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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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172 }); | 172 }); |
173 thread_sync_event.Wait(rtc::Event::kForever); | 173 thread_sync_event.Wait(rtc::Event::kForever); |
174 | 174 |
175 ScopedVoEInterface<VoEBase> base(voice_engine()); | 175 ScopedVoEInterface<VoEBase> base(voice_engine()); |
176 int error = base->StopSend(config_.voe_channel_id); | 176 int error = base->StopSend(config_.voe_channel_id); |
177 if (error != 0) { | 177 if (error != 0) { |
178 LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error; | 178 LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error; |
179 } | 179 } |
180 } | 180 } |
181 | 181 |
182 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event, | 182 bool AudioSendStream::SendTelephoneEvent(int payload_type, |
| 183 int payload_frequency, int event, |
183 int duration_ms) { | 184 int duration_ms) { |
184 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 185 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
185 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) && | 186 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type, |
| 187 payload_frequency) && |
186 channel_proxy_->SendTelephoneEventOutband(event, duration_ms); | 188 channel_proxy_->SendTelephoneEventOutband(event, duration_ms); |
187 } | 189 } |
188 | 190 |
189 void AudioSendStream::SetMuted(bool muted) { | 191 void AudioSendStream::SetMuted(bool muted) { |
190 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 192 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
191 channel_proxy_->SetInputMute(muted); | 193 channel_proxy_->SetInputMute(muted); |
192 } | 194 } |
193 | 195 |
194 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { | 196 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
195 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 197 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
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433 LOG_RTCERR2(SetVADStatus, channel, true, base->LastError()); | 435 LOG_RTCERR2(SetVADStatus, channel, true, base->LastError()); |
434 return false; | 436 return false; |
435 } | 437 } |
436 } | 438 } |
437 } | 439 } |
438 return true; | 440 return true; |
439 } | 441 } |
440 | 442 |
441 } // namespace internal | 443 } // namespace internal |
442 } // namespace webrtc | 444 } // namespace webrtc |
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