1bcd827e9b
Merged FrameBuffer3 {Next,Last}DecodableTemporalUnitRtpTimestamp() function.
...
Bug: webrtc:13343
Change-Id: Ic21eddd38466e6b5fd8b912b3ee2d9dc47a0ba35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260981
Reviewed-by: Evan Shrubsole <eshr@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36756}
2022-05-04 11:29:57 +00:00
44be579b4a
Make all VideoReceiveStream2Test use simulated time
...
Adds matchers to webrtc::VideoFrame to help with the tests.
Bug: webrtc:14003
Change-Id: I62fc1c577bb76b21a96741ba829f6dcd53a308c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Evan Shrubsole <eshr@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36755}
2022-05-04 11:27:16 +00:00
49850c328c
Fork test/ios/google_test_runner.mm from Chromium's //base
...
WebRTC should not depend on Chromium's //base.
Bug: webrtc:13662
Change-Id: I7ce7036c96b26a79de46a519ed24dd660a298433
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260925
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36754}
2022-05-04 11:26:07 +00:00
b0eb25f01c
Roll chromium_revision 5d85a975a5..6effd442e8 (999193:999303)
...
Change log: 5d85a975a5..6effd442e85d85a975a5..6effd442e8e706c2f931..1eefedb688c812b8d227..d7c4eab611d1ce413bae..e0fba2731c42bb98d9f6..3ac36968b234049d3f74..dea6018ecahttps://chromium.googlesource.com/catapult.git/+log/d3c44146a6..05b7cd33df
* src/tools: f5ac564f9d..42973cc9775d85a975a5..6effd442e8https://webrtc-review.googlesource.com/c/src/+/261140
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2022-05-04 11:24:57 +00:00
55c55c3c43
Reorder mb_config.pyl to fix 'mb.py validate'
...
This is a follow up of this CL: https://crrev.com/c/3611486 .
Bug: b/231404336
Change-Id: I3b6f4b780aa1939b6b2d0f1d08d1746a3a6f86dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261080
Commit-Queue: Jeremy Leconte <jleconte@google.com >
Reviewed-by: Christoffer Jansson <jansson@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36752}
2022-05-04 08:31:29 +00:00
3466511ee1
Migrate iOS simulator tests to Mac-12 machines.
...
Bug: b/227442116
Change-Id: I7b5157a735596616f9785391c40b8f26974c222d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260985
Reviewed-by: Christoffer Jansson <jansson@webrtc.org >
Commit-Queue: Jeremy Leconte <jleconte@google.com >
Cr-Commit-Position: refs/heads/main@{#36751}
2022-05-04 07:27:25 +00:00
0dc2b7095f
Update WebRTC code version (2022-05-04T04:04:14).
...
Bug: None
Change-Id: Ibbd8db1953d8406f864679b45563c644f2ef3fdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261026
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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2022-05-04 05:16:37 +00:00
1cc01c3b18
Roll chromium_revision b59a320a3b..5d85a975a5 (999089:999193)
...
Change log: b59a320a3b..5d85a975a5b59a320a3b..5d85a975a515a28feff9..e706c2f931c5dc2c9629..c812b8d227bd8b86ed2b..d1ce413bae66e2a04447..42bb98d9f690343e1d57..34049d3f74fb8cf9cc78..13acea364532b2d1d526..e481fc655a3743fbdb22..f5ac564f9db59a320a3b..5d85a975a5https://webrtc-review.googlesource.com/c/src/+/261024
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Cr-Commit-Position: refs/heads/main@{#36749}
2022-05-04 02:29:08 +00:00
75dc112338
Roll chromium_revision cb4b2debf2..b59a320a3b (998895:999089)
...
