Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 1 | // Copyright 2013 The Chromium Authors. All rights reserved. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 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_DEVICE_IMPL_H_ |
| 6 | #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
| 7 | |
| 8 | #include <string> |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 9 | #include <vector> |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 10 | |
| 11 | #include "base/basictypes.h" |
| 12 | #include "base/compiler_specific.h" |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 13 | #include "base/logging.h" |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 14 | #include "base/memory/ref_counted.h" |
| 15 | #include "base/memory/scoped_ptr.h" |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 16 | #include "base/threading/thread_checker.h" |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 17 | #include "content/common/content_export.h" |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 18 | #include "content/renderer/media/webrtc_audio_capturer.h" |
| 19 | #include "content/renderer/media/webrtc_audio_device_not_impl.h" |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 20 | #include "content/renderer/media/webrtc_audio_renderer.h" |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 21 | #include "media/base/audio_capturer_source.h" |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 22 | #include "media/base/audio_renderer_sink.h" |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 23 | |
| 24 | // A WebRtcAudioDeviceImpl instance implements the abstract interface |
| 25 | // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc:: |
| 26 | // VoiceEngine) to register this class as an external AudioDeviceModule (ADM). |
| 27 | // Then WebRtcAudioDeviceImpl::SetSessionId() needs to be called to set the |
Torne (Richard Coles) | 868fa2f | 2013-06-11 10:57:03 +0100 | [diff] [blame] | 28 | // session id that tells which device to use. The user can then call |
| 29 | // WebRtcAudioDeviceImpl::StartPlayout() and |
| 30 | // WebRtcAudioDeviceImpl::StartRecording() from the render process to initiate |
| 31 | // and start audio rendering and capturing in the browser process. IPC is |
| 32 | // utilized to set up the media streams. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 33 | // |
| 34 | // Usage example: |
| 35 | // |
| 36 | // using namespace webrtc; |
| 37 | // |
| 38 | // { |
| 39 | // scoped_refptr<WebRtcAudioDeviceImpl> external_adm; |
| 40 | // external_adm = new WebRtcAudioDeviceImpl(); |
Torne (Richard Coles) | 868fa2f | 2013-06-11 10:57:03 +0100 | [diff] [blame] | 41 | // external_adm->SetSessionId(session_id); |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 42 | // VoiceEngine* voe = VoiceEngine::Create(); |
| 43 | // VoEBase* base = VoEBase::GetInterface(voe); |
| 44 | // base->Init(external_adm); |
| 45 | // int ch = base->CreateChannel(); |
| 46 | // ... |
| 47 | // base->StartReceive(ch) |
| 48 | // base->StartPlayout(ch); |
| 49 | // base->StartSending(ch); |
| 50 | // ... |
| 51 | // <== full-duplex audio session with AGC enabled ==> |
| 52 | // ... |
| 53 | // base->DeleteChannel(ch); |
| 54 | // base->Terminate(); |
| 55 | // base->Release(); |
| 56 | // VoiceEngine::Delete(voe); |
| 57 | // } |
| 58 | // |
| 59 | // webrtc::VoiceEngine::Init() calls these ADM methods (in this order): |
| 60 | // |
| 61 | // RegisterAudioCallback(this) |
| 62 | // webrtc::VoiceEngine is an webrtc::AudioTransport implementation and |
| 63 | // implements the RecordedDataIsAvailable() and NeedMorePlayData() callbacks. |
| 64 | // |
| 65 | // Init() |
| 66 | // Creates and initializes the AudioOutputDevice and AudioInputDevice |
| 67 | // objects. |
| 68 | // |
| 69 | // SetAGC(true) |
| 70 | // Enables the adaptive analog mode of the AGC which ensures that a |
| 71 | // suitable microphone volume level will be set. This scheme will affect |
| 72 | // the actual microphone control slider. |
| 73 | // |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 74 | // AGC overview: |
| 75 | // |
| 76 | // It aims to maintain a constant speech loudness level from the microphone. |
| 77 | // This is done by both controlling the analog microphone gain and applying |
| 78 | // digital gain. The microphone gain on the sound card is slowly |
| 79 | // increased/decreased during speech only. By observing the microphone control |
| 80 | // slider you can see it move when you speak. If you scream, the slider moves |
| 81 | // downwards and then upwards again when you return to normal. It is not |
| 82 | // uncommon that the slider hits the maximum. This means that the maximum |
