c06dc4df80
Add Fuchsia Perf bot
...
Bug: b/232740856
Change-Id: I901c283262cc444f835a099a99ee9a6e09d6b759
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285442
Reviewed-by: Jeremy Leconte <jleconte@google.com >
Commit-Queue: Christoffer Jansson <jansson@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38758}
2022-11-29 13:27:52 +00:00
2076af4673
APM: InputVolumeController tests simplified
...
Bug: webrtc:7494
Change-Id: I8f622b950aed8f1d5c42fcb8eb0c37c86532b6fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285440
Reviewed-by: Hanna Silen <silen@webrtc.org >
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38757}
2022-11-29 12:45:46 +00:00
bf2f605e03
Add more information to RTPVideoHeader::GetAsMetadata().
...
Update GetAsMetadata() to include more of the RTPVideoHeader metadata.
The intent is to be able to both get and set all of these from
JavaScript behind a flag.
Planned follow-up CLs:
1. Also get codecs-specifics, starting with VP8.
2. Test refactoring/rename: Move tests to RTPVideoHeaderTest.
3. Add RTPVideoHeader::SetFromMetadata() covering everything gettable.
4. Chrome plumbing.
Bug: webrtc:14709
Change-Id: I78679b9ff4ca749d50f309a1713e71ceabb826dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285084
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38756}
2022-11-29 12:30:46 +00:00
c5aac4ec1f
Fix crash in PacketArrivalMap::EraseTo() when using missing seq number
...
There was a DCHECK in PacketArrivalMap::EraseTo() that the
seqeuence number that is used as argument has been received.
However, this is not necessarily the case since it's cleared upon
a request of a feedback report from the sender.
Bug: webrtc:14679
Change-Id: I908b4bf1f2a4355593f0a361e1733fc91527366d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283741
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38755}
2022-11-29 12:29:43 +00:00
ef7618a8f9
Run Android perf testers on more performant bots.
...
Change-Id: I412d83beed78d887a974bf1d52c98814456b749f
Bug: b/260684409
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285441
Reviewed-by: Christoffer Jansson <jansson@webrtc.org >
Commit-Queue: Jeremy Leconte <jleconte@google.com >
Cr-Commit-Position: refs/heads/main@{#38754}
2022-11-29 12:02:17 +00:00
fd9a1e1d98
modules/video_capture: add NV12 support on Linux
...
Add native NV12 support on Linux v4l2 video_capture module.
Bug: webrtc:14650
Change-Id: I97e2010be4f15168b218da4855be8b0e985008a5
Signed-off-by: Dimitri John Ledkov <dimitri.ledkov@canonical.com >
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282841
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38753}
2022-11-29 11:59:52 +00:00
e03862bcbf
Avoid replaying excessive padding generation.
...
If the real-time clock makes a sudden and large shift forward while
there is at least one packet in the queue, the pacer would previously
try to "replay" the behavior it would have done while it was asleep.
This can lead to grand bursts of padding packets, which is
undesireable.
This CL mitigates this problem by forwarding the internal state clock
to (now - 50ms) if there is nothing to do but generate padding.
Bug: webrtc:11340, b/258509536
Change-Id: I5b8a130d938dd2566f72c1946c139f50e099e5a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285380
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38752}
2022-11-29 09:56:48 +00:00
7184016a6a
Roll chromium_revision 422b5c30cb..8113154ab7 (1076590:1076706)
...
Change log: 422b5c30cb..8113154ab7422b5c30cb..8113154ab74546310f77..b4ae821cf67362046cc8..9231726c615870472fdd..fd49ee4ee9df6e08138d..31d059e03a9fbafd799e..760f5a33d61dd824ccd1..2f339b3151https://chromium.googlesource.com/catapult.git/+log/4f0d759433..41982a4a32
* src/third_party/depot_tools: 84edf22d0e..fc04ff1628e6109b2345..30a9474044422b5c30cb..8113154ab7https://webrtc-review.googlesource.com/c/src/+/285401
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38751}
2022-11-29 08:29:45 +00:00
29a8d525be
Roll chromium_revision 7280ad8a34..422b5c30cb (1076450:1076590)
...
Change log: 7280ad8a34..422b5c30cb7280ad8a34..422b5c30cb4d0695e7f3..4546310f77984e227265..df6e08138dd1d306b60a..9fbafd799e2c68614b8f..1dd824ccd1https://chromium.googlesource.com/catapult.git/+log/07489847d9..4f0d759433
* src/third_party/depot_tools: d83509c0c3..84edf22d0ec2fa74a858..36fb00a3c383b4c2357a..e6109b23457280ad8a34..422b5c30cbhttps://webrtc-review.googlesource.com/c/src/+/285366
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Cr-Commit-Position: refs/heads/main@{#38750}
2022-11-29 02:30:36 +00:00
f25076751b
Roll chromium_revision 6ec5a4cf6c..7280ad8a34 (1076269:1076450)
...
Change log: 6ec5a4cf6c..7280ad8a346ec5a4cf6c..7280ad8a34df723f67ee..0efbd89bb54a1666ec51..f2fe6dda1e0488dfb230..c2e10323ccb2a75aecfe..984e227265a61f62b4b4..d1d306b60af69e353f8e..2c68614b8f0115386a26..d83509c0c38713ba3f0b..4a3c79cb311af6f10d7c..c2fa74a8584c0e1e5df4..83b4c2357a6ec5a4cf6c..7280ad8a34https://webrtc-review.googlesource.com/c/src/+/285364
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38749}
2022-11-28 23:13:43 +00:00
370ca9c52c
Enable sharding for fuchsia bots.
...
* Add '--quick' argument to 'low_bandwidth_audio_test' even though it doesn't look like it makes much timing difference.
* Add sharding for 'svc_tests' and 'video_engine_tests'.
Change-Id: I6e3357954d18ad03ea9f62912dd77e0e1a74b97d
Bug: webrtc:14713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285100
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Jeremy Leconte <jleconte@google.com >
Cr-Commit-Position: refs/heads/main@{#38748}
2022-11-28 19:39:08 +00:00
6358cbf7bb
Roll chromium_revision 5589fdd2c5..6ec5a4cf6c (1076152:1076269)
...
Change log: 5589fdd2c5..6ec5a4cf6c5589fdd2c5..6ec5a4cf6cba01a6701b..4d0695e7f3405f7d9671..df723f67ee6c1006503f..b2a75aecfee12e87908e..a61f62b4b47e029979e9..f69e353f8e5f99acc936..1af6f10d7c8403af59c7..4c0e1e5df45589fdd2c5..6ec5a4cf6chttps://webrtc-review.googlesource.com/c/src/+/285362
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Cr-Commit-Position: refs/heads/main@{#38747}
2022-11-28 19:12:25 +00:00
06cba44d7a
WebRTC APM: Add missing channel format check
...
The check was lost in CL https://webrtc-review.googlesource.com/c/src/+/276920
Bug: webrtc:5298
Change-Id: Ic5f072ebef4ad0bdef5446cad0536728b4ad610e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284560
Reviewed-by: Per Åhgren <peah@webrtc.org >
Commit-Queue: Sam Zackrisson <saza@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38746}
2022-11-28 18:19:06 +00:00
7daa6787fa
Roll chromium_revision 77d975d36f..5589fdd2c5 (1076026:1076152)
...
Change log: 77d975d36f..5589fdd2c577d975d36f..5589fdd2c566766af1db..ba01a6701b66b602d833..405f7d9671942f6a0db4..6c1006503f426e0dd3d3..e12e87908e74b3b05a6a..7e029979e9https://aomedia.googlesource.com/aom.git/+log/407ee57e28..d7fc49b8b6
* src/third_party/perfetto: 32e057826c..5f99acc9360cc590f941..8403af59c777d975d36f..5589fdd2c5https://webrtc-review.googlesource.com/c/src/+/285279
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38745}
2022-11-28 12:18:42 +00:00
569af3e80f
[Fuchsia] - Add more tests that pass locally
...
Create bugs and add comments for failures.
Bug: b/232740856
Change-Id: Ic81477de73c3e848569dbd0c63c1a44542b90821
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285081
Reviewed-by: Jeremy Leconte <jleconte@google.com >
Commit-Queue: Christoffer Jansson <jansson@google.com >
Cr-Commit-Position: refs/heads/main@{#38744}
2022-11-28 09:16:51 +00:00
0eea00c77b
Update WebRTC code version (2022-11-28T04:02:58).
...
Bug: None
Change-Id: Idaaa8c3075cb7ad4b52f9ec5b51287ec134f2bce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285274
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@{#38743}
2022-11-28 05:11:28 +00:00
840ea0f703
Roll chromium_revision 749a201e62..77d975d36f (1075924:1076026)
...
Change log: 749a201e62..77d975d36f749a201e62..77d975d36fd079fd08ee..66766af1db3542a5de2b..66b602d8338641823a80..4a1666ec518a91ae4de9..0488dfb2306d56a9fb5e..942f6a0db4acaeeff5f4..426e0dd3d331b56b4546..74b3b05a6ae807b5d3c8..0cc590f941749a201e62..77d975d36fhttps://webrtc-review.googlesource.com/c/src/+/285272
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38742}
2022-11-28 00:37:57 +00:00
95b556f022
Add jleconte@ and mbonadei@ as test OWNERS.
...
No-Try: True
Bug: None
Change-Id: I3c1c1d45315f316227f1e75a7764bbafaabb7403
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285280
Reviewed-by: Andrey Logvin <landrey@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38741}
2022-11-27 14:49:58 +00:00
f71e87a71d
Support --webrtc_test_metrics_output_path in test main().
...
Bug: b/260493525
Change-Id: Ic0ba5683abf467fe3671f2e673ce02867f3caf73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284700
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38740}
2022-11-27 12:34:03 +00:00
53eb544c10
Roll chromium_revision 9b137a19df..749a201e62 (1075807:1075924)
...
Change log: 9b137a19df..749a201e629b137a19df..749a201e62803c3ad4fa..d079fd08ee6f7741c787..3542a5de2b9ca2fd2711..8641823a80cd0a050474..8a91ae4de983cce21b4b..7362046cc8b67d0b07eb..6d56a9fb5e18a9afb8e0..acaeeff5f4d2e98fe0f4..31b56b454673a26246e5..0115386a26be5960bcdf..32e057826c432e14b26d..e807b5d3c89b137a19df..749a201e62https://webrtc-review.googlesource.com/c/src/+/285221
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Cr-Commit-Position: refs/heads/main@{#38739}
2022-11-26 23:10:51 +00:00
802ccfef21
Roll chromium_revision 54ff700461..9b137a19df (1075689:1075807)
...
Change log: 54ff700461..9b137a19df54ff700461..9b137a19df591898ab4e..803c3ad4faf57e5d4150..6f7741c787a4b9182bf7..b67d0b07ebd7d358c209..18a9afb8e0f40607ea4f..d2e98fe0f4c194bc190d..be5960bcdf6e54adbfd5..432e14b26d54ff700461..9b137a19dfhttps://webrtc-review.googlesource.com/c/src/+/285061
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38738}
2022-11-25 16:18:04 +00:00
c7fc01269e
Fix IvfFileReader to support different time scales
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Bug: b/237998511
Change-Id: I8e7766b38cfe19b2fb58853c9614e4d32ea34715
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285087
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38737}
2022-11-25 15:31:53 +00:00
bc43fe3a50
Remove field trial string WebRTC-AdaptiveBweThreshold and cleanup
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Removed old disabled tests
enable test on android
Bug: webrtc:4711
Change-Id: Ic9adbdadc9e847bdf31b8be4ce116a3695499944
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284922
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Commit-Queue: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38736}
2022-11-25 14:47:23 +00:00
158d5e3078
Add RTPVideoHeader::GetAsMetadata().
...
In preparation of adding RTPVideoHeader::SetFromMetadata() method, the
VideoFrameMetadata construct-from-RTPVideoHeader is replaced by
RTPVideoHeader::GetAsMetadata(). This serves two purposes:
1. Having "GetAs" and "SetFrom" in the same file reduces the risk of
these two methods getting out of sync as we expand its usage.
2. This is necessary to avoid a circular dependency that would
otherwise be introduced by RTPVideoHeader::SetFromMetadata().
Bug: webrtc:14709
Change-Id: I127b3d15f9a8c6af210449a5a50d414c9ba79930
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285080
Reviewed-by: Tony Herre <herre@google.com >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38735}
2022-11-25 14:40:30 +00:00
e862da376f
generatekeyframe: allow simulcast_encoder_adapter to request per-layer
...
in the case of separate encoders.
drive-by: add helper function to expect per-layer keyframes
BUG=chromium:1354101
Change-Id: Ib645a621add899f035bea319f035dcb0b2617510
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281002
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Philipp Hancke <phancke@microsoft.com >
Cr-Commit-Position: refs/heads/main@{#38734}
2022-11-25 12:12:05 +00:00
5c4509a604
Add a clone method to the video frame transformer API.
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This will clone an encoded video frame into a sender frame,
preserving metadata as much as possible.
Bug: webrtc:14708
Change-Id: I6f68d2ee65ef85c32cc3c142a41346b81ba73533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284701
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Guido Urdaneta <guidou@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38733}
2022-11-25 11:18:22 +00:00
4a44e0ef40
Update webrtc build command instruction to autoninja.
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`autoninja` is a wrapper that automatically provides optimal values
for the arguments passed to `ninja`. For example, more cores can be
deployed to make webrtc build faster.
Bug: None
Change-Id: I1c62141c10c176e30bbb75994a7a241c0938e6bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285044
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38732}
2022-11-25 09:19:37 +00:00
e40bb38faa
Revert "Do not log on stderr on Android tests."
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This reverts commit c48a2653466f926ecc1d259a16ede304333fc14b.
Reason for revert: logcat issue should be fixed with https://crrev.com/c/4055461 .
Original change's description:
> Do not log on stderr on Android tests.
>
> On Pixel 2, this causes an increase in flakiness. This needs to be
> reenabled once the root cause is fixed.
>
> Bug: chromium:1384172, b/259113795
> Change-Id: Ie94d3e2daad3a2de5af673c763362ea1b42fde7d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283522
> Reviewed-by: Jeremy Leconte <jleconte@google.com >
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org >
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#38623}
Bug: chromium:1384172, b/259113795
Change-Id: Iadd7c484f4e73deea952df7980acc0164c96a592
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Jeremy Leconte <jleconte@google.com >
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Cr-Commit-Position: refs/heads/main@{#38731}
2022-11-25 06:34:22 +00:00
b5e5b8a6c4
Roll chromium_revision 450a6cd6b4..54ff700461 (1075587:1075689)
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Change log: 450a6cd6b4..54ff700461450a6cd6b4..54ff7004616246c9fdab..591898ab4e8a5c1d25af..f57e5d41501a32724f72..83cce21b4b1731b7c979..a4b9182bf7fb09c79e1e..d7d358c209d68aa45336..f40607ea4fe7decdf989..6e54adbfd5450a6cd6b4..54ff700461https://webrtc-review.googlesource.com/c/src/+/285041
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Cr-Commit-Position: refs/heads/main@{#38730}
2022-11-25 04:28:11 +00:00
5a65f9e0aa
Roll chromium_revision 0b34fd9343..450a6cd6b4 (1075397:1075587)
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Change log: 0b34fd9343..450a6cd6b40b34fd9343..450a6cd6b451c5a6a363..6246c9fdab9c7229eca3..8a5c1d25afe5d207cb13..1731b7c9795b5ec2409e..fb09c79e1e1060ed10cd..d68aa45336https://chromium.googlesource.com/catapult.git/+log/ddb6643965..07489847d9
* src/third_party/depot_tools: 6fde0fbe92..73a26246e51c44de209c..aca4ec5907197b7113ea..c194bc190dd27cf2a4ce..e7decdf9890b34fd9343..450a6cd6b4https://webrtc-review.googlesource.com/c/src/+/284962
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2022-11-24 19:14:43 +00:00
893c0e449d
Allow Video Sender OnTransformedFrame() before TransformFrame()
...
Lazily initialize the RTPSenderVideoFrameTransformerDelegate's
encoder_queue_ on either OnTransformedFrame() or TransformFrame(), to
allow apps to write to an encoded insertable stream's writable before
reading from its readable.
Bug: chromium:1393373
Change-Id: I08f11682fa142884b575bb207d7d7044e80bbb9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284921
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Tony Herre <herre@google.com >
Auto-Submit: Tony Herre <herre@google.com >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38728}
2022-11-24 18:55:30 +00:00
d6b330ea77
Remove unused field trial string kRTCFieldTrialAudioForceNoTWCCKey
...
Bug: webrtc:8243
Change-Id: I9c24999e44a749669208e19574d1251dcbd22d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284941
Auto-Submit: Per Kjellander <perkj@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38727}
2022-11-24 15:11:55 +00:00
a7013ee650
Remove unused field trial WebRTC-Bwe-LossBasedStableRate
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Originally submitted here: https://webrtc.googlesource.com/src/+/350a82aec3556cfab385e41b67ab4f26f2fb0151
Bug: None
Change-Id: Id464770b089122e2cf13ce2d841f7114aa9eb9d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284942
Commit-Queue: Per Kjellander <perkj@webrtc.org >
Reviewed-by: Jonas Oreland <jonaso@webrtc.org >
Commit-Queue: Jonas Oreland <jonaso@webrtc.org >
Auto-Submit: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38726}
2022-11-24 15:02:11 +00:00
75170be4ac
Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
...
This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc.
Reason for revert: Tentative revert due to possible perf regression. b/260123362
Original change's description:
> Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream
>
> VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream.
> Therefore this cl:
> - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience.
> - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing.
> - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used.
>
> Bug: none
> Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660
> Reviewed-by: Erik Språng <sprang@webrtc.org >
> Commit-Queue: Per Kjellander <perkj@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#38698}
Bug: none
Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Auto-Submit: Per Kjellander <perkj@webrtc.org >
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38725}
2022-11-24 14:18:45 +00:00
b05968e5ec
Do not create DtmfSender for video sender.
...
On Android bindings, do not build a DtmfSender instance in a
RtpSender if its video kind is Video.
This will prevent showing an error when trying to access
that DtmfSender instance that has no native reference
Bug: webrtc:14680
Change-Id: Iba67a12cae8604c032915156b581af269f6ed265
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283742
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38724}
2022-11-24 10:58:17 +00:00
d409621f28
Fix SetCurrentThreadName() for Fuchsia.
...
CL suggested by lingxueluo@.
Bug: b/236891952
Change-Id: I960d5b016186e9784e0af5e86608891a06e78b7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38723}
2022-11-24 09:56:39 +00:00
6e55319b5d
Roll chromium_revision 26dbfd3991..0b34fd9343 (1075289:1075397)
...
Change log: 26dbfd3991..0b34fd934326dbfd3991..0b34fd9343a56456850a..51c5a6a363cd53ecaafb..9c7229eca326c26b7124..e5d207cb13b7a356c920..1060ed10cdhttps://chromium.googlesource.com/catapult.git/+log/cce68bc50e..ddb6643965
* src/third_party/perfetto: cf25b47c40..197b7113ea3bb6762707..d27cf2a4ce26dbfd3991..0b34fd9343https://webrtc-review.googlesource.com/c/src/+/284902
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38722}
2022-11-24 00:29:26 +00:00
7216b27406
screencast_portal: Add option to choose cursor capture mode
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Change adds a flag that can be used with desktop capture options
to specify how the cursor capture should be handled.
Bug: chromium:1291247
Change-Id: If8150f8412ade2b6216a65dd026ca528654f52bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284780
Commit-Queue: Salman Malik <salmanmalik@chromium.org >
Reviewed-by: Alexander Cooper <alcooper@chromium.org >
Cr-Commit-Position: refs/heads/main@{#38721}
2022-11-23 23:12:46 +00:00
fc5d709e41
Roll chromium_revision 74a29f2053..26dbfd3991 (1075167:1075289)
...
Change log: 74a29f2053..26dbfd399174a29f2053..26dbfd3991aeaee00014..a56456850a88913de942..cd53ecaafb3c8fef071e..9ca2fd27119542e93f96..26c26b71246d05100169..5b5ec2409e5c6533ad2a..b7a356c920https://boringssl.googlesource.com/boringssl.git/+log/f0518d4511..2fd8de6534
* src/tools: 4624b091db..3bb676270774a29f2053..26dbfd3991https://webrtc-review.googlesource.com/c/src/+/284900
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@{#38720}
2022-11-23 21:33:35 +00:00
514dff834b
Fix crash when setting RTCRtpEncodingParameter.maxBitrate to a non-positive value.
...
This will result in using a default value instead.
The crash is caused by configuring audio send stream with a non-positive max_bitrate_bps and it expects a positive value (or special value -1).
Also remove support for setting min bitrate since it is not in the spec and there are no tests.
Bug: chromium:1377286
Change-Id: I19af29bff79eda5c0fbb1e528fced7976f5c2511
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284640
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38719}
2022-11-23 18:38:00 +00:00
4bee365c82
Roll chromium_revision 6669f72cdc..74a29f2053 (1075042:1075167)
...
Change log: 6669f72cdc..74a29f20536669f72cdc..74a29f20536e71e09519..aeaee000148961725eee..88913de942866ec86ecb..9542e93f961a716cbded..6d0510016973f7282fa2..5c6533ad2a2fc7e1ffd5..6fde0fbe9228b1da6e19..cf25b47c40fff0ab378c..4624b091db6669f72cdc..74a29f2053https://webrtc-review.googlesource.com/c/src/+/284828
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38718}
2022-11-23 16:27:32 +00:00
46e2d103b4
Use ScopedFieldTrials instead of InitFieldTrialsFromString in InternalDecoderFactoryTest.
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Bug: webrtc:14698
Change-Id: I47ec906b258cd2b613c863cad6a0318d71a1fa32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284840
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38717}
2022-11-23 10:08:20 +00:00
6a8776a108
[DVQA] Provide more precise time for qp
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Bug: None
Change-Id: Ic7b6323c296b20e164b7ff0aca861c439bb86c89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284721
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38716}
2022-11-23 10:02:29 +00:00
bbc8fc165c
Roll chromium_revision d3d85d6b43..6669f72cdc (1074816:1075042)
...
Change log: d3d85d6b43..6669f72cdcd3d85d6b43..6669f72cdc41677446eb..6e71e095199c35b43202..8961725eee60d1ee9478..1a716cbdeda6efa878cf..73f7282fa2https://chromium.googlesource.com/catapult.git/+log/2b857d73ff..cce68bc50e
* src/third_party/depot_tools: 30e3ce8b1c..2fc7e1ffd54f6803dfe0..28b1da6e191f8033911d..fff0ab378cd3d85d6b43..6669f72cdchttps://webrtc-review.googlesource.com/c/src/+/284822
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#38715}
2022-11-23 07:33:07 +00:00
19d96365b2
Update WebRTC code version (2022-11-23T04:04:56).
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Bug: None
Change-Id: I1b9eeddbc4fa14c57314faf0e953a00d6b88534f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284820
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@{#38714}
2022-11-23 05:56:37 +00:00
447b9f3fde
Unvirtualize more wasteful functions
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Cleaning up instances of virtual functions with no overrides.
Savings of 1.4kb for chrome.dll file.
Note: These are the savings for Windows, relocation savings are
probably larger on other platforms.
GN args for builds:
use_goma=true
is_debug=false
target_cpu="x64"
use_lld=false
fatal_linker_warnings=false
symbol_level=2
dcheck_always_on = false
pe_summarize analysis pre-change -> change:
Size of out\Default\chrome.dll is 188.844544 MB
Size of out\SessionDescription\chrome.dll is 188.843520 MB
Memory size change from out\Default\chrome.dll
to out\SessionDescription\chrome.dll
.text: -704 bytes change
.rdata: -512 bytes change
.pdata: -48 bytes change
.reloc: -168 bytes change
Total change: -1432 bytes
Bug: chromium:1371503
Change-Id: I51ad0a8acf3595fc499dbbcde2fab2d1bdf90fb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283940
Commit-Queue: Ivan Rosales <rosalesi@google.com >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38713}
2022-11-22 22:44:21 +00:00
8a8c455cce
Roll chromium_revision e6090767ac..d3d85d6b43 (1074663:1074816)
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Change log: e6090767ac..d3d85d6b43e6090767ac..d3d85d6b43a7d382c97b..41677446eb00b05f4262..9c35b4320293196a4539..866ec86ecbb5de475a1c..60d1ee9478a27a0809a9..a6efa878cfhttps://chromium.googlesource.com/catapult.git/+log/c996ebbd3d..2b857d73ff
* src/third_party/perfetto: 51908f273f..4f6803dfe0ce8c4663d8..1f8033911de6090767ac..d3d85d6b43https://webrtc-review.googlesource.com/c/src/+/284761
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Cr-Commit-Position: refs/heads/main@{#38712}
2022-11-22 20:52:47 +00:00
2c2d2c75b5
Roll chromium_revision f2dad4225a..e6090767ac (1074534:1074663)
...
Change log: f2dad4225a..e6090767acf2dad4225a..e6090767acdfd53becd8..a7d382c97bb9179a2e57..00b05f4262e571bff1fe..93196a45396a779efd18..b5de475a1cd11fab37a6..a27a0809a9c5fb93e710..51908f273f3a307285aa..ce8c4663d8f2dad4225a..e6090767achttps://webrtc-review.googlesource.com/c/src/+/284740
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Cr-Commit-Position: refs/heads/main@{#38711}
2022-11-22 16:45:27 +00:00
27fed4513f
InputVolumeController: Make speech_probability non-optional
...
Make the argument speech_probability non-optional in
InputVolumeController::Process() and
MonoInputVolumeController::Process().
Additional clean-up: Remove the flag enabled in the
config. Add unit tests for MonoInputVolumeController.
Bug: webrtc:7494
Change-Id: Ie28af77dc628bf71d09ce1ff033d39031f77a21e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283700
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org >
Commit-Queue: Hanna Silen <silen@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#38710}
2022-11-22 15:19:02 +00:00
310e0624aa
Add Cisco AnyConnect IPSec MAC to VPN detection
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and set the underlying type to unknown for mac-based detection.
BUG=webrtc:13097,webrtc:13172
Co-authored by: amir.pirogovsky@microsoft.com
Change-Id: I177214f9eb5c2990faa65262f77b3d0ce9d5eab7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284500
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Jonas Oreland <jonaso@webrtc.org >
Commit-Queue: Philipp Hancke <phancke@microsoft.com >
Cr-Commit-Position: refs/heads/main@{#38709}
2022-11-22 11:55:26 +00:00
