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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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| 927 } | 927 } |
| 928 | 928 |
| 929 const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() { | 929 const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() { |
| 930 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); | 930 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
| 931 return codecs_; | 931 return codecs_; |
| 932 } | 932 } |
| 933 | 933 |
| 934 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { | 934 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { |
| 935 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); | 935 RTC_DCHECK(signal_thread_checker_.CalledOnValidThread()); |
| 936 RtpCapabilities capabilities; | 936 RtpCapabilities capabilities; |
| 937 capabilities.header_extensions.push_back(RtpHeaderExtension( | |
| 938 kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId)); | |
| 939 capabilities.header_extensions.push_back( | 937 capabilities.header_extensions.push_back( |
| 940 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, | 938 webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, |
| 941 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); | 939 webrtc::RtpExtension::kAudioLevelDefaultId)); |
| 940 capabilities.header_extensions.push_back( |
| 941 webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri, |
| 942 webrtc::RtpExtension::kAbsSendTimeDefaultId)); |
| 942 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == | 943 if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == |
| 943 "Enabled") { | 944 "Enabled") { |
| 944 capabilities.header_extensions.push_back(RtpHeaderExtension( | 945 capabilities.header_extensions.push_back(webrtc::RtpExtension( |
| 945 kRtpTransportSequenceNumberHeaderExtension, | 946 webrtc::RtpExtension::kTransportSequenceNumberUri, |
| 946 kRtpTransportSequenceNumberHeaderExtensionDefaultId)); | 947 webrtc::RtpExtension::kTransportSequenceNumberDefaultId)); |
| 947 } | 948 } |
| 948 return capabilities; | 949 return capabilities; |
| 949 } | 950 } |
| 950 | 951 |
| 951 int WebRtcVoiceEngine::GetLastEngineError() { | 952 int WebRtcVoiceEngine::GetLastEngineError() { |
| 952 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 953 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| 953 return voe_wrapper_->error(); | 954 return voe_wrapper_->error(); |
| 954 } | 955 } |
| 955 | 956 |
| 956 void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace, | 957 void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace, |
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| 2560 } | 2561 } |
| 2561 } else { | 2562 } else { |
| 2562 LOG(LS_INFO) << "Stopping playout for channel #" << channel; | 2563 LOG(LS_INFO) << "Stopping playout for channel #" << channel; |
| 2563 engine()->voe()->base()->StopPlayout(channel); | 2564 engine()->voe()->base()->StopPlayout(channel); |
| 2564 } | 2565 } |
| 2565 return true; | 2566 return true; |
| 2566 } | 2567 } |
| 2567 } // namespace cricket | 2568 } // namespace cricket |
| 2568 | 2569 |
| 2569 #endif // HAVE_WEBRTC_VOICE | 2570 #endif // HAVE_WEBRTC_VOICE |
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