Change log: cb4b2debf2..b59a320a3bcb4b2debf2..b59a320a3b31d89ad0a1..15a28feff9dae9260114..c5dc2c9629a7a192af80..bd8b86ed2b95047fa2c0..66e2a04447ec34c6fb1e..90343e1d57https://chromium.googlesource.com/catapult.git/+log/eebb0a0e63..d3c44146a6
* src/third_party/depot_tools: e06e307750..fb8cf9cc78f233f88ebb..f8b405298bce04993a19..3743fbdb22cb4b2debf2..b59a320a3bhttps://webrtc-review.googlesource.com/c/src/+/261022
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Cr-Commit-Position: refs/heads/main@{#36748}
2022-05-03 23:51:27 +00:00
b9f634497e
Roll chromium_revision d1899b8072..cb4b2debf2 (998754:998895)
...
Change log: d1899b8072..cb4b2debf2d1899b8072..cb4b2debf23767de1d65..31d89ad0a115a156bdb5..dae9260114c7888dd707..ce96725bddf274b4fe3c..a7a192af8056c3743b21..95047fa2c05b43f0d4da..ec34c6fb1e02f7958f65..f233f88ebb25ad985bf2..ce04993a19d1899b8072..cb4b2debf2https://webrtc-review.googlesource.com/c/src/+/261020
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Cr-Commit-Position: refs/heads/main@{#36747}
2022-05-03 16:53:06 +00:00
25adc8e36b
Eliminate channel.h from rtp_transmission_manager.cc
...
This also hides the existence of the classes VideoChannel and
VoiceChannel from anything that does not include "channel.h".
Bug: webrtc:13931
Change-Id: I080a692b6acfd5d2d0401ec20d59c3a684eddb05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260944
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36746}
2022-05-03 14:46:36 +00:00
00579e8bce
Use AlwaysValidPointer in connection_context
...
This extends AlwaysValidPointer to take a lambda for its default
rather than requesting a constructor.
Bug: none
Change-Id: Ied97968c3f511af15422a1eef9801d14d4ec5b96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260580
Reviewed-by: Jonas Oreland <jonaso@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36745}
2022-05-03 13:39:06 +00:00
9a743179cf
Eliminate need for stats to know of channel.h
...
Also eliminate FillBitrateInfo from the Channel object.
Bug: webrtc:13931
Change-Id: I5265b7629413a1ed04898272adf26708e2ee9b8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260469
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36744}
2022-05-03 12:06:56 +00:00
a429ad06bc
Roll chromium_revision 8adb72660d..d1899b8072 (998650:998754)
...
Change log: 8adb72660d..d1899b80728adb72660d..d1899b807228bb4c36ce..3767de1d65be173d9ec5..15a156bdb56b184c2967..f274b4fe3cd9eac53afa..56c3743b21e4157b9c7a..5b43f0d4dahttps://chromium.googlesource.com/catapult.git/+log/9ba02ee5ca..eebb0a0e63
* src/third_party/depot_tools: 5b13afcae3..e06e307750e5064e056f..25ad985bf28adb72660d..d1899b8072https://webrtc-review.googlesource.com/c/src/+/260911
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36743}
2022-05-03 08:53:47 +00:00
eeffb6aaf7
Remove deprecated audio option residual_echo_detector
...
The feature is now enabled in other ways. See PSA or linked Monorail issue for details.
https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
Bug: webrtc:11539
Change-Id: I0f5816baf2bfa1508a1c85ddbd7b775417434c62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260107
Auto-Submit: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org >
Commit-Queue: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36742}
2022-05-03 07:07:58 +00:00
3a9e6877e5
Cleanup rtc_base_approved target
...
Removed all unused or transitional parts
Bug: webrtc:9838
Change-Id: I6bc14e4e0ac09ab27e4d739ac76bb6a4dfff2cb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260140
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36741}
2022-05-03 06:58:27 +00:00
777c48b553
Update WebRTC code version (2022-05-03T04:02:12).
...
Bug: None
Change-Id: Ibd8c9420236e6132eeb39dd6a96503ad14e9a678
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260908
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36740}
2022-05-03 05:23:26 +00:00
c01a037105
Roll chromium_revision 423039e609..8adb72660d (998528:998650)
...
