910446f945
Roll chromium_revision 134690e638..31213dddfe (1006082:1006183)
...
Change log: 134690e638..31213dddfe134690e638..31213dddfe775ec9d2f3..e6f782f39144ff734808..b8694ed10678ffa10dff..e44b1a0d89e982f3e038..d369e411e6b36fb6563f..e2d2260e040e9a7d2643..2f988479fb9f7cf67e7b..cdbe4e9aae134690e638..31213dddfehttps://webrtc-review.googlesource.com/c/src/+/263041
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36956}
2022-05-22 00:44:41 +00:00
1def899931
Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling
...
Bug: none
Change-Id: I14164546950cc63c37e54544cdc80bfd4eddf211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262962
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36955}
2022-05-21 23:06:21 +00:00
e508ebc645
Update WebRTC code version (2022-05-21T04:04:39).
...
Bug: None
Change-Id: I072c54f3afd47f1cb3320dab9150a48dbc2091b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263021
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@{#36954}
2022-05-21 05:29:51 +00:00
3f6c634394
Roll chromium_revision 2ca99fd94b..134690e638 (1005981:1006082)
...
Change log: 2ca99fd94b..134690e6382ca99fd94b..134690e6383edc3dbedd..775ec9d2f35807be4980..44ff734808c9b2288d0c..d03f56b8fa1f5ee59496..e982f3e03890f2ca5ebd..b36fb6563fhttps://chromium.googlesource.com/catapult.git/+log/811104909b..a1cf7a299e
* src/third_party/depot_tools: cf9a776713..0e9a7d26431c67b4e7d3..4c1a86872547f77dd3a5..9f7cf67e7b2ca99fd94b..134690e6382ca99fd94b..134690e638https://webrtc-review.googlesource.com/c/src/+/263000
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36953}
2022-05-21 03:00:51 +00:00
fc412ed853
Roll chromium_revision 97155e4126..2ca99fd94b (1005869:1005981)
...
Change log: 97155e4126..2ca99fd94b97155e4126..2ca99fd94b33fc67acc9..3edc3dbedd3e4d3831b8..4ad92ece128f163ea16e..78ffa10dffc11866ec3f..1f5ee594961d34e64bd0..90f2ca5ebd5deea0a6bc..47f77dd3a597155e4126..2ca99fd94bhttps://webrtc-review.googlesource.com/c/src/+/262922
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36952}
2022-05-20 22:28:51 +00:00
213d389d56
Roll chromium_revision 04a304ade5..97155e4126 (1005734:1005869)
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Change log: 04a304ade5..97155e412604a304ade5..97155e4126267dcac3ed..33fc67acc923d59a15cb..5807be498032d46891d4..8f163ea16e083840820b..c11866ec3fd18b96b235..1d34e64bd0e481fc655a..abfc2628f206f2aed74d..5deea0a6bc04a304ade5..97155e4126https://webrtc-review.googlesource.com/c/src/+/262908
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36951}
2022-05-20 18:40:32 +00:00
83830f316e
Delete TestListener and top-level thread wrapping.
...
Instead use rtc::AutoThread in tests that need that.
Bug: webrtc:9714
Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36950}
2022-05-20 15:21:21 +00:00
be2fb41b32
Delete codec-specific frameDroppingOn settings.
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Followup to https://webrtc-review.googlesource.com/c/src/+/262244
Bug: webrtc:6883
Change-Id: Iefac43709f14424c74470aa878ec512b7dacc68a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262258
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36949}
2022-05-20 14:54:31 +00:00
2ee3e4db22
Use resolution bitrate limits if one spatial layer is configured via scalability mode.
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Bug: webrtc:13960
Change-Id: Ie9238f3352b0d9d92fda97a250de0792e6bbfc9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261721
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36948}
2022-05-20 13:25:51 +00:00
6fb674ea5a
Rename MediaReceiveStream to MediaReceiveStreamInterface
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Bug: webrtc:7484
Change-Id: I0bc4bc57e8c4450c503ae4d5a41f9bbe243b00e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262768
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36947}
2022-05-20 13:17:52 +00:00
c0a9f35248
Define SimulcastStream as an alias for SpatialLayer
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Step one in making it a separate type, that will be done as a
followup, after downstream code is updated to use the new name.
