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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 1984983002: Remove use of RtpHeaderExtension and clean up (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed nit Created 4 years, 6 months ago
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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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