Change log: 423039e609..8adb72660d423039e609..8adb72660dd01af52469..28bb4c36ce56e01bffd1..be173d9ec55b971a05e7..6b184c296711573d49c6..d9eac53afac8d7344da7..e4157b9c7a5a642f93bc..e5064e056f423039e609..8adb72660dhttps://webrtc-review.googlesource.com/c/src/+/260906
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Cr-Commit-Position: refs/heads/main@{#36739}
2022-05-03 01:00:36 +00:00
8dc10ffba5
Roll chromium_revision 6ab3fd62f5..423039e609 (998390:998528)
...
Change log: 6ab3fd62f5..423039e6096ab3fd62f5..423039e6097d24fc96eb..d01af5246935387279e4..56e01bffd1867a173cf3..5b971a05e7e49c8a1245..11573d49c6cf748b0908..c8d7344da79a7b7675ba..5b13afcae3178dda6301..5a642f93bc6ab3fd62f5..423039e609https://webrtc-review.googlesource.com/c/src/+/260904
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36738}
2022-05-02 21:10:30 +00:00
b73c058702
Add new prioritized packet queue.
...
This queue is a more strict round robing queue, unlike the class
named RoundRobinPacketQueue. That is, we don't have the same logic to
prioritize lower-bitrate streams.
The queue time mechanism is essentially directly copied from the
previous implementation however.
Bug: webrtc:11340
Change-Id: Ie38ba8ce27c985f5f1e907cec068d6a365089bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260562
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36737}
2022-05-02 20:49:35 +00:00
869c87a2b9
Revert "Make deletion of Connection objects more deterministic."
...
This reverts commit 942cac2e9e6a205fd673dc003a051cfb3867f2e3.
Reason for revert: Reverting while downstream updates are made.
Original change's description:
> Make deletion of Connection objects more deterministic.
>
> This changes most deletion paths of Connection objects to go through
> the owner class of the Connection instances, Port.
>
> In situations where Connection objects still need to be deleted
> asynchronously, `async = true` can be passed to
> `Port::DestroyConnection` and get the same behavior as
> `Connection::Destroy` formerly gave.
>
> The `Destroy()` method still exists for downstream compatibility, but
> instead of deleting connection objects asynchronously, the deletion
> now happens synchronously via the Port class.
>
> Bug: webrtc:13892, webrtc:13865
> Change-Id: I07edb7bb5e5d93b33542581b4b09def548de9e12
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259826
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#36676}
Bug: webrtc:13892, webrtc:13865
Change-Id: I37a15692c8201716402ba5c10f249e4d3754ce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260862
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Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36736}
2022-05-02 20:41:39 +00:00
edeeef24a6
Revert "Clear port_ before firing destroyed event."
...
This reverts commit ca94696ae257395b452048535b3b86e95dbf47c4.
Reason for revert: Downstream needs updating.
Original change's description:
> Clear port_ before firing destroyed event.
>
> This reverts a change introduced last week in [1] whereby the port_
> pointer would be valid while firing the `Destroyed` event.
>
> [1] https://webrtc-review.googlesource.com/c/src/+/259826
>
> Bug: webrtc:13892, webrtc:13865
> Change-Id: I9c7be8fa9a5603fbdbf0debd91e2d4e21b303270
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260860
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#36728}
Bug: webrtc:13892, webrtc:13865
Change-Id: Ic7571fa8433897b348ca9a8f73ceae68014fe0c6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260921
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
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Cr-Commit-Position: refs/heads/main@{#36735}
2022-05-02 19:06:37 +00:00
0a16276290
Restore FiredDirection and maybe fire OnTrack in Rollback.
...
Prior to this CL, rollback did not restore FiredDirection and remote
streams were only sometimes restored. This resulted in not firing
ontrack if a track was rolled back and then added again on the same
transceiver.
