namespace content {
+class MediaStreamAudioBus;
+class MediaStreamAudioFifo;
class RTCMediaConstraints;
using webrtc::AudioProcessorInterface;
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
+ // Called when the 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.
+ // Pushes capture data in |audio_source| to the internal FIFO. Each call to
+ // this method should be followed by calls to ProcessAndConsumeData() while
+ // it returns false, to pull out all available data.
// 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.
+ // of the data represeted by |out| is guaranteed 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 new microphone 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.
+ //
+ // TODO(ajm): Don't we want this to output float?
bool ProcessAndConsumeData(base::TimeDelta capture_delay,
int volume,
bool key_pressed,
// this method.
void Stop();
- // The audio format of the input to the processor.
+ // The audio formats of the capture input to and output from the processor.
+ // Must only be called on the main render or audio capture threads.
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.
private:
friend class MediaStreamAudioProcessorTest;
-
- class MediaStreamAudioConverter;
+ FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
+ GetAecDumpMessageFilter);
// WebRtcPlayoutDataSource::Sink implementation.
virtual void OnPlayoutData(media::AudioBus* audio_bus,
const blink::WebMediaConstraints& constraints, int effects);
// Helper to initialize the capture converter.
- void InitializeCaptureConverter(const media::AudioParameters& source_params);
+ void InitializeCaptureFifo(const media::AudioParameters& input_format);
// Helper to initialize the render converter.
- void InitializeRenderConverterIfNeeded(int sample_rate,
- int number_of_channels,
- int frames_per_buffer);
+ void InitializeRenderFifoIfNeeded(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,
+ int ProcessData(const float* const* process_ptrs,
+ int process_frames,
base::TimeDelta capture_delay,
int volume,
- bool key_pressed);
+ bool key_pressed,
+ float* const* output_ptrs);
// 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.
+ // Module to handle processing and format conversion.
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_;
+ // FIFO to provide 10 ms capture chunks.
+ scoped_ptr<MediaStreamAudioFifo> capture_fifo_;
+ // Receives processing output.
+ scoped_ptr<MediaStreamAudioBus> output_bus_;
+ // Receives interleaved int16 data for output.
+ scoped_ptr<int16[]> output_data_;
- // AudioFrame used to hold the output of |render_converter_|.
- webrtc::AudioFrame render_frame_;
+ // FIFO to provide 10 ms render chunks when the AEC is enabled.
+ scoped_ptr<MediaStreamAudioFifo> render_fifo_;
- // Data bus to help converting interleaved data to an AudioBus.
- scoped_ptr<media::AudioBus> render_data_bus_;
+ // These are mutated on the main render thread in OnCaptureFormatChanged().
+ // The caller guarantees this does not run concurrently with accesses on the
+ // capture audio thread.
+ media::AudioParameters input_format_;
+ media::AudioParameters output_format_;
+ // Only used on the render audio thread.
+ media::AudioParameters render_format_;
// 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.
+ // 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.
+ // Used to DCHECK that some methods are called on the render audio thread.
base::ThreadChecker render_thread_checker_;
- // Flag to enable the stereo channels mirroring.
+ // Flag to enable stereo channel 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().
projects / platform / framework / web / crosswalk.git / blobdiff