Bug: webrtc:11607
Change-Id: I6fa664a0729b1cfd71b7f02b6441880beee0e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262806
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36946}
2022-05-20 13:12:21 +00:00
1c633c8b87
Roll chromium_revision d198633052..04a304ade5 (1005612:1005734)
...
Change log: d198633052..04a304ade5d198633052..04a304ade503a9b994b7..267dcac3ed399520d50a..23d59a15cb2b5dc23af3..32d46891d4900b72a5b4..083840820bf3ee75faa3..d18b96b235bd80a1be23..cf9a776713a742a5ba95..06f2aed74dd198633052..04a304ade5https://webrtc-review.googlesource.com/c/src/+/262906
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36945}
2022-05-20 13:08:21 +00:00
9e6ebfe59c
Remove AcmReceiverBitExactnessOldApi tests.
...
AcmReceiver basically only does resampling, which is not something we need to test for bit-exactness.
NetEq bit-exactness is already tested with the same rtp input file as these tests.
Bug: None
Change-Id: Ibb3936c86098e0eea944860d33e2c13bf046e40b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262816
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36944}
2022-05-20 13:05:01 +00:00
afe1680609
Add cluster rules, fix bug instance
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Bug: b/227310186
Change-Id: I39fa3204f0dd6b514dc468b29faa819c644486a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262942
Commit-Queue: Christoffer Jansson <jansson@webrtc.org >
Reviewed-by: Jeremy Leconte <jleconte@google.com >
Cr-Commit-Position: refs/heads/main@{#36943}
2022-05-20 12:00:46 +00:00
f8c81ca469
Revert "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate."
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This reverts commit 256733c78af655029cb04afae2c404afb41ea685.
Reason for revert: <breaks downstream>
Original change's description:
> Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate.
>
> Bug: webrtc:13573
> Change-Id: Iae649e7922a67f70c53d33f3b87ea62bb6a3490c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262812
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Commit-Queue: Philip Eliasson <philipel@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#36940}
Bug: webrtc:13573
Change-Id: I3341b6b96a56de63058c38943611b8c1629294ce
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262941
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36942}
2022-05-20 11:45:47 +00:00
ae2f39ca3b
Refactor ResolutionBitrateLimitsTest
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Use the DataRate type, and designated initializers to construct expectations.
Bug: None
Change-Id: I9f3a64faf7afffb1c2efebeda84f3ef6d4e71dee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262940
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36941}
2022-05-20 11:26:37 +00:00
256733c78a
Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate.
...
Bug: webrtc:13573
Change-Id: Iae649e7922a67f70c53d33f3b87ea62bb6a3490c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262812
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36940}
2022-05-20 11:14:36 +00:00
5b8dc1dbad
Add RTC_EXPORT to CurrentTaskQueueSetter
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This allows for its use in test targets in Chromium Windows, which fixes
the compiled errors found in https://chromium-review.googlesource.com/c/chromium/src/+/3649679
Change-Id: I738b2eaab8eca73c40e847ede67ff5e7757ec512
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262811
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36939}
2022-05-20 10:38:20 +00:00
fa6ec69232
Restructure RtcpTransceiver tests to use fake time instead of real time
...
to make tests faster and more determenistic.
Bug: webrtc:8239
Change-Id: I18067251a1f1a349fda28bbfbb59bce333bfddca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201737
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36938}
2022-05-20 07:52:29 +00:00
f2c710852e
Update WebRTC code version (2022-05-20T04:02:10).
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Bug: None
Change-Id: I7e157362c58193fa64316f7bbe02da6ad94f1f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262902
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@{#36937}
2022-05-20 05:23:07 +00:00
0a7f045934
Roll chromium_revision 7d47e8b461..d198633052 (1005487:1005612)
...