Rollback also never performed OnTrack, which is incorrect because a
transceiver that goes from sendrecv to inactive will cause OnRemoveTrack
and if this is rolled back (so we become sendrecv again) then we need
OnTrack to fire.
This CL improves rollback's "memory", fires ontrack in Rollback() and
adds test coverage.
Needed to solve similar bugs in the Chromium layers as well:
https://chromium-review.googlesource.com/c/chromium/src/+/3613313
Bug: chromium:1320669
Change-Id: I655dd7d8a6b86080fe0e7c32c9e8c6434062ae91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260330
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36734}
2022-05-02 18:07:24 +00:00
548102642d
Roll chromium_revision b0579bf7df..6ab3fd62f5 (998259:998390)
...
Change log: b0579bf7df..6ab3fd62f5b0579bf7df..6ab3fd62f593f872454a..7d24fc96ebd8a47466e3..393e3eee99fe8021fccc..867a173cf3e04b1ba07e..e49c8a1245e83b9e5544..cf748b0908fccf35cb8e..9a7b7675ba6fb7b7a09d..e8ebfe988b22ebf0451e..178dda6301b0579bf7df..6ab3fd62f5https://webrtc-review.googlesource.com/c/src/+/260902
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Cr-Commit-Position: refs/heads/main@{#36733}
2022-05-02 16:39:23 +00:00
a16a6a6341
stats: implement inbound-rtp totalProcessingDelay for video
...
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
BUG=webrtc:13984
Change-Id: Ifd821bd8553add46218f09a11366096d62f5d09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259768
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36732}
2022-05-02 10:56:22 +00:00
a63b6b7d40
[PCLF] Allow configuring RtpEncodingParameters with singlecast
...
With the encoding parameters in the SimulcastConfig objects, it wasn't
possible to configure explicit encoding parameters when using singlecast,
required for example to use the spec standard SVC API.
Bug: webrtc:11607
Change-Id: I92b1446e772e2ecec93379dc91a3da159b8bc209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260002
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36731}
2022-05-02 10:30:14 +00:00
69c1df2f44
stats: add dtlsRole to transport
...
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-dtlsrole
BUG=webrtc:13978
Change-Id: Ib158427d2df0307884381bdd46c411f60f56a371
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259761
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/main@{#36730}
2022-05-02 10:13:54 +00:00
8c354882f8
Updated libaom AV1 encoder configuration.
...
New configuration parameters are:
AV1E_SET_DISABLE_TRELLIS_QUANT = 1
AV1E_SET_ENABLE_DIST_WTD_COMP = 0
AV1E_SET_ENABLE_DIFF_WTD_COMP = 0
AV1E_SET_ENABLE_DUAL_FILTER = 0
AV1E_SET_ENABLE_INTERINTRA_COMP = 0
AV1E_SET_ENABLE_INTERINTRA_WEDGE = 0
AV1E_SET_ENABLE_INTRA_EDGE_FILTER = 0
AV1E_SET_ENABLE_INTRABC = 0
AV1E_SET_ENABLE_MASKED_COMP = 0
AV1E_SET_ENABLE_PAETH_INTRA = 0
AV1E_SET_ENABLE_QM = 0
AV1E_SET_ENABLE_RECT_PARTITIONS = 0
AV1E_SET_ENABLE_RESTORATION = 0
AV1E_SET_ENABLE_SMOOTH_INTERINTRA = 0
AV1E_SET_ENABLE_TX64 = 0
AV1E_SET_MAX_REFERENCE_FRAMES = 3
Also added a SET_ENCODER_PARAM_OR_RETURN_ERROR convenience macro.
Bug: none
Change-Id: I7a683ec4ad36f33e13e669ba25db2ad81b9b5c86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260463
Reviewed-by: Marco Paniconi <marpan@google.com >
Reviewed-by: Marco Paniconi <marpan@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36729}
2022-05-02 10:12:52 +00:00
ca94696ae2
Clear port_ before firing destroyed event.
