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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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204 stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate); | 204 stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate); |
205 stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate); | 205 stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate); |
206 | 206 |
207 auto ds = channel_proxy_->GetDecodingCallStatistics(); | 207 auto ds = channel_proxy_->GetDecodingCallStatistics(); |
208 stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator; | 208 stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator; |
209 stats.decoding_calls_to_neteq = ds.calls_to_neteq; | 209 stats.decoding_calls_to_neteq = ds.calls_to_neteq; |
210 stats.decoding_normal = ds.decoded_normal; | 210 stats.decoding_normal = ds.decoded_normal; |
211 stats.decoding_plc = ds.decoded_plc; | 211 stats.decoding_plc = ds.decoded_plc; |
212 stats.decoding_cng = ds.decoded_cng; | 212 stats.decoding_cng = ds.decoded_cng; |
213 stats.decoding_plc_cng = ds.decoded_plc_cng; | 213 stats.decoding_plc_cng = ds.decoded_plc_cng; |
| 214 stats.decoding_muted_output = ds.decoded_muted_output; |
214 | 215 |
215 return stats; | 216 return stats; |
216 } | 217 } |
217 | 218 |
218 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) { | 219 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) { |
219 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 220 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
220 channel_proxy_->SetSink(std::move(sink)); | 221 channel_proxy_->SetSink(std::move(sink)); |
221 } | 222 } |
222 | 223 |
223 void AudioReceiveStream::SetGain(float gain) { | 224 void AudioReceiveStream::SetGain(float gain) { |
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272 | 273 |
273 VoiceEngine* AudioReceiveStream::voice_engine() const { | 274 VoiceEngine* AudioReceiveStream::voice_engine() const { |
274 internal::AudioState* audio_state = | 275 internal::AudioState* audio_state = |
275 static_cast<internal::AudioState*>(audio_state_.get()); | 276 static_cast<internal::AudioState*>(audio_state_.get()); |
276 VoiceEngine* voice_engine = audio_state->voice_engine(); | 277 VoiceEngine* voice_engine = audio_state->voice_engine(); |
277 RTC_DCHECK(voice_engine); | 278 RTC_DCHECK(voice_engine); |
278 return voice_engine; | 279 return voice_engine; |
279 } | 280 } |
280 } // namespace internal | 281 } // namespace internal |
281 } // namespace webrtc | 282 } // namespace webrtc |
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