Peter Hanspers
3fef8b27db
Adding an error callback to AudioDeviceModuleIOS.
...
This adds an optional callback closure and an enum representing the error.
Bug: webrtc:390314937
Change-Id: If9a22dd6d90d5c4d94175e021511766ea49acec2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374420
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43780}
2025-01-21 07:56:58 -08:00
Harald Alvestrand
32f3c6cef1
Add AbslStringify for RtcErrorType and RtcErrorDetail
...
Drive-by: fix MakeVal in rtc_base/checks.h to ensure that StrCat is
used for types that have AbslStringify.
Bug: None
Change-Id: Ia78c65da18b4a826365a6a2c741f11809640197f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374345
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43779}
2025-01-21 07:10:50 -08:00
Jeremy Leconte
a6bccab358
[DVQA] Dont try to render a 'superfluous' frame.
...
Change-Id: I3427cecab30b1705e5fbec110494f58cb1c599b5
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374861
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43778}
2025-01-21 03:19:31 -08:00
Henrik Boström
283a84d92a
Add matchers for RTCError, rename old matcher for RTCErrorOr.
...
Needed for testing in a follow-up CL.
Using ToString rather than absl::StrCat because I want the name of the
enum (e.g. "INVALID_MODIFICATION") as opposed to the enum value (int).
Bug: none
Change-Id: I45a925fad65395d1e6a886a9f787c2f360fb8604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374343
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43777}
2025-01-21 02:21:58 -08:00
Henrik Boström
860a13c6fd
Misc improvements to RtpTransceiver unit tests and test utils.
...
In order to reduce the size and scope of a follow-up CL, this CL makes
some cleaning up and improvements to existing tests and adds some minor
test utility methods that will be used in the follow-up.
No change in behavior, this CL...
- Makes use of NiceMock in RtpTransceiver tests to avoid wall of text
spam for various "uninteresting" method calls in all tests in this
file.
- Refactors creating senders, receivers and transceivers to allow the
follow-up CL to create such objects for kind "video" as well.
- Exposes cricket::FakeVideoEngine* to RtpTranscieverTest and allows
adding unidirectional video codecs in the fake engine, to be used by
the follow-up CL's tests.
- Allows creating fake video engine codecs from SdpVideoFormat in the
fake decoder factory (already possible in the fake encoder factory).
Bug: chromium:381407888
Change-Id: Ie07eff79d832dd21800b95fd584891ebf4520798
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43776}
2025-01-20 23:55:17 -08:00
webrtc-version-updater
ee7371f1f8
Update WebRTC code version (2025-01-21T04:06:38).
...
Bug: None
Change-Id: I7c85901d82ed77f395c502e0e4b7894c2efa9ccf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374828
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@{#43775}
2025-01-20 22:01:42 -08:00
chromium-webrtc-autoroll
b4127b5597
Roll chromium_revision dcda5ff9c0..3462a5bab8 (1408528:1408687)
...
Change log: dcda5ff9c0..3462a5bab8dcda5ff9c0..3462a5bab8a7c5912a8c..f83303423ae52ba4f90d..63412fdcdfb11145bc07..d212a71416e3a3b39dd0..a059b237bb47ddac2996..10592b60c0943a0d8992..a0c2bc79195b0a775998..6820cc03ccdcda5ff9c0..3462a5bab8https://webrtc-review.googlesource.com/c/src/+/374848
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Cr-Commit-Position: refs/heads/main@{#43774}
2025-01-20 08:37:36 -08:00
Erik Språng
d621b419a3
Make WebRTC-Video-AV1EvenPayloadSizes default-on.
...
Experimentation shows only a slight increase in bitrate due to improved
BWE. There's no negative side-effects we have been able to see so far.
This CL flips the experiment to default-on but is kept around as a
kill-switch until the next milestone just in case something unexpected
is discovered.