...
This reverts a change introduced last week in [1] whereby the port_
pointer would be valid while firing the `Destroyed` event.
[1] https://webrtc-review.googlesource.com/c/src/+/259826
Bug: webrtc:13892, webrtc:13865
Change-Id: I9c7be8fa9a5603fbdbf0debd91e2d4e21b303270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260860
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36728}
2022-05-02 09:56:42 +00:00
a4e9480279
Eliminate channel.h from rtc_stats_collector
...
This reduces the visibility of the implementation details
of cricket::ChannelInterface implementations.
Bug: webrtc:13931
Change-Id: Ia720a297821c1ddc242af2b04da4f52b1e04ab6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260560
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36727}
2022-05-02 09:28:34 +00:00
4c29ca654b
Remove mention of MSVC support from docs
...
Since MSVC bots has been removed, we can't say that MSVC is supported.
Bug: webrtc:13232
Change-Id: I39c6b8d9ef7af2aca6c6e5f2e5c44c9b1146145b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com >
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36726}
2022-05-02 09:26:32 +00:00
2c761b2212
Eliminate channel.h from peer_connection.cc
...
This limits the exposure of the implementation of ChannelInterface.
Bug: webrtc:13931
Change-Id: Ifc0fa496c210413d81ad71f44fa4040b881d092c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260561
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36725}
2022-05-02 09:04:32 +00:00
5e354d9969
dcsctp: Improve fast retransmission support
...
Before this CL, fast retransmission didn't follow the SHOULDs:
https://datatracker.ietf.org/doc/html/rfc4960#section-7.2.4
* "the sender SHOULD ignore the value of cwnd (...)"
* "(...) and SHOULD NOT delay retransmission for this single
packet."
With this CL, chunks that are eligible for fast retransmission (limited
to what can fit in a single packet) will be sent just after having
received the SACK that reported them missing and transitioned the socket
into fast recovery, and they will be sent even if the congestion window
is full.
Bug: webrtc:13969
Change-Id: I12c7e191a8ffd67973db7f083bad8a6061549fa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259866
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36724}
2022-05-02 08:29:52 +00:00
8fcc79b3d5
Add missing overload of LogSink::OnLogMessage()
...
Bug: webrtc:13579
Change-Id: If2ddff90f404361745257b0bed0951d9de0f08bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260470
Commit-Queue: Ali Tofigh <alito@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36723}
2022-05-02 08:15:05 +00:00
5df5b12ed3
Roll chromium_revision 62b50a78d7..b0579bf7df (998144:998259)
...
Change log: 62b50a78d7..b0579bf7df62b50a78d7..b0579bf7df86eacc9ca2..93f872454a06b5f68974..35387279e4405a1c108a..fe8021fccc3e4e4e96fb..e04b1ba07e1bfb6b5d04..e83b9e554462b50a78d7..b0579bf7dfhttps://webrtc-review.googlesource.com/c/src/+/260840
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36722}
2022-05-02 07:04:51 +00:00
a5fecb3917
dcsctp: Add proper fast retransmission support
...
This CL makes OutstandingData keep track of chunks that are eligible for
fast retransmission. When the socket goes into fast recovery, the
reported missing chunks can be retransmitted quickly (ignoring the
congestion window) according to
https://datatracker.ietf.org/doc/html/rfc4960#section-7.2.4 .
The CL also adds the new API to OutstandingData to retrieve only the
chunks that are eligible for fast retransmission, and moves the
remaining chunks to the ordinary list of chunks to be retransmitted
later.
This solves an issue where the retransmission timer wouldn't start if
there wouldn't be any chunks to fast-retransmit.
It doesn't, however, make sure that chunks that should be fast
retransmitted can send even when the congestion window is full. That
will be solved in the follow-up CL.
