Update WebRTC to m112 #21

mininny · 2023-05-06T16:40:18Z

No description provided.

This CL removes the iOS sim bots 12, 13, 14 and replaces them with a unique bot that runs the test on iOS version 14, 15 and 16. Change-Id: I46673bad44c2c5539fbbf266cc9d5d468557022e Bug: b/264630045 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290999 Reviewed-by: Mirko Bonadei <[email protected]> Commit-Queue: Jeremy Leconte <[email protected]> Cr-Commit-Position: refs/heads/main@{#39131}

This reverts commit b802369. Reason for revert: Breaking WPT tests in Chrome. Example build https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/1263191/overview Original change's description: > Ensure RTCRtpSenders are always created with one encoding > > It is possible to have AddTransceiver calls with an empty array > of encodings or AddTrack calls. In both cases, before negotiation, > the sender's encodings array would be empty and it was not possible > to update any value. > > Now, a default entry should be created in those cases, allowing to > update the configuration before negotiation. > > Bug: webrtc:10567 > Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994 > Auto-Submit: Florent Castelli <[email protected]> > Reviewed-by: Henrik Boström <[email protected]> > Commit-Queue: Florent Castelli <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39126} Bug: webrtc:10567 Change-Id: Ib8931b38182251baac616540788a02a5fafaf670 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291120 Bot-Commit: [email protected] <[email protected]> Commit-Queue: Evan Shrubsole <[email protected]> Auto-Submit: Evan Shrubsole <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Cr-Commit-Position: refs/heads/main@{#39132}

Change-Id: I9313b9ab034be8cf3933d82f72c4c8e8858ca6a5 Bug: b/264630045 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291100 Reviewed-by: Mirko Bonadei <[email protected]> Commit-Queue: Jeremy Leconte <[email protected]> Cr-Commit-Position: refs/heads/main@{#39133}

… sent Bug: webrtc:5540, webrtc:10198 Change-Id: I408b471cbd14c12bdb98606999807cc7f2b56c3f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289100 Commit-Queue: Danil Chapovalov <[email protected]> Reviewed-by: Erik Språng <[email protected]> Cr-Commit-Position: refs/heads/main@{#39134}

This is a reland of commit fa962ff Original change's description: > Move leb128 helper functions into own build target > > to remove duplicated implementation of these functions between av1 packetizer, av1 depacketizer and video allocation rtp header extension > > Bug: None > Change-Id: I30049f31c289bdb9e0aad6520f5145d1f999e635 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290731 > Commit-Queue: Danil Chapovalov <[email protected]> > Reviewed-by: Philip Eliasson <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39069} Bug: None Change-Id: I091276868599a6716407db2972457507ddd46a8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290961 Commit-Queue: Danil Chapovalov <[email protected]> Reviewed-by: Philip Eliasson <[email protected]> Cr-Commit-Position: refs/heads/main@{#39135}

Receivers no longer need to set extensions in the configuration. That field will be removed in a follow up. Tested with: video_loopback --rtp_dump_name="./my.rtpdump" --duration=10 video_replay --input_file=./my.rtpdump Bug: webrtc:14795 Change-Id: I952cd487cb2f3be8be01a90f6a2312f1fef5d93e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290995 Auto-Submit: Per Kjellander <[email protected]> Commit-Queue: Mirko Bonadei <[email protected]> Reviewed-by: Mirko Bonadei <[email protected]> Reviewed-by: Philip Eliasson <[email protected]> Cr-Commit-Position: refs/heads/main@{#39136}

…iver::DeliverRtcp PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. Therefore DirectTransport is provided with the extension mapping. CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 Bug: webrtc:7135, webrtc:14795 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 Reviewed-by: Jakob Ivarsson‎ <[email protected]> Commit-Queue: Per Kjellander <[email protected]> Reviewed-by: Erik Språng <[email protected]> Cr-Commit-Position: refs/heads/main@{#39137}

modernizing the code a bit. BUG=None Change-Id: I380e9c2c4b20e3d6fc75d5963b0ed129e722099f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290997 Reviewed-by: Henrik Boström <[email protected]> Reviewed-by: Ilya Nikolaevskiy <[email protected]> Commit-Queue: Philipp Hancke <[email protected]> Cr-Commit-Position: refs/heads/main@{#39138}

Bug: webrtc:7135, webrtc:14795 Change-Id: I0242a3600d4a156eae2315966e5e59e03be8aeab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290998 Commit-Queue: Per Kjellander <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Cr-Commit-Position: refs/heads/main@{#39139}

Bug: b/263477303 Change-Id: Ifabfd1c015788e944d1b78ba2a0454c29426c5e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290993 Commit-Queue: Christoffer Jansson <[email protected]> Reviewed-by: Jeremy Leconte <[email protected]> Cr-Commit-Position: refs/heads/main@{#39140}

Bug: chromium:1330308, b/234414450 Change-Id: Iad9d38048b62d2fb99e5c76b072dd929c5e24954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291101 Reviewed-by: Erik Språng <[email protected]> Commit-Queue: Philip Eliasson <[email protected]> Cr-Commit-Position: refs/heads/main@{#39142}

This is used by CRD and export is required for component builds to work properly. Bug: chromium:1291247 Change-Id: I281e490b7d00cbd074b96eac905976af38400f8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291200 Reviewed-by: Alexander Cooper <[email protected]> Auto-Submit: Salman Malik <[email protected]> Commit-Queue: Alexander Cooper <[email protected]> Commit-Queue: Salman Malik <[email protected]> Cr-Commit-Position: refs/heads/main@{#39143}

Bug: None Change-Id: I1bd3f9b5e930df7375313014cdffa19ff1692d15 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291164 Bot-Commit: [email protected] <[email protected]> Commit-Queue: [email protected] <[email protected]> Cr-Commit-Position: refs/heads/main@{#39144}

This is a reland of commit b802369 Original change's description: > Ensure RTCRtpSenders are always created with one encoding > > It is possible to have AddTransceiver calls with an empty array > of encodings or AddTrack calls. In both cases, before negotiation, > the sender's encodings array would be empty and it was not possible > to update any value. > > Now, a default entry should be created in those cases, allowing to > update the configuration before negotiation. > > Bug: webrtc:10567 > Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994 > Auto-Submit: Florent Castelli <[email protected]> > Reviewed-by: Henrik Boström <[email protected]> > Commit-Queue: Florent Castelli <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39126} Bug: webrtc:10567 Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140 Reviewed-by: Henrik Boström <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Commit-Queue: Florent Castelli <[email protected]> Cr-Commit-Position: refs/heads/main@{#39145}

…cketReceiver::DeliverRtcp" This reverts commit 3b96f2c. Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio Original change's description: > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > Therefore DirectTransport is provided with the extension mapping. > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > Bug: webrtc:7135, webrtc:14795 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > Reviewed-by: Jakob Ivarsson‎ <[email protected]> > Commit-Queue: Per Kjellander <[email protected]> > Reviewed-by: Erik Språng <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39137} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 Owners-Override: Björn Terelius <[email protected]> Auto-Submit: Per Kjellander <[email protected]> Reviewed-by: Björn Terelius <[email protected]> Commit-Queue: Björn Terelius <[email protected]> Bot-Commit: [email protected] <[email protected]> Cr-Commit-Position: refs/heads/main@{#39146}

…ding"" This reverts commit fc5d627. Reason for revert: Breaks upstream WPT tests Original change's description: > Reland "Ensure RTCRtpSenders are always created with one encoding" > > This is a reland of commit b802369 > > Original change's description: > > Ensure RTCRtpSenders are always created with one encoding > > > > It is possible to have AddTransceiver calls with an empty array > > of encodings or AddTrack calls. In both cases, before negotiation, > > the sender's encodings array would be empty and it was not possible > > to update any value. > > > > Now, a default entry should be created in those cases, allowing to > > update the configuration before negotiation. > > > > Bug: webrtc:10567 > > Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994 > > Auto-Submit: Florent Castelli <[email protected]> > > Reviewed-by: Henrik Boström <[email protected]> > > Commit-Queue: Florent Castelli <[email protected]> > > Cr-Commit-Position: refs/heads/main@{#39126} > > Bug: webrtc:10567 > Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140 > Reviewed-by: Henrik Boström <[email protected]> > Reviewed-by: Harald Alvestrand <[email protected]> > Commit-Queue: Florent Castelli <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39145} Bug: webrtc:10567 Change-Id: If9b5adb5debb7c87a15662a8d0f232405a0e8136 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291221 Auto-Submit: Evan Shrubsole <[email protected]> Bot-Commit: [email protected] <[email protected]> Commit-Queue: Harald Alvestrand <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Cr-Commit-Position: refs/heads/main@{#39147}

