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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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188 | 188 |
189 // Interface of the audio processor used by the audio track to collect | 189 // Interface of the audio processor used by the audio track to collect |
190 // statistics. | 190 // statistics. |
191 class AudioProcessorInterface : public rtc::RefCountInterface { | 191 class AudioProcessorInterface : public rtc::RefCountInterface { |
192 public: | 192 public: |
193 struct AudioProcessorStats { | 193 struct AudioProcessorStats { |
194 AudioProcessorStats() : typing_noise_detected(false), | 194 AudioProcessorStats() : typing_noise_detected(false), |
195 echo_return_loss(0), | 195 echo_return_loss(0), |
196 echo_return_loss_enhancement(0), | 196 echo_return_loss_enhancement(0), |
197 echo_delay_median_ms(0), | 197 echo_delay_median_ms(0), |
| 198 echo_delay_std_ms(0), |
198 aec_quality_min(0.0), | 199 aec_quality_min(0.0), |
199 echo_delay_std_ms(0), | 200 residual_echo_likelihood(0.0f), |
200 aec_divergent_filter_fraction(0.0) {} | 201 aec_divergent_filter_fraction(0.0) {} |
201 ~AudioProcessorStats() {} | 202 ~AudioProcessorStats() {} |
202 | 203 |
203 bool typing_noise_detected; | 204 bool typing_noise_detected; |
204 int echo_return_loss; | 205 int echo_return_loss; |
205 int echo_return_loss_enhancement; | 206 int echo_return_loss_enhancement; |
206 int echo_delay_median_ms; | 207 int echo_delay_median_ms; |
| 208 int echo_delay_std_ms; |
207 float aec_quality_min; | 209 float aec_quality_min; |
208 int echo_delay_std_ms; | 210 float residual_echo_likelihood; |
209 float aec_divergent_filter_fraction; | 211 float aec_divergent_filter_fraction; |
210 }; | 212 }; |
211 | 213 |
212 // Get audio processor statistics. | 214 // Get audio processor statistics. |
213 virtual void GetStats(AudioProcessorStats* stats) = 0; | 215 virtual void GetStats(AudioProcessorStats* stats) = 0; |
214 | 216 |
215 protected: | 217 protected: |
216 virtual ~AudioProcessorInterface() {} | 218 virtual ~AudioProcessorInterface() {} |
217 }; | 219 }; |
218 | 220 |
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263 virtual bool RemoveTrack(AudioTrackInterface* track) = 0; | 265 virtual bool RemoveTrack(AudioTrackInterface* track) = 0; |
264 virtual bool RemoveTrack(VideoTrackInterface* track) = 0; | 266 virtual bool RemoveTrack(VideoTrackInterface* track) = 0; |
265 | 267 |
266 protected: | 268 protected: |
267 virtual ~MediaStreamInterface() {} | 269 virtual ~MediaStreamInterface() {} |
268 }; | 270 }; |
269 | 271 |
270 } // namespace webrtc | 272 } // namespace webrtc |
271 | 273 |
272 #endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_ | 274 #endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_ |
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