Some clients will not count audio packets into the bandwidth estimate despite negotiating e.g. abs-send-time for that SSRC. If padding is sent on such an RTP module, we might get stuck in a low resolution. This CL works around that by preferring to send padding on video SSRCs. Bug: webrtc:11196 Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30066} |
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| .. | ||
| flexfec_receiver.h | ||
| flexfec_sender.h | ||
| receive_statistics.h | ||
| remote_ntp_time_estimator.h | ||
| report_block_data.cc | ||
| report_block_data.h | ||
| rtcp_statistics.h | ||
| rtp_cvo.h | ||
| rtp_header_extension_map.h | ||
| rtp_packet_sender.h | ||
| rtp_rtcp_defines.cc | ||
| rtp_rtcp_defines.h | ||
| rtp_rtcp.h | ||
| ulpfec_receiver.h | ||