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Issue 2539813003: Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. (Closed)
Patch Set: Remove the default parameter. Created 4 years ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 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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1712 bundle_transport ? *bundle_transport : content->name; 1712 bundle_transport ? *bundle_transport : content->name;
1713 quic_data_transport_->SetTransport(transport_name); 1713 quic_data_transport_->SetTransport(transport_name);
1714 return true; 1714 return true;
1715 } 1715 }
1716 #endif // HAVE_QUIC 1716 #endif // HAVE_QUIC
1717 bool sctp = (data_channel_type_ == cricket::DCT_SCTP); 1717 bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
1718 bool require_rtcp_mux = 1718 bool require_rtcp_mux =
1719 rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire; 1719 rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
1720 bool create_rtcp_transport_channel = !sctp && !require_rtcp_mux; 1720 bool create_rtcp_transport_channel = !sctp && !require_rtcp_mux;
1721 data_channel_.reset(channel_manager_->CreateDataChannel( 1721 data_channel_.reset(channel_manager_->CreateDataChannel(
1722 transport_controller_.get(), content->name, bundle_transport, 1722 transport_controller_.get(), media_controller_, content->name,
1723 create_rtcp_transport_channel, data_channel_type_)); 1723 bundle_transport, create_rtcp_transport_channel, data_channel_type_));
1724 if (!data_channel_) { 1724 if (!data_channel_) {
1725 return false; 1725 return false;
1726 } 1726 }
1727 if (require_rtcp_mux) { 1727 if (require_rtcp_mux) {
1728 data_channel_->ActivateRtcpMux(); 1728 data_channel_->ActivateRtcpMux();
1729 } 1729 }
1730 1730
1731 if (sctp) { 1731 if (sctp) {
1732 data_channel_->SignalDataReceived.connect( 1732 data_channel_->SignalDataReceived.connect(
1733 this, &WebRtcSession::OnDataChannelMessageReceived); 1733 this, &WebRtcSession::OnDataChannelMessageReceived);
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2079 } 2079 }
2080 2080
2081 void WebRtcSession::OnDtlsHandshakeError(rtc::SSLHandshakeError error) { 2081 void WebRtcSession::OnDtlsHandshakeError(rtc::SSLHandshakeError error) {
2082 if (metrics_observer_) { 2082 if (metrics_observer_) {
2083 metrics_observer_->IncrementEnumCounter( 2083 metrics_observer_->IncrementEnumCounter(
2084 webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error), 2084 webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
2085 static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE)); 2085 static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
2086 } 2086 }
2087 } 2087 }
2088 } // namespace webrtc 2088 } // namespace webrtc
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