Bug: webrtc:42226301
Change-Id: I4a0b1c85e912b909d7bff58d78966cf161857f7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374182
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43773}
2025-01-20 07:23:58 -08:00
Anna Lemehova
d48113627a
Fix hw decoder rendering delay after frame resize
...
Need to clear frameInfos in case of reinit, as outdated items produce
incorrect decode time. This leds to render timestamps 'in future'
(VCMTiming::RenderTime) and rendering delays (low fps).
Bug: None
Change-Id: Iee569ff74fe3e0ff3610877472756cbbd59aba7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374680
Auto-Submit: Anna Lemehova <anna.lemehova@gmail.com>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Commit-Queue: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43772}
2025-01-20 03:12:34 -08:00
Evan Shrubsole
fa73a2ed79
Convert timeouts in integration_test_helpers to TimeDelta
...
Bug: webrtc:42223979
Change-Id: Ia77b34c5c30a32fcb520359b993ff0b976be378c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374880
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43771}
2025-01-20 02:58:26 -08:00
Evan Shrubsole
f1b3e3e115
Replace gunit.h macros with WaitUntil in modules/
...
Bug: webrtc:381524905
Change-Id: Ifde77639d720b404eda2e21275d0c3682ac933c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374222
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43770}
2025-01-20 02:17:10 -08:00
Evan Shrubsole
2a858e21f6
Migrate last uses of gunit.h macros
...
Bug: webrtc:381524905
Change-Id: I9bf00a61dfcc00355e81fea34625119ef3ac61b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43769}
2025-01-20 02:06:48 -08:00
chromium-webrtc-autoroll
4f56e15075
Roll chromium_revision 48223dfc0a..dcda5ff9c0 (1408397:1408528)
...
Change log: 48223dfc0a..dcda5ff9c048223dfc0a..dcda5ff9c0772674e3fd..e0914b760642cfea3f59..a7c5912a8cbd43e8a355..e52ba4f90d71c33544a8..b11145bc0748223dfc0a..dcda5ff9c0https://webrtc-review.googlesource.com/c/src/+/374847
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43768}
2025-01-20 00:31:56 -08:00
Philipp Hancke
9165a9b436
Disable OpenSSL tests needing a fake clock when boringssl is not used
...
since OpenSSL lacks that feature so broke as a result of
https://chromiumdash.appspot.com/commit/130bdcea7e69368cae66e10a99c5e86dc4e5770a
BUG=webrtc:375552698
Change-Id: I55da67a7dcf5815f7190bd43847a9ad6f7c1b93d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43767}
2025-01-19 22:33:02 -08:00
webrtc-version-updater
e6890adc3a
Update WebRTC code version (2025-01-20T04:06:42).
...
Bug: None
Change-Id: I86c608e9ac3d63685f2e17d666baf7ead7651be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374846
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43766}
2025-01-19 21:54:55 -08:00
webrtc-version-updater
0908c9b90a
Update WebRTC code version (2025-01-19T04:03:25).
...
Bug: None
Change-Id: Ida110cbe7fbb1c5272a1cef8027121035908e652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374804
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@{#43765}
2025-01-18 21:29:36 -08:00
chromium-webrtc-autoroll
9aeeb6123d
Roll chromium_revision bb864a5b8d..48223dfc0a (1408296:1408397)
...
Change log: bb864a5b8d..48223dfc0abb864a5b8d..48223dfc0a5080dff087..42cfea3f5953f19d08f2..71c33544a8bb864a5b8d..48223dfc0ahttps://webrtc-review.googlesource.com/c/src/+/374768
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43764}
2025-01-18 20:58:14 -08:00
Fabian Reddig
a85040ffd9
Revert "Reland "Use Payload Type suggester for all codec merging""
...
This reverts commit f5d13267aee114aa60e9718fc6f5032c8a5450f3.
Reason for revert: Caused downstream test failures
Original change's description:
> Reland "Use Payload Type suggester for all codec merging"
>
> This reverts commit b7abaee819771ca297bac4c51ec0daf62bd9a3fc.
>
> Reason for revert: Suspicion that suspected breakage wasn't real
>
> Original change's description:
> > Revert "Use Payload Type suggester for all codec merging"
> >
> > This reverts commit 0bac2aae596771db020f01a57fee4828081fbc38.
> >
> > Reason for revert: Suspected breakages downstream
> >
> > Original change's description:
> > > Use Payload Type suggester for all codec merging
> > >
> > > Bug: webrtc:360058654
> > > Change-Id: Id475762253c427c1800c2352a60fc0121c2dc388
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364783
> > > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#43267}
> >
> > Bug: webrtc:360058654, b/375132036
> > Change-Id: Ieda626270193e7e6c93903b3c03a691b2bf0c1e2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366540
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43290}
>
> Bug: webrtc:360058654, b/375132036
> Change-Id: Id6e72f7aac81023da43de7627c24dd1a792ea461
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374304
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43739}
Bug: webrtc:360058654, b/375132036
Change-Id: I3fb302d6ddb7d9e4b0acc3eefdac74edf55ca01a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374700
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43763}
2025-01-17 23:49:26 -08:00
chromium-webrtc-autoroll
48ca8b3d13
Roll chromium_revision 62907d98e8..bb864a5b8d (1408002:1408296)
...
Change log: 62907d98e8..bb864a5b8d62907d98e8..bb864a5b8dc2ccc824eb..64e296c42af0a46a623b..5080dff087eb8f5d1b81..bd43e8a355924b14f340..53f19d08f2https://boringssl.googlesource.com/boringssl.git/+log/bca2d72c5e..d4b6eb542d
* src/third_party/depot_tools: ea75de4c3b..80d19694223fec72bd56..12809bfa85c28325f698..943a0d89926910068b59..5b0a77599862907d98e8..bb864a5b8dhttps://webrtc-review.googlesource.com/c/src/+/374761
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43762}
2025-01-17 22:58:11 -08:00
webrtc-version-updater
1305eb9083
Update WebRTC code version (2025-01-18T04:08:19).
...
Bug: None
Change-Id: I1d52b180fda4b842ec8012a15a246f0c451bd827
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374760
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@{#43761}
2025-01-17 21:33:48 -08:00
chromium-webrtc-autoroll
5f27925a5b
Roll chromium_revision 8bf5d05e2b..62907d98e8 (1407755:1408002)
...
Change log: 8bf5d05e2b..62907d98e88bf5d05e2b..62907d98e8d89a69dc9d..772674e3fd287333d912..c2ccc824eb26eb3d1408..f0a46a623bc61fa789bb..eb8f5d1b816a6b5e89a3..924b14f340d4631a2eb3..e3a3b39dd04414ade7ba..3fec72bd56022c528998..c28325f6984b1293dd0a..6910068b598bf5d05e2b..62907d98e8https://webrtc-review.googlesource.com/c/src/+/374642
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Cr-Commit-Position: refs/heads/main@{#43760}
2025-01-17 11:56:19 -08:00
Henrik Boström
9f68535e68
Fix setParameters() throwing when level-id does not match.
...
In order to align with this PR[1], setParameters() should not throw if
the H265 level ID we're trying to send does not match what was
negotiated. This was believed to be fixed by [2] but we were still
throwing due to a check on a different layer (media_engine.cc).
In order to reproduce the issue despite WebRTC lacking SW
encoder/decoder for H265, peer_connection_encodings_integrationtest.cc
gets a new test with real stack but fake encoder/decoder factory. This
allows negotiating H265 and doing SetParameters() even though the codec
is not processing any frames.
- Basic test coverage is added for singlecast and simulcast H265.
- Test coverage for the bug being fixed added.
- In Chrome the equivalent WPTs exists for when real HW is available
here[3]. Those tests PASS with this CL (currently FAIL).
[1] https://github.com/w3c/webrtc-pc/pull/3023
[2] https://webrtc-review.googlesource.com/c/src/+/368781
[3] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/h265-level-id.https.html
Bug: chromium:381407888
Change-Id: I3619a124586b8b26d3695cfad8890cf40bd475db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43759}
2025-01-17 09:17:11 -08:00
Evan Shrubsole
b0038dd14a
Replace gunit.h macros with WaitUntil in P2P
...
Bug: webrtc:381524905
Change-Id: Ia2614729b17a40cd2f5f226e351b3f82c2038977
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374181
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43758}
2025-01-17 06:53:50 -08:00
Evan Shrubsole
d9593037dd
Replace gunit.h macros with WaitUntil in rtc_base/
...
Bug: webrtc:381524905
Change-Id: I7a78269586ee3556c0b3de63f5add393f12f4fa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374223
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43757}
2025-01-17 06:44:49 -08:00
Tommi
762753d0a2
Slight restriction of access to ContentInfo and prefer mid to name.
...
As a first step, use .mid() instead of .name in JsepTransportController
Bug: webrtc:42233761
Change-Id: I23ab97609175f8dbfdf59ee41c4db42f21a9e9ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374660
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43756}
2025-01-17 06:19:38 -08:00
Åsa Persson
88833e6d22
Update video stats documentation.
...
Bug: b/363185297
Change-Id: I70d33f40c3dab09e49076a503069b8ce0eab8eb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374307
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43755}
2025-01-17 06:01:14 -08:00
chromium-webrtc-autoroll
ceb5a3b11e
Roll chromium_revision 839b9b8bb4..8bf5d05e2b (1406733:1407755)
...
Change log: 839b9b8bb4..8bf5d05e2b839b9b8bb4..8bf5d05e2b28b89c75d5..d89a69dc9dce61bc84f9..287333d91265bfd734ce..26eb3d14085720363cd0..c61fa789bbdca9e8a29a..6a6b5e89a3https://boringssl.googlesource.com/boringssl.git/+log/571a7432a1..bca2d72c5e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d25caed4b9..b15611f18b
* src/third_party/depot_tools: a90906c74d..ea75de4c3bee1310ab5c..d4631a2eb3be49f904cc..00edfa0dd9504ea69cf7..2b6b042a77https://aomedia.googlesource.com/aom.git/+log/0c13a5d540..d737ca3ebe
* src/third_party/libunwind/src: d1e95b102f..e55d8cf51c0eadb2fa14..4414ade7ba3227982dba..022c528998c5f991404f..4b1293dd0a839b9b8bb4..8bf5d05e2bhttps://webrtc-review.googlesource.com/c/src/+/374640
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43754}
2025-01-17 01:22:51 -08:00
thebongy
d23d04163d
Fix to allow small negative jumps due to out of order packets in packet buffer
...
This resolves an issue where when packets appear out of order at the
beginning of a stream, packet_buffer.cc might drop the entire packet
buffer because it detects a "large negative jump" even though the
difference in sequence numbers is very minor and is caused by network
congestion / packet re-ordering. Currently, when the issue occurs, this
can cause video corruption/artifacts. More details and reproduction is
available on the attached webrtc bug report 390329776.
Bug: webrtc:390329776
Change-Id: Idb56eb2e066d596d8afd7ec904359baf0cb3feef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374540
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43753}
2025-01-17 00:52:36 -08:00
Philipp Hancke
3cc17eed68
dtls-stun-piggybacking: add missing configuration guard
...
the lack of this led to a DTLS restart in the default case.
Also reduce the log level from "ERROR" to "INFO".
BUG=webrtc:367395350
Change-Id: I52cd74f508ca4b4c28b11220ea82d95081558eee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374503
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43752}
2025-01-16 23:49:38 -08:00
Sergey Silkin
a65c453f9e
Reduce default max QP for AV1 from 56 to 52
...
