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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| .. | ||
| audio_coding | ||
| audio_device | ||
| audio_mixer | ||
| audio_processing | ||
| congestion_controller | ||
| desktop_capture | ||
| include | ||
| pacing | ||
| remote_bitrate_estimator | ||
| rtp_rtcp | ||
| third_party | ||
| utility | ||
| video_capture | ||
| video_coding | ||
| video_processing | ||
| BUILD.gn | ||
| module_common_types_unittest.cc | ||
| OWNERS | ||