TimestampExtrapolator maps RTP timestamps of received video frames
to local timestamps. As part of this mapping, the clock drift
between the local and remote clock is estimated.
Add the histogram WebRTC.Video.EstimatedClockDrift_ppm to log the
relative clock drift in points per million.
Bug: b/363166487
Change-Id: I0c2e628ef72c05a93e1f3138c8f71c77467130b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368342
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43413}
This is a reland of commit 775639e930f14a619974944594b40c633cc574a3
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
Change-Id: Idedf6249130bd01dd31261672c624b88c3f4c1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43412}
This seems to have no effect on tests, so it appears that these were
not used after all.
The goal is to make transport-cc a media-section-level attribute.
Bug: webrtc:378698658
Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43411}
Instead use the preprocessor to avoid compiling Perfetto related code
when RTC_USE_PERFETTO is not defined.
Bug: None
Change-Id: I85b37cb0287327035ac2e8feb3caf9505486a1e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43410}
stun_prober will fail on Windows and return RESOLVE_FAILED
Bug: webrtc:378688998
Change-Id: I3b957f6b2adf6658a0f6b83c8ff427ffd9779f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43405}
H.265 should have limits probably between VP9 and AV1, instead of using
VP8 tables. For now we reuse VP9 tables but eventually we may create
table for H.265.
Bug: chromium:41480904
Change-Id: I6dc2ec629142ee06f1c82a2df30d753ec1353496
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368240
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43404}
The gn target for rtc_pc_unittests cleared the "configs" that is by
default set for rtc_test. Restore it back so we get RTC_ENABLE_H265
macro when rtc_use_h265 is configured.
BUG: chromium:41480904
Change-Id: If172482776e5be2ad99d976db12dcfa556ee8a24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368183
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43403}
which is already gone from the code.
BUG=webrtc:40644300
Change-Id: Ic4a53d7895fa49d8417f11778d128740cecaee49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43401}
Dependency is required for Chromium roll in WebRTC.
Change-Id: I284c55f97bae3eee638d7a9f9fb5319fa1ae24e8
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43399}
This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated.
This should eliminate the need for the "force" flag in the field trial.
Bug: webrtc:42225697
Change-Id: Iec7a894c244a417a8499200861550a33f89966a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43398}
This reverts commit 775639e930f14a619974944594b40c633cc574a3.
Reason for revert: Breaks internal tests.
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
No-Try: true
Change-Id: I5485b1abfd5f586ec187cc57817940aa2efd72af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368200
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43396}
The workaround in https://webrtc-review.googlesource.com/c/src/+/367740
is incomplete because it does not fix the issue for the first decoded
mono packet after CN/PLC. This CL extends the workaround to such a case
and adds a unit test for it.
Note: it was verified that the 2nd packet after CN/PLC is trivial
stereo.
Credits: jakobi@webrtc.org for raising the concern
Bug: webrtc:376493209
Change-Id: Ide27e411781693f14629cf9db8b6c0c0fc762a17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368160
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43393}
This commit fixes the issue of video playback in slow motion caused by VCMTiming being unable to provide the correct rendering time in
scenarios of continuous network packet loss
WANT_LGTM=mbonadei
Bug: webrtc:376183208
Change-Id: I63617068506e536c4b812215ea084eec18e8ee06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367000
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43392}
H.265 does not have software fallback, and it may have issue supporting
more than 1 temporal layers on some devices. Set default to L1T1 when
scalability is not configured, or if a scalability mode is reported as
not supported by encoder.
Bug: chromium:41480904
Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43389}
Originally we used node id from PipeWire as an unique device name and
while this works, it will change everytime PipeWire is restarted. This
has an impact on default camera selection, where for example Firefox can
automatically request a camera device that was used before, but this can
break with the next PipeWire restart.
Bug: webrtc:42225999
Change-Id: I9440ee065ffeaa1ffb911a4dc7c405d57c9416dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367880
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43387}
The problem is that in some opt builds we remove RTC_DCHECK, and hence the test that expects death through RTC_DCHECK fails. Adding RTC_DCHECK_IS_ON ensures that test is coupled with builds that include RTC_DCHECK.
Bug: webrtc:358039777
Change-Id: I7721849dc76ce976ac4a084273cedad44f96fef2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366524
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43386}
The Android bots and the libfuzzer bots have an indirect dependency on base. This CL downloads the rust toolchain for these bots so that rust can be used in Chromium base/ without guards to prevent WebRTC from breaking.
Change-Id: I81e0a32827e8eee29f333d933d0fb21dc0b7dc23
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367921
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43384}
which launched a while back.
BUG=webrtc:40644399,webrtc:364825888
Change-Id: Ied1d76d8ab2cbb395e09c08f6354d99b4e082cef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367840
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43383}
This adds support of H.265 into the RTP video frame assembler, which is
now a public interface.
Bug: chromium:41480904
Change-Id: I74fd761949d0b095ba4526d2fa887e963f48abcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367603
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43374}
by going from cartesian product of combinations to an explicit
minimal list.
BUG=webrtc:375552698
Change-Id: I99b9afd7376f19abde54dafd8917954617d8c255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367504
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43373}
Adding a temporary workaround in the WebRTC Opus decoder wrapper to fix
https://issues.webrtc.org/376493209. Once the issue is fixed in libopus,
the workaround must be removed (TODO added in the code).
The workaround keeps track of the number of channels for the last
decoded packet and, if the decoder operates in stereo mode and the last
packet was a mono one, the left channel is copied into the right one
when comfort noise / PLC audio is generated.
Bug: webrtc:376493209
Change-Id: Iad3bfb1b393bd68833decf51b69b5238cb0ec4b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367740
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43371}
Also updated the test to cover IsTemporalLayersSupported() for all types
of codecs.
Bug: chromium:41480904
Change-Id: I25788a87737aba7308b1d6980ad5b2c26b0e225f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367570
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43369}
The std::lcm and std::gcd functions are part of the C++ standard
library. The existing functions are marked as deprecated rather than
deleted in the case of possible third party uses.
#rtc_cleanup
Bug: webrtc:377205743
Change-Id: I174e663f152d750c984a35dc7136bc18dc01bc8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367440
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43368}
Inactive encoders are included in the string when they are paused due to
bitrate allocation being 0 for that simulcast layer.
#rtc_ktlo
Bug: webrtc:376804631
Change-Id: I4234b452b60fee58981907380df41962fda5bf40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43367}
