95c30413db
Update WebRTC code version (2021-06-18T04:03:27).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: Icd2faaa16bee1217af5c8e22db125aff1efb5f7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223000
Reviewed-by: 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/master@{#34328}
2021-06-18 06:30:14 +00:00
42dacda82c
AGC analog clipping predictor: integrate evaluator
...
Integrate ClippingPredictorEvaluator in AgcManagerDirect adding the
possibility to run the predictor without affecting the analog gain
adjustment process.
The evaluator is used to compute precision, recall and F1 score.
F1 score and the measured clipping prediction intervals are logged as
`WebRTC.Audio.Agc.ClippingPredictor.F1Score` and `.PredictionInterval`
histograms respectively.
Bug: webrtc:12774
Change-Id: I708dcda9321f92d5bd17ec4c36ebce1165ead57f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221921
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Hanna Silen <silen@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34327}
2021-06-17 16:16:53 +00:00
7d5418233d
Avoid assembling complicated but unused video rtp header extensions
...
Bug: chromium:1219407
Change-Id: I017de10813a1e80f4af0ba55d8d1aa73077dd131
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222615
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34326}
2021-06-17 16:09:13 +00:00
afb28116d3
Catch possible RuntimeException from getCameraCharacteristics
...
Also changed the logging of exceptions to give more details
Bug: webrtc:10804
Change-Id: Ifba6dee3d1c8ba4ecab408ca7715c3b792d9c004
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222641
Commit-Queue: Xavier Lepaul <xalep@webrtc.org >
Reviewed-by: Paulina Hensman <phensman@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34325}
2021-06-17 15:32:42 +00:00
11b92cfa65
Refactoring: Move groups-by-mid into Bundle manager
...
Bug: webrtc:12837
Change-Id: I2431dbbf8cc291b5f3848d81cf290fd3e97ec15d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222614
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34324}
2021-06-17 14:35:33 +00:00
de22ab2850
Apply IWYU to jsep_transport_controller/collection
...
Bug: None
Change-Id: I7b584099fcc26d7d74f2e08a0939a89a9196cd00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222612
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34323}
2021-06-17 14:01:11 +00:00
d354ced5ac
Mark VideoSendTiming flags as invalid by default.
...
Bug: none
Change-Id: I962df8a55c022193cb3ec036c3cf35f34f9b2412
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222611
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34322}
2021-06-17 12:39:34 +00:00
ada810aab2
Reland "Deprecate microsecond timestamps in RTC event log."
...
This is a reland of e6ee8fab7eac915b2b6abc9b71b6d33ad086f3d1
Original change's description:
> Deprecate microsecond timestamps in RTC event log.
>
> (Microsecond timestamps are only used in the legacy wire-format,
> and the clocks only have microsecond resolution on some platforms.)
>
> Also convert structs on the parsing side to use a Timestamp instead
> of a uint64_t to represent the log time.
>
> Bug: webrtc:11933
> Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
> Commit-Queue: Björn Terelius <terelius@webrtc.org >
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org >
> Reviewed-by: Sebastian Jansson <srte@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34097}
Bug: webrtc:11933
Change-Id: I295be966ee96b50719ceb4690dad7e7ce958dbac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221361
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34321}
2021-06-17 12:08:54 +00:00
1bb36d2c77
Change YuvConverter.convert to catch GLExceptions and return null.
...
With https://webrtc-review.googlesource.com/c/src/+/222582 ,
I420 conversion is allowed to fail.
Bug: webrtc:12877
Change-Id: Iadae21ad889f084b8027206af4478223d7733d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222653
Reviewed-by: Xavier Lepaul <xalep@webrtc.org >
Commit-Queue: Fabian Bergmark <fabianbergmark@google.com >
Cr-Commit-Position: refs/heads/master@{#34320}
2021-06-17 11:33:54 +00:00
ac82bd386a
Add timestamp to log message in generic_decoder.cc
...
Bug: None
Change-Id: Ib558247d887aff880853ef824f8d80d8e7e4feee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222610
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34319}
2021-06-17 10:14:14 +00:00
41c700dbd1
Remove unnused build configs for M1 builder
...
Bug: webrtc:12882
Change-Id: I3aa95d2305bee28c3ea5333b641ac1657a87e0ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222643
Reviewed-by: Tommi <tommi@webrtc.org >
Reviewed-by: Andrey Logvin <landrey@webrtc.org >
Commit-Queue: Christoffer Jansson <jansson@google.com >
Cr-Commit-Position: refs/heads/master@{#34318}
2021-06-17 09:37:50 +00:00
82f21fdb99
Make WebRtcAudioReceiveStream::stream_ const.
...
This builds on a few other CLs that avoid recreating the audio receive
streams on config changes and removes redundant config state in
WebRtcAudioReceiveStream, constructs the embedded receive stream in the
initializer list and keeps it const.
Bug: webrtc:11993
Change-Id: Iad28e0170bee6bf1e08713a89af7c81435b4265e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222100
Commit-Queue: Tommi <tommi@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34317}
2021-06-17 08:39:44 +00:00
b4100ad06a
Avoid using legacy rtp parser in neteq test::Packet
...
Bug: None
Change-Id: I9184954d9c99f0a34ae335d03843171864071e5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222648
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34316}
2021-06-17 08:38:14 +00:00
35b21ba8d4
In RtcpTransceiver avoid extra PostTask during construction
...
it is not required because during construction members can be set on
wrong thread, and in some corner cases it may even cause a crash.
Bug: none
Change-Id: I37d7f2a7772b6ab5e574077d3f53bca2529f9ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222651
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34315}
2021-06-17 08:36:34 +00:00
f9d5e55a31
Revert "Avoid video stream allocation on configuration change after timeout."
...
This reverts commit 10814873c584df17e560462adcc2b72e488ada91.
Reason for revert: Breaks down stream project
Original change's description:
> Avoid video stream allocation on configuration change after timeout.
>
> This is to prevent the video stream to get in a state where it is
> allocated but there is no activity.
>
> Bug: b/189842675
> Change-Id: I0793bd4cbf2a4faed92cf811550437ae75742102
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221618
> Reviewed-by: Erik Språng <sprang@webrtc.org >
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34276}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: b/189842675
Change-Id: If930360000f5ca290100920a4f215358b8c3e644
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222652
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34314}
2021-06-17 08:33:24 +00:00
a3796c8090
Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap.
...
The behavior was changed on https://webrtc-review.googlesource.com/c/src/+/219696 . The revert is due to unknown implications for a downstream project. As REMB caps are not used with send-side bandwidth estimation it should be a noop.
Bug: webrtc:12306
Change-Id: Idecc49fda007f72512a8fc1e35d62e673b00df3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222607
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34313}
2021-06-17 07:44:02 +00:00
ce3b3ba3cd
Update WebRTC code version (2021-06-17T04:05:50).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: Id614e3f216e5bcd9870ec6bac74fc44264c9e1a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222843
Reviewed-by: 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/master@{#34312}
2021-06-17 05:28:35 +00:00
4b629521b9
Roll chromium_revision 6ade74989a..6f7025c98c (893176:893293)
...
Change log: 6ade74989a..6f7025c98c6ade74989a..6f7025c98cb400b802fd..39aab38bd49ccd2d2400..a6379d4f30be7dcbc361..466954eda3701dca6ebd..941fd54fffde894a1492..57d2a56d14https://chromium.googlesource.com/catapult.git/+log/70bf08ecdb..96bc38d7d5
* src/third_party/depot_tools: 592d5ec077..74ef838a402e51eecb00..680815db186ade74989a..6f7025c98cTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Ib1e008619c60a83a30f70e19bafcbf9f96630bcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222842
Reviewed-by: 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/master@{#34311}
2021-06-17 05:00:35 +00:00
e0c73656fa
Roll chromium_revision 19c2bebe7d..6ade74989a (893060:893176)
...
Change log: 19c2bebe7d..6ade74989a19c2bebe7d..6ade74989a46146bca12..b400b802fd94441aacf6..9ccd2d2400b9f393bf29..9e4ba8b69f1c1a1626e9..701dca6ebd3bdcf36fcf..de894a14928b07d9bbd0..d57b60b2a9dd9e6218fa..2e51eecb0019c2bebe7d..6ade74989a19c2bebe7d..6ade74989aTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I58a03ad0dcfa7215ea9ef6c3d745d20c48418804
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222860
Reviewed-by: 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/master@{#34310}
2021-06-16 23:03:54 +00:00
a2a073bc0b
Reformat pc/g3doc/rtp.md
...
Bug: None
Change-Id: I9d528ea414b8214b4f7e193b56ad399f0b8f562c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222649
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34309}
2021-06-16 22:47:55 +00:00
55107c8507
Update the sync_group id without recreating audio receive streams.
...
Bug: webrtc:11993
Change-Id: I7aaff6d6f89332e60967fba741252b630fd941cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222043
Commit-Queue: Tommi <tommi@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34308}
2021-06-16 19:34:18 +00:00
25029c4135
Roll chromium_revision b452ca696d..19c2bebe7d (892948:893060)
...
Change log: b452ca696d..19c2bebe7db452ca696d..19c2bebe7d5728cd31d7..46146bca1296bb81b691..94441aacf67e85c7a26b..a38ef11ab6f3b597ed94..b9f393bf2972756fe569..1c1a1626e9da46404503..3bdcf36fcfhttps://chromium.googlesource.com/catapult.git/+log/6434229377..70bf08ecdb
* src/third_party/depot_tools: e319aba2ba..592d5ec0778336d53cff..c6fcd6122873cf27bf12..8b07d9bbd06d465fb61f..dd9e6218fab452ca696d..19c2bebe7dTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I0121ce62748a4e0b362cee40626faa7787f41df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222822
Reviewed-by: 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/master@{#34307}
2021-06-16 18:35:08 +00:00
355c47309d
Fix VideoRtpDepacketizerVp{8,9} copy assignment signature.
...
Bug: none
Change-Id: I4adca8b4cbf4ffa15172fabc1eaba8c2b65c6fb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222650
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34306}
2021-06-16 17:09:05 +00:00
5b9d0c70c2
AGC1 add clipping predictor evaluator
...
Observes clipping predictions and detections and computes evaluation
metrics for the predictor.
Bug: webrtc:12774
Change-Id: I83f5942a3b6491de288510f2200f2f5c0e099bf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221619
Reviewed-by: Hanna Silen <silen@webrtc.org >
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34305}
2021-06-16 15:33:51 +00:00
808f49428f
LOG DTLS (failed) handshake retransmission
...
Add logging of DTLS handshake retransmission,
either when it happens or when it fails.
Note that is only for the handshake messages,
which are retransmitted with exponential back off.
This patch aim to help rare DTLS hanging problems.
BUG=None
Change-Id: Iae808190711dd80dd8a43ff22757dd69919d63ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222647
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Jonas Oreland <jonaso@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34304}
2021-06-16 13:13:52 +00:00
d579e6bc7b
dcsctp: Do explicit bounds checking in bounded IO
...
The previous approach was that the caller was responsible for ensuring
that any buffer passed in to the Bounded IO wrappers, and that any
offset from where sub-readers were created were valid. The called would
always do a validation of the data and return proper error messages
if they were not.
This didn't pan out. https://crbug.com/1216758 found an overflow that
fooled the validation logic and the fuzzer could read out-of-bounds,
although it would always crash in that particular case.
There was already bounds checking, but under DCHECKs. This CL changes
that so that any bounds checking is done with CHECKS, as would've been
done in Rust. It's better to crash than to read arbitrary memory.
Bug: chromium:1216758
Change-Id: I89b52f0758495b5fe46f926c142870a263b96314
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221743
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Reviewed-by: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34303}
2021-06-16 13:02:32 +00:00
72b79986a5
Remove the createDecoder(String) overload
...
