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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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128 }); | 128 }); |
129 thread_sync_event.Wait(rtc::Event::kForever); | 129 thread_sync_event.Wait(rtc::Event::kForever); |
130 | 130 |
131 ScopedVoEInterface<VoEBase> base(voice_engine()); | 131 ScopedVoEInterface<VoEBase> base(voice_engine()); |
132 int error = base->StopSend(config_.voe_channel_id); | 132 int error = base->StopSend(config_.voe_channel_id); |
133 if (error != 0) { | 133 if (error != 0) { |
134 LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error; | 134 LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error; |
135 } | 135 } |
136 } | 136 } |
137 | 137 |
138 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event, | 138 bool AudioSendStream::SendTelephoneEvent(int payload_type, |
| 139 int payload_frequency, int event, |
139 int duration_ms) { | 140 int duration_ms) { |
140 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 141 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
141 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) && | 142 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type, |
| 143 payload_frequency) && |
142 channel_proxy_->SendTelephoneEventOutband(event, duration_ms); | 144 channel_proxy_->SendTelephoneEventOutband(event, duration_ms); |
143 } | 145 } |
144 | 146 |
145 void AudioSendStream::SetMuted(bool muted) { | 147 void AudioSendStream::SetMuted(bool muted) { |
146 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 148 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
147 channel_proxy_->SetInputMute(muted); | 149 channel_proxy_->SetInputMute(muted); |
148 } | 150 } |
149 | 151 |
150 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { | 152 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
151 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 153 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
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386 LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError(); | 388 LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError(); |
387 return false; | 389 return false; |
388 } | 390 } |
389 } | 391 } |
390 } | 392 } |
391 return true; | 393 return true; |
392 } | 394 } |
393 | 395 |
394 } // namespace internal | 396 } // namespace internal |
395 } // namespace webrtc | 397 } // namespace webrtc |
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