Before this CL VP8 and AV1 used the same max QP=56. Tests show that at this QP AV1 delivers a worse PSNR than VP8. We want AV1 min quality to be not worse than VP8. This CL reduces the default max QP for AV1 to 52. With this value libaom AV1 encoder delivers PSNR close to libvpx VP8 at QP 56.
Bug: webrtc:351644568, b/369540380
Change-Id: I2e27ddab562f9c9710b11dc09076b03d7b308bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374041
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43751}
2025-01-16 10:55:53 -08:00
Philipp Hancke
da1c9e08ac
video_replay: fix wait when using start_timestamp
...
which should not wait for the amount of time between the initial packet
(which is ignored) and the packet video_replay was told to start at.
BUG=webrtc:382396709
Change-Id: Ic9c465cfa3e0ab66d9c2ff2e8e56a5bf419b8687
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370712
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43750}
2025-01-16 09:21:14 -08:00
Jeremy Leconte
808c9a66cd
Re-enable junit_tests (they haven't run since june 2024).
...
* Fix generate_buildbot_json script (https://chromium-review.googlesource.com/c/angle/angle/+/5792035 )
* Copy mixins that are removed from Chromium in mixins_webrtc.pyl
* Replace 'junit_test' by 'raw' test_type
Change-Id: Ie2642fc97c04d88e226b6c9f65b46e43942946a1
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43749}
2025-01-16 07:38:02 -08:00
Andrew Grieve
9135a02378
Delete reference to "no_build_hooks" GN variable.
...
It's a no-op and I'm going to remove it.
Bug: chromium:40114668
Change-Id: I165cf0132ef3f7505c8ffb0ab0611e644acbe6f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374380
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Auto-Submit: Andrew Grieve <agrieve@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43748}
2025-01-16 07:22:39 -08:00
Jeremy Leconte
0e642639a9
Add 'rtc_p2p_unittests' missing dependencies.
...
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/362480 .
Change-Id: Idba0153f3770a3de50d82b2dac62b345e3e2c9c8
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374480
Auto-Submit: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43747}
2025-01-16 05:11:41 -08:00
Danil Chapovalov
23b95d4fe4
Propagate field trials to aec3 sub components
...
Bug: webrtc:369904700
Change-Id: I17264de11346838b70ab2c47d6f6dc768e74b41a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374361
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43746}
2025-01-16 04:02:36 -08:00
Liad Rubin
1ba220a816
Migrate from MediaCodecList.getCodecCount to iterate getCodecInfos member function in MediaCodecVideoDecoderFactory
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GetCodecCount is deprecated in API 21+ and getCodecInfos is encouraged to be used instead.
See guidance: https://developer.android.com/reference/android/media/MediaCodecList#getCodecCount()
Bug: webrtc:386857228
Change-Id: I3578809a550ef069562c521e47ce58e42ab2f5a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374163
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43745}
2025-01-16 03:31:48 -08:00
webrtc-version-updater
1dcd3b4ae2
Update WebRTC code version (2025-01-16T04:03:05).
...
Bug: None
Change-Id: Iaf2b3aebb2066108c69aef91519e70548279cbda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374460
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@{#43744}
2025-01-15 22:15:08 -08:00
chromium-webrtc-autoroll
f422eeaa35
Roll chromium_revision e4f24f6380..839b9b8bb4 (1406593:1406733)
...
Change log: e4f24f6380..839b9b8bb4e4f24f6380..839b9b8bb402a2254b55..65bfd734ce86b5f23890..5720363cd07fee96e94b..dca9e8a29ab90022fc6a..0eadb2fa144d929882d7..3227982dbab8f83af38d..c5f991404fe4f24f6380..839b9b8bb4https://webrtc-review.googlesource.com/c/src/+/374248
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43743}
2025-01-15 08:39:06 -08:00
Philipp Hancke
bce57cda1e
Piggyback DTLS handshake in initial STUN packets
...
This change puts the DTLS handshake as payload of STUN packets with a custom STUN attribute (registered with the IANA) and starts the DTLS handshake before the ICE transport becomes writable. Effectively, STUN acts as a transport layer for DTLS during the handshake phase.
This will theoretically reduce the call setup time by one RTT for aggressive nomination or two RTTs for regular nomination.
The latest DTLS packet (flight) is cached and sent on every STUN request or response. DTLS packets are extracted from every authenticated STUN request or response and handled to the DTLS layer for processing.
The caching also increases the resilience to packet loss as STUN pacing is more aggressive (every 20ms) than the exponential backoff used by DTLS which should reduce call setup time in lossy networks.
If the other side of the connection does not support this feature the fallback to normal DTLS happens as soon as the ICE transport becomes writable. This also handles edge-cases like fragmentation of the DTLS handshake.
The feature is only supported when ECDSA certificates are used since RSA certificates are too large to transport as STUN attributes. The observed attributes for the server and client flights with the certificates were around 600 to 650 bytes. This may be further reduced by using raw public keys defined in RFC 7250.
This feature is disabled by default and guarded by the field trial
WebRTC-IceHandshakeDtls
and requires experimentation and standardization before roll-out in the browser.
Parts of this landed in
https://webrtc-review.googlesource.com/c/src/+/370679
BUG=webrtc:367395350
Change-Id: I4809438b2a267c4690a9b2bd6f1766d2f959500d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362480
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43742}
2025-01-15 07:54:23 -08:00
Philipp Hancke
cfaba8fd2d
Measure SDP munging
...
by storing
[[LastCreatedOffer]] / [[LastCreatedAnswer]]
which are similar to the W3C equivalent but as
description objects instead of serialized SDP strings.
While rejecting all SDP munging is not feasible, this lets us
measure and reject certain modifications gradually.
Chromium metrics CL:
https://chromium-review.googlesource.com/c/chromium/src/+/6089633
This is measured at three points during the lifetime of a peerconnection:
* for the first SLD call
* when the connection is first established
* when the connection was established and is being closed
Note that the "first" SDP munging detected is returned which may hide that something uses more than one modification.
BUG=chromium:40567530
Change-Id: I964e3ee6e75f73b777d90556fac8691a6f3dc27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43741}
2025-01-15 07:38:45 -08:00
chromium-webrtc-autoroll
07349dc819
Roll chromium_revision 2c3bf1f0cb..e4f24f6380 (1406418:1406593)
...
Change log: 2c3bf1f0cb..e4f24f63802c3bf1f0cb..e4f24f6380f929f8df72..28b89c75d5031e7e777b..ce61bc84f970c34fdc34..02a2254b55ef37f6f688..86b5f23890891f1edbbb..7fee96e94b9a9142793a..a90906c74d17e065a2f2..4d929882d7062de17659..b8f83af38d2c3bf1f0cb..e4f24f6380https://webrtc-review.googlesource.com/c/src/+/374328
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43740}
2025-01-15 02:34:25 -08:00
Harald Alvestrand
f5d13267ae
Reland "Use Payload Type suggester for all codec merging"
...
This reverts commit b7abaee819771ca297bac4c51ec0daf62bd9a3fc.
Reason for revert: Suspicion that suspected breakage wasn't real
Original change's description:
> Revert "Use Payload Type suggester for all codec merging"
>
> This reverts commit 0bac2aae596771db020f01a57fee4828081fbc38.
>
> Reason for revert: Suspected breakages downstream
>
> Original change's description:
> > Use Payload Type suggester for all codec merging
> >
> > Bug: webrtc:360058654
> > Change-Id: Id475762253c427c1800c2352a60fc0121c2dc388
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364783
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43267}
>
> Bug: webrtc:360058654, b/375132036
> Change-Id: Ieda626270193e7e6c93903b3c03a691b2bf0c1e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366540
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43290}
Bug: webrtc:360058654, b/375132036
Change-Id: Id6e72f7aac81023da43de7627c24dd1a792ea461
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374304
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43739}
2025-01-15 02:17:56 -08:00
webrtc-version-updater
3e384c7436
Update WebRTC code version (2025-01-15T04:05:49).
...
Bug: None
Change-Id: Icdd102267f9179329af657325d0919aa2b38e1f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374327
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@{#43738}
2025-01-14 21:08:48 -08:00
chromium-webrtc-autoroll
7cb91b2e81
Roll chromium_revision fedc00ae77..2c3bf1f0cb (1406283:1406418)
...
Change log: fedc00ae77..2c3bf1f0cbfedc00ae77..2c3bf1f0cb826afad130..031e7e777b555ebb1ed0..70c34fdc346157451a6a..ef37f6f6884b3b2479cb..891f1edbbbhttps://boringssl.googlesource.com/boringssl.git/+log/305e5c570b..571a7432a1
* src/third_party/depot_tools: 4c54361841..9a9142793acbada99a33..8205ccf0f296cdfc15f0..b90022fc6aefc96bb324..17e065a2f24233a3bd57..062de17659fedc00ae77..2c3bf1f0cbhttps://webrtc-review.googlesource.com/c/src/+/374326
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43737}
2025-01-14 16:41:00 -08:00
chromium-webrtc-autoroll
cffff6f30d
Roll chromium_revision 18e51f86e8..fedc00ae77 (1406109:1406283)
...
Change log: 18e51f86e8..fedc00ae7718e51f86e8..fedc00ae77ae1f3dd505..f929f8df72f1eca0eef3..826afad130e212dc9e53..555ebb1ed0cee2aaad61..6157451a6a1ab59ebcf0..4b3b2479cbhttps://boringssl.googlesource.com/boringssl.git/+log/bf4cf6938a..305e5c570b
* src/third_party/depot_tools: b576ab3b78..4c54361841b86e98ff11..be49f904cc823412c530..96cdfc15f04596ae7456..efc96bb324bdbb5e679f..4233a3bd5718e51f86e8..fedc00ae77https://webrtc-review.googlesource.com/c/src/+/374325
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Cr-Commit-Position: refs/heads/main@{#43736}
2025-01-14 13:05:11 -08:00
chromium-webrtc-autoroll
6daec622c3
Roll chromium_revision b0a681ce87..18e51f86e8 (1405977:1406109)
...
Change log: b0a681ce87..18e51f86e8b0a681ce87..18e51f86e8ee46c04a01..e212dc9e5378eec180e1..cee2aaad61e4564d246c..1ab59ebcf0597b2e8c42..823412c5302eab41ff13..4596ae7456feddf11766..bdbb5e679fb0a681ce87..18e51f86e8https://webrtc-review.googlesource.com/c/src/+/374323
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43735}
2025-01-14 08:36:56 -08:00
Mirko Bonadei
a4f557f840
Update target_sdk_version from 23 to 24.
...
Bug: chromium:385432898
Change-Id: I39a2976f666692e9fdff334adbedd4aae61bf04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374120
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43734}
2025-01-14 07:42:28 -08:00
Danil Chapovalov
c329350008
Propagate field trials into EchoCanceller3
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Bug: webrtc:369904700
Change-Id: I698dd126f1627f84abe2633bde215c06aeef6299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43733}
2025-01-14 07:41:22 -08:00
Danil Chapovalov
63b914d178
Relax FieldTrials construction parameter to string_view
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Bug: webrtc:42223790
Change-Id: I8ae85a9870aee86f021249a2652555f9c882b338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374162
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43732}
2025-01-14 06:07:10 -08:00
Philipp Hancke
11edd41c04
dtls_utils: skip over DTLSCiphertext from DTLS 1.3
...
where we can not extract any message_seq for piggybacking.
BUG=webrtc:367395350
Change-Id: I2e42d3b1304b5cf204c34bc8a136f17c363e011c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43731}
2025-01-14 03:37:29 -08:00
