content/renderer/media/media_stream_audio_processor.h

   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_MEDIA_STREAM_AUDIO_PROCESSOR_H_
   6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
   7
   8 #include "base/atomicops.h"
   9 #include "base/files/file.h"
  10 #include "base/synchronization/lock.h"
  11 #include "base/threading/thread_checker.h"
  12 #include "base/time/time.h"
  13 #include "content/common/content_export.h"
  14 #include "content/renderer/media/aec_dump_message_filter.h"
  15 #include "content/renderer/media/webrtc_audio_device_impl.h"
  16 #include "media/base/audio_converter.h"
  17 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  18 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
  19 #include "third_party/webrtc/modules/interface/module_common_types.h"
  20
  21 namespace blink {
  22 class WebMediaConstraints;
  23 }
  24
  25 namespace media {
  26 class AudioBus;
  27 class AudioFifo;
  28 class AudioParameters;
  29 }  // namespace media
  30
  31 namespace webrtc {
  32 class AudioFrame;
  33 class TypingDetection;
  34 }
  35
  36 namespace content {
  37
  38 class MediaStreamAudioBus;
  39 class MediaStreamAudioFifo;
  40 class RTCMediaConstraints;
  41
  42 using webrtc::AudioProcessorInterface;
  43
  44 // This class owns an object of webrtc::AudioProcessing which contains signal
  45 // processing components like AGC, AEC and NS. It enables the components based
  46 // on the getUserMedia constraints, processes the data and outputs it in a unit
  47 // of 10 ms data chunk.
  48 class CONTENT_EXPORT MediaStreamAudioProcessor :
  49     NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
  50     NON_EXPORTED_BASE(public AudioProcessorInterface),
  51     NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
  52  public:
  53   // Returns false if |kDisableAudioTrackProcessing| is set to true, otherwise
  54   // returns true.
  55   static bool IsAudioTrackProcessingEnabled();
  56
  57   // |playout_data_source| is used to register this class as a sink to the
  58   // WebRtc playout data for processing AEC. If clients do not enable AEC,
  59   // |playout_data_source| won't be used.
  60   MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints,
  61                             int effects,
  62                             WebRtcPlayoutDataSource* playout_data_source);
  63
  64   // Called when the format of the capture data has changed.
  65   // Called on the main render thread. The caller is responsible for stopping
  66   // the capture thread before calling this method.
  67   // After this method, the capture thread will be changed to a new capture
  68   // thread.
  69   void OnCaptureFormatChanged(const media::AudioParameters& source_params);
  70
  71   // Pushes capture data in |audio_source| to the internal FIFO. Each call to
  72   // this method should be followed by calls to ProcessAndConsumeData() while
  73   // it returns false, to pull out all available data.
  74   // Called on the capture audio thread.
  75   void PushCaptureData(const media::AudioBus* audio_source);
  76
  77   // Processes a block of 10 ms data from the internal FIFO and outputs it via
  78   // |out|. |out| is the address of the pointer that will be pointed to
  79   // the post-processed data if the method is returning a true. The lifetime
  80   // of the data represeted by |out| is guaranteed until this method is called
  81   // again.
  82   // |new_volume| receives the new microphone volume from the AGC.
  83   // The new microphone volume range is [0, 255], and the value will be 0 if
  84   // the microphone volume should not be adjusted.
  85   // Returns true if the internal FIFO has at least 10 ms data for processing,
  86   // otherwise false.
  87   // Called on the capture audio thread.
  88   //
  89   // TODO(ajm): Don't we want this to output float?
  90   bool ProcessAndConsumeData(base::TimeDelta capture_delay,
  91                              int volume,
  92                              bool key_pressed,
  93                              int* new_volume,
  94                              int16** out);
  95
  96   // Stops the audio processor, no more AEC dump or render data after calling
  97   // this method.
  98   void Stop();
  99
 100   // The audio formats of the capture input to and output from the processor.
 101   // Must only be called on the main render or audio capture threads.
 102   const media::AudioParameters& InputFormat() const;
 103   const media::AudioParameters& OutputFormat() const;
 104
 105   // Accessor to check if the audio processing is enabled or not.
 106   bool has_audio_processing() const { return audio_processing_ != NULL; }
 107
 108   // AecDumpMessageFilter::AecDumpDelegate implementation.
 109   // Called on the main render thread.