Bug: webrtc:14833 Change-Id: I584524cca81e17ac91d581daab6030705ad68dac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291104 Reviewed-by: Erik Språng <[email protected]> Commit-Queue: Erik Språng <[email protected]> Auto-Submit: Per Kjellander <[email protected]> Commit-Queue: Per Kjellander <[email protected]> Cr-Commit-Position: refs/heads/main@{#39148}

Bug: webrtc:14653 Change-Id: I0c59604b218d0839a126c02914626b8ed2bee76c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291040 Commit-Queue: Fredrik Hernqvist <[email protected]> Reviewed-by: Henrik Boström <[email protected]> Cr-Commit-Position: refs/heads/main@{#39149}

This is a reland of commit b802369 Original change's description: > Ensure RTCRtpSenders are always created with one encoding > > It is possible to have AddTransceiver calls with an empty array > of encodings or AddTrack calls. In both cases, before negotiation, > the sender's encodings array would be empty and it was not possible > to update any value. > > Now, a default entry should be created in those cases, allowing to > update the configuration before negotiation. > > Bug: webrtc:10567 > Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994 > Auto-Submit: Florent Castelli <[email protected]> > Reviewed-by: Henrik Boström <[email protected]> > Commit-Queue: Florent Castelli <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39126} Bug: webrtc:10567 Change-Id: I558a95f7b587302b5e95f6ec26d1eb1fedf3dbed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291240 Reviewed-by: Harald Alvestrand <[email protected]> Commit-Queue: Florent Castelli <[email protected]> Cr-Commit-Position: refs/heads/main@{#39150}

Bug: webrtc:13757 Change-Id: I5244ed1148f628df9482f934fdfb509e511a9856 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291103 Commit-Queue: Danil Chapovalov <[email protected]> Reviewed-by: Emil Lundmark <[email protected]> Cr-Commit-Position: refs/heads/main@{#39151}

This should help to debug problems with DNS resolution of turn server name. E.g. DNS server returns fake IP address to block turn server. Similar change to stun_port.cc were done in https://webrtc-review.googlesource.com/c/src/+/215926 Bug: None Change-Id: If1cc410f4cd0f89620d3678aabf05a0f1b22a393 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290992 Reviewed-by: Harald Alvestrand <[email protected]> Commit-Queue: Yury Yarashevich <[email protected]> Cr-Commit-Position: refs/heads/main@{#39152}

Implement GetProcessResidentSizeBytes for Fuchsia. Bug: webrtc:14825 Change-Id: I64582ce0da72d3bb0fa61ff64799a1a165e1192f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290922 Commit-Queue: Sarah Pham <[email protected]> Reviewed-by: Mirko Bonadei <[email protected]> Cr-Commit-Position: refs/heads/main@{#39154}

Bug: None Change-Id: I7a24b900ad511df687693339516cc6d757f9145d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291173 Commit-Queue: [email protected] <[email protected]> Bot-Commit: [email protected] <[email protected]> Cr-Commit-Position: refs/heads/main@{#39156}

…cketReceiver::DeliverRtcp" This reverts commit 3e61f88. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <[email protected]> > > Commit-Queue: Per Kjellander <[email protected]> > > Reviewed-by: Erik Språng <[email protected]> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <[email protected]> > Auto-Submit: Per Kjellander <[email protected]> > Reviewed-by: Björn Terelius <[email protected]> > Commit-Queue: Björn Terelius <[email protected]> > Bot-Commit: [email protected] <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <[email protected]> Reviewed-by: Jakob Ivarsson‎ <[email protected]> Bot-Commit: [email protected] <[email protected]> Owners-Override: Per Kjellander <[email protected]> Cr-Commit-Position: refs/heads/main@{#39157}

Certificates being missing is a sign of a bug (e.g. webrtc:14844, to be fixed separately) which is why we have a DCHECK. But this DCHECK does not protect against accessing the invalid iterator if it is a release build. This CL makes that safe. Bug: chromium:1408392 Change-Id: I97a82786028e41c58ef8ef15002c3f959bbec7f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291109 Reviewed-by: Mirko Bonadei <[email protected]> Commit-Queue: Henrik Boström <[email protected]> Cr-Commit-Position: refs/heads/main@{#39159}

Creating a data channel or negotiating it can make the SCTP transport name go from nothing (empty string) to something. Inside the RTCStatsCollector this is relevant because which transports we have affect which certificates we should cache, so this is an instance of having to call ClearStatsCache(). The bug is that we don't. This CL fixes the bug. I tried to create unittests to cover this, but I was unable to reproduce the race in a testing environment (if I did it would have hit an RTC_DCHECK). Not ideal... but I hope we can land it anyway since the fix is trivial and clearing the cache in response to API calls is worst case harmless. Bug: webrtc:14844 Change-Id: Ia7174cde040839e5555237db6de285297120b123 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291112 Reviewed-by: Mirko Bonadei <[email protected]> Commit-Queue: Henrik Boström <[email protected]> Cr-Commit-Position: refs/heads/main@{#39160}

