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| 1 // Copyright (c) 2012 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_AUDIO_CAPTURER_H_ | |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_ | |
| 7 | |
| 8 #include <list> | |
| 9 #include <string> | |
| 10 | |
| 11 #include "base/callback.h" | |
| 12 #include "base/files/file.h" | |
| 13 #include "base/macros.h" | |
| 14 #include "base/memory/ref_counted.h" | |
| 15 #include "base/synchronization/lock.h" | |
| 16 #include "base/threading/thread_checker.h" | |
| 17 #include "base/time/time.h" | |
| 18 #include "content/common/media/media_stream_options.h" | |
| 19 #include "content/renderer/media/tagged_list.h" | |
| 20 #include "media/audio/audio_input_device.h" | |
| 21 #include "media/base/audio_capturer_source.h" | |
| 22 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | |
| 23 | |
| 24 namespace media { | |
| 25 class AudioBus; | |
| 26 } | |
| 27 | |
| 28 namespace content { | |
| 29 | |
| 30 class MediaStreamAudioProcessor; | |
| 31 class MediaStreamAudioSource; | |
| 32 class WebRtcAudioDeviceImpl; | |
| 33 class WebRtcLocalAudioRenderer; | |
| 34 class WebRtcLocalAudioTrack; | |
| 35 | |
| 36 // This class manages the capture data flow by getting data from its | |
| 37 // |source_|, and passing it to its |tracks_|. | |
| 38 // The threading model for this class is rather complex since it will be | |
| 39 // created on the main render thread, captured data is provided on a dedicated | |
| 40 // AudioInputDevice thread, and methods can be called either on the Libjingle | |
| 41 // thread or on the main render thread but also other client threads | |
| 42 // if an alternative AudioCapturerSource has been set. | |
| 43 class CONTENT_EXPORT WebRtcAudioCapturer | |
| 44 : public base::RefCountedThreadSafe<WebRtcAudioCapturer>, | |
| 45 NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) { | |
| 46 public: | |
| 47 // Used to construct the audio capturer. |render_frame_id| specifies the | |
| 48 // RenderFrame consuming audio for capture; -1 is used for tests. | |
| 49 // |device_info| contains all the device information that the capturer is | |
| 50 // created for. |constraints| contains the settings for audio processing. | |
| 51 // TODO(xians): Implement the interface for the audio source and move the | |
| 52 // |constraints| to ApplyConstraints(). Called on the main render thread. | |
| 53 static scoped_refptr<WebRtcAudioCapturer> CreateCapturer( | |
| 54 int render_frame_id, | |
| 55 const StreamDeviceInfo& device_info, | |
| 56 const blink::WebMediaConstraints& constraints, | |
| 57 WebRtcAudioDeviceImpl* audio_device, | |
| 58 MediaStreamAudioSource* audio_source); | |
| 59 | |
| 60 // Add a audio track to the sinks of the capturer. | |
| 61 // WebRtcAudioDeviceImpl calls this method on the main render thread but | |
| 62 // other clients may call it from other threads. The current implementation | |
| 63 // does not support multi-thread calling. | |
| 64 // The first AddTrack will implicitly trigger the Start() of this object. | |
| 65 void AddTrack(WebRtcLocalAudioTrack* track); | |
| 66 | |
| 67 // Remove a audio track from the sinks of the capturer. | |
| 68 // If the track has been added to the capturer, it must call RemoveTrack() | |
| 69 // before it goes away. | |
| 70 // Called on the main render thread or libjingle working thread. | |
| 71 void RemoveTrack(WebRtcLocalAudioTrack* track); | |
| 72 | |
| 73 // Called when a stream is connecting to a peer connection. This will set | |
| 74 // up the native buffer size for the stream in order to optimize the | |
| 75 // performance for peer connection. | |
| 76 void EnablePeerConnectionMode(); | |
| 77 | |
| 78 // Volume APIs used by WebRtcAudioDeviceImpl. | |
| 79 // Called on the AudioInputDevice audio thread. | |
| 80 void SetVolume(int volume); | |
| 81 int Volume() const; | |
| 82 int MaxVolume() const; | |
| 83 | |
| 84 // Audio parameters utilized by the source of the audio capturer. | |
| 85 // TODO(phoglund): Think over the implications of this accessor and if we can | |
| 86 // remove it. | |
| 87 media::AudioParameters source_audio_parameters() const; | |
| 88 | |
| 89 // Gets information about the paired output device. Returns true if such a | |
| 90 // device exists. | |
| 91 bool GetPairedOutputParameters(int* session_id, | |
| 92 int* output_sample_rate, | |
| 93 int* output_frames_per_buffer) const; | |
| 94 | |
| 95 const std::string& device_id() const { return device_info_.device.id; } | |
| 96 int session_id() const { return device_info_.session_id; } | |
| 97 | |
| 98 // Stops recording audio. This method will empty its track lists since | |
| 99 // stopping the capturer will implicitly invalidate all its tracks. | |
| 100 // This method is exposed to the public because the MediaStreamAudioSource can | |
| 101 // call Stop() | |
| 102 void Stop(); | |
| 103 | |
