Danil Chapovalov
e567d8a112
Remove unused AudioFrameProcessor* parameter from WebRtcVoiceEngine constructor
...
Bug: webrtc:15111
Change-Id: Ia55e55f98ffeceeb91fb9b4fc2323a4fd7bc1046
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326523
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41118}
2023-11-09 16:13:24 +00:00
Stephan Hartmann
fa4d7c92b7
libstdc++: replace deprecated std::is_pod<T>
...
std::is_pod is deprecated since C++20. Replace with std::trivial and
std::is_standard_layout. Avoids a lot of warnings.
Bug: chromium:957519
Change-Id: Idb4bde7401c14c0896a84c357ec668b9916f613e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325484
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41117}
2023-11-09 15:16:00 +00:00
Mirko Bonadei
be23ea4bb9
Revert "Make frame transformer MimeType pure virtual again"
...
This reverts commit 3ea9fc4cd8135555360aafbfe788571d9e2f23f9.
Reason for revert: Breaks downstream project.
Original change's description:
> Make frame transformer MimeType pure virtual again
>
> after both audio and video have been implemented.
>
> BUG=webrtc:15579
>
> Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tony Herre <herre@google.com>
> Cr-Commit-Position: refs/heads/main@{#41114}
BUG=webrtc:15579
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I9b4c9753e260aca98d24a40f32ce57d86a181ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326525
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
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Cr-Commit-Position: refs/heads/main@{#41116}
2023-11-09 14:24:53 +00:00
chromium-webrtc-autoroll
2d43014acb
Roll chromium_revision 0f76018f3e..2b21cf7f67 (1222068:1222194)
...
Change log: 0f76018f3e..2b21cf7f670f76018f3e..2b21cf7f67050b7db780..951d4c4c6759a4989748..7f3a1ab7f2f30600a48a..9f2e0d0687297bfa5893..04c5f4b01ca0e595b1d4..27da5284733a66263cd2..5828285134978c4586ae..7b24689f35e09886135f..57e930ef1f0f76018f3e..2b21cf7f67https://webrtc-review.googlesource.com/c/src/+/326800
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Cr-Commit-Position: refs/heads/main@{#41115}
2023-11-09 12:36:17 +00:00
Philipp Hancke
3ea9fc4cd8
Make frame transformer MimeType pure virtual again
...
after both audio and video have been implemented.
BUG=webrtc:15579
Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41114}
2023-11-09 12:28:10 +00:00
Raman Budny
36ed560339
Fall back to software encoding on wrong HW video encoder configuration
...
Catch one more IllegalArgumentException to avoid crashes on some devices.
Bug: webrtc:15636
Change-Id: I396473b409a1ceba8f4a91d5e4aa66d5fe3b0f44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326002
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41113}
2023-11-09 12:21:11 +00:00
Per K
698e0bec17
Add Connection::OnReadPacket(const rtc::PacketReceived&)
...
Deprecate old version but keep it for the time being.
Bug: webrtc:15368
Change-Id: Icbd2078a00d877ff948a2441c2027a12c85d4f2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326104
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41112}
2023-11-09 09:15:02 +00:00
webrtc-version-updater
0ece6706c8
Update WebRTC code version (2023-11-09T04:08:00).
...
Bug: None
Change-Id: I846946f473a5a849f28dd8f8f0b794f4439bac15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326720
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@{#41111}
2023-11-09 06:01:21 +00:00
chromium-webrtc-autoroll
534374ad57
Roll chromium_revision a4610eda72..0f76018f3e (1221833:1222068)
...
Change log: a4610eda72..0f76018f3ea4610eda72..0f76018f3edfec8b1c5a..050b7db78048ab3bd053..32d5d565047b3c5c09a5..f30600a48a86d489a621..297bfa5893c19e21aeff..a0e595b1d4https://chromium.googlesource.com/catapult.git/+log/84fb1a2dd0..5a57691454
* src/third_party/depot_tools: a83e690662..3a66263cd2424723dc02..0d3ef6ffd2ed90e27354..978c4586ae61b420ca1b..e09886135fa4610eda72..0f76018f3ehttps://webrtc-review.googlesource.com/c/src/+/326700
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Cr-Commit-Position: refs/heads/main@{#41110}
2023-11-09 05:01:44 +00:00
chromium-webrtc-autoroll
d3414d9688
Roll chromium_revision 7ae4a9b4a4..a4610eda72 (1221714:1221833)
...
Change log: 7ae4a9b4a4..a4610eda727ae4a9b4a4..a4610eda7227922da2b6..dfec8b1c5ab4bda7dc8c..59a4989748a0be9f91de..7b3c5c09a56f5ca081fc..86d489a6212db36d5cef..c19e21aeffhttps://chromium.googlesource.com/catapult.git/+log/96c026ec93..84fb1a2dd0
* src/third_party/perfetto: 86ba297f43..ed90e273541f92be0b8a..61b420ca1b7ae4a9b4a4..a4610eda72https://webrtc-review.googlesource.com/c/src/+/326620
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Cr-Commit-Position: refs/heads/main@{#41109}
2023-11-08 20:50:32 +00:00
chromium-webrtc-autoroll
ce627d4f44
Roll chromium_revision 157ae1ccb9..7ae4a9b4a4 (1221460:1221714)
...
Change log: 157ae1ccb9..7ae4a9b4a4157ae1ccb9..7ae4a9b4a4df6338f68f..b4bda7dc8cc2e026aaa7..a0be9f91deafe6533c33..6f5ca081fce5f0721968..2db36d5cefeb35cbb649..86ba297f43a1aa2fa48b..1f92be0b8a157ae1ccb9..7ae4a9b4a4https://webrtc-review.googlesource.com/c/src/+/326580
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Cr-Commit-Position: refs/heads/main@{#41108}
2023-11-08 18:59:01 +00:00
qwu16
bd396fdffa
Add rtp packetizer for H265
...
Bug: webrtc:13485
Change-Id: I4e7e29a7661d51e12bb2ee12e319f6cef49482d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318005
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41107}
2023-11-08 15:49:37 +00:00
chromium-webrtc-autoroll
916ec25eed
Roll chromium_revision 337b838893..157ae1ccb9 (1221347:1221460)
...
Change log: 337b838893..157ae1ccb9337b838893..157ae1ccb90394b45670..27922da2b6a724b3ac94..48ab3bd05331d0fd470e..c2e026aaa720306afb46..afe6533c3360388eb234..e5f0721968https://chromium.googlesource.com/catapult.git/+log/124950ee2c..96c026ec93
* src/third_party/depot_tools: 197399a8ac..a83e6906622364ae8b07..278060665f5cee89ee99..a1aa2fa48b337b838893..157ae1ccb9https://webrtc-review.googlesource.com/c/src/+/326440
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Cr-Commit-Position: refs/heads/main@{#41106}
2023-11-08 10:33:23 +00:00
Philipp Hancke
977b56c9e9
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
...
This uses libSRTPs srtp_remove_stream()
https://github.com/cisco/libsrtp/blob/main/include/srtp.h#L597
method to remove SSRCs from the libSRTP session when they are removed
from the RTP demuxer. This works even when the stream was added
automatically via the ssrc_any_inbound mechanism.
Only streams for inbound SSRCs that were added explicitly via SDP negotiation are removed.
Guarded by WebRTC-SrtpRemoveReceiveStream field trial.
BUG=webrtc:15604
Change-Id: I655bde5f8ddf26ac91395ef54bd1b3c598813380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41105}
2023-11-08 10:24:10 +00:00
webrtc-version-updater
80056062f5
Update WebRTC code version (2023-11-08T04:03:41).
...
Bug: None
Change-Id: I8e8385057584cad78ebbbf59db68192c3ad3b815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326361
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@{#41104}
2023-11-08 06:01:50 +00:00
chromium-webrtc-autoroll
3e436d4ee4
Roll chromium_revision bad490e614..337b838893 (1221203:1221347)
...
Change log: bad490e614..337b838893bad490e614..337b8388938656d160d3..0394b45670292639dc3d..df6338f68f2aee7b6b4b..31d0fd470eea2d7d64ac..20306afb46079a66a7b1..60388eb234e72395e5f2..197399a8ac7e1ff25009..eb35cbb64924d460a9db..974f44c8d451086b018a..5cee89ee99bad490e614..337b838893https://webrtc-review.googlesource.com/c/src/+/326360
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#41103}
2023-11-08 03:17:15 +00:00
chromium-webrtc-autoroll
8ed4b4d314
Roll chromium_revision 3df2d0c8e6..bad490e614 (1221033:1221203)
...
Change log: 3df2d0c8e6..bad490e6143df2d0c8e6..bad490e614eaa652f6a0..8656d160d3ab8815d434..292639dc3dedce87a922..2aee7b6b4bbea08cc1fd..ea2d7d64acc90ff621c5..079a66a7b1ccc34d2b44..e72395e5f272cd237dbe..7e1ff250096f2aa72349..51086b018a3df2d0c8e6..bad490e614https://webrtc-review.googlesource.com/c/src/+/326320
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Cr-Commit-Position: refs/heads/main@{#41102}
2023-11-07 23:19:00 +00:00
Victor Boivie
4397482d71
dcsctp: Convert timers to rtc::TimeDelta
...
With this, the code base should be mostly converted from using
DurationMs to rtc::TimeDelta, and the work can continue to replace
TimeMs with rtc::Timestamp.
Bug: webrtc:15593
Change-Id: I083fee6eccb173efc0232bb8d46e2554a5fbee5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326161
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41101}
2023-11-07 21:42:15 +00:00
henrika
644025c51f
FrameCadenceAdapter: keeps max_fps in sync with constraints during 0Hz
...
Before this change, the FCA did not not update its cadence when
max_fps was changed and zero-hertz was already enabled.
See https://paste.googleplex.com/6300124249587712 for more details.
Bug: chromium:1400204
Change-Id: I95d80bdfa85ecac8681784b2b29e98d1a587ba53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326105
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41100}
2023-11-07 19:21:47 +00:00
chromium-webrtc-autoroll
ad9c4773f4
Roll chromium_revision fb6b63f0cb..3df2d0c8e6 (1220864:1221033)
...
Change log: fb6b63f0cb..3df2d0c8e6fb6b63f0cb..3df2d0c8e6879c868190..eaa652f6a0e467eae792..edce87a922e8307c2a9b..bea08cc1fdae4d577ee6..c90ff621c56f755e28a2..72cd237dbead2dd019f1..6f2aa72349fb6b63f0cb..3df2d0c8e6https://webrtc-review.googlesource.com/c/src/+/326300
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Cr-Commit-Position: refs/heads/main@{#41099}
2023-11-07 18:56:50 +00:00
Mirko Bonadei
795a2ca30f
Roll chromium_revision 15aec93e66..fb6b63f0cb (1220619:1220864)
...
Change log: 15aec93e66..fb6b63f0cb15aec93e66..fb6b63f0cb44f4ed53b3..879c868190d1c7080c73..ab8815d434e1fa8831f1..a724b3ac944909d73b1a..e467eae79205d2066c10..e8307c2a9ba8c098943d..ae4d577ee6https://chromium.googlesource.com/catapult.git/+log/c6db8519c0..124950ee2c
* src/third_party/depot_tools: 0c5e8652fe..ccc34d2b44e920273774..6338f2a681d781e6e1b8..2364ae8b0733063c9403..6f755e28a2fdfc133180..ad2dd019f115aec93e66..fb6b63f0cbhttps://webrtc-review.googlesource.com/c/src/+/326261
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41098}
2023-11-07 14:16:44 +00:00
Markus Handell
03bc3a0fa6
VideoStreamEncoder: exclude screencast from capture time measurement.
...
This CL avoids measurement for screencast encoding work. The reason is
screencast can cling on to and re-encode old video frames for which
webrtc::VideoFrame::reference_time() is unchanged.
Bug: chromium:1498378
Change-Id: I5bf79d29ef7f57ddff2622cbb6c3436480bd16ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41097}
2023-11-07 14:14:56 +00:00
Per K
03457f6891
Remove unused Connection::SignalReadPacket
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Bug: webrtc:11943
Change-Id: I44bbc023a52c9a7499a4d66db14037180f075b30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326102
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41096}
2023-11-07 11:25:12 +00:00
Jakob Ivarsson
e925db88c1
Make stats member of packet buffer.
...
Bug: none
Change-Id: Ide88e895ea27fdfe5c68aa45295df45bf72bc292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325532
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41095}
2023-11-07 10:19:25 +00:00
Linus Nilsson
bd523afd3a
Add unit test to RenderSynchronizer
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Bug: b/307672498
Change-Id: Ia50f514ff9ecbd273953fc1eab55340b75728a4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325185
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41094}
2023-11-07 09:58:37 +00:00
Philipp Hancke
7946be7429
Refactor audio/video offer/answer creation helpers
...
BUG=webrtc:15214
Change-Id: I35dcac465221760e54b09bc6c5e4126df4193289
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326141
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41093}
2023-11-07 06:23:26 +00:00
webrtc-version-updater
5dbfe9621a
Update WebRTC code version (2023-11-07T04:11:54).
...
Bug: None
Change-Id: Icd9a8ca5b6cf9c419e73bb839bd9a0e15675a13e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326221
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@{#41092}
2023-11-07 05:59:57 +00:00
chromium-webrtc-autoroll
eba274a63b
Roll chromium_revision d51ff9503d..15aec93e66 (1220376:1220619)
...
Change log: d51ff9503d..15aec93e66d51ff9503d..15aec93e6654337a07a5..44f4ed53b3f4f9306736..d1c7080c7370a0dc5735..4909d73b1a9d709bef7e..05d2066c10e73ff98615..a8c098943dhttps://chromium.googlesource.com/catapult.git/+log/448b931d9a..c6db8519c0
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/1dbe1c7fae..7b5f665231
* src/third_party/perfetto: 79b45f60df..33063c94032e673c45f4..fdfc133180d51ff9503d..15aec93e66https://webrtc-review.googlesource.com/c/src/+/326075
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Cr-Commit-Position: refs/heads/main@{#41091}
2023-11-07 01:29:29 +00:00
Victor Boivie
be04c98d64
dcsctp: Migrate non-Timer related to rtc::TimeDelta
...
This does the bulk of the remaining refactoring, except timers since
they are an even bigger part - but more isolated.
Bug: webrtc:15593
Change-Id: I7afa349e2119be7592797ee6b3b198e6de4f697a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326160
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41090}
2023-11-06 19:55:22 +00:00
chromium-webrtc-autoroll
0a33589db1
Roll chromium_revision 1204aeca4d..d51ff9503d (1220240:1220376)
...
Change log: 1204aeca4d..d51ff9503d1204aeca4d..d51ff9503dae6082baab..54337a07a565cd2ac234..f4f93067369a13e5ba21..70a0dc5735da5a47c57c..9d709bef7e3ff0a6225a..e73ff98615https://chromium.googlesource.com/catapult.git/+log/f3e882d631..448b931d9a
* src/third_party/depot_tools: b4e466e610..0c5e8652fe6d6607b8b3..e920273774f26fd69d85..920c40cd43eaa388ea81..79b45f60df97c493c880..2e673c45f41204aeca4d..d51ff9503dhttps://webrtc-review.googlesource.com/c/src/+/326123
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@{#41089}
2023-11-06 18:34:15 +00:00
Danil Chapovalov
2b58ec2938
Deprecate call_factory and media_engine in PeerConnectionFactoryDependencies
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Bug: webrtc:15574
Change-Id: Ia97ad0853196fea5c20fc0c0d58a9305b72c515b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326001
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41088}
2023-11-06 15:53:39 +00:00
Jakob Ivarsson
0873faae00
Remove smart flushing experiment.
...
It did not result in big quality improvements.
Bug: webrtc:12201
Change-Id: I9728469a388ee179d6069af8521bfc5571870bd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325533
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41087}
2023-11-06 15:38:04 +00:00
Mirko Bonadei
9272771418
Roll chromium_revision c89d7a6d7f..1204aeca4d (1216881:1220240)
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Change log: c89d7a6d7f..1204aeca4dc89d7a6d7f..1204aeca4d1546e3adb6..ae6082baaba21fc60651..65cd2ac2346f834e2039..e1fa8831f15139a7efd4..9a13e5ba2146366a7e4d..da5a47c57c64d9ec3158..3ff0a6225ahttps://boringssl.googlesource.com/boringssl.git/+log/c38dc29860..a96e7377fc
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/47efdb4b14..f3e882d631
* src/third_party/depot_tools: 9f3b33a275..b4e466e61055d0287cfc..6d6607b8b3a429c26ae2..d781e6e1b82ca9f38714..0226cb1cdf7686b5d38c..69b8c64697cefa83de08..eaa388ea81d7f60c3fd2..97c493c880c89d7a6d7f..1204aeca4dhttps://webrtc-review.googlesource.com/c/src/+/326072
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41086}
2023-11-06 15:12:45 +00:00
Mirko Bonadei
24f1ff77e5
Add third_party/re2/src to DEPS.
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Bug: b/309389865
Change-Id: Idc565340a2f095b243bd8f7c6c1407a111443d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326181
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41085}
2023-11-06 13:08:46 +00:00
Mirko Bonadei
2bb48727a0
Add third_party/fuzztest/src to DEPS.
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Bug: b/309389865
Change-Id: I290d09bed28591abb645739b1e916313abf9240e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326180
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41084}
2023-11-06 11:29:31 +00:00
Harald Alvestrand
99ea7c3eaa
Use pass-by-value in delayed scheduling of OnSentPacket
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The change in the test failed to trigger an error on msan, but making
the change anyway out of an abundance of caution.
Bug: chromium:1496240
Change-Id: Ifa1b632f4e9ddb413f0eb23aba3f5b321b287b06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326080
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41083}
2023-11-06 09:52:43 +00:00
Philipp Hancke
7fbcc8cef7
Rename PlanB helper function in tests
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which is not doing anything specific to plan-b.
BUG=None
Change-Id: Ic214b10a9c3021a8ca93601453d6eb42b84f2d84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325529
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41082}
2023-11-06 07:08:36 +00:00
webrtc-version-updater
386873b7cc
Update WebRTC code version (2023-11-05T04:02:21).
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Bug: None
Change-Id: I1d00f1d24eeaabb18c27e1304acee2765a5cf8b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326051
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@{#41081}
2023-11-05 05:37:33 +00:00
webrtc-version-updater
37df1c07b2
Update WebRTC code version (2023-11-04T04:12:04).
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Bug: None
Change-Id: I22d43f0d5657fdef1c8845756ba260a5affc0d6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325985
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@{#41080}
2023-11-04 05:48:53 +00:00
henrika
4be5927dc7
Reduces rate at which TryGetNextFrame returns NULL for WGC
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This CL is a follow-up of work done in
https://webrtc-review.googlesource.com/c/src/+/323882 where the goal
was to reduce the amount of FrameDropped error logs in
WebRTC.DesktopCapture.Win.WgcCaptureSessionGetFrameResult.
The previous work avoids FrameDropped logs for a minimized window
being captured with WGC but we still se a large amount of these error
(or rather warning) logs. See [1] which comes from Canary.
This CL does two different things to improve the situation:
1) It adds kFramePoolEmpty to the existing
GetFrameResult::kFrameDropped enum to point out that the warning
comes from the frame pool not being able to return a valid new frame.
It also makes it more clear that it does not cause an outer/final
error as WgcCapturerResult::kFrameDropped. We still keep the inner
GetFrameResult::kFrameDropped but it is only produced when the frame
pool returns NULL and our external queue is empty. Hence, a real
frame-drop error. Note that, it is still easy to provoke
kFramePoolEmpty simply by asking for a high resolution at a high rate.
The example in [2] comes from a 4K screen @30fps. Hence, we have not
fixed anything yet.
2) It also increases the size of the internal frame pool from 1 to 2.
This does lead to an almost zero rate of kFramePoolEmpt
warnings at the expense of a slightly reduced max capture rate. BUT,
with 1 as size, we can "see" a higher max capture rate but it is not
a true rate since it comes with a high rate of kFramePoolEmpty
errors. Hence, we "emulate" a high capture rate by simply feeding
copies of the last frame that we had stored in the external queue.
Using 2 leads to a more "true" rate of what we actually can capture
in terms of *new* frames and also a substantially lower rate of
kFramePoolEmpty.
In addition, with 1 as size, if we ask at a too high rate and provide
a copy of the last frame, our CPU adaptation will not reduce its rate
since we think that things are OK when it is actually not.
Also, the samples in [3] and [4] both use 2 as numberOfBuffers
as well.
Let me also mention that with this small change, I a have not been
able to provoke any kFramePoolEmpty error messages.
Finally, geDisplayMedia can be called called with constraints where
min and max framerate is defined. The mechanism which maintains the
min rate is implemented via the RequestRefreshFrame API and it can
be sent to the source (DesktopCaptureDevice) back to back with a
previous timer interrupt for a capture request. Without this change,
these RRFs were also a source of a large amount of
kFramePoolEmpty error logs. With 2 buffers instead; this is no
longer the case.
[1] https://screenshot.googleplex.com/7sfv6HdGXLwyxdj
[2] https://paste.googleplex.com/4795680001359872
[3] https://github.com/robmikh/Win32CaptureSample/blob/master/Win32CaptureSample/SimpleCapture.cpp
[4] https://learn.microsoft.com/en-us/windows/uwp/audio-video-camera/screen-capture#add-the-screen-capture-capability
Bug: chromium:1314868
Change-Id: I73b823b31a993fd2cd6e007b212826dfe1a80012
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325521
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41079}
2023-11-03 18:05:17 +00:00
Harald Alvestrand
2931ddd2e9
Fire SentPacket in a PostTask when recursive
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Speculative fix; test included.
Bug: chromium:1496240
Change-Id: I9cb8953653e9d45adbc8694b67b0d5399cf9fde9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326020
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41078}
2023-11-03 16:41:34 +00:00
Harald Alvestrand
97f3fb08a0
Assert some pointers that might be flaky.
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Speculative fix; replaces nullpointer following with test failure.
Bug: b/307520538
Change-Id: I294d5747c852a8cde43a0acad973dca12712c69e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326000
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41077}
2023-11-03 14:47:46 +00:00
Danil Chapovalov
4fd1cc70da
Add EnableMediaWithDefaultsAndTimeController
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To replace CreateTimeControllerBasedCallFactory
Update webrtc tests to use this new function
Bug: webrtc:15574
Change-Id: I2b74cd930ecc4f72dd1e7aa853764ca298b66ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325527
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41076}
2023-11-03 13:26:32 +00:00
Danil Chapovalov
c63120a092
Migrate PeerConnection tests to EnableMedia api
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Add test helper to inject fake media engine for those tests.
Bug: webrtc:15574
Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41075}
2023-11-03 12:32:14 +00:00
Victor Boivie
37f5172f6e
dcsctp: Use TimeDelta in RetransmissionTimeout
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It's still doing the calculations in milliseconds, which will be updated
in follow-up CLs.
Bug: webrtc:15593
Change-Id: I7fb93d4687d58f409db182db40db720d82feb3dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325524
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41074}
2023-11-03 11:56:18 +00:00
Philipp Hancke
d2098933e1
Expose audio mimeType for insertable streams
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Split from
https://webrtc-review.googlesource.com/c/src/+/318283
to reduce CL size. Takes a different and (hopefully) simpler
approach.
BUG=webrtc:15579
Change-Id: I8517ffbeb0f0a76db80e3e367de727fb6976211d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325023
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41073}
2023-11-03 09:49:12 +00:00
Harald Alvestrand
23cecc1d43
Move scoped_refptr from rtc:: to webrtc::
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leaving a compatible alias.
Bug: webrtc:15622
Change-Id: Ie25d87fa372cc71eaf52882454f4dd24c7c33789
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325462
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41072}
2023-11-03 07:36:07 +00:00
Markus Handell
414ffac4b7
Add RTC_EXPORT for Voucher::ScopedSetter and Voucher::Attachment.
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Bug: chromium:1498378
Change-Id: I606079411c71ac46cc5d261a86f6a11c473d1278
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41071}
2023-11-03 07:25:10 +00:00
webrtc-version-updater
c68da75d04
Update WebRTC code version (2023-11-03T04:11:45).
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Bug: None
Change-Id: Ia946554696121721295015253ef742d960eed71b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325844
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@{#41070}
2023-11-03 05:33:19 +00:00
Danil Chapovalov
779c9dede9
Migrate CreatePeerConnectionFactory implementation to EnableMedia api
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Bug: webrtc:15574
Change-Id: I2e109a62a9069f37a580fa64cacdd5a86a293203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41069}
2023-11-02 23:01:31 +00:00