Bug: webrtc:14918 Change-Id: I4f521c8db7432739d09b05371879f4814ce2f59b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294288 Reviewed-by: Christoffer Jansson <[email protected]> Reviewed-by: Jeremy Leconte <[email protected]> Commit-Queue: Björn Terelius <[email protected]> Cr-Commit-Position: refs/heads/main@{#39354}

[email protected], [email protected] Bug: webrtc:14913 Change-Id: I1ea9d5bda798ea01fa9ec2a9b8d96cb50ccb9ec2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293920 Reviewed-by: Erik Språng <[email protected]> Reviewed-by: Henrik Boström <[email protected]> Commit-Queue: Evan Shrubsole <[email protected]> Cr-Commit-Position: refs/heads/main@{#39356}

This makes the max bitrate configured in RTP parameter to still be respected, instead of being overridden by defaults in the case that an encoder with untrusted QP is reconfigured (e.g. due to some adaptation). Bug: webrtc:14914 Change-Id: I2d3ff645c069b80ec2e36887e6ce0ecd09a7ecbf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293944 Reviewed-by: Philip Eliasson <[email protected]> Commit-Queue: Erik Språng <[email protected]> Cr-Commit-Position: refs/heads/main@{#39357}

…lback)." This is a reland of commit 8ad4924 See diff between latest Patch Set and PS1. Fixes include: - VideoStreamEncoder's call to bitrate_adjuster_->OnEncodedFrame() is updated to take stream index (spatial or simulcast index) instead of only looking at SpatialIndex(). - Migrate test-only helpers to use Spatial/SimulcastIndex correctly. The fixes are to migrate some test-only helpers that we had forgot to fix that are used by external tests. Original change's description: > Make SimulcastIndex() and SpatialIndex() distinct (remove fallback). > > This CL removes the fallback logic to return the other index when the > one requested has not been set. This means we can remove the codec gates > that was previously needed because SpatialIndex() had multiple meanings, > resolving the TODOs previously added in > https://webrtc-review.googlesource.com/c/src/+/293343. > > We have already migrated all known external dependencies from > SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL. > > PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY > > Bug: webrtc:14884 > Change-Id: I82787505ab10be151e5f64965b270c45465d63a9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740 > Reviewed-by: Ilya Nikolaevskiy <[email protected]> > Reviewed-by: Erik Språng <[email protected]> > Commit-Queue: Henrik Boström <[email protected]> > Reviewed-by: Philip Eliasson <[email protected]> > Reviewed-by: Evan Shrubsole <[email protected]> > Cr-Commit-Position: refs/heads/main@{#39343} Bug: webrtc:14884 Change-Id: Ib966924efca1a040dae881599f0789a7f2ab24a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294284 Reviewed-by: Philip Eliasson <[email protected]> Reviewed-by: Erik Språng <[email protected]> Reviewed-by: Evan Shrubsole <[email protected]> Commit-Queue: Henrik Boström <[email protected]> Cr-Commit-Position: refs/heads/main@{#39358}

This CL gives the ability to explicitly set the network labels in the network quality metrics reporter, so that reported metrics can be aligned with peer names in case "alice" and "bob" are not used as peer names. Bug: webrtc:14897 Change-Id: Ib66d4683af71ff4302696aff4dcb9932b47e4efd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293161 Reviewed-by: Harald Alvestrand <[email protected]> Reviewed-by: Artem Titov <[email protected]> Reviewed-by: Andrey Logvin <[email protected]> Reviewed-by: Mirko Bonadei <[email protected]> Commit-Queue: Artem Titov <[email protected]> Cr-Commit-Position: refs/heads/main@{#39360}

…creencast_portal.(cc|h) Bug: webrtc:14917 Change-Id: I40e8f011b7263675aab99c452cda8f89ad137cc5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294283 Auto-Submit: Björn Terelius <[email protected]> Reviewed-by: Alexander Cooper <[email protected]> Commit-Queue: Alexander Cooper <[email protected]> Cr-Commit-Position: refs/heads/main@{#39361}

verifying these are transferred to the individual m-lines. Also verify that mixed usage both at session level as well as media level is not allowed as described in https://www.rfc-editor.org/rfc/rfc5285#section-6 BUG=None Change-Id: Iade387817c9f31362d0a26c5f13a3012c72b51b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294360 Reviewed-by: Florent Castelli <[email protected]> Commit-Queue: Philipp Hancke <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Cr-Commit-Position: refs/heads/main@{#39363}

