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_WEBRTC_LOCAL_AUDIO_TRACK_H_
6#define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
7
8#include <list>
9#include <string>
10
11#include "base/memory/ref_counted.h"
12#include "base/synchronization/lock.h"
13#include "base/threading/thread_checker.h"
14#include "content/renderer/media/media_stream_track.h"
15#include "content/renderer/media/tagged_list.h"
16#include "content/renderer/media/webrtc_audio_device_impl.h"
17
18namespace content {
19
20class MediaStreamAudioLevelCalculator;
21class MediaStreamAudioProcessor;
22class MediaStreamAudioSink;
23class MediaStreamAudioSinkOwner;
24class MediaStreamAudioTrackSink;
25class PeerConnectionAudioSink;
26class WebAudioCapturerSource;
27class WebRtcAudioCapturer;
28class WebRtcLocalAudioTrackAdapter;
29
30// A WebRtcLocalAudioTrack instance contains the implementations of
31// MediaStreamTrackExtraData.
32// When an instance is created, it will register itself as a track to the
33// WebRtcAudioCapturer to get the captured data, and forward the data to
34// its |sinks_|. The data flow can be stopped by disabling the audio track.
35class CONTENT_EXPORT WebRtcLocalAudioTrack
36    : NON_EXPORTED_BASE(public MediaStreamTrack) {
37 public:
38  WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter,
39                        const scoped_refptr<WebRtcAudioCapturer>& capturer,
40                        WebAudioCapturerSource* webaudio_source);
41
42  virtual ~WebRtcLocalAudioTrack();
43
44  // Add a sink to the track. This function will trigger a OnSetFormat()
45  // call on the |sink|.
46  // Called on the main render thread.
47  void AddSink(MediaStreamAudioSink* sink);
48
49  // Remove a sink from the track.
50  // Called on the main render thread.
51  void RemoveSink(MediaStreamAudioSink* sink);
52
53  // Add/remove PeerConnection sink to/from the track.
54  // TODO(xians): Remove these two methods after PeerConnection can use the
55  // same sink interface as MediaStreamAudioSink.
56  void AddSink(PeerConnectionAudioSink* sink);
57  void RemoveSink(PeerConnectionAudioSink* sink);
58
59  // Starts the local audio track. Called on the main render thread and
60  // should be called only once when audio track is created.
61  void Start();
62
63  // Stops the local audio track. Called on the main render thread and
64  // should be called only once when audio track going away.
65  virtual void Stop() OVERRIDE;
66
67  // Method called by the capturer to deliver the capture data.
68  // Called on the capture audio thread.
69  void Capture(const int16* audio_data,
70               base::TimeDelta delay,
71               int volume,
72               bool key_pressed,
73               bool need_audio_processing,
74               bool force_report_nonzero_energy);
75
76  // Method called by the capturer to set the audio parameters used by source
77  // of the capture data..
78  // Called on the capture audio thread.
79  void OnSetFormat(const media::AudioParameters& params);
80
81  // Method called by the capturer to set the processor that applies signal
82  // processing on the data of the track.
83  // Called on the capture audio thread.
84  void SetAudioProcessor(
85      const scoped_refptr<MediaStreamAudioProcessor>& processor);
86
87 private:
88  typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
89
90  // All usage of libjingle is through this adapter. The adapter holds
91  // a reference on this object, but not vice versa.
92  WebRtcLocalAudioTrackAdapter* adapter_;
93
94  // The provider of captured data to render.
95  scoped_refptr<WebRtcAudioCapturer> capturer_;
96
97  // The source of the audio track which is used by WebAudio, which provides
98  // data to the audio track when hooking up with WebAudio.
99  scoped_refptr<WebAudioCapturerSource> webaudio_source_;
100
101  // A tagged list of sinks that the audio data is fed to. Tags
102  // indicate tracks that need to be notified that the audio format
103  // has changed.
104  SinkList sinks_;
105
106  // Used to DCHECK that some methods are called on the main render thread.
107  base::ThreadChecker main_render_thread_checker_;
108
109  // Used to DCHECK that some methods are called on the capture audio thread.
110  base::ThreadChecker capture_thread_checker_;
111
112  // Protects |params_| and |sinks_|.
113  mutable base::Lock lock_;
114
115  // Audio parameters of the audio capture stream.
116  // Accessed on only the audio capture thread.
117  media::AudioParameters audio_parameters_;
118
119  // Used to calculate the signal level that shows in the UI.
120  // Accessed on only the audio thread.
121  scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
122
123  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
124};
125
126}  // namespace content
127
128#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
129