content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

   1 // Copyright 2014 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_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
   6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
   7
   8 #include <vector>
   9
  10 #include "base/memory/ref_counted.h"
  11 #include "base/memory/scoped_vector.h"
  12 #include "base/synchronization/lock.h"
  13 #include "content/common/content_export.h"
  14 #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
  15 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
  16
  17 namespace cricket {
  18 class AudioRenderer;
  19 }
  20
  21 namespace webrtc {
  22 class AudioSourceInterface;
  23 class AudioProcessorInterface;
  24 }
  25
  26 namespace content {
  27
  28 class MediaStreamAudioProcessor;
  29 class WebRtcAudioSinkAdapter;
  30 class WebRtcLocalAudioTrack;
  31
  32 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
  33     : NON_EXPORTED_BASE(public cricket::AudioRenderer),
  34       NON_EXPORTED_BASE(
  35           public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
  36  public:
  37   static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
  38       const std::string& label,
  39       webrtc::AudioSourceInterface* track_source);
  40
  41   WebRtcLocalAudioTrackAdapter(
  42       const std::string& label,
  43       webrtc::AudioSourceInterface* track_source);
  44
  45   virtual ~WebRtcLocalAudioTrackAdapter();
  46
  47   void Initialize(WebRtcLocalAudioTrack* owner);
  48
  49   std::vector<int> VoeChannels() const;
  50
  51   // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal
  52   // level of the audio data.
  53   void SetSignalLevel(int signal_level);
  54
  55   // Method called by the WebRtcLocalAudioTrack to set the processor that
  56   // applies signal processing on the data of the track.
  57   // This class will keep a reference of the |processor|.
  58   // Called on the main render thread.
  59   void SetAudioProcessor(
  60       const scoped_refptr<MediaStreamAudioProcessor>& processor);
  61
  62  private:
  63   // webrtc::MediaStreamTrack implementation.
  64   virtual std::string kind() const override;
  65
  66   // webrtc::AudioTrackInterface implementation.
  67   virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
  68   virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
  69   virtual bool GetSignalLevel(int* level) override;
  70   virtual rtc::scoped_refptr<webrtc::AudioProcessorInterface>
  71       GetAudioProcessor() override;
  72
  73   // cricket::AudioCapturer implementation.
  74   virtual void AddChannel(int channel_id) override;
  75   virtual void RemoveChannel(int channel_id) override;
  76
  77   // webrtc::AudioTrackInterface implementation.
  78   virtual webrtc::AudioSourceInterface* GetSource() const override;
  79   virtual cricket::AudioRenderer* GetRenderer() override;
  80
  81   // Weak reference.
  82   WebRtcLocalAudioTrack* owner_;
  83
  84   // The source of the audio track which handles the audio constraints.
  85   // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
  86   rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
  87
  88   // The audio processsor that applies audio processing on the data of audio
  89   // track.
  90   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
  91
  92   // A vector of WebRtc VoE channels that the capturer sends data to.
  93   std::vector<int> voe_channels_;
  94
  95   // A vector of the peer connection sink adapters which receive the audio data
  96   // from the audio track.
  97   ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
  98
  99   // The amplitude of the signal.
 100   int signal_level_;
 101
 102   // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
 103   mutable base::Lock lock_;
 104 };
 105
 106 }  // namespace content
 107
 108 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_