Bug: webrtc:14919 Change-Id: Iba00b2782b7e7c3dbd345a94aba541fad8c979ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294289 Commit-Queue: Jesus de Vicente Pena <[email protected]> Reviewed-by: Gustaf Ullberg <[email protected]> Cr-Commit-Position: refs/heads/main@{#39364}

This test verifies perkj's fixes in https://crbug.com/webrtc/14817. I ran the test 6000 times locally and it didn't fail once. Bug: webrtc:14817 Change-Id: I3f78f3ae2ca09b328cbfa12a89ad228d3de899c2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294522 Reviewed-by: Per Kjellander <[email protected]> Commit-Queue: Per Kjellander <[email protected]> Auto-Submit: Henrik Boström <[email protected]> Commit-Queue: Henrik Boström <[email protected]> Cr-Commit-Position: refs/heads/main@{#39365}

in favor of the variant with (or returning) a Timestamp object. BUG=webrtc:14813,webrtc:13756 Change-Id: I7b40f48f640a8be40a134b380a7a1b99cc99913b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294287 Reviewed-by: Henrik Boström <[email protected]> Reviewed-by: Danil Chapovalov <[email protected]> Commit-Queue: Philipp Hancke <[email protected]> Cr-Commit-Position: refs/heads/main@{#39366}

This class is used for both simulcast and SVC use cases. Update variable names and code comments to reflect this fact. Also add TODOs that we'll need to address for VP9 simulcast. Bug: webrtc:14884 Change-Id: I814c8fa0097306b16d552f55ca391ac8f716348a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294383 Commit-Queue: Henrik Boström <[email protected]> Reviewed-by: Ilya Nikolaevskiy <[email protected]> Reviewed-by: Erik Språng <[email protected]> Cr-Commit-Position: refs/heads/main@{#39367}

Bug: webrtc:11031 Change-Id: I44405d0d15e885307b3134b1b88dcb74b96381fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294400 Reviewed-by: Erik Språng <[email protected]> Commit-Queue: Danil Chapovalov <[email protected]> Cr-Commit-Position: refs/heads/main@{#39368}

Replace the existing variable capture_time_identifier_ms_ with capture_time_identifier_ in webrtc::VideoFrame and webrtc::VideoFrame::Builder. This variable uses webrtc::Timestamp as its type versus using int64_t which creates confusion about whether it is being recorded in milliseconds/microseconds. Change-Id: I0b83a6235fb1d5732f7afe2c14d7d6121f1e985b Bug: webrtc:14878 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293520 Reviewed-by: Harald Alvestrand <[email protected]> Commit-Queue: Palak Agarwal <[email protected]> Cr-Commit-Position: refs/heads/main@{#39369}

With LTR and SVC etc., H.264 should be able to skip lost frames, and continue to play from the new frames. With DependencyDescriptor, it is allowed to reference the previous frames, even there is a gap in the middle. However, we found there is a special logic for H.264 in packet_buffer.cc, which requires no gap for H.264. We should allow gaps if the packet has GenericDescriptorInfo (either GenericDescriptor or DependencyDescriptor header extension). Bug: webrtc:14887 Change-Id: Id66726bab33229bd883f257136ff2e8523fb44c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294062 Commit-Queue: Philip Eliasson <[email protected]> Reviewed-by: Philip Eliasson <[email protected]> Cr-Commit-Position: refs/heads/main@{#39370}

This uses the field trial introduced is crbug.com/1406331 and extends the usage to OpenH264. This simplifies experimentation whether this change improves performance without requiring multi-slice encoding. BUG=webrtc:14368 Change-Id: I0031e59059f7113dd5453234869c957d46f311bb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294340 Reviewed-by: Erik Språng <[email protected]> Reviewed-by: Sergey Silkin <[email protected]> Reviewed-by: Henrik Boström <[email protected]> Commit-Queue: Philipp Hancke <[email protected]> Cr-Commit-Position: refs/heads/main@{#39371}

Without the flag enabled: https://screenshot.googleplex.com/BZ6fqsNHQAUxbyU With the flag enabled: https://screenshot.googleplex.com/4GKrfvAdUpsQDx9 Bug: webrtc:12707 Change-Id: Ia1a9761aeaedc57cff6a2d1eca3c61519b9dd26e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293660 Commit-Queue: Diep Bui <[email protected]> Reviewed-by: Per Kjellander <[email protected]> Cr-Commit-Position: refs/heads/main@{#39372}

