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| 1 /* | 1 /* |
| 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 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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| 944 } | 944 } |
| 945 | 945 |
| 946 const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() { | 946 const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() { |
| 947 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); | 947 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
| 948 return codecs_; | 948 return codecs_; |
| 949 } | 949 } |
| 950 | 950 |
| 951 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { | 951 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { |
| 952 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); | 952 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
| 953 RtpCapabilities capabilities; | 953 RtpCapabilities capabilities; |
| 954 capabilities.header_extensions.push_back(RtpHeaderExtension( | |
| 955 kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId)); | |
| 956 capabilities.header_extensions.push_back( | 954 capabilities.header_extensions.push_back( |
| 957 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, | 955 webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, |
| 958 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); | 956 webrtc::RtpExtension::kAudioLevelDefaultId)); |
| 957 capabilities.header_extensions.push_back( |
| 958 webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri, |
| 959 webrtc::RtpExtension::kAbsSendTimeDefaultId)); |
| 959 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == | 960 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == |
| 960 "Enabled") { | 961 "Enabled") { |
| 961 capabilities.header_extensions.push_back(RtpHeaderExtension( | 962 capabilities.header_extensions.push_back(webrtc::RtpExtension( |
| 962 kRtpTransportSequenceNumberHeaderExtension, | 963 webrtc::RtpExtension::kTransportSequenceNumberUri, |
| 963 kRtpTransportSequenceNumberHeaderExtensionDefaultId)); | 964 webrtc::RtpExtension::kTransportSequenceNumberDefaultId)); |
| 964 } | 965 } |
| 965 return capabilities; | 966 return capabilities; |
| 966 } | 967 } |
| 967 | 968 |
| 968 int WebRtcVoiceEngine::GetLastEngineError() { | 969 int WebRtcVoiceEngine::GetLastEngineError() { |
| 969 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 970 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 970 return voe_wrapper_->error(); | 971 return voe_wrapper_->error(); |
| 971 } | 972 } |
| 972 | 973 |
| 973 void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace, | 974 void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace, |
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| 2627 } | 2628 } |
| 2628 } else { | 2629 } else { |
| 2629 LOG(LS_INFO) << "Stopping playout for channel #" << channel; | 2630 LOG(LS_INFO) << "Stopping playout for channel #" << channel; |
| 2630 engine()->voe()->base()->StopPlayout(channel); | 2631 engine()->voe()->base()->StopPlayout(channel); |
| 2631 } | 2632 } |
| 2632 return true; | 2633 return true; |
| 2633 } | 2634 } |
| 2634 } // namespace cricket | 2635 } // namespace cricket |
| 2635 | 2636 |
| 2636 #endif // HAVE_WEBRTC_VOICE | 2637 #endif // HAVE_WEBRTC_VOICE |
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