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![]() 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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mock_recovered_packet_receiver.cc | ||
mock_recovered_packet_receiver.h | ||
mock_rtcp_bandwidth_observer.cc | ||
mock_rtcp_bandwidth_observer.h | ||
mock_rtcp_rtt_stats.cc | ||
mock_rtcp_rtt_stats.h | ||
mock_rtp_rtcp.cc | ||
mock_rtp_rtcp.h |