| 83 | // analog gain is not large enough to give the desired loudness. Nevertheless, |
| 84 | // we can in general still attain the desired loudness. If the microphone |
| 85 | // control slider is moved manually, the gain adaptation restarts and returns |
| 86 | // to roughly the same position as before the change if the circumstances are |
| 87 | // still the same. When the input microphone signal causes saturation, the |
| 88 | // level is decreased dramatically and has to re-adapt towards the old level. |
| 89 | // The adaptation is a slowly varying process and at the beginning of capture |
| 90 | // this is noticed by a slow increase in volume. Smaller changes in microphone |
| 91 | // input level is leveled out by the built-in digital control. For larger |
| 92 | // differences we need to rely on the slow adaptation. |
| 93 | // See http://en.wikipedia.org/wiki/Automatic_gain_control for more details. |
| 94 | // |
| 95 | // AGC implementation details: |
| 96 | // |
| 97 | // The adaptive analog mode of the AGC is always enabled for desktop platforms |
| 98 | // in WebRTC. |
| 99 | // |
| 100 | // Before recording starts, the ADM enables AGC on the AudioInputDevice. |
| 101 | // |
| 102 | // A capture session with AGC is started up as follows (simplified): |
| 103 | // |
| 104 | // [renderer] |
| 105 | // | |
| 106 | // ADM::StartRecording() |
| 107 | // AudioInputDevice::InitializeOnIOThread() |
| 108 | // AudioInputHostMsg_CreateStream(..., agc=true) [IPC] |
| 109 | // | |
| 110 | // [IPC to the browser] |
| 111 | // | |
| 112 | // AudioInputRendererHost::OnCreateStream() |
| 113 | // AudioInputController::CreateLowLatency() |
| 114 | // AudioInputController::DoSetAutomaticGainControl(true) |
| 115 | // AudioInputStream::SetAutomaticGainControl(true) |
| 116 | // | |
| 117 | // AGC is now enabled in the media layer and streaming starts (details omitted). |
| 118 | // The figure below illustrates the AGC scheme which is active in combination |
| 119 | // with the default media flow explained earlier. |
| 120 | // | |
| 121 | // [browser] |
| 122 | // | |
| 123 | // AudioInputStream::(Capture thread loop) |
Torne (Richard Coles) | 90dce4d | 2013-05-29 14:40:03 +0100 | [diff] [blame] | 124 | // AgcAudioStream<AudioInputStream>::GetAgcVolume() => get latest mic volume |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 125 | // AudioInputData::OnData(..., volume) |
| 126 | // AudioInputController::OnData(..., volume) |
| 127 | // AudioInputSyncWriter::Write(..., volume) |
| 128 | // | |
| 129 | // [volume | size | data] is sent to the renderer [shared memory] |
| 130 | // | |
| 131 | // [renderer] |
| 132 | // | |
| 133 | // AudioInputDevice::AudioThreadCallback::Process() |
| 134 | // WebRtcAudioDeviceImpl::Capture(..., volume) |
| 135 | // AudioTransport::RecordedDataIsAvailable(...,volume, new_volume) |
| 136 | // | |
| 137 | // The AGC now uses the current volume input and computes a suitable new |
| 138 | // level given by the |new_level| output. This value is only non-zero if the |
| 139 | // AGC has take a decision that the microphone level should change. |
| 140 | // | |
| 141 | // if (new_volume != 0) |
| 142 | // AudioInputDevice::SetVolume(new_volume) |
| 143 | // AudioInputHostMsg_SetVolume(new_volume) [IPC] |
| 144 | // | |
| 145 | // [IPC to the browser] |
| 146 | // | |
| 147 | // AudioInputRendererHost::OnSetVolume() |
| 148 | // AudioInputController::SetVolume() |
| 149 | // AudioInputStream::SetVolume(scaled_volume) |
| 150 | // | |
| 151 | // Here we set the new microphone level in the media layer and at the same time |
| 152 | // read the new setting (we might not get exactly what is set). |
| 153 | // | |
| 154 | // AudioInputData::OnData(..., updated_volume) |
| 155 | // AudioInputController::OnData(..., updated_volume) |
| 156 | // | |
| 157 | // | |
| 158 | // This process repeats until we stop capturing data. Note that, a common |
| 159 | // steady state is that the volume control reaches its max and the new_volume |
| 160 | // value from the AGC is zero. A loud voice input is required to break this |
| 161 | // state and start lowering the level again. |
| 162 | // |
| 163 | // Implementation notes: |
| 164 | // |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 165 | // - This class must be created and destroyed on the main render thread and |
| 166 | // most methods are called on the same thread. However, some methods are |
| 167 | // also called on a Libjingle worker thread. RenderData is called on the |
| 168 | // AudioOutputDevice thread and CaptureData on the AudioInputDevice thread. |
| 169 | // To summarize: this class lives on four different threads. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 170 | // - The webrtc::AudioDeviceModule is reference counted. |