 110   virtual void OnAecDumpFile(
 111       const IPC::PlatformFileForTransit& file_handle) OVERRIDE;
 112   virtual void OnDisableAecDump() OVERRIDE;
 113   virtual void OnIpcClosing() OVERRIDE;
 114
 115  protected:
 116   friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>;
 117   virtual ~MediaStreamAudioProcessor();
 118
 119  private:
 120   friend class MediaStreamAudioProcessorTest;
 121   FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
 122                            GetAecDumpMessageFilter);
 123
 124   // WebRtcPlayoutDataSource::Sink implementation.
 125   virtual void OnPlayoutData(media::AudioBus* audio_bus,
 126                              int sample_rate,
 127                              int audio_delay_milliseconds) OVERRIDE;
 128   virtual void OnPlayoutDataSourceChanged() OVERRIDE;
 129
 130   // webrtc::AudioProcessorInterface implementation.
 131   // This method is called on the libjingle thread.
 132   virtual void GetStats(AudioProcessorStats* stats) OVERRIDE;
 133
 134   // Helper to initialize the WebRtc AudioProcessing.
 135   void InitializeAudioProcessingModule(
 136       const blink::WebMediaConstraints& constraints, int effects);
 137
 138   // Helper to initialize the capture converter.
 139   void InitializeCaptureFifo(const media::AudioParameters& input_format);
 140
 141   // Helper to initialize the render converter.
 142   void InitializeRenderFifoIfNeeded(int sample_rate,
 143                                     int number_of_channels,
 144                                     int frames_per_buffer);
 145
 146   // Called by ProcessAndConsumeData().
 147   // Returns the new microphone volume in the range of |0, 255].
 148   // When the volume does not need to be updated, it returns 0.
 149   int ProcessData(const float* const* process_ptrs,
 150                   int process_frames,
 151                   base::TimeDelta capture_delay,
 152                   int volume,
 153                   bool key_pressed,
 154                   float* const* output_ptrs);
 155
 156   // Cached value for the render delay latency. This member is accessed by
 157   // both the capture audio thread and the render audio thread.
 158   base::subtle::Atomic32 render_delay_ms_;
 159
 160   // Module to handle processing and format conversion.
 161   scoped_ptr<webrtc::AudioProcessing> audio_processing_;
 162
 163   // FIFO to provide 10 ms capture chunks.
 164   scoped_ptr<MediaStreamAudioFifo> capture_fifo_;
 165   // Receives processing output.
 166   scoped_ptr<MediaStreamAudioBus> output_bus_;
 167   // Receives interleaved int16 data for output.
 168   scoped_ptr<int16[]> output_data_;
 169
 170   // FIFO to provide 10 ms render chunks when the AEC is enabled.
 171   scoped_ptr<MediaStreamAudioFifo> render_fifo_;
 172
 173   // These are mutated on the main render thread in OnCaptureFormatChanged().
 174   // The caller guarantees this does not run concurrently with accesses on the
 175   // capture audio thread.
 176   media::AudioParameters input_format_;
 177   media::AudioParameters output_format_;
 178   // Only used on the render audio thread.
 179   media::AudioParameters render_format_;
 180
 181   // Raw pointer to the WebRtcPlayoutDataSource, which is valid for the
 182   // lifetime of RenderThread.
 183   WebRtcPlayoutDataSource* playout_data_source_;
 184
 185   // Used to DCHECK that some methods are called on the main render thread.
 186   base::ThreadChecker main_thread_checker_;
 187   // Used to DCHECK that some methods are called on the capture audio thread.
 188   base::ThreadChecker capture_thread_checker_;
 189   // Used to DCHECK that some methods are called on the render audio thread.
 190   base::ThreadChecker render_thread_checker_;
 191
 192   // Flag to enable stereo channel mirroring.
 193   bool audio_mirroring_;
 194
 195   scoped_ptr<webrtc::TypingDetection> typing_detector_;
 196   // This flag is used to show the result of typing detection.
 197   // It can be accessed by the capture audio thread and by the libjingle thread
 198   // which calls GetStats().
 199   base::subtle::Atomic32 typing_detected_;
 200
 201   // Communication with browser for AEC dump.
 202   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
 203
 204   // Flag to avoid executing Stop() more than once.
 205   bool stopped_;
 206 };
 207
 208 }  // namespace content
 209
 210 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_