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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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120 } | 120 } |
121 } | 121 } |
122 | 122 |
123 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event, | 123 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event, |
124 int duration_ms) { | 124 int duration_ms) { |
125 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 125 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
126 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) && | 126 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) && |
127 channel_proxy_->SendTelephoneEventOutband(event, duration_ms); | 127 channel_proxy_->SendTelephoneEventOutband(event, duration_ms); |
128 } | 128 } |
129 | 129 |
| 130 void AudioSendStream::SetMuted(bool muted) { |
| 131 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| 132 channel_proxy_->SetInputMute(muted); |
| 133 } |
| 134 |
130 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { | 135 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
131 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 136 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
132 webrtc::AudioSendStream::Stats stats; | 137 webrtc::AudioSendStream::Stats stats; |
133 stats.local_ssrc = config_.rtp.ssrc; | 138 stats.local_ssrc = config_.rtp.ssrc; |
134 ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine()); | 139 ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine()); |
135 ScopedVoEInterface<VoECodec> codec(voice_engine()); | 140 ScopedVoEInterface<VoECodec> codec(voice_engine()); |
136 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); | 141 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); |
137 | 142 |
138 webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics(); | 143 webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics(); |
139 stats.bytes_sent = call_stats.bytesSent; | 144 stats.bytes_sent = call_stats.bytesSent; |
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229 | 234 |
230 VoiceEngine* AudioSendStream::voice_engine() const { | 235 VoiceEngine* AudioSendStream::voice_engine() const { |
231 internal::AudioState* audio_state = | 236 internal::AudioState* audio_state = |
232 static_cast<internal::AudioState*>(audio_state_.get()); | 237 static_cast<internal::AudioState*>(audio_state_.get()); |
233 VoiceEngine* voice_engine = audio_state->voice_engine(); | 238 VoiceEngine* voice_engine = audio_state->voice_engine(); |
234 RTC_DCHECK(voice_engine); | 239 RTC_DCHECK(voice_engine); |
235 return voice_engine; | 240 return voice_engine; |
236 } | 241 } |
237 } // namespace internal | 242 } // namespace internal |
238 } // namespace webrtc | 243 } // namespace webrtc |
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