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signed-to-unsigned overflow in send_side_bandwidth_estimation.cc #872
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Thanks @fippo, we'll do some tests next week and see if there is any difference on the resulting BWE. |
The change does not make any better in the local tests. I can even see worst results with it applied :-/ The tests is as simple as using mediasoup-demo with several consumers and periodically checking the mediasoup transport stats in order to see the available outgoing bitrate. |
@jmillan We found this issue several months ago, and apply a similar fix as what libwebrtc do. |
By the way, this issue is in |
@sarumjanuch can you consider these changes into your ongoing PR #922? UPDATE: Ignore, since it's already done in your PR: https://github.com/versatica/mediasoup/pull/922/files#diff-531cc543a44ef9ae28d7eaea4e06c92879598eeac34186fcf2c5c225c3ad6833R395 |
- Fix calculation of feedback min_pending_time in goog_cc - Fixes #849 - Commit in libwebrtc: https://webrtc.googlesource.com/src/+/d65dc979b17cdc7cd359aada59e5bce8a6f1b8ce%5E%21/ - Fix signed-to-unsigned overflow in send_side_bandwidth_estimation.cc - Fixes #872 - Issue in libwebrtc: https://bugs.chromium.org/p/webrtc/issues/detail?id=14272 - Commit in libwebrtc: https://webrtc.googlesource.com/src/+/9804aa5f6ad26a45338d685da66497c3bbd88ca6%5E%21/ NOTE: Some changes are already present in ongoing PR #922 but it's not yet merged.
Anyway changes are applied into current v3 in this PR: #944 |
- Fix calculation of feedback min_pending_time in goog_cc - Fixes #849 - Commit in libwebrtc: https://webrtc.googlesource.com/src/+/d65dc979b17cdc7cd359aada59e5bce8a6f1b8ce%5E%21/ - Fix signed-to-unsigned overflow in send_side_bandwidth_estimation.cc - Fixes #872 - Issue in libwebrtc: https://bugs.chromium.org/p/webrtc/issues/detail?id=14272 - Commit in libwebrtc: https://webrtc.googlesource.com/src/+/9804aa5f6ad26a45338d685da66497c3bbd88ca6%5E%21/ NOTE: Some changes are already present in ongoing PR #922 but it's not yet merged.
- Fix calculation of feedback min_pending_time in goog_cc - Fixes versatica#849 - Commit in libwebrtc: https://webrtc.googlesource.com/src/+/d65dc979b17cdc7cd359aada59e5bce8a6f1b8ce%5E%21/ - Fix signed-to-unsigned overflow in send_side_bandwidth_estimation.cc - Fixes versatica#872 - Issue in libwebrtc: https://bugs.chromium.org/p/webrtc/issues/detail?id=14272 - Commit in libwebrtc: https://webrtc.googlesource.com/src/+/9804aa5f6ad26a45338d685da66497c3bbd88ca6%5E%21/ NOTE: Some changes are already present in ongoing PR versatica#922 but it's not yet merged.
Issue description
From libwebrtc upstream:
https://bugs.chromium.org/p/webrtc/issues/detail?id=14272
mediasoup has the similar (but not exactly the same, probably an old version) code here, the fix should still apply
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