This method has been deprecated since 2018-07:
https://webrtc-review.googlesource.com/c/src/+/88368/
It is never called by WebRTC itself.
Custom `VideoDecoderFactory` implementations overriding this method can
switch to the overload accepting a `VideoCodecInfo` object.
This is also adding a `toString()` implementation to `VideoCodecInfo`,
to make logging of the value more useful.
Bug: webrtc:7925
Change-Id: I70ec07a0cd4ffa07d165c9851e393439fcc5870b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221960
Commit-Queue: Xavier Lepaul <xalep@webrtc.org >
Reviewed-by: Paulina Hensman <phensman@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34302}
2021-06-16 12:31:52 +00:00
130e031010
Roll chromium_revision 570a173256..b452ca696d (892156:892948)
...
Change log: 570a173256..b452ca696d570a173256..b452ca696d5861d7d03a..5728cd31d740d08dfe70..96bb81b691e5f25a39b9..f4328ad7c0fb190fe345..f3b597ed94d4bd8dddb7..72756fe56902b80969a2..da46404503https://chromium.googlesource.com/catapult.git/+log/2a1d8a231a..6434229377
* src/third_party/depot_tools: ef579a1192..e319aba2ba8d191d6395..73cf27bf1273b2a35554..6d465fb61f570a173256..b452ca696dTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I8ccf3968adaaa02ff716cb51d7d9954b69ad4351
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222800
Reviewed-by: 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/master@{#34301}
2021-06-16 10:50:17 +00:00
98ff0280ce
AGC analog ClippingPredictor refactoring 2/2
...
Uunit test code readability improvements.
Bug: webrtc:12774
Change-Id: I66f552d23680ddb03824618dab869946e0940334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Hanna Silen <silen@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34300}
2021-06-16 10:28:57 +00:00
08be9baaa3
Don't recreate the audio receive stream when updating the local_ssrc.
...
Bug: webrtc:11993
Change-Id: Ic5d8a8a8b7c12fb1d906e0b3cbdf657fd9e8eafc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222042
Commit-Queue: Tommi <tommi@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34299}
2021-06-16 10:03:31 +00:00
bc03259de7
Define generate_location_tags gn arg
...
This is needed as of https://chromium-review.googlesource.com/c/chromium/src/+/2957335
Bug: webrtc:12881
Change-Id: Ica345dfa6e73b0096cfa5643c06b85e628c8708f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222606
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34298}
2021-06-16 09:37:01 +00:00
6a0a55907b
Reland "Correctly handle retransmissions/padding in early loss detection."
...
This is a reland of e9ae4729e03f60dbe3b1828dd9009b401097cd3f
TBR=philipel@webrtc.org ,terelius@webrtc.org
Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org >
> Reviewed-by: Philip Eliasson <philipel@webrtc.org >
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Commit-Queue: Erik Språng <sprang@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34293}
Bug: webrtc:12713
Change-Id: Iec123d71edafea98fe289acde007b57e212681f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222640
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34297}
2021-06-16 08:14:27 +00:00
c03d6e965e
Support Java_Buffer_toI420 returning null
...
Bug: webrtc:12877
Change-Id: Ib9c8c62b350e644b17a3194e4208df6b06a4d9b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222582
Reviewed-by: Xavier Lepaul <xalep@webrtc.org >
Reviewed-by: Evan Shrubsole <eshr@google.com >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Fabian Bergmark <fabianbergmark@google.com >
Cr-Commit-Position: refs/heads/master@{#34296}
2021-06-16 07:27:47 +00:00
cd430c84d9
Update WebRTC code version (2021-06-16T04:05:58).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I3a8bc38e99b1ad78a73021c215c96357c0ab45d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222760
Reviewed-by: 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/master@{#34295}
2021-06-16 05:11:06 +00:00
d6957c2eed
Revert "Correctly handle retransmissions/padding in early loss detection."
...
This reverts commit e9ae4729e03f60dbe3b1828dd9009b401097cd3f.
Reason for revert: Internal test failure
Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org >
> Reviewed-by: Philip Eliasson <philipel@webrtc.org >
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Commit-Queue: Erik Språng <sprang@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34293}
TBR=danilchap@webrtc.org ,terelius@webrtc.org ,sprang@webrtc.org ,philipel@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Iaca6dc7739d953e97add5f5d516139b4819e43ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222601
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34294}
2021-06-15 15:59:10 +00:00
e9ae4729e0
Correctly handle retransmissions/padding in early loss detection.
...
This CL makes sure we don't cull packets from the history based on
incorrect ack mapping, just like it's predecessor:
https://webrtc-review.googlesource.com/c/src/+/218000
It also changes the logic to make sure retransmits counts towards
history pruning - and properly ignores padding/fec.
Bug: webrtc:12713
Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34293}
2021-06-15 15:39:19 +00:00
e3ceb88c72
Sanitize hostname literals when mDNS obfuscation is on.
...
Also applies sanitizing to prflx candidates, not just local ones.
Also add tests for the port allocator Sanitize function.
Bug: chromium:1218346
Change-Id: Ifbc7843cd6e289c09ca72b6ec610a34bbbf7e04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222581
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34292}
2021-06-15 14:41:46 +00:00
be53049555
Reland "Avoid sending empty receiver reports with RtcpTransceiver"
...
This reverts commit 48420fa947cea4c618d51dc5f87908765a3a69db.
Reason for revert: downstream unittests adjusted
Original change's description:
> Revert "Avoid sending empty receiver reports with RtcpTransceiver"
>
> This reverts commit e5f1a3992e3bbfa0445b90f317576c8229524d74.
>
> Reason for revert: Speculative revert due to failing downstream unittest.
>
> Original change's description:
> > Avoid sending empty receiver reports with RtcpTransceiver
> >
> > Bug: None
> > Change-Id: Ia017c2df285febefb72ba88ba43366455bde5a78
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222402
> > Reviewed-by: Per Kjellander <perkj@webrtc.org >
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#34281}
>
> TBR=danilchap@webrtc.org ,perkj@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I895317ad0381756e97e501a36d6440f83a68b6f8
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222440
> Reviewed-by: Björn Terelius <terelius@webrtc.org >
> Commit-Queue: Björn Terelius <terelius@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34284}
# Not skipping CQ checks because this is a reland.
Bug: None
Change-Id: I3481b9b12ddabaef7303ba80e9cd885930988caa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34291}
2021-06-15 12:57:56 +00:00
7a2db8acbe
Modify Bundle logic to not add & destroy extra transport at add-track
...
If a bundle is established, then per JSEP, the offerer is required to
include the new track in the bundle, and per BUNDLE, the answerer has
to either accept the track as part of the bundle or reject the track;
it cannot move it out of the group. So we will never need the transport.
Bug: webrtc:12837
Change-Id: I41cae74605fecf454900a958776b95607ccf3745
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221601
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34290}
2021-06-15 09:44:36 +00:00
e4eb8af08f
libstdc++: fix ostream operator<< usage in JsepTransportCollection
...
it.second in JsepTransportCollection::IsConsistent() is of type
std::unique_ptr, which has no operator<< in libstdc++. Even if it
would exist, it would just return the pointer value and not the
content.
Bug: chromium:957519
Change-Id: I17c75db47d63f42b33a551ee2b7afbf5c9a3cfde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222401
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34289}
2021-06-15 08:00:26 +00:00
07bf5b5d78
Update WebRTC code version (2021-06-15T04:04:38).
...
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com ,mbonadei@webrtc.org
Bug: None
Change-Id: I0fe97b075adfe06b14b167111e727734060fce90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222501
Reviewed-by: 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/master@{#34288}
2021-06-15 05:14:20 +00:00
3008bcd588
Don't recreate audio receive streams on header extension update.
...
Bug: webrtc:11993
Change-Id: Ibf45cb846713a6dd991a73bc72b4c5f59e3e26e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222041
Commit-Queue: Tommi <tommi@webrtc.org >
Reviewed-by: Markus Handell <handellm@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34287}
2021-06-14 19:02:30 +00:00
6bbe1a47e1
Roll chromium_revision e9261a56ad..570a173256 (892013:892156)
...
Change log: e9261a56ad..570a173256e9261a56ad..570a17325654fedec3aa..5861d7d03ab59724c60a..40d08dfe709bf9e6fb50..fb190fe3453c85e0a05c..d4bd8dddb79c4fcf3744..02b80969a22@3.11 .0.cr0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2f46970950..2a1d8a231a
* src/third_party/freetype/src: 7833e308ef..8336d53cff9834c0b426..8d191d639524587d48b2..73b2a35554e9261a56ad..570a173256TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I2db347965d73349821552d28a13c24282c69b2bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222421
Reviewed-by: 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/master@{#34286}
2021-06-14 18:55:40 +00:00
d350006b70
Add rtp_config() accessor to ReceiveStream.
...
This is a consistent way to get to common config parameters for
all receive streams and avoids storing a copy of the extension
headers inside of Call. This is needed to get rid of the need of
keeping config and copies in sync, which currently is part of why
we repeatedly delete and recreate audio receive streams on config
changes.
Bug: webrtc:11993
Change-Id: Ia356b6cac1425c8c6766abd2e52fdeb73c4a4b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222040
Commit-Queue: Tommi <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34285}
2021-06-14 17:57:57 +00:00
48420fa947
Revert "Avoid sending empty receiver reports with RtcpTransceiver"
...
This reverts commit e5f1a3992e3bbfa0445b90f317576c8229524d74.
Reason for revert: Speculative revert due to failing downstream unittest.
Original change's description:
> Avoid sending empty receiver reports with RtcpTransceiver
>
> Bug: None
> Change-Id: Ia017c2df285febefb72ba88ba43366455bde5a78
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222402
> Reviewed-by: Per Kjellander <perkj@webrtc.org >
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34281}
TBR=danilchap@webrtc.org ,perkj@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I895317ad0381756e97e501a36d6440f83a68b6f8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222440
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34284}
2021-06-14 17:29:09 +00:00
1c1f540487
Factor out common receive stream methods to a common interface.
...
Also including common Rtp config members.
Follow up changes will remove the ReceiveRtpConfig class in Call
and copy of extension headers, instead use the config directly
from the receive streams and not require stream recreation for changing
the headers.
Bug: webrtc:11993
Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34283}
2021-06-14 16:54:07 +00:00
e097282f11
Avoid recreating the audio stream when a frame decryptor is set.
...
This is to be consistent with how things work on the video side but
also much less drastic than the current implementation. Aim is to
remove RecreateAudioReceiveStream(), which would improve efficiency
as well as allow for specific handling of the cases that currently
trigger recreation.
Bug: webrtc:11993
Change-Id: Ia81a5e66d44e41ea4eb2bff800e0b1583821c96a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221860
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Markus Handell <handellm@webrtc.org >
Commit-Queue: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34282}
2021-06-14 16:32:17 +00:00
e5f1a3992e
Avoid sending empty receiver reports with RtcpTransceiver
...
Bug: None
Change-Id: Ia017c2df285febefb72ba88ba43366455bde5a78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222402
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34281}
2021-06-14 16:19:47 +00:00
8b6929081e
Fix VideoStreamEncoder QP tests to not use SetHasInternalSource
...
The has_internal_source feature is deprecated, and unrelated to the
tests of QP parsing.
Bug: webtc:12875
Change-Id: Ib43063ebf49e6e0bd7a5328a04ba2816f3a7ecb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222400
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34280}
2021-06-14 14:07:46 +00:00
b237a87a25
AGC analog ClippingPredictor refactoring 1/2
...
- ClippingPredictor API and docstring changes
- Unified ClippingPredictor factory function
Bug: webrtc:12774
Change-Id: Iafaddae52addc00eb790ac165bf407a4bdd1cb52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221540
Reviewed-by: Hanna Silen <silen@webrtc.org >
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34279}
2021-06-14 12:21:31 +00:00