Change log: 7d47e8b461..d1986330527d47e8b461..d198633052b72c83ea2b..03a9b994b75b375fc783..399520d50a0d5a6e1f01..2b5dc23af3d456033f79..900b72a5b4961b157acf..f3ee75faa3a99c70ac33..a742a5ba957d47e8b461..d198633052https://webrtc-review.googlesource.com/c/src/+/262920
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36936}
2022-05-20 04:51:47 +00:00
f4fcdf1ee1
Roll chromium_revision f6c3a5b15b..7d47e8b461 (1005363:1005487)
...
Change log: f6c3a5b15b..7d47e8b461f6c3a5b15b..7d47e8b46136820df997..b72c83ea2bced2ca7c67..5b375fc78375a3853b9d..3e4d3831b8837a94efb7..c9b2288d0ce99a25fe6e..0d5a6e1f0179650df435..d456033f79b193687bd4..961b157acf632bbc0cb1..bd80a1be236ebcd58ed9..370def59dc4ebf4a3e6e..a99c70ac33f6c3a5b15b..7d47e8b461https://webrtc-review.googlesource.com/c/src/+/262900
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Cr-Commit-Position: refs/heads/main@{#36935}
2022-05-20 01:01:26 +00:00
d4ad96e578
Roll chromium_revision 524559feaa..f6c3a5b15b (1004707:1005363)
...
Change log: 524559feaa..f6c3a5b15b524559feaa..f6c3a5b15b48dbbda0ac..36820df99725a9e63316..ced2ca7c677208eddba1..a5fa4652f6b682786415..75a3853b9d44c86bbb7c..837a94efb70f7e70fedb..e99a25fe6ec5ec396eb3..79650df4357e38141447..b193687bd48fb649c425..632bbc0cb15d49473f85..a4c4566b6d585942f33d..1c67b4e7d3ec1d6d0d51..6ebcd58ed9d20a3fff87..4ebf4a3e6e524559feaa..f6c3a5b15bhttps://webrtc-review.googlesource.com/c/src/+/262880
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36934}
2022-05-19 20:43:06 +00:00
ab8c1e50a3
Manual roll of linux64 chromium build tools
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Bug: b/233227148
Change-Id: I69b80696af9f99fca8803d72250b571909d2456d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262815
Auto-Submit: Andrey Logvin <landrey@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36933}
2022-05-19 19:17:07 +00:00
65b2d8ad21
Move RunLoop test class to its own build target
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To make it usable in tests without depending on all of CallTest.
Bug: None
Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36932}
2022-05-19 15:51:39 +00:00
8139cda2ec
Enable encoder frame drop for call tests and video quality tests
...
It was accidentally disabled by
https://webrtc-review.googlesource.com/c/src/+/262244 , resulting in
lots of unintended changes in performance tests.
Bug: webrtc:6883, webrtc:14075
Change-Id: Ie414f729ec2248f0eef99922e9704f4c4a0b1aa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262813
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36931}
2022-05-19 14:55:46 +00:00
6c81a42eb1
Simulate generic dependency structure for VP8
...
This will be used as a fall-back when the encoder adapter doesn't
provide any dependency structure. This ensures we can always generate a
dependency descriptor RTP header extension for VP8.
Before, when switching between encoder adapters where the old one
generated a dependency structure but the new one didn't we had to make
sure the structure was cleared so that packets weren't sent with the
dependency structure from the previous adapter. This will not be a
problem anymore since the new adapter will use the simulated dependency
structure.
Bug: b/227749056
Change-Id: I8463c48a9dcde4b8d32c519819dd8a92acd8e43b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262765
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36930}
2022-05-19 11:53:08 +00:00
b1ba85385e
Eliminate unnecessary RTC_TRACE_EVENTS_ENABLED
...
