Index: content/renderer/media/webrtc_audio_processor.h |
diff --git a/content/renderer/media/webrtc_audio_processor.h b/content/renderer/media/webrtc_audio_processor.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..fa0c5c1be37590013e16b9d5ea043cb993f43a31 |
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+// 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_WEBRTC_AUDIO_PROCESSOR_H_ |
+#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ |
+ |
+#include "base/atomicops.h" |
+#include "base/synchronization/lock.h" |
+#include "base/threading/thread_checker.h" |
+#include "base/time/time.h" |
+#include "content/common/content_export.h" |
+#include "media/base/audio_converter.h" |
+#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" |
+#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h" |
+#include "third_party/webrtc/modules/interface/module_common_types.h" |
+ |
+namespace media { |
+class AudioBus; |
+class AudioFifo; |
+class AudioParameters; |
+} // namespace media |
+ |
+namespace webrtc { |
+class AudioFrame; |
+} |
+ |
+namespace content { |
+ |
+// 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 WebRtcAudioProcessor { |
+ public: |
+ explicit WebRtcAudioProcessor( |
+ const webrtc::MediaConstraintsInterface* constraints); |
+ ~WebRtcAudioProcessor(); |
+ |
+ // Pushes capture data in |audio_source| to the internal FIFO. |
+ // Called on the capture audio thread. |
+ void PushCaptureData(media::AudioBus* audio_source); |
+ |
+ // Push the render audio to webrtc::AudioProcessing for analysis. This is |
+ // needed iff echo processing is enabled. |
+ // |render_audio| is the pointer to the render audio data, its format |
+ // is specified by |sample_rate|, |number_of_channels| and |number_of_frames|. |
+ // Called on the render audio thread. |
+ void PushRenderData(const int16* render_audio, |
+ int sample_rate, |
+ int number_of_channels, |
+ int number_of_frames, |
+ base::TimeDelta render_delay); |
+ |
+ // 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. |
+ // 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, |
+ int16** out); |
+ |
+ // Called when the format of the capture data has changed. |
+ // This has to be called before PushCaptureData() and ProcessAndConsumeData(). |
+ // Called on the main render thread. |
+ void SetCaptureFormat(const media::AudioParameters& source_params); |
+ |
+ // 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_.get() != NULL; } |
+ |
+ private: |
+ class WebRtcAudioConverter; |
+ |
+ // Helper to initialize the WebRtc AudioProcessing. |
+ void InitializeAudioProcessingModule( |
+ const webrtc::MediaConstraintsInterface* constraints); |
+ |
+ // Helper to initialize the render converter. |
+ void InitializeRenderConverterIfNeeded(int sample_rate, |
+ int number_of_channels, |
+ int frames_per_buffer); |
+ |
+ // Called by ProcessAndConsumeData(). |
+ void ProcessData(webrtc::AudioFrame* audio_frame, |
+ base::TimeDelta capture_delay, |
+ int volume, |
+ bool key_pressed); |
+ |
+ // Called when the processor is going away. |
+ void StopAudioProcessing(); |
+ |
+ // 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<WebRtcAudioConverter> 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<WebRtcAudioConverter> 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_; |
+ |
+ // Used to DCHECK that some methods are 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_; |
+}; |
+ |
+} // namespace content |
+ |
+#endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_ |