| 171 | // - AGC is only supported in combination with the WASAPI-based audio layer |
| 172 | // on Windows, i.e., it is not supported on Windows XP. |
| 173 | // - All volume levels required for the AGC scheme are transfered in a |
| 174 | // normalized range [0.0, 1.0]. Scaling takes place in both endpoints |
| 175 | // (WebRTC client a media layer). This approach ensures that we can avoid |
| 176 | // transferring maximum levels between the renderer and the browser. |
| 177 | // |
| 178 | |
| 179 | namespace content { |
| 180 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 181 | class WebRtcAudioCapturer; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 182 | class WebRtcAudioRenderer; |
| 183 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 184 | // TODO(xians): Move the following two interfaces to webrtc so that |
| 185 | // libjingle can own references to the renderer and capturer. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 186 | class WebRtcAudioRendererSource { |
| 187 | public: |
| 188 | // Callback to get the rendered interleaved data. |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 189 | // TODO(xians): Change uint8* to int16*. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 190 | virtual void RenderData(uint8* audio_data, |
| 191 | int number_of_channels, |
| 192 | int number_of_frames, |
| 193 | int audio_delay_milliseconds) = 0; |
| 194 | |
| 195 | // Set the format for the capture audio parameters. |
| 196 | virtual void SetRenderFormat(const media::AudioParameters& params) = 0; |
| 197 | |
| 198 | // Callback to notify the client that the renderer is going away. |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 199 | virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 200 | |
| 201 | protected: |
| 202 | virtual ~WebRtcAudioRendererSource() {} |
| 203 | }; |
| 204 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 205 | class WebRtcAudioCapturerSink { |
| 206 | public: |
| 207 | // Callback to deliver the captured interleaved data. |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 208 | // |channels| contains a vector of WebRtc VoE channels. |
| 209 | // |audio_data| is the pointer to the audio data. |
| 210 | // |sample_rate| is the sample frequency of audio data. |
| 211 | // |number_of_channels| is the number of channels reflecting the order of |
| 212 | // surround sound channels. |
| 213 | // |audio_delay_milliseconds| is recording delay value. |
| 214 | // |current_volume| is current microphone volume, in range of |0, 255]. |
| 215 | // |need_audio_processing| indicates if the audio needs WebRtc AEC/NS/AGC |
| 216 | // audio processing. |
| 217 | // The return value is the new microphone volume, in the range of |0, 255]. |
| 218 | // When the volume does not need to be updated, it returns 0. |
| 219 | virtual int CaptureData(const std::vector<int>& channels, |
| 220 | const int16* audio_data, |
| 221 | int sample_rate, |
| 222 | int number_of_channels, |
| 223 | int number_of_frames, |
| 224 | int audio_delay_milliseconds, |
| 225 | int current_volume, |
| 226 | bool need_audio_processing) = 0; |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 227 | |
| 228 | // Set the format for the capture audio parameters. |
| 229 | virtual void SetCaptureFormat(const media::AudioParameters& params) = 0; |
| 230 | |
| 231 | protected: |
| 232 | virtual ~WebRtcAudioCapturerSink() {} |
| 233 | }; |
| 234 | |
| 235 | // Note that this class inherits from webrtc::AudioDeviceModule but due to |
| 236 | // the high number of non-implemented methods, we move the cruft over to the |
| 237 | // WebRtcAudioDeviceNotImpl. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 238 | class CONTENT_EXPORT WebRtcAudioDeviceImpl |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 239 | : NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl), |
| 240 | NON_EXPORTED_BASE(public WebRtcAudioCapturerSink), |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 241 | NON_EXPORTED_BASE(public WebRtcAudioRendererSource) { |
| 242 | public: |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 243 | // The maximum volume value WebRtc uses. |
| 244 | static const int kMaxVolumeLevel = 255; |
| 245 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 246 | // Instances of this object are created on the main render thread. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 247 | WebRtcAudioDeviceImpl(); |
| 248 | |
| 249 | // webrtc::RefCountedModule implementation. |
| 250 | // The creator must call AddRef() after construction and use Release() |
