2 main reasons:
1. Packet sizes are much different thus a lost audio packet should not be treated similar to a lost video packet. In low bandwidth/traffic policing scenario, the number of send packet is few, thus the computed loss can be imprecise.
2. Given a candidate bandwidth estimate, the objective function (how good the candidate is) is computed by recomputing loss rate = send rate/estimate bandwith + inherent loss. It means the objective function is byte based rather than packet based.
Potential risk: the current algorithm params are tuned based on packet count, thus it might not work with byte count, which is much higher than packet count.
The change is under field trial and disabled by default.
Bug: webrtc:12707
Change-Id: I8b832e7920d2b4cadcd4a072b3a4d4f26a213a20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325065
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41013}
1f2f5dc951