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}
This collision can occur when we have
asymetrical send and receive codecs. This is the case in the current
code base with the VP9 codec familly but is not visible untill more
codecs are added.
Added Nutanix Inc. to AUTHORS.
Bug: chromium:1291956
Change-Id: I09d3f76161d984d2a3edf721639753bffd4947b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250034
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35944}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
This reverts commit 3babb8af238a531cbff27951604b09bb78b762cd.
Reason for revert:
- Causes regressions to transceivers, see https://crbug.com/1291956 for more information, including tests to reproduce the issue.
This CL is not a pure revert. While it reverts everything else, it does
keep the new enum value (kProfilePredictiveHigh444). This is as to not
break Chromium which already depend on it. It is not listed in the
kProfilePatterns though so the enum value should never be applicable.
Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
>
> Bug: chromium:1251096
> Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> 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@{#35684}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1251096, chromium:1291956
Change-Id: Ib4d8ea4898f9832914d88e7076e6b39da0c804ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249791
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35835}
Context: The timer precision of PostDelayedTask() is about to be lowered
to include up to 17 ms leeway. In order not to break use cases that
require high precision timers, PostDelayedHighPrecisionTask() will
continue to have the same precision that PostDelayedTask() has today.
webrtc::TaskQueueBase has an enum (kLow, kHigh) to decide which
precision to use when calling PostDelayedTaskWithPrecision().
See go/postdelayedtask-precision-in-webrtc for motivation and a table of
delayed task use cases in WebRTC that are "high" or "low" precision.
Most timers in DCSCTP are believed to only be needing low precision (see
table), but the delayed_ack_timer_ of DataTracker[1] is an example of a
use case that is likely to break if the timer precision is lowered (if
ACK is sent too late, retransmissions may occur). So this is considered
a high precision use case.
This CL makes it possible to specify the precision of dcsctp::Timer.
In a follow-up CL we will update delayed_ack_timer_ to kHigh precision.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/net/dcsctp/rx/data_tracker.cc;l=340
Bug: webrtc:13604
Change-Id: I8eec5ce37044096978b5dd1985fbb00bc0d8fb7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249081
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35809}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
The interface is implemented by the ChannelManager and contains methods
to create and destroy media channel objects as used by a transceiver.
This will subsequently allow us to delete the channel objects from
the transceiver class where ownership really lies rather than from
the outside - which is currently required by some tests that keep
channel objects on the stack. We'll furthermore be able to do the
destruction asynchronously without additional Invoke()s as we do now
which will remove an Invoke when making sdp changes.
With introducing the interface, the following simplifications were made:
* ChannelManager constructed on the signaling thread.
Before, there was an Invoke in the context class, which existed
for the purposes of calling MediaEngine::Init() (which in turn is
only needed for the VoiceEngine). This Invoke has now been moved
into the CM (more tbd).
* The CM now has a pointer to the signaling thread (since that's the
construction thread). That allows us to remove the signaling thread
parameter from the CreateFooChannel methods.
* The ssrc_generator (UniqueRandomIdGenerator) instance for SSRCs moved
from SdpOfferAnswerHandler to the CM, as it's always used in
combination with the CM. This simplifies the CreateFooChannel methods
as well as a couple of other classes that have a CM dependency.
* Removed DestroyFooChannel related code from SdpOfferAnswerHandler since
the channel type detail can be taken care of by the CM.
Bug: webrtc:11992, webrtc:13540
Change-Id: I04938a803734de8489ba31e6212d9eaecc244126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247904
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35766}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
Currently if encoder initialization fails WebRTC doesn't send any video.
This CL adds functionality that changes encoder type in such case and
restores the video. If encoder selector is available we switch to
encoder it recommends. Otherwise, VP8 is used as the default fallback
encoder.
Bug: webrtc:13572
Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35761}
WebRTC can switch encoder on-fly when encoder fails or by request from
encoder selector. Putting the current send codec to the front of the
codecs list provides a simple way for apps to know what is actually
used without retrieving stats.
Bug: webrtc:13572
Change-Id: Iaaa5f7ad8667f59016dc92bff9e9a57a7425ef44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246500
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35723}
This is a simple check for upcoming changes for media channels to
be able to check if the state on the network thread is consistent.
Bug: webrtc:11992
Change-Id: I8ed2d091ecf3869a66970fc4733aebf209c4ef82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246681
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35706}
This can cause issues on Android S if this initialization happens when
the app doesn't have permission to access the microphone.
Bug: b/197461765
Change-Id: Iebccff9d15f5bb12a7b2c78e1c373e379b37a127
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246104
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35689}
Added Nutanix Inc. to the AUTHORS file.
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
Bug: chromium:1251096
Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
Reviewed-by: Niels Moller <nisse@webrtc.org>
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@{#35684}
