Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(36)

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

Issue 1493403003: modules/rtp_rtcp/include folder cleared of lint warnings (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 782 matching lines...) Expand 10 before | Expand all | Expand 10 after
793 return rtp_sender_.SetAudioLevel(level_d_bov); 793 return rtp_sender_.SetAudioLevel(level_d_bov);
794 } 794 }
795 795
796 // Set payload type for Redundant Audio Data RFC 2198. 796 // Set payload type for Redundant Audio Data RFC 2198.
797 int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType( 797 int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType(
798 const int8_t payload_type) { 798 const int8_t payload_type) {
799 return rtp_sender_.SetRED(payload_type); 799 return rtp_sender_.SetRED(payload_type);
800 } 800 }
801 801
802 // Get payload type for Redundant Audio Data RFC 2198. 802 // Get payload type for Redundant Audio Data RFC 2198.
803 int32_t ModuleRtpRtcpImpl::SendREDPayloadType( 803 int32_t ModuleRtpRtcpImpl::SendREDPayloadType(int8_t* payload_type) const {
804 int8_t& payload_type) const { 804 return rtp_sender_.RED(payload_type);
805 return rtp_sender_.RED(&payload_type);
806 } 805 }
807 806
808 void ModuleRtpRtcpImpl::SetTargetSendBitrate(uint32_t bitrate_bps) { 807 void ModuleRtpRtcpImpl::SetTargetSendBitrate(uint32_t bitrate_bps) {
809 rtp_sender_.SetTargetBitrate(bitrate_bps); 808 rtp_sender_.SetTargetBitrate(bitrate_bps);
810 } 809 }
811 810
812 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( 811 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
813 const KeyFrameRequestMethod method) { 812 const KeyFrameRequestMethod method) {
814 key_frame_req_method_ = method; 813 key_frame_req_method_ = method;
815 return 0; 814 return 0;
(...skipping 15 matching lines...) Expand all
831 GetFeedbackState(), kRtcpSli, 0, 0, false, picture_id); 830 GetFeedbackState(), kRtcpSli, 0, 0, false, picture_id);
832 } 831 }
833 832
834 void ModuleRtpRtcpImpl::SetGenericFECStatus( 833 void ModuleRtpRtcpImpl::SetGenericFECStatus(
835 const bool enable, 834 const bool enable,
836 const uint8_t payload_type_red, 835 const uint8_t payload_type_red,
837 const uint8_t payload_type_fec) { 836 const uint8_t payload_type_fec) {
838 rtp_sender_.SetGenericFECStatus(enable, payload_type_red, payload_type_fec); 837 rtp_sender_.SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
839 } 838 }
840 839
841 void ModuleRtpRtcpImpl::GenericFECStatus(bool& enable, 840 void ModuleRtpRtcpImpl::GenericFECStatus(bool* enable,
842 uint8_t& payload_type_red, 841 uint8_t* payload_type_red,
843 uint8_t& payload_type_fec) { 842 uint8_t* payload_type_fec) {
844 rtp_sender_.GenericFECStatus(&enable, &payload_type_red, 843 rtp_sender_.GenericFECStatus(enable, payload_type_red, payload_type_fec);
845 &payload_type_fec);
846 } 844 }
847 845
848 int32_t ModuleRtpRtcpImpl::SetFecParameters( 846 int32_t ModuleRtpRtcpImpl::SetFecParameters(
849 const FecProtectionParams* delta_params, 847 const FecProtectionParams* delta_params,
850 const FecProtectionParams* key_params) { 848 const FecProtectionParams* key_params) {
851 return rtp_sender_.SetFecParameters(delta_params, key_params); 849 return rtp_sender_.SetFecParameters(delta_params, key_params);
852 } 850 }
853 851
854 void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) { 852 void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
855 // Inform about the incoming SSRC. 853 // Inform about the incoming SSRC.
(...skipping 135 matching lines...) Expand 10 before | Expand all | Expand 10 after
991 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 989 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
992 StreamDataCountersCallback* callback) { 990 StreamDataCountersCallback* callback) {
993 rtp_sender_.RegisterRtpStatisticsCallback(callback); 991 rtp_sender_.RegisterRtpStatisticsCallback(callback);
994 } 992 }
995 993
996 StreamDataCountersCallback* 994 StreamDataCountersCallback*
997 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 995 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
998 return rtp_sender_.GetRtpStatisticsCallback(); 996 return rtp_sender_.GetRtpStatisticsCallback();
999 } 997 }
1000 } // namespace webrtc 998 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h ('k') | webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698