Bug: webrtc:14073
Change-Id: I6365cc17393be52c11312dfa954783a3e135cb8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262263
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Johannes Kron <kron@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36929}
2022-05-19 09:52:47 +00:00
d68a06aedf
Move pinpoint acl to the perf bucket.
...
The perf builders have been moved from the 'ci' to the 'perf' bucket.
Bug: b/233159334
Change-Id: Ic5de4489892599d3a9cf94696a4db8a708c1f0e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262808
Commit-Queue: Jeremy Leconte <jleconte@google.com >
Reviewed-by: Christoffer Jansson <jansson@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36928}
2022-05-19 07:22:00 +00:00
42cf83cc61
Update WebRTC code version (2022-05-19T04:04:47).
...
Bug: None
Change-Id: I8eff6cd60b9416f81b32a1a51ca56286fedb7919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262736
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@{#36927}
2022-05-19 05:49:10 +00:00
cf2c8915f4
Delete H264EncoderSpecificSettings
...
Production code always use the default settings.
Bug: webrtc:6883
Change-Id: I213fc6433bb1cd0a6623ad523fee2df1506588e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261903
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36926}
2022-05-18 13:53:20 +00:00
34fc3f406b
Roll chromium_revision 99a40fa7f7..524559feaa (1003838:1004707)
...
Change log: 99a40fa7f7..524559feaa99a40fa7f7..524559feaa8db0ea4764..48dbbda0ac4b864c708c..25a9e63316e112e5ec5b..b682786415cf88e03ee0..44c86bbb7cd534abaffc..0f7e70fedb060c3c7ca1..c5ec396eb3f4a34bfe66..7e38141447https://chromium.googlesource.com/catapult.git/+log/25ac622b73..811104909b
* src/third_party/depot_tools: f8300fba9d..8fb649c425ee2b75e337..d1338d8751https://aomedia.googlesource.com/aom.git/+log/a1d676b44f..ed81a4095c
* src/third_party/libvpx/source/libvpx: cb1abee145..ca89bed50dc585628f45..ec1d6d0d51442185b165..d20a3fff8799a40fa7f7..524559feaahttps://webrtc-review.googlesource.com/c/src/+/262860
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36925}
2022-05-18 12:43:10 +00:00
304b78d3d9
Delete a few unused methods on DecoderDatabase
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Bug: none
Change-Id: Ic0a20036b92e0f1d088bae88724a777eca93760d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262763
Reviewed-by: Minyue Li <minyue@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36924}
2022-05-18 11:22:21 +00:00
0ac50b9dfd
Move ownership of objects from channel_manager to connection_context
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This is a preparatory step in deleting the ChannelManager class.
Also delete some declarations whose implementation was previously removed.
Bug: webrtc:13931
Change-Id: I8764c00fa696932e79fcfe17550ef2490d6a1ed1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262804
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36923}
2022-05-18 09:17:24 +00:00
0359ba2225
stats: add frame assembly time stats
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implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36922}
2022-05-18 09:16:10 +00:00
94c09169a2
Improve RTC event log parser.
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(Support parsing TWCC which nack sequnce numbers greater than the last
one received. Don't silently drop unrecognized/malformed RTCP packets.)
Bug: webrtc:14078
Change-Id: I34a0deabfdb5f36b988919cfcc9159197435756c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262800
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36921}
2022-05-18 09:14:40 +00:00
2f3168ff38
peerconnection: reject content if there are no common media codecs
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for video dealing with both the case where there is no common media
codec as well as only a red/ulpfec/flexfec codec in common for video
and only RED/CN in common for audio
BUG=webrtc:4957,webrtc:14069
Change-Id: I1c888b4f77199aade8122051c31b690dc2fd5925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262642
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36920}
2022-05-18 09:00:00 +00:00
cde992ddad
Add support for VP9 configuration through scalability mode.
