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| 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
| 7 | 7 |
| 8 #include <string> | 8 #include <string> |
| 9 | 9 |
| 10 #include "base/basictypes.h" | 10 #include "base/basictypes.h" |
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| 205 // Callback to deliver the captured interleaved data. | 205 // Callback to deliver the captured interleaved data. |
| 206 virtual void CaptureData(const int16* audio_data, | 206 virtual void CaptureData(const int16* audio_data, |
| 207 int number_of_channels, | 207 int number_of_channels, |
| 208 int number_of_frames, | 208 int number_of_frames, |
| 209 int audio_delay_milliseconds, | 209 int audio_delay_milliseconds, |
| 210 double volume) = 0; | 210 double volume) = 0; |
| 211 | 211 |
| 212 // Set the format for the capture audio parameters. | 212 // Set the format for the capture audio parameters. |
| 213 virtual void SetCaptureFormat(const media::AudioParameters& params) = 0; | 213 virtual void SetCaptureFormat(const media::AudioParameters& params) = 0; |
| 214 | 214 |
| 215 // Callback to notify the client that the capturer is being stopped. | |
| 216 virtual void OnCaptureDeviceStopped() = 0; | |
| 217 | |
| 218 protected: | 215 protected: |
| 219 virtual ~WebRtcAudioCapturerSink() {} | 216 virtual ~WebRtcAudioCapturerSink() {} |
| 220 }; | 217 }; |
| 221 | 218 |
| 222 class CONTENT_EXPORT WebRtcAudioDeviceImpl | 219 class CONTENT_EXPORT WebRtcAudioDeviceImpl |
| 223 : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule), | 220 : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule), |
| 224 NON_EXPORTED_BASE(public WebRtcAudioCapturerSink), | 221 NON_EXPORTED_BASE(public WebRtcAudioCapturerSink), |
| 225 NON_EXPORTED_BASE(public WebRtcAudioRendererSource) { | 222 NON_EXPORTED_BASE(public WebRtcAudioRendererSource) { |
| 226 public: | 223 public: |
| 227 // Methods called on main render thread. | 224 // Methods called on main render thread. |
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| 241 virtual void SetRenderFormat(const media::AudioParameters& params) OVERRIDE; | 238 virtual void SetRenderFormat(const media::AudioParameters& params) OVERRIDE; |
| 242 virtual void RemoveRenderer(WebRtcAudioRenderer* renderer) OVERRIDE; | 239 virtual void RemoveRenderer(WebRtcAudioRenderer* renderer) OVERRIDE; |
| 243 | 240 |
| 244 // WebRtcAudioCapturerSink implementation. | 241 // WebRtcAudioCapturerSink implementation. |
| 245 virtual void CaptureData(const int16* audio_data, | 242 virtual void CaptureData(const int16* audio_data, |
| 246 int number_of_channels, | 243 int number_of_channels, |
| 247 int number_of_frames, | 244 int number_of_frames, |
| 248 int audio_delay_milliseconds, | 245 int audio_delay_milliseconds, |
| 249 double volume) OVERRIDE; | 246 double volume) OVERRIDE; |
| 250 virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; | 247 virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; |
| 251 virtual void OnCaptureDeviceStopped() OVERRIDE; | |
| 252 | 248 |
| 253 // webrtc::Module implementation. | 249 // webrtc::Module implementation. |
| 254 virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE; | 250 virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE; |
| 255 virtual int32_t TimeUntilNextProcess() OVERRIDE; | 251 virtual int32_t TimeUntilNextProcess() OVERRIDE; |
| 256 virtual int32_t Process() OVERRIDE; | 252 virtual int32_t Process() OVERRIDE; |
| 257 | 253 |
| 258 // webrtc::AudioDeviceModule implementation. | 254 // webrtc::AudioDeviceModule implementation. |
| 259 virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const OVERRIDE; | 255 virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const OVERRIDE; |
| 260 virtual ErrorCode LastError() const OVERRIDE; | 256 virtual ErrorCode LastError() const OVERRIDE; |
| 261 | 257 |
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| 456 // Used for histograms of total recording and playout times. | 452 // Used for histograms of total recording and playout times. |
| 457 base::Time start_capture_time_; | 453 base::Time start_capture_time_; |
| 458 base::Time start_render_time_; | 454 base::Time start_render_time_; |
| 459 | 455 |
| 460 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); | 456 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); |
| 461 }; | 457 }; |
| 462 | 458 |
| 463 } // namespace content | 459 } // namespace content |
| 464 | 460 |
| 465 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ | 461 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ |
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