| 251 | // to release the reference and delete this object. |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 252 | // Called on the main render thread. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 253 | virtual int32_t AddRef() OVERRIDE; |
| 254 | virtual int32_t Release() OVERRIDE; |
| 255 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 256 | // webrtc::AudioDeviceModule implementation. |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 257 | // All implemented methods are called on the main render thread unless |
| 258 | // anything else is stated. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 259 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 260 | virtual int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback) |
| 261 | OVERRIDE; |
| 262 | |
| 263 | virtual int32_t Init() OVERRIDE; |
| 264 | virtual int32_t Terminate() OVERRIDE; |
| 265 | virtual bool Initialized() const OVERRIDE; |
| 266 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 267 | virtual int32_t PlayoutIsAvailable(bool* available) OVERRIDE; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 268 | virtual bool PlayoutIsInitialized() const OVERRIDE; |
| 269 | virtual int32_t RecordingIsAvailable(bool* available) OVERRIDE; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 270 | virtual bool RecordingIsInitialized() const OVERRIDE; |
| 271 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 272 | // All Start/Stop methods are called on a libJingle worker thread. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 273 | virtual int32_t StartPlayout() OVERRIDE; |
| 274 | virtual int32_t StopPlayout() OVERRIDE; |
| 275 | virtual bool Playing() const OVERRIDE; |
| 276 | virtual int32_t StartRecording() OVERRIDE; |
| 277 | virtual int32_t StopRecording() OVERRIDE; |
| 278 | virtual bool Recording() const OVERRIDE; |
| 279 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 280 | // Called on the main render thread and libJingle worker thread. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 281 | virtual int32_t SetAGC(bool enable) OVERRIDE; |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 282 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 283 | virtual bool AGC() const OVERRIDE; |
| 284 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 285 | // Called on the AudioInputDevice worker thread. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 286 | virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE; |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 287 | |
| 288 | // TODO(henrika): sort out calling thread once we start using this API. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 289 | virtual int32_t MicrophoneVolume(uint32_t* volume) const OVERRIDE; |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 290 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 291 | virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const OVERRIDE; |
| 292 | virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const OVERRIDE; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 293 | virtual int32_t StereoPlayoutIsAvailable(bool* available) const OVERRIDE; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 294 | virtual int32_t StereoRecordingIsAvailable(bool* available) const OVERRIDE; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 295 | virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE; |
| 296 | virtual int32_t RecordingDelay(uint16_t* delay_ms) const OVERRIDE; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 297 | virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const OVERRIDE; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 298 | virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE; |
| 299 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 300 | // Sets the |renderer_|, returns false if |renderer_| already exists. |
| 301 | // Called on the main renderer thread. |
| 302 | bool SetAudioRenderer(WebRtcAudioRenderer* renderer); |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 303 | |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 304 | // Adds the capturer to the ADM. |
| 305 | void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer); |
| 306 | |
| 307 | // Gets the default capturer, which is the capturer in the list with |
| 308 | // a valid |device_id|. Microphones are represented by capturers with a valid |
| 309 | // |device_id|, since only one microphone is supported today, only one |
| 310 | // capturer in the |capturers_| can have a valid |device_id|. |
| 311 | scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const; |
| 312 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 313 | const scoped_refptr<WebRtcAudioRenderer>& renderer() const { |
| 314 | return renderer_; |
| 315 | } |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 316 | int output_buffer_size() const { |
| 317 | return output_audio_parameters_.frames_per_buffer(); |
| 318 | } |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 319 | int output_channels() const { |
| 320 | return output_audio_parameters_.channels(); |
| 321 | } |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 322 | int output_sample_rate() const { |
| 323 | return output_audio_parameters_.sample_rate(); |
| 324 | } |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 325 | |
| 326 | private: |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 327 | typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList; |
| 328 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 329 | // Make destructor private to ensure that we can only be deleted by Release(). |
| 330 | virtual ~WebRtcAudioDeviceImpl(); |
| 331 | |
Torne (Richard Coles) | c2e0dbd | 2013-05-09 18:35:53 +0100 | [diff] [blame] | 332 | // WebRtcAudioCapturerSink implementation. |
| 333 | |
| 334 | // Called on the AudioInputDevice worker thread. |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 335 | virtual int CaptureData(const std::vector<int>& channels, |
| 336 | const int16* audio_data, |
| 337 | int sample_rate, |
| 338 | int number_of_channels, |
| 339 | int number_of_frames, |
| 340 | int audio_delay_milliseconds, |
| 341 | int current_volume, |
| 342 | bool need_audio_processing) OVERRIDE; |
Torne (Richard Coles) | c2e0dbd | 2013-05-09 18:35:53 +0100 | [diff] [blame] | 343 | |
| 344 | // Called on the main render thread. |
| 345 | virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; |
| 346 | |
| 347 | // WebRtcAudioRendererSource implementation. |
| 348 | |
| 349 | // Called on the AudioInputDevice worker thread. |
| 350 | virtual void RenderData(uint8* audio_data, |
| 351 | int number_of_channels, |
| 352 | int number_of_frames, |
| 353 | int audio_delay_milliseconds) OVERRIDE; |
| 354 | |
| 355 | // Called on the main render thread. |
| 356 | virtual void SetRenderFormat(const media::AudioParameters& params) OVERRIDE; |
| 357 | virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE; |
| 358 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 359 | // Used to DCHECK that we are called on the correct thread. |
| 360 | base::ThreadChecker thread_checker_; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 361 | |
| 362 | int ref_count_; |
| 363 | |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 364 | // List of captures which provides access to the native audio input layer |
| 365 | // in the browser process. |
| 366 | CapturerList capturers_; |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 367 | |
| 368 | // Provides access to the audio renderer in the browser process. |
| 369 | scoped_refptr<WebRtcAudioRenderer> renderer_; |
| 370 | |
| 371 | // Weak reference to the audio callback. |
| 372 | // The webrtc client defines |audio_transport_callback_| by calling |
| 373 | // RegisterAudioCallback(). |
| 374 | webrtc::AudioTransport* audio_transport_callback_; |
| 375 | |
Ben Murdoch | bb1529c | 2013-08-08 10:24:53 +0100 | [diff] [blame^] | 376 | // Cached values of used output audio parameters. Platform dependent. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 377 | media::AudioParameters output_audio_parameters_; |
| 378 | |
| 379 | // Cached value of the current audio delay on the input/capture side. |
| 380 | int input_delay_ms_; |
| 381 | |
| 382 | // Cached value of the current audio delay on the output/renderer side. |
| 383 | int output_delay_ms_; |
| 384 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 385 | // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|, |renderer_| |
| 386 | // |recording_| and |microphone_volume_|. |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 387 | mutable base::Lock lock_; |
| 388 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 389 | bool initialized_; |
| 390 | bool playing_; |
| 391 | bool recording_; |
| 392 | |
| 393 | // Local copy of the current Automatic Gain Control state. |
| 394 | bool agc_is_enabled_; |
| 395 | |
| 396 | // Used for histograms of total recording and playout times. |
| 397 | base::Time start_capture_time_; |
| 398 | base::Time start_render_time_; |
| 399 | |
Torne (Richard Coles) | 2a99a7e | 2013-03-28 15:31:22 +0000 | [diff] [blame] | 400 | // Stores latest microphone volume received in a CaptureData() callback. |
| 401 | // Range is [0, 255]. |
| 402 | uint32_t microphone_volume_; |
| 403 | |
Torne (Richard Coles) | 5821806 | 2012-11-14 11:43:16 +0000 | [diff] [blame] | 404 | DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); |
| 405 | }; |
| 406 | |
| 407 | } // namespace content |
| 408 | |
| 409 | #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |