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Issue 1181653002: Base A/V synchronization on sync_labels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 5 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2014 Google Inc. 3 * Copyright 2014 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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1148 1148
1149 if (!ValidateReceiveSsrcAvailability(sp)) 1149 if (!ValidateReceiveSsrcAvailability(sp))
1150 return false; 1150 return false;
1151 1151
1152 for (uint32 used_ssrc : sp.ssrcs) 1152 for (uint32 used_ssrc : sp.ssrcs)
1153 receive_ssrcs_.insert(used_ssrc); 1153 receive_ssrcs_.insert(used_ssrc);
1154 1154
1155 webrtc::VideoReceiveStream::Config config; 1155 webrtc::VideoReceiveStream::Config config;
1156 ConfigureReceiverRtp(&config, sp); 1156 ConfigureReceiverRtp(&config, sp);
1157 1157
1158 // Set up A/V sync if there is a VoiceChannel. 1158 // Set up A/V sync group based on sync label.
1159 // TODO(pbos): The A/V is synched by the receiving channel. So we need to know 1159 config.sync_group = sp.sync_label;
1160 // the SSRC of the remote audio channel in order to sync the correct webrtc
1161 // VoiceEngine channel. For now sync the first channel in non-conference to
1162 // match existing behavior in WebRtcVideoEngine.
1163 if (voice_channel_id_ != -1 && receive_streams_.empty() &&
1164 !options_.conference_mode.GetWithDefaultIfUnset(false)) {
1165 config.audio_channel_id = voice_channel_id_;
1166 }
1167 1160
1168 config.rtp.remb = false; 1161 config.rtp.remb = false;
1169 VideoCodecSettings send_codec; 1162 VideoCodecSettings send_codec;
1170 if (send_codec_.Get(&send_codec)) { 1163 if (send_codec_.Get(&send_codec)) {
1171 config.rtp.remb = HasRemb(send_codec.codec); 1164 config.rtp.remb = HasRemb(send_codec.codec);
1172 } 1165 }
1173 1166
1174 receive_streams_[ssrc] = new WebRtcVideoReceiveStream( 1167 receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
1175 call_.get(), sp, external_decoder_factory_, default_stream, config, 1168 call_.get(), sp, external_decoder_factory_, default_stream, config,
1176 recv_codecs_); 1169 recv_codecs_);
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2610 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2603 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2611 } 2604 }
2612 } 2605 }
2613 2606
2614 return video_codecs; 2607 return video_codecs;
2615 } 2608 }
2616 2609
2617 } // namespace cricket 2610 } // namespace cricket
2618 2611
2619 #endif // HAVE_WEBRTC_VIDEO 2612 #endif // HAVE_WEBRTC_VIDEO
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