File: webrtc_audio_module.h

package info (click to toggle)
chromium-browser 57.0.2987.98-1~deb8u1
  • links: PTS, VCS
  • area: main
  • in suites: jessie
  • size: 2,637,852 kB
  • ctags: 2,544,394
  • sloc: cpp: 12,815,961; ansic: 3,676,222; python: 1,147,112; asm: 526,608; java: 523,212; xml: 286,794; perl: 92,654; sh: 86,408; objc: 73,271; makefile: 27,698; cs: 18,487; yacc: 13,031; tcl: 12,957; pascal: 4,875; ml: 4,716; lex: 3,904; sql: 3,862; ruby: 1,982; lisp: 1,508; php: 1,368; exp: 404; awk: 325; csh: 117; jsp: 39; sed: 37
file content (164 lines) | stat: -rw-r--r-- 7,369 bytes parent folder | download
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
// Copyright 2016 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#ifndef REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_
#define REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_

#include "base/memory/ref_counted.h"
#include "base/synchronization/lock.h"
#include "base/timer/timer.h"
#include "third_party/webrtc/modules/audio_device/include/audio_device.h"

namespace base {
class SingleThreadTaskRunner;
}  // namespace base

namespace remoting {
namespace protocol {

// Audio module passed to WebRTC. It doesn't access actual audio devices, but it
// provides all functionality we need to ensure that audio streaming works
// properly in WebRTC. Particularly it's responsible for calling AudioTransport
// on regular intervals when playback is active. This ensures that all incoming
// audio data is processed and passed to webrtc::AudioTrackSinkInterface
// connected to the audio track.
class WebrtcAudioModule : public webrtc::AudioDeviceModule {
 public:
  WebrtcAudioModule();
  ~WebrtcAudioModule() override;

  void SetAudioTaskRunner(
      scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner);

  // webrtc::AudioDeviceModule implementation.
  int64_t TimeUntilNextProcess() override;
  void Process() override;
  int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
  ErrorCode LastError() const override;
  int32_t RegisterEventObserver(
      webrtc::AudioDeviceObserver* event_callback) override;
  int32_t RegisterAudioCallback(
      webrtc::AudioTransport* audio_callback) override;
  int32_t Init() override;
  int32_t Terminate() override;
  bool Initialized() const override;
  int16_t PlayoutDevices() override;
  int16_t RecordingDevices() override;
  int32_t PlayoutDeviceName(uint16_t index,
                            char name[webrtc::kAdmMaxDeviceNameSize],
                            char guid[webrtc::kAdmMaxGuidSize]) override;
  int32_t RecordingDeviceName(uint16_t index,
                              char name[webrtc::kAdmMaxDeviceNameSize],
                              char guid[webrtc::kAdmMaxGuidSize]) override;
  int32_t SetPlayoutDevice(uint16_t index) override;
  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
  int32_t SetRecordingDevice(uint16_t index) override;
  int32_t SetRecordingDevice(WindowsDeviceType device) override;
  int32_t PlayoutIsAvailable(bool* available) override;
  int32_t InitPlayout() override;
  bool PlayoutIsInitialized() const override;
  int32_t RecordingIsAvailable(bool* available) override;
  int32_t InitRecording() override;
  bool RecordingIsInitialized() const override;
  int32_t StartPlayout() override;
  int32_t StopPlayout() override;
  bool Playing() const override;
  int32_t StartRecording() override;
  int32_t StopRecording() override;
  bool Recording() const override;
  int32_t SetAGC(bool enable) override;
  bool AGC() const override;
  int32_t SetWaveOutVolume(uint16_t volume_left,
                           uint16_t volume_right) override;
  int32_t WaveOutVolume(uint16_t* volume_left,
                        uint16_t* volume_right) const override;
  int32_t InitSpeaker() override;
  bool SpeakerIsInitialized() const override;
  int32_t InitMicrophone() override;
  bool MicrophoneIsInitialized() const override;
  int32_t SpeakerVolumeIsAvailable(bool* available) override;
  int32_t SetSpeakerVolume(uint32_t volume) override;
  int32_t SpeakerVolume(uint32_t* volume) const override;
  int32_t MaxSpeakerVolume(uint32_t* max_volume) const override;
  int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
  int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
  int32_t SetMicrophoneVolume(uint32_t volume) override;
  int32_t MicrophoneVolume(uint32_t* volume) const override;
  int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
  int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
  int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
  int32_t SpeakerMuteIsAvailable(bool* available) override;
  int32_t SetSpeakerMute(bool enable) override;
  int32_t SpeakerMute(bool* enabled) const override;
  int32_t MicrophoneMuteIsAvailable(bool* available) override;
  int32_t SetMicrophoneMute(bool enable) override;
  int32_t MicrophoneMute(bool* enabled) const override;
  int32_t MicrophoneBoostIsAvailable(bool* available) override;
  int32_t SetMicrophoneBoost(bool enable) override;
  int32_t MicrophoneBoost(bool* enabled) const override;
  int32_t StereoPlayoutIsAvailable(bool* available) const override;
  int32_t SetStereoPlayout(bool enable) override;
  int32_t StereoPlayout(bool* enabled) const override;
  int32_t StereoRecordingIsAvailable(bool* available) const override;
  int32_t SetStereoRecording(bool enable) override;
  int32_t StereoRecording(bool* enabled) const override;
  int32_t SetRecordingChannel(const ChannelType channel) override;
  int32_t RecordingChannel(ChannelType* channel) const override;
  int32_t SetPlayoutBuffer(const BufferType type, uint16_t size_ms) override;
  int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
  int32_t PlayoutDelay(uint16_t* delay_ms) const override;
  int32_t RecordingDelay(uint16_t* delay_ms) const override;
  int32_t CPULoad(uint16_t* load) const override;
  int32_t StartRawOutputFileRecording(
      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
  int32_t StopRawOutputFileRecording() override;
  int32_t StartRawInputFileRecording(
      const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
  int32_t StopRawInputFileRecording() override;
  int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
  int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
  int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
  int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
  int32_t ResetAudioDevice() override;
  int32_t SetLoudspeakerStatus(bool enable) override;
  int32_t GetLoudspeakerStatus(bool* enabled) const override;
  bool BuiltInAECIsAvailable() const override;
  bool BuiltInAGCIsAvailable() const override;
  bool BuiltInNSIsAvailable() const override;
  int32_t EnableBuiltInAEC(bool enable) override;
  int32_t EnableBuiltInAGC(bool enable) override;
  int32_t EnableBuiltInNS(bool enable) override;

// Only supported on iOS.
#if defined(WEBRTC_IOS)
  int GetPlayoutAudioParameters(webrtc::AudioParameters* params) const override;
  int GetRecordAudioParameters(webrtc::AudioParameters* params) const override;
#endif  // WEBRTC_IOS

 private:
  void StartPlayoutOnAudioThread();
  void StopPlayoutOnAudioThread();

  void PollFromSource();

  scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner_;

  // |lock_| must be locked when accessing |initialized_|, |playing_| and
  // |audio_transport_|.
  mutable base::Lock lock_;

  bool initialized_ = false;
  bool playing_ = false;
  webrtc::AudioTransport* audio_transport_ = nullptr;

  // Timer running on the |audio_task_runner_| that polls audio from
  // |audio_transport_|.
  base::RepeatingTimer poll_timer_;
};

}  // namespace protocol
}  // namespace remoting

#endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_