Bug: webrtc:13969
Change-Id: If4012d1cb284ef4a2d815683ed60cbbbff5b3c3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259865
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36721}
2022-05-02 07:03:42 +00:00
c706220cbd
Update WebRTC code version (2022-05-02T04:04:05).
...
Bug: None
Change-Id: Ia7ceabc10b15f378ced28b8a83ecbdd26fe31039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260821
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36720}
2022-05-02 04:57:13 +00:00
07999d2b6b
Roll chromium_revision bd9f2a5e9a..62b50a78d7 (998044:998144)
...
Change log: bd9f2a5e9a..62b50a78d7bd9f2a5e9a..62b50a78d706a6995dbf..06b5f68974e56fb2bb40..405a1c108aff930f4c1a..3e4e4e96fb2f85cdfe51..1bfb6b5d04https://chromium.googlesource.com/catapult.git/+log/ef89d1327c..9ba02ee5ca
* src/third_party/icu: 5fb93cb43c..85814e1af54d9708c007..22ebf0451ebd9f2a5e9a..62b50a78d7bd9f2a5e9a..62b50a78d7https://webrtc-review.googlesource.com/c/src/+/260662
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36719}
2022-05-01 06:53:41 +00:00
3af79d1768
Move ownership of the Channel class to RTCRtpTransceiver
...
This makes the channel manager object into a factory, not a manager.
Bug: webrtc:13931
Change-Id: I59f7d818a739797a7c0a7a32e6583450834df122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260467
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36718}
2022-04-30 19:21:11 +00:00
249382e79d
Update WebRTC code version (2022-04-30T04:02:31).
...
Bug: None
Change-Id: I779009d5e864e6cca54f7ae060c05161870d8091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260542
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36717}
2022-04-30 05:51:30 +00:00
64951c0f42
Roll chromium_revision 77c4ffe161..bd9f2a5e9a (997940:998044)
...
Change log: 77c4ffe161..bd9f2a5e9a77c4ffe161..bd9f2a5e9a0963de918a..86eacc9ca253c76d1464..06a6995dbf0d43d42517..e56fb2bb40b047aae9aa..ff930f4c1a9ecf61e269..2f85cdfe518a87603683..fccf35cb8ed833206dc8..4d9708c00777c4ffe161..bd9f2a5e9ahttps://webrtc-review.googlesource.com/c/src/+/260543
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36716}
2022-04-30 04:52:10 +00:00
a156b90ea0
Roll chromium_revision 2c71305f8d..77c4ffe161 (997792:997940)
...
Change log: 2c71305f8d..77c4ffe1612c71305f8d..77c4ffe1611936293b59..0963de918ac393df7e73..53c76d1464c055932162..c7888dd70778c796fee8..0d43d425174a2f544bfa..b047aae9aa98d31af5e3..9ecf61e269b384c7aa6b..02f7958f65fc36b66b31..d833206dc82c71305f8d..77c4ffe161https://webrtc-review.googlesource.com/c/src/+/260501
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36715}
2022-04-29 22:59:30 +00:00
d7de6e5ea4
Fix rendering tests of RTCMTLVideoViewTests to work as expected
...
In https://webrtc-review.googlesource.com/c/src/+/244420 ,
I added sanity check of RTCVideoFrame in RTCMTLVideoView, but I forgot
to modify the related tests.
Fix this by adding the appropriate property stubs to RTCVideoFrame stubs
created in RTCMTLVideoViewTests.
Bug: webrtc:13990, webrtc:13490
Change-Id: I21f0f75cd052e4255e1eee5f39ffdd50c2f8e61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260420
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org >
Reviewed-by: Jeremy Leconte <jleconte@google.com >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com >
Cr-Commit-Position: refs/heads/main@{#36714}
2022-04-29 19:27:22 +00:00
e8aa4155a3
Roll chromium_revision 6d7878cb33..2c71305f8d (997007:997792)
...