| 104 // Returns the output format. | |
| 105 // Called on the main render thread. | |
| 106 media::AudioParameters GetOutputFormat() const; | |
| 107 | |
| 108 // Used by clients to inject their own source to the capturer. | |
| 109 void SetCapturerSource( | |
| 110 const scoped_refptr<media::AudioCapturerSource>& source, | |
| 111 media::AudioParameters params); | |
| 112 | |
| 113 protected: | |
| 114 friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>; | |
| 115 ~WebRtcAudioCapturer() override; | |
| 116 | |
| 117 private: | |
| 118 class TrackOwner; | |
| 119 typedef TaggedList<TrackOwner> TrackList; | |
| 120 | |
| 121 WebRtcAudioCapturer(int render_frame_id, | |
| 122 const StreamDeviceInfo& device_info, | |
| 123 const blink::WebMediaConstraints& constraints, | |
| 124 WebRtcAudioDeviceImpl* audio_device, | |
| 125 MediaStreamAudioSource* audio_source); | |
| 126 | |
| 127 // AudioCapturerSource::CaptureCallback implementation. | |
| 128 // Called on the AudioInputDevice audio thread. | |
| 129 void Capture(const media::AudioBus* audio_source, | |
| 130 int audio_delay_milliseconds, | |
| 131 double volume, | |
| 132 bool key_pressed) override; | |
| 133 void OnCaptureError(const std::string& message) override; | |
| 134 | |
| 135 // Initializes the default audio capturing source using the provided render | |
| 136 // frame id and device information. Return true if success, otherwise false. | |
| 137 bool Initialize(); | |
| 138 | |
| 139 // SetCapturerSourceInternal() is called if the client on the source side | |
| 140 // desires to provide their own captured audio data. Client is responsible | |
| 141 // for calling Start() on its own source to get the ball rolling. | |
| 142 // Called on the main render thread. | |
| 143 // buffer_size is optional. Set to 0 to let it be chosen automatically. | |
| 144 void SetCapturerSourceInternal( | |
| 145 const scoped_refptr<media::AudioCapturerSource>& source, | |
| 146 media::ChannelLayout channel_layout, | |
| 147 int sample_rate, | |
| 148 int buffer_size); | |
| 149 | |
| 150 // Starts recording audio. | |
| 151 // Triggered by AddSink() on the main render thread or a Libjingle working | |
| 152 // thread. It should NOT be called under |lock_|. | |
| 153 void Start(); | |
| 154 | |
| 155 // Helper function to get the buffer size based on |peer_connection_mode_| | |
| 156 // and sample rate; | |
| 157 int GetBufferSize(int sample_rate) const; | |
| 158 | |
| 159 // Used to DCHECK that we are called on the correct thread. | |
| 160 base::ThreadChecker thread_checker_; | |
| 161 | |
| 162 // Protects |source_|, |audio_tracks_|, |running_|, |loopback_fifo_|, | |
| 163 // |params_| and |buffering_|. | |
| 164 mutable base::Lock lock_; | |
| 165 | |
| 166 // A tagged list of audio tracks that the audio data is fed | |
| 167 // to. Tagged items need to be notified that the audio format has | |
| 168 // changed. | |
| 169 TrackList tracks_; | |
| 170 | |
| 171 // The audio data source from the browser process. | |
| 172 scoped_refptr<media::AudioCapturerSource> source_; | |
| 173 | |
| 174 // Cached audio constraints for the capturer. | |
| 175 blink::WebMediaConstraints constraints_; | |
| 176 | |
| 177 // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output | |
| 178 // data is in a unit of 10 ms data chunk. | |
| 179 scoped_refptr<MediaStreamAudioProcessor> audio_processor_; | |
| 180 | |
| 181 bool running_; | |
| 182 | |
| 183 int render_frame_id_; | |
| 184 | |
| 185 // Cached information of the device used by the capturer. | |
| 186 const StreamDeviceInfo device_info_; | |
| 187 | |
| 188 // Stores latest microphone volume received in a CaptureData() callback. | |
| 189 // Range is [0, 255]. | |
| 190 int volume_; | |
| 191 | |
| 192 // Flag which affects the buffer size used by the capturer. | |
| 193 bool peer_connection_mode_; | |
| 194 | |
| 195 // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime | |
| 196 // of RenderThread. | |
| 197 WebRtcAudioDeviceImpl* audio_device_; | |
| 198 | |
| 199 // Raw pointer to the MediaStreamAudioSource object that holds a reference | |
| 200 // to this WebRtcAudioCapturer. | |
| 201 // Since |audio_source_| is owned by a blink::WebMediaStreamSource object and | |
| 202 // blink guarantees that the blink::WebMediaStreamSource outlives any | |
| 203 // blink::WebMediaStreamTrack connected to the source, |audio_source_| is | |
| 204 // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this | |
| 205 // WebRtcAudioCapturer. | |
| 206 MediaStreamAudioSource* const audio_source_; | |
| 207 | |
| 208 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer); | |
| 209 }; | |
| 210 | |
| 211 } // namespace content | |
| 212 | |
| 213 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_ | |
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Chromium Code Reviews
Issue 1647773002:
MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master