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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 comments Created 4 years, 7 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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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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