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Bug: webrtc:13960
Change-Id: Ia930647b15f624a4d10d8d335519b69ffdae6636
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260983
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36919}
2022-05-18 08:21:00 +00:00
15a73be295
Update apprtc example code to support macos 10.13
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It currently blocks chromium to webrtc roll after https://crrev.com/c/3646050
Bug: b/232899082
Change-Id: I87e178d91037d844fc98a28491bdfe69fa3bc9b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262802
Auto-Submit: Andrey Logvin <landrey@webrtc.org >
Commit-Queue: Kári Helgason <kthelgason@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36918}
2022-05-18 07:44:20 +00:00
0601db9a48
Rename ReceiveStream to ReceiveStreamInterface
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Bug: webrtc:7484
Change-Id: I41176a66b8399f6c8cf568630f2808eb95cf6247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262767
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36917}
2022-05-18 07:26:50 +00:00
f92a0846c3
Fix build in C++20 mode.
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Structs with user-declared constructors are not aggregates and cannot
be initialized with designated initializers. Remove declarations that
don't actually affect anything.
Bug: chromium:1284275
Change-Id: Ib45ea334d7be28bfa7bbce132985612f0e6ecd10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262820
Auto-Submit: Peter Kasting <pkasting@chromium.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36916}
2022-05-18 06:47:30 +00:00
79dc0a223e
Update WebRTC code version (2022-05-18T04:01:44).
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Bug: None
Change-Id: Ib10b8b8236ab5905ed6c524d058a39a356eebebd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262732
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36915}
2022-05-18 05:16:30 +00:00
c5aa5bea49
Delete deprecated VP8Encoder::Create function
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Bug: None
Change-Id: I42e5d846d3d012f8b85eede743202b84239ceef1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262764
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36914}
2022-05-17 13:23:24 +00:00
a9f10c8189
Make fake decode from file produce 10 ms comfort noise frames.
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This is to more accurately simulate Opus CNG.
Bug: None
Change-Id: I3244d88e1f7410190551b6fa24cdd08599b5771e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262661
Reviewed-by: Minyue Li <minyue@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36913}
2022-05-17 13:11:34 +00:00
4da317f0bb
Remove complexity parameter from video codec specific structs.
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Now only using the complexity from the main VideoCodec settings.
Bug: webrtc:13694
Change-Id: I5a29df0fac0c0686bf5ea0b677f8946d23ef9888
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262762
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36912}
2022-05-17 12:45:04 +00:00
88af20356f
Use ProbeClusterConfig in BitrateProber from GoogCC
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Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.
To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>
Deprecates field trial "WebRTC-Bwe-ProbingConfiguration"
Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36911}
2022-05-17 12:29:25 +00:00
641a1b11b6
Adopt absl::string_view in call/
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Bug: webrtc:13579
Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Ali Tofigh <alito@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36910}
2022-05-17 12:00:45 +00:00
d01e692e01
Fix flakiness of CheckLogLevelUpdateStatsEmpty.
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Bug: b/232719223
Change-Id: I26bdb10bb7bae89f8a99b8c4db14609ae9bfda50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262620
Reviewed-by: Hanna Silen <silen@webrtc.org >
Commit-Queue: Jeremy Leconte <jleconte@google.com >
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36909}
2022-05-17 11:57:35 +00:00
39bc030e0b
Delete deprecated functions in RempoteNtpTimeEstimator
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Bug: webrtc:13757
Change-Id: Ie9cdb41de956df1b6877c8b86fd1829fdcd35043
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262761
Reviewed-by: Erik Språng <sprang@webrtc.org >
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36908}
2022-05-17 11:12:25 +00:00
1331c1821c
Reland: Update local_ssrc without needing to recreate video streams.
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This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042
This is a reland of commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca
with an additional fix in Patchset 2. Another problem turned out to be
in RTCPReceiver, which is fixed in:
https://webrtc-review.googlesource.com/c/src/+/262663
Bug: webrtc:11993
Change-Id: I63c7cf62a6dd50f88b491fea3ba866697552ef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262665
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36907}
2022-05-17 10:59:54 +00:00