Bug: webrtc:13931, webrtc:14920 Change-Id: I451869e295e099a1d08c0c80e481decd53149f1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294382 Reviewed-by: Henrik Boström <[email protected]> Commit-Queue: Harald Alvestrand <[email protected]> Cr-Commit-Position: refs/heads/main@{#39373}

This CL propagates capture_time_identifier introduced in webrtc::VideoFrame and propagates it to EncodedImage. For use cases involving EncodedTransforms, this identifier is further propagated to TransformableVideoSenderFrame. VideoEncoder::Encode function is overriden by each encoder. Each of these overriden functions needs to be changed so that they can handle this new identifier and propagate its value in the created EncodedImage. Change-Id: I5bea4c5a3fe714f1198e497a4bcb5fd059afe516 Bug: webrtc:14878 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291800 Reviewed-by: Tony Herre <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Commit-Queue: Palak Agarwal <[email protected]> Cr-Commit-Position: refs/heads/main@{#39374}

Bug: webrtc:14708 Change-Id: I7926b3cfa6530e02eb13c31fecbc9e2e73f78f71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293744 Reviewed-by: Tove Petersson <[email protected]> Reviewed-by: Danil Chapovalov <[email protected]> Commit-Queue: Tony Herre <[email protected]> Cr-Commit-Position: refs/heads/main@{#39375}

The Xenial instrumented libraries are no longer used and can be removed. Bug: chromium:1260217 Change-Id: If3e7cb97f8e6d1b1b482e43885b215d5b199333e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294640 Reviewed-by: Mirko Bonadei <[email protected]> Auto-Submit: Thomas Anderson <[email protected]> Commit-Queue: Mirko Bonadei <[email protected]> Cr-Commit-Position: refs/heads/main@{#39376}

…back Since AddTrack now has an implicit init_encodings value, it will also have a StableState saved when associating a transceiver. That state may not have a saved mid and mline_index, and so on a rollback, it could blindly reset the mid and mline_index of an associated transceiver. This is wrong, the mid and mline_index of associated transceivers should only be updated when the StableState objects actually have one saved. (cherry picked from commit b3d424c) Bug: chromium:1424238 Change-Id: I8e80a04cd072d90200ca7643de892c0ef29b1f1a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297920 Commit-Queue: Florent Castelli <[email protected]> Reviewed-by: Henrik Boström <[email protected]> Cr-Original-Commit-Position: refs/heads/main@{#39577} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297983 Reviewed-by: Per Kjellander <[email protected]> Commit-Queue: Henrik Boström <[email protected]> Cr-Commit-Position: refs/branch-heads/5615@{#1} Cr-Branched-From: cdfeb4f-refs/heads/main@{#39376}

This reverts commit be03c09. Causes regression in web projects that 1/ add a stopped-by-default extension in SRD 2/ call createAnswer 3/ munge the stopped-by-default extension back in SLD 4/ create a subsequent offer and expect the extension to be present BUG=chromium:1427442,chromium:1051821 Change-Id: I2e48831e92384963a254d873398f54eaee96739a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299143 Reviewed-by: Harald Alvestrand <[email protected]> Commit-Queue: Philipp Hancke <[email protected]> Reviewed-by: Henrik Boström <[email protected]> Cr-Commit-Position: refs/branch-heads/5615@{#2} Cr-Branched-From: cdfeb4f-refs/heads/main@{#39376}

…to-m112 # Conflicts: # call/rtp_payload_params.cc # call/rtp_video_sender.cc # call/rtp_video_sender_unittest.cc # net/dcsctp/tx/outstanding_data_test.cc # pc/rtp_sender_receiver_unittest.cc # sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h # sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm # sdk/objc/native/src/audio/audio_device_ios.h # tools_webrtc/ios/build_ios_libs.py

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it was a very hard review 🤣

jleconte2 and others added 30 commits January 18, 2023 10:28

Björn Terelius and others added 27 commits February 21, 2023 09:24

Fix merge conflict

6f7cdda

x-0o0 self-requested a review May 9, 2023 02:09

x-0o0 approved these changes May 9, 2023

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mininny merged commit 685f7cb into develop May 9, 2023

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Update WebRTC to m112 #21

Update WebRTC to m112 #21

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Update WebRTC to m112 #21

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