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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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| 203 ScopedVoEInterface<VoEBase> base(voice_engine()); | 203 ScopedVoEInterface<VoEBase> base(voice_engine()); |
| 204 RTC_DCHECK(base->audio_processing()); | 204 RTC_DCHECK(base->audio_processing()); |
| 205 auto audio_processing_stats = base->audio_processing()->GetStatistics(); | 205 auto audio_processing_stats = base->audio_processing()->GetStatistics(); |
| 206 stats.echo_delay_median_ms = audio_processing_stats.delay_median; | 206 stats.echo_delay_median_ms = audio_processing_stats.delay_median; |
| 207 stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation; | 207 stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation; |
| 208 stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant(); | 208 stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant(); |
| 209 stats.echo_return_loss_enhancement = | 209 stats.echo_return_loss_enhancement = |
| 210 audio_processing_stats.echo_return_loss_enhancement.instant(); | 210 audio_processing_stats.echo_return_loss_enhancement.instant(); |
| 211 stats.residual_echo_likelihood = | 211 stats.residual_echo_likelihood = |
| 212 audio_processing_stats.residual_echo_likelihood; | 212 audio_processing_stats.residual_echo_likelihood; |
| 213 stats.residual_echo_likelihood_recent_max = |
| 214 audio_processing_stats.residual_echo_likelihood_recent_max; |
| 213 | 215 |
| 214 internal::AudioState* audio_state = | 216 internal::AudioState* audio_state = |
| 215 static_cast<internal::AudioState*>(audio_state_.get()); | 217 static_cast<internal::AudioState*>(audio_state_.get()); |
| 216 stats.typing_noise_detected = audio_state->typing_noise_detected(); | 218 stats.typing_noise_detected = audio_state->typing_noise_detected(); |
| 217 | 219 |
| 218 return stats; | 220 return stats; |
| 219 } | 221 } |
| 220 | 222 |
| 221 void AudioSendStream::SignalNetworkState(NetworkState state) { | 223 void AudioSendStream::SignalNetworkState(NetworkState state) { |
| 222 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 224 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
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| 381 LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError(); | 383 LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError(); |
| 382 return false; | 384 return false; |
| 383 } | 385 } |
| 384 } | 386 } |
| 385 } | 387 } |
| 386 return true; | 388 return true; |
| 387 } | 389 } |
| 388 | 390 |
| 389 } // namespace internal | 391 } // namespace internal |
| 390 } // namespace webrtc | 392 } // namespace webrtc |
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