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Side by Side Diff: webrtc/audio/audio_send_stream.cc

Issue 2392883002: Multi frequency DTMF support - sender side (Closed)
Patch Set: WVoMC unittests for multi rate DTMF send Created 4 years, 2 months ago
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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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