
| // of 10 ms data chunk. |
| class CONTENT_EXPORT MediaStreamAudioProcessor : |
| NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink), |
| NON_EXPORTED_BASE(public AudioProcessorInterface), |
| NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) { |
| public: |
| // Returns false if |kDisableAudioTrackProcessing| is set to true, otherwise |
| // returns true. |
| static bool IsAudioTrackProcessingEnabled(); |
| |
| // |playout_data_source| is used to register this class as a sink to the |
| // WebRtc playout data for processing AEC. If clients do not enable AEC, |
| // |playout_data_source| won't be used. |
| MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, |
| int effects, |
| WebRtcPlayoutDataSource* playout_data_source); |
| |
| // Called when format of the capture data has changed. |
| // Called on the main render thread. The caller is responsible for stopping |
| // the capture thread before calling this method. |
| // After this method, the capture thread will be changed to a new capture |
| // thread. |
| void OnCaptureFormatChanged(const media::AudioParameters& source_params); |
| |
| // Pushes capture data in |audio_source| to the internal FIFO. |
| // Called on the capture audio thread. |
| void PushCaptureData(const media::AudioBus* audio_source); |
| |
| // Processes a block of 10 ms data from the internal FIFO and outputs it via |
| // |out|. |out| is the address of the pointer that will be pointed to |
| // the post-processed data if the method is returning a true. The lifetime |
| // of the data represeted by |out| is guaranteed to outlive the method call. |
| // That also says *|out| won't change until this method is called again. |
| // |new_volume| receives the new microphone volume from the AGC. |
| // The new microphoen volume range is [0, 255], and the value will be 0 if |
| // the microphone volume should not be adjusted. |
| // Returns true if the internal FIFO has at least 10 ms data for processing, |
| // otherwise false. |
| // |capture_delay|, |volume| and |key_pressed| will be passed to |
| // webrtc::AudioProcessing to help processing the data. |
| // Called on the capture audio thread. |
| bool ProcessAndConsumeData(base::TimeDelta capture_delay, |
| int volume, |
| bool key_pressed, |
| int* new_volume, |
| int16** out); |
| |
| // Stops the audio processor, no more AEC dump or render data after calling |
| // this method. |
| void Stop(); |
| |
| // The audio format of the input to the processor. |
| const media::AudioParameters& InputFormat() const; |
| |
| // The audio format of the output from the processor. |
| const media::AudioParameters& OutputFormat() const; |
| |
| // Accessor to check if the audio processing is enabled or not. |
| bool has_audio_processing() const { return audio_processing_ != NULL; } |
| |
| // AecDumpMessageFilter::AecDumpDelegate implementation. |
| // Called on the main render thread. |
| virtual void OnAecDumpFile( |
| const IPC::PlatformFileForTransit& file_handle) OVERRIDE; |
| virtual void OnDisableAecDump() OVERRIDE; |
| virtual void OnIpcClosing() OVERRIDE; |
| |
| protected: |
| friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>; |
| virtual ~MediaStreamAudioProcessor(); |
| |
| private: |
| friend class MediaStreamAudioProcessorTest; |
| |
| class MediaStreamAudioConverter; |
| |
| // WebRtcPlayoutDataSource::Sink implementation. |
| virtual void OnPlayoutData(media::AudioBus* audio_bus, |
| int sample_rate, |
| int audio_delay_milliseconds) OVERRIDE; |
| virtual void OnPlayoutDataSourceChanged() OVERRIDE; |
| |
| // webrtc::AudioProcessorInterface implementation. |
| // This method is called on the libjingle thread. |
| virtual void GetStats(AudioProcessorStats* stats) OVERRIDE; |
| |
| // Helper to initialize the WebRtc AudioProcessing. |
| void InitializeAudioProcessingModule( |
| const blink::WebMediaConstraints& constraints, int effects); |
| |
| // Helper to initialize the capture converter. |
| void InitializeCaptureConverter(const media::AudioParameters& source_params); |
| |
| // Helper to initialize the render converter. |
| void InitializeRenderConverterIfNeeded(int sample_rate, |
| int number_of_channels, |
| int frames_per_buffer); |
| |
| // Called by ProcessAndConsumeData(). |
| // Returns the new microphone volume in the range of |0, 255]. |
| // When the volume does not need to be updated, it returns 0. |
| int ProcessData(webrtc::AudioFrame* audio_frame, |
| base::TimeDelta capture_delay, |
| int volume, |
| bool key_pressed); |
| |
| // Cached value for the render delay latency. This member is accessed by |
| // both the capture audio thread and the render audio thread. |
| base::subtle::Atomic32 render_delay_ms_; |
| |
| // webrtc::AudioProcessing module which does AEC, AGC, NS, HighPass filter, |
| // ..etc. |
| scoped_ptr<webrtc::AudioProcessing> audio_processing_; |
| |
| // Converter used for the down-mixing and resampling of the capture data. |
| scoped_ptr<MediaStreamAudioConverter> capture_converter_; |
| |
| // AudioFrame used to hold the output of |capture_converter_|. |
| webrtc::AudioFrame capture_frame_; |
| |
| // Converter used for the down-mixing and resampling of the render data when |
| // the AEC is enabled. |
| scoped_ptr<MediaStreamAudioConverter> render_converter_; |
| |
| // AudioFrame used to hold the output of |render_converter_|. |
| webrtc::AudioFrame render_frame_; |
| |
| // Data bus to help converting interleaved data to an AudioBus. |
| scoped_ptr<media::AudioBus> render_data_bus_; |
| |
| // Raw pointer to the WebRtcPlayoutDataSource, which is valid for the |
| // lifetime of RenderThread. |
| WebRtcPlayoutDataSource* playout_data_source_; |
| |
| // Used to DCHECK that the destructor is called on the main render thread. |
| base::ThreadChecker main_thread_checker_; |
| |
| // Used to DCHECK that some methods are called on the capture audio thread. |
| base::ThreadChecker capture_thread_checker_; |
| |
| // Used to DCHECK that PushRenderData() is called on the render audio thread. |
| base::ThreadChecker render_thread_checker_; |
| |
| // Flag to enable the stereo channels mirroring. |
| bool audio_mirroring_; |
| |
| // Used by the typing detection. |
| scoped_ptr<webrtc::TypingDetection> typing_detector_; |
| |
| // This flag is used to show the result of typing detection. |
| // It can be accessed by the capture audio thread and by the libjingle thread |
| // which calls GetStats(). |
| base::subtle::Atomic32 typing_detected_; |
| |
| // Communication with browser for AEC dump. |
| scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_; |
| |
| // Flag to avoid executing Stop() more than once. |
| bool stopped_; |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |