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 EchoInformation;
  39 class MediaStreamAudioBus;
  40 class MediaStreamAudioFifo;
  41 class RTCMediaConstraints;
  42
  43 using webrtc::AudioProcessorInterface;
  44
  45 // This class owns an object of webrtc::AudioProcessing which contains signal
  46 // processing components like AGC, AEC and NS. It enables the components based
  47 // on the getUserMedia constraints, processes the data and outputs it in a unit
  48 // of 10 ms data chunk.
  49 class CONTENT_EXPORT MediaStreamAudioProcessor :
  50     NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
  51     NON_EXPORTED_BASE(public AudioProcessorInterface),
  52     NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
  53  public:
  54   // |playout_data_source| is used to register this class as a sink to the
  55   // WebRtc playout data for processing AEC. If clients do not enable AEC,
  56   // |playout_data_source| won't be used.
  57   MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints,
  58                             int effects,
  59                             WebRtcPlayoutDataSource* playout_data_source);
  60
  61   // Called when the format of the capture data has changed.
  62   // Called on the main render thread. The caller is responsible for stopping
  63   // the capture thread before calling this method.
  64   // After this method, the capture thread will be changed to a new capture
  65   // thread.
  66   void OnCaptureFormatChanged(const media::AudioParameters& source_params);
  67
  68   // Pushes capture data in |audio_source| to the internal FIFO. Each call to
  69   // this method should be followed by calls to ProcessAndConsumeData() while
  70   // it returns false, to pull out all available data.
  71   // Called on the capture audio thread.
  72   void PushCaptureData(const media::AudioBus& audio_source,
  73                        base::TimeDelta capture_delay);
  74
  75   // Processes a block of 10 ms data from the internal FIFO, returning true if
  76   // |processed_data| contains the result. Returns false and does not modify the
  77   // outputs if the internal FIFO has insufficient data. The caller does NOT own
  78   // the object pointed to by |*processed_data|.
  79   // |capture_delay| is an adjustment on the |capture_delay| value provided in
  80   // the last call to PushCaptureData().
  81   // |new_volume| receives the new microphone volume from the AGC.
  82   // The new microphone volume range is [0, 255], and the value will be 0 if
  83   // the microphone volume should not be adjusted.
  84   // Called on the capture audio thread.
  85   bool ProcessAndConsumeData(
  86       int volume,
  87       bool key_pressed,
  88       media::AudioBus** processed_data,
  89       base::TimeDelta* capture_delay,
  90       int* new_volume);
  91
  92   // Stops the audio processor, no more AEC dump or render data after calling
  93   // this method.
  94   void Stop();
  95
  96   // The audio formats of the capture input to and output from the processor.
  97   // Must only be called on the main render or audio capture threads.
  98   const media::AudioParameters& InputFormat() const;
  99   const media::AudioParameters& OutputFormat() const;
 100
 101   // Accessor to check if the audio processing is enabled or not.
 102   bool has_audio_processing() const { return audio_processing_ != NULL; }
 103
 104   // AecDumpMessageFilter::AecDumpDelegate implementation.
 105   // Called on the main render thread.
 106   void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
 107   void OnDisableAecDump() override;
 108   void OnIpcClosing() override;
 109
 110  protected:
 111   ~MediaStreamAudioProcessor() override;
 112
 113  private:
 114   friend class MediaStreamAudioProcessorTest;
 115   FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
 116                            GetAecDumpMessageFilter);
 117
 118   // WebRtcPlayoutDataSource::Sink implementation.
 119   void OnPlayoutData(media::AudioBus* audio_bus,
 120                      int sample_rate,
 121                      int audio_delay_milliseconds) override;
 122   void OnPlayoutDataSourceChanged() override;
 123
 124   // webrtc::AudioProcessorInterface implementation.
 125   // This method is called on the libjingle thread.
 126   void GetStats(AudioProcessorStats* stats) override;
 127
 128   // Helper to initialize the WebRtc AudioProcessing.
 129   void InitializeAudioProcessingModule(
 130       const blink::WebMediaConstraints& constraints, int effects);
 131   void ConfigureBeamforming(webrtc::Config* config);
 132
 133   // Helper to initialize the capture converter.
 134   void InitializeCaptureFifo(const media::AudioParameters& input_format);
 135
 136   // Helper to initialize the render converter.
 137   void InitializeRenderFifoIfNeeded(int sample_rate,
 138                                     int number_of_channels,
 139                                     int frames_per_buffer);
 140
 141   // Called by ProcessAndConsumeData().
 142   // Returns the new microphone volume in the range of |0, 255].
 143   // When the volume does not need to be updated, it returns 0.
 144   int ProcessData(const float* const* process_ptrs,
 145                   int process_frames,
 146                   base::TimeDelta capture_delay,
 147                   int volume,
 148                   bool key_pressed,
 149                   float* const* output_ptrs);
 150
 151   // Cached value for the render delay latency. This member is accessed by
 152   // both the capture audio thread and the render audio thread.
 153   base::subtle::Atomic32 render_delay_ms_;
 154
 155   // Module to handle processing and format conversion.
 156   scoped_ptr<webrtc::AudioProcessing> audio_processing_;
 157
 158   // FIFO to provide 10 ms capture chunks.
 159   scoped_ptr<MediaStreamAudioFifo> capture_fifo_;
 160   // Receives processing output.
 161   scoped_ptr<MediaStreamAudioBus> output_bus_;
 162
 163   // FIFO to provide 10 ms render chunks when the AEC is enabled.
 164   scoped_ptr<MediaStreamAudioFifo> render_fifo_;
 165
 166   // These are mutated on the main render thread in OnCaptureFormatChanged().
 167   // The caller guarantees this does not run concurrently with accesses on the
 168   // capture audio thread.
 169   media::AudioParameters input_format_;
 170   media::AudioParameters output_format_;
 171   // Only used on the render audio thread.
 172   media::AudioParameters render_format_;
 173
 174   // Raw pointer to the WebRtcPlayoutDataSource, which is valid for the
 175   // lifetime of RenderThread.
 176   WebRtcPlayoutDataSource* playout_data_source_;
 177
 178   // Used to DCHECK that some methods are called on the main render thread.
 179   base::ThreadChecker main_thread_checker_;
 180   // Used to DCHECK that some methods are called on the capture audio thread.
 181   base::ThreadChecker capture_thread_checker_;
 182   // Used to DCHECK that some methods are called on the render audio thread.
 183   base::ThreadChecker render_thread_checker_;
 184
 185   // Flag to enable stereo channel mirroring.
 186   bool audio_mirroring_;
 187
 188   scoped_ptr<webrtc::TypingDetection> typing_detector_;
 189   // This flag is used to show the result of typing detection.
 190   // It can be accessed by the capture audio thread and by the libjingle thread
 191   // which calls GetStats().
 192   base::subtle::Atomic32 typing_detected_;
 193
 194   // Communication with browser for AEC dump.
 195   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
 196
 197   // Flag to avoid executing Stop() more than once.
 198   bool stopped_;
 199
 200   // Object for logging echo information when the AEC is enabled. Accessible by
 201   // the libjingle thread through GetStats().
 202   scoped_ptr<EchoInformation> echo_information_;
 203
 204   // Flag is enabled if AudioProcessing supports 48kHz sample rate.
 205   bool audio_proc_48kHz_support_;
 206
 207   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 208 };
 209
 210 }  // namespace content
 211
 212 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_