commit 1f2f5dc95131db779455813108ee2f84449f10c5
author Diep Bui <diepbp@webrtc.org> Thu Oct 26 07:58:26 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Oct 26 09:32:27 2023
tree 9dec53dde8bc192058c6b00d4c89ee4d31b4c7a6
parent 40ce7674c481df6a7cd9095efef369444428b3ff

Compute loss rate based on byte count rather than packet count in loss based BWE.

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}