`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.
Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.
Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
The state machine for handling resets couldn't handle resets
happening from both sides at the same time.
Bug: webrtc:13994
Change-Id: I2c268e54f4c5c9858913faef91ff00f6af956e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261305
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36799}
This reverts commit 45361f78ed18c350b3edcaef19ae4c7cf167e95b.
Reason for revert: Perf alerts galore.
Original change's description:
> Calculate video stream max bitrate using expression.
>
> This replaces the ealier table-based caps.
> Apart from the VGA cap (now 1600kbps instead of 1700kbps), or if using
> "in between" resolutions, the caps are unchanged - but now cover high
> resolutions better.
>
> Bug: webrtc:14017
> Change-Id: I8649b528495d6c917e38ea8cb1a272df6c464c03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260940
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36776}
Bug: webrtc:14017
Change-Id: I18ebc81c6054713c58d49bd227e37090686958c9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261309
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36794}
This replaces the ealier table-based caps.
Apart from the VGA cap (now 1600kbps instead of 1700kbps), or if using
"in between" resolutions, the caps are unchanged - but now cover high
resolutions better.
Bug: webrtc:14017
Change-Id: I8649b528495d6c917e38ea8cb1a272df6c464c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260940
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36776}
When an SCTP stream is closing, a stream reset needs
to be sent from both ends.
The remote was not sending a stream reset and quickly
opening another stream with the same StreamID could
cause SCTP errors.
Bug: webrtc:13994
Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36773}
DSCP is controlled by the spec-compliant API
RTCRtpEncodingParameters.networkPriority[1]. It already has a default
value that is the same as when DSCP is disabled.
- If you want non-default DSCP default values, you need to set
networkPriority and shouldn't need to set a non-standard googDscp flag
for it to have an effect.
- If you want the default DSCP value, you wouldn't change
networkPriority and so you don't care if enable_dscp is true... you'll
get the default regardless.
Drive-by: This CL also adds crbug references to other goog flags.
[1] https://w3c.github.io/webrtc-priority/#dom-rtcrtpencodingparameters-networkpriority
Bug: chromium:1315574
Change-Id: I15a0470fa04f55e2534cee0d240eeb03446c2de6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36550}
Recent Clang versions have enhanced -Wunused-but-set-variable which now
warns about this.
Bug: chromium:1309955
Change-Id: Ie70df85f5a6d2cbabb4e10960bfd926ff7bf32cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257162
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Hans Wennborg <hans@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36381}
`enable_non_sender_rtt_` is updated a few lines below.
This seems to have been missed in the code review.
Bug: webrtc:12951, webrtc:13853
Change-Id: Idc06421362f6b2d831b5a828f296142aab9a46e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256860
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36380}
This patch takes a stab at modules/video_coding,
but reaches only about half.
Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.
Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.
Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
convert almost all of video/ (and the collateral)
Bug: webrtc:10335
Change-Id: Ic94e05937f54d11ee8a635b6b66fd146962d9f11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36192}
this has been enable by default since M96
BUG=webrtc:11640
Change-Id: I5d310d3929882007211eae12bc3ac1366107ca87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36123}
The variable instance_count might be accessed from multiple threads when
different PeerConnectionFactory objects are used, which may create
multiple network threads. This is a pattern mostly noticed in tests.
This fixes issues with logging when run under TSAN, it should not have
any production impact.
Bug: chromium:1243702
Change-Id: Iab1412a7907545811a309cab27a3ae23b4718606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251983
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36045}
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.
Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/
Also remove a hidden no-break space in dcSCTP logging causing issues in
some log parsing.
Bug: chromium:1243702
Change-Id: I46136a8913a6d803a3c63c710f3ed29523e4d773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251867
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36027}
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/
Bug: chromium:1243702
Change-Id: Ia90b796562245558a61481317bcded437400b045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251800
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36018}
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/235340
Bug: chromium:1251096
Change-Id: Icd997c7f7732229954d5494343b4e7a70deb09d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251303
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35964}
