Change of msid in remote description should update receiver's streams and fire ontrack |
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Issue descriptionThis is the chromium layer version of https://crbug.com/webrtc/10083 . When the streams are updated on third_party/webrtc, chromium also needs to update its code in order to surface the state changes and make them visible in JavaScript.
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Dec 20
The following revision refers to this bug: https://webrtc.googlesource.com/src.git/+/afa07dda42c9253a37d1c220e71e2c79df90d8d7 commit afa07dda42c9253a37d1c220e71e2c79df90d8d7 Author: Henrik Boström <hbos@webrtc.org> Date: Thu Dec 20 13:01:58 2018 [Unified Plan] SRD: Always set associated remote streams. This fixes a bug where the streams are not updated if the "msid" changes without triggering "ontrack", such as if the streams associated with a receiver changes while the receiver is active. Bug: webrtc:10083 , chromium:916934 Change-Id: Ic7b19ad5ef648ed6880cae4157bf49f8435467ae Reviewed-on: https://webrtc-review.googlesource.com/c/114161 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26069} [modify] https://crrev.com/afa07dda42c9253a37d1c220e71e2c79df90d8d7/pc/peerconnection.cc [modify] https://crrev.com/afa07dda42c9253a37d1c220e71e2c79df90d8d7/pc/peerconnection.h [modify] https://crrev.com/afa07dda42c9253a37d1c220e71e2c79df90d8d7/pc/peerconnection_rtp_unittest.cc
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Dec 20
The following revision refers to this bug: https://chromium.googlesource.com/chromium/src.git/+/c33c286e565fc89faaa492178a3b4ba7cf666ae8 commit c33c286e565fc89faaa492178a3b4ba7cf666ae8 Author: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com> Date: Thu Dec 20 18:34:55 2018 Roll src/third_party/webrtc 9a4f38ec5c4a..3793bb447ae5 (8 commits) https://webrtc.googlesource.com/src.git/+log/9a4f38ec5c4a..3793bb447ae5 git log 9a4f38ec5c4a..3793bb447ae5 --date=short --no-merges --format='%ad %ae %s' 2018-12-20 nisse@webrtc.org Refactor TestVideoCapturer to support multiple sinks. 2018-12-20 ssilkin@webrtc.org Update number of spatial layers if SS is available. 2018-12-20 srte@webrtc.org Adds flag to customize output root dir in scenario tests. 2018-12-20 srte@webrtc.org Adds support for empty key fields in field trial parser. 2018-12-20 hbos@webrtc.org [Unified Plan] SRD: Always set associated remote streams. 2018-12-20 srte@webrtc.org Stop using deprecated PacedSender method from RtpTransportControllerSend. 2018-12-20 nisse@webrtc.org Refactor NetEqDecoderPlc to use AudioDecoderProxyFactory 2018-12-20 srte@webrtc.org Removes return value and Try prefix from TryDeliverPacket. Created with: gclient setdep -r src/third_party/webrtc@3793bb447ae5 The AutoRoll server is located here: https://autoroll.skia.org/r/webrtc-chromium-autoroll Documentation for the AutoRoller is here: https://skia.googlesource.com/buildbot/+/master/autoroll/README.md If the roll is causing failures, please contact the current sheriff, who should be CC'd on the roll, and stop the roller if necessary. CQ_INCLUDE_TRYBOTS=luci.chromium.try:linux_chromium_archive_rel_ng;luci.chromium.try:mac_chromium_archive_rel_ng BUG= chromium:916934 TBR=webrtc-chromium-sheriffs-robots@google.com Change-Id: Idaf600f08455b6b520d6cdae738c8cf881d30baa Reviewed-on: https://chromium-review.googlesource.com/c/1387186 Reviewed-by: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com> Commit-Queue: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#618278} [modify] https://crrev.com/c33c286e565fc89faaa492178a3b4ba7cf666ae8/DEPS
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Dec 21
This was the second CL that landed for this issue, I forgot to reference this bug from it: https://chromium-review.googlesource.com/c/chromium/src/+/1384257
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Dec 21
Requesting to merge third_party/webrtc CL in #2 and chromium CL in #4 to M-72.
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Dec 21
This bug requires manual review: DEPS changes referenced in bugdroid comments. Please contact the milestone owner if you have questions. Owners: govind@(Android), kariahda@(iOS), djmm@(ChromeOS), abdulsyed@(Desktop) For more details visit https://www.chromium.org/issue-tracking/autotriage - Your friendly Sheriffbot
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Dec 21
Note this merge has priority for WebRTC as we have planned to switch simultaneously with Safari to the new SDP format for web compact reasons. See https://webrtc.org/web-apis/chrome/unified-plan/
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Jan 2
Pls apply appropriate OSs label.
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Jan 3
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Jan 3
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Jan 3
Pls merge your change to M72 branch 3626 ASAP so we can pick it up for next Beta release. Thank you.
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Jan 4
Thank you, will do. Is the next Beta release the 9th?
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Jan 4
The following revision refers to this bug: https://chromium.googlesource.com/chromium/src.git/+/97dd55252be171ede829e9d57acdb199ddc6832f commit 97dd55252be171ede829e9d57acdb199ddc6832f Author: Henrik Boström <hbos@chromium.org> Date: Fri Jan 04 11:32:33 2019 Surface associated remote stream MSID changes for active receivers. Previously, associated media streams were only updated on the "addition" and "removal" of a track. But MSIDs can change for active receivers as well. In this CL, we always update the associated media streams based on the state of the webrtc-layer receiver. TBR=hbos@chromium.org (cherry picked from commit 05c7d3b938defb3ce4043e09d1ca839264b4c2e9) Bug: webrtc:10083 , chromium:916934 Change-Id: I54bb00a5077a0b7b78964faf13882055c8101427 Reviewed-on: https://chromium-review.googlesource.com/c/1384257 Commit-Queue: Henrik Boström <hbos@chromium.org> Reviewed-by: Harald Alvestrand <hta@chromium.org> Cr-Original-Commit-Position: refs/heads/master@{#618333} Reviewed-on: https://chromium-review.googlesource.com/c/1396142 Reviewed-by: Henrik Boström <hbos@chromium.org> Cr-Commit-Position: refs/branch-heads/3626@{#559} Cr-Branched-From: d897fb137fbaaa9355c0c93124cc048824eb1e65-refs/heads/master@{#612437} [modify] https://crrev.com/97dd55252be171ede829e9d57acdb199ddc6832f/third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.cc [modify] https://crrev.com/97dd55252be171ede829e9d57acdb199ddc6832f/third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.h
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Jan 4
#13 is the chromium merge and this is the webrtc merge: https://webrtc-review.googlesource.com/c/src/+/116180 |
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Comment 1 by hbos@chromium.org
, Dec 20