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Side by Side Diff: webrtc/media/engine/webrtcvideoengine2.cc

Issue 1904063003: Fixing the interaction between codec bitrate limit and "b=AS". (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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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763 763
764 // Handle RTP header extensions. 764 // Handle RTP header extensions.
765 std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( 765 std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
766 params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true); 766 params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
767 if (send_rtp_extensions_ != filtered_extensions) { 767 if (send_rtp_extensions_ != filtered_extensions) {
768 changed_params->rtp_header_extensions = 768 changed_params->rtp_header_extensions =
769 rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); 769 rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
770 } 770 }
771 771
772 // Handle max bitrate. 772 // Handle max bitrate.
773 if (params.max_bandwidth_bps != bitrate_config_.max_bitrate_bps && 773 if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps &&
774 params.max_bandwidth_bps >= 0) { 774 params.max_bandwidth_bps >= 0) {
775 // 0 uncaps max bitrate (-1). 775 // 0 uncaps max bitrate (-1).
776 changed_params->max_bandwidth_bps = rtc::Optional<int>( 776 changed_params->max_bandwidth_bps = rtc::Optional<int>(
777 params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps); 777 params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps);
778 } 778 }
779 779
780 // Handle conference mode. 780 // Handle conference mode.
781 if (params.conference_mode != send_params_.conference_mode) { 781 if (params.conference_mode != send_params_.conference_mode) {
782 changed_params->conference_mode = 782 changed_params->conference_mode =
783 rtc::Optional<bool>(params.conference_mode); 783 rtc::Optional<bool>(params.conference_mode);
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798 } 798 }
799 799
800 bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) { 800 bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
801 TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters"); 801 TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters");
802 LOG(LS_INFO) << "SetSendParameters: " << params.ToString(); 802 LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
803 ChangedSendParameters changed_params; 803 ChangedSendParameters changed_params;
804 if (!GetChangedSendParameters(params, &changed_params)) { 804 if (!GetChangedSendParameters(params, &changed_params)) {
805 return false; 805 return false;
806 } 806 }
807 807
808 bool bitrate_config_changed = false;
809
810 if (changed_params.codec) { 808 if (changed_params.codec) {
811 const VideoCodecSettings& codec_settings = *changed_params.codec; 809 const VideoCodecSettings& codec_settings = *changed_params.codec;
812 send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings); 810 send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings);
813
814 LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString(); 811 LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString();
815 // TODO(holmer): Changing the codec parameters shouldn't necessarily mean
816 // that we change the min/max of bandwidth estimation. Reevaluate this.
817 bitrate_config_ = GetBitrateConfigForCodec(codec_settings.codec);
818 bitrate_config_changed = true;
819 } 812 }
820 813
821 if (changed_params.rtp_header_extensions) { 814 if (changed_params.rtp_header_extensions) {
822 send_rtp_extensions_ = *changed_params.rtp_header_extensions; 815 send_rtp_extensions_ = *changed_params.rtp_header_extensions;
823 } 816 }
824 817
825 if (changed_params.max_bandwidth_bps) { 818 // TODO(holmer): Changing the codec parameters shouldn't necessarily mean
819 // that we change the min/max of bandwidth estimation. Reevaluate this.
820 if (changed_params.codec || changed_params.max_bandwidth_bps) {
821 if (send_codec_) {
822 bitrate_config_ = GetBitrateConfigForCodec(send_codec_->codec);
823 }
826 // TODO(pbos): Figure out whether b=AS means max bitrate for this 824 // TODO(pbos): Figure out whether b=AS means max bitrate for this
827 // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC), in 825 // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC), in
828 // which case this should not set a Call::BitrateConfig but rather 826 // which case this should not set a Call::BitrateConfig but rather
829 // reconfigure all senders. 827 // reconfigure all senders.
830 int max_bitrate_bps = *changed_params.max_bandwidth_bps; 828 int max_bandwidth_bps =
831 bitrate_config_.start_bitrate_bps = -1; 829 params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
832 bitrate_config_.max_bitrate_bps = max_bitrate_bps; 830 bitrate_config_.max_bitrate_bps =
833 if (max_bitrate_bps > 0 && 831 send_codec_
834 bitrate_config_.min_bitrate_bps > max_bitrate_bps) { 832 ? MinNonNegative(bitrate_config_.max_bitrate_bps, max_bandwidth_bps)
835 bitrate_config_.min_bitrate_bps = max_bitrate_bps; 833 : max_bandwidth_bps;
834 if (!changed_params.codec) {
835 // If the codec isn't changing, set the start bitrate to -1 which means
836 // "unchanged" so that BWE isn't affected.
837 bitrate_config_.start_bitrate_bps = -1;
836 } 838 }
837 bitrate_config_changed = true;
838 }
839
840 if (bitrate_config_changed) {
841 call_->SetBitrateConfig(bitrate_config_); 839 call_->SetBitrateConfig(bitrate_config_);
842 } 840 }
843 841
844 { 842 {
845 rtc::CritScope stream_lock(&stream_crit_); 843 rtc::CritScope stream_lock(&stream_crit_);
846 for (auto& kv : send_streams_) { 844 for (auto& kv : send_streams_) {
847 kv.second->SetSendParameters(changed_params); 845 kv.second->SetSendParameters(changed_params);
848 } 846 }
849 if (changed_params.codec || changed_params.rtcp_mode) { 847 if (changed_params.codec || changed_params.rtcp_mode) {
850 // Update receive feedback parameters from new codec or RTCP mode. 848 // Update receive feedback parameters from new codec or RTCP mode.
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2576 rtx_mapping[video_codecs[i].codec.id] != 2574 rtx_mapping[video_codecs[i].codec.id] !=
2577 fec_settings.red_payload_type) { 2575 fec_settings.red_payload_type) {
2578 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2576 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2579 } 2577 }
2580 } 2578 }
2581 2579
2582 return video_codecs; 2580 return video_codecs;
2583 } 2581 }
2584 2582
2585 } // namespace cricket 2583 } // namespace cricket
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