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

Issue 1702043002: Remove video-codec max bitrate from TMMBN. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 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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243 return rtp_sender_.RegisterPayload( 243 return rtp_sender_.RegisterPayload(
244 voice_codec.plname, 244 voice_codec.plname,
245 voice_codec.pltype, 245 voice_codec.pltype,
246 voice_codec.plfreq, 246 voice_codec.plfreq,
247 voice_codec.channels, 247 voice_codec.channels,
248 (voice_codec.rate < 0) ? 0 : voice_codec.rate); 248 (voice_codec.rate < 0) ? 0 : voice_codec.rate);
249 } 249 }
250 250
251 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const VideoCodec& video_codec) { 251 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const VideoCodec& video_codec) {
252 send_video_codec_ = video_codec; 252 send_video_codec_ = video_codec;
253 return rtp_sender_.RegisterPayload(video_codec.plName, 253 return rtp_sender_.RegisterPayload(video_codec.plName, video_codec.plType,
254 video_codec.plType, 254 90000, 0, 0);
255 90000,
256 0,
257 video_codec.maxBitrate);
258 } 255 }
259 256
260 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) { 257 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
261 return rtp_sender_.DeRegisterSendPayload(payload_type); 258 return rtp_sender_.DeRegisterSendPayload(payload_type);
262 } 259 }
263 260
264 int8_t ModuleRtpRtcpImpl::SendPayloadType() const { 261 int8_t ModuleRtpRtcpImpl::SendPayloadType() const {
265 return rtp_sender_.SendPayloadType(); 262 return rtp_sender_.SendPayloadType();
266 } 263 }
267 264
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667 // (TMMBR) Temporary Max Media Bit Rate. 664 // (TMMBR) Temporary Max Media Bit Rate.
668 bool ModuleRtpRtcpImpl::TMMBR() const { 665 bool ModuleRtpRtcpImpl::TMMBR() const {
669 return rtcp_sender_.TMMBR(); 666 return rtcp_sender_.TMMBR();
670 } 667 }
671 668
672 void ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) { 669 void ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) {
673 rtcp_sender_.SetTMMBRStatus(enable); 670 rtcp_sender_.SetTMMBRStatus(enable);
674 } 671 }
675 672
676 int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) { 673 int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) {
677 uint32_t max_bitrate_kbit = 674 return rtcp_sender_.SetTMMBN(bounding_set);
678 rtp_sender_.MaxConfiguredBitrateVideo() / 1000;
679 return rtcp_sender_.SetTMMBN(bounding_set, max_bitrate_kbit);
680 } 675 }
681 676
682 // Returns the currently configured retransmission mode. 677 // Returns the currently configured retransmission mode.
683 int ModuleRtpRtcpImpl::SelectiveRetransmissions() const { 678 int ModuleRtpRtcpImpl::SelectiveRetransmissions() const {
684 return rtp_sender_.SelectiveRetransmissions(); 679 return rtp_sender_.SelectiveRetransmissions();
685 } 680 }
686 681
687 // Enable or disable a retransmission mode, which decides which packets will 682 // Enable or disable a retransmission mode, which decides which packets will
688 // be retransmitted if NACKed. 683 // be retransmitted if NACKed.
689 int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) { 684 int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) {
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989 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 984 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
990 StreamDataCountersCallback* callback) { 985 StreamDataCountersCallback* callback) {
991 rtp_sender_.RegisterRtpStatisticsCallback(callback); 986 rtp_sender_.RegisterRtpStatisticsCallback(callback);
992 } 987 }
993 988
994 StreamDataCountersCallback* 989 StreamDataCountersCallback*
995 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 990 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
996 return rtp_sender_.GetRtpStatisticsCallback(); 991 return rtp_sender_.GetRtpStatisticsCallback();
997 } 992 }
998 } // namespace webrtc 993 } // namespace webrtc
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