Change log: 6d7878cb33..2c71305f8d6d7878cb33..2c71305f8d8261266302..1936293b5987b04ad665..c393df7e73f0d740e4e2..113378f9b3a53022fa7e..c05593216243a7a256a0..d8a47466e3ac2b722342..78c796fee8b926234b2f..4a2f544bfa9bef587db2..98d31af5e3381db68adc..8a87603683https://aomedia.googlesource.com/aom.git/+log/e24a83a72b..7977bb492a
* src/third_party/perfetto: 2f647cd610..b384c7aa6b8ca1cca79e..fc36b66b316d7878cb33..2c71305f8d6d7878cb33..2c71305f8dhttps://webrtc-review.googlesource.com/c/src/+/260500
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36713}
2022-04-29 18:56:52 +00:00
0bb7cbc278
Lower av1 test psnr threshold
...
AV1 Realtime encoder stats changed
Bug: None
Change-Id: I50a8d36c45a775b3c0127476fb32c3d68d288508
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Marco Paniconi <marpan@google.com >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Jerome Jiang <jianj@google.com >
Cr-Commit-Position: refs/heads/main@{#36712}
2022-04-29 16:12:32 +00:00
13f9c62ec8
Fix comparator bugs which are not compliant to strict weak ordering.
...
See a full explanation of the problem on this blog [1] post about changing
std::sort in LLVM and relative issues uncovered.
The CompareNetwork function was violating the 4th rule of "strict weak
ordering" (Transitivity of incomparability: x == y and y == z imply x == z, where x == y means x < y and y < x are both false).
[1] - https://danlark.org/2022/04/20/changing-stdsort-at-googles-scale-and-beyond/
Bug: None
Change-Id: I7e893f0a30da31403766284823f75c45c4db91c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251681
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36711}
2022-04-29 16:00:53 +00:00
d3890781be
Adopt absl::string_view in some fakes under rtc_base/
...
Bug: webrtc:13579
Change-Id: I52462a0edb4fb309cea3bc27f60dc81c5ea50522
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260464
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Ali Tofigh <alito@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36710}
2022-04-29 15:33:12 +00:00
daee870a35
Remove ability to do SetChannel() without ClearChannel()
...
This calls out the fact that SetChannel() is only used on M-section activation; ClearChannel is called on deactivation, and we never change the channel while a transceiver is active.
Bug: webrtc:13931
Change-Id: I3a3bfeec7c1d27d98c3f94a9401bee2130754ed7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260461
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36709}
2022-04-29 14:28:02 +00:00
d4fce5a361
Use playout sample rate for audio unit.
...
Fixing a race condition where session.sampleRate changes before AudioDeviceIOS::HandleValidRouteChange() finishes.
session.sampleRate is read into session_sample_rate at 576 and used at 623 to initialize the audio unit. However, in the call to SetupAudioBuffersForActiveAudioSession() the session.sampleRate is read again and may have changed, resulting in different sample rates used for the buffers and the audio unit. The consequence is a sample rate mismatch with either high pitched or low pitched audio.
The fix is to always use the buffer sample rate for the audio unit.
The DCHECK at 622 would save us in debug, but not in production, hence removed.
Change-Id: I562f1bf7f94d7447139ada2636b02ade7fcd6371
Bug: webrtc:14011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260329
Reviewed-by: Henrik Andreasson <henrika@google.com >
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36708}
2022-04-29 14:07:02 +00:00
8ec4a2eca3
dcsctp: Refactor chunk lifecycle state flag
...
This CL replaces two booleans, that could never be active at the same
time (there is no such thing as an abandoned chunk that is scheduled
for retransmission), with a single enum.
Just for increased readability, and to understand that there is no such
thing as an abandoned chunk that will be retransmitted.
Bug: None
Change-Id: I1682c383aed692db07fd4ae1f84c0166db86c062
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259864
Reviewed-by: Florent Castelli <orphis@webrtc.org >
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36707}
2022-04-29 13:38:41 +00:00
