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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 2066473002: RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove-red3
Patch Set: rebase Created 4 years, 4 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 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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779 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( 779 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
780 const uint16_t packet_size_samples) { 780 const uint16_t packet_size_samples) {
781 return rtp_sender_.SetAudioPacketSize(packet_size_samples); 781 return rtp_sender_.SetAudioPacketSize(packet_size_samples);
782 } 782 }
783 783
784 int32_t ModuleRtpRtcpImpl::SetAudioLevel( 784 int32_t ModuleRtpRtcpImpl::SetAudioLevel(
785 const uint8_t level_d_bov) { 785 const uint8_t level_d_bov) {
786 return rtp_sender_.SetAudioLevel(level_d_bov); 786 return rtp_sender_.SetAudioLevel(level_d_bov);
787 } 787 }
788 788
789 // Set payload type for Redundant Audio Data RFC 2198.
790 int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType(
791 const int8_t payload_type) {
792 return rtp_sender_.SetRED(payload_type);
793 }
794
795 // Get payload type for Redundant Audio Data RFC 2198.
796 int32_t ModuleRtpRtcpImpl::SendREDPayloadType(int8_t* payload_type) const {
797 return rtp_sender_.RED(payload_type);
798 }
799
800 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( 789 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
801 const KeyFrameRequestMethod method) { 790 const KeyFrameRequestMethod method) {
802 key_frame_req_method_ = method; 791 key_frame_req_method_ = method;
803 return 0; 792 return 0;
804 } 793 }
805 794
806 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { 795 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
807 switch (key_frame_req_method_) { 796 switch (key_frame_req_method_) {
808 case kKeyFrameReqPliRtcp: 797 case kKeyFrameReqPliRtcp:
809 return SendRTCP(kRtcpPli); 798 return SendRTCP(kRtcpPli);
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969 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 958 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
970 StreamDataCountersCallback* callback) { 959 StreamDataCountersCallback* callback) {
971 rtp_sender_.RegisterRtpStatisticsCallback(callback); 960 rtp_sender_.RegisterRtpStatisticsCallback(callback);
972 } 961 }
973 962
974 StreamDataCountersCallback* 963 StreamDataCountersCallback*
975 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 964 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
976 return rtp_sender_.GetRtpStatisticsCallback(); 965 return rtp_sender_.GetRtpStatisticsCallback();
977 } 966 }
978 } // namespace webrtc 967 } // namespace webrtc
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