This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
This is part of the project to delete the class entirely.
The CL also adds an "use_rtx" parameter to the function for listing
video codecs, rather than filtering those away afterwards.
Bug: webrtc:13931
Change-Id: I96b9b18c694a1c0986ccf22face76ef4c704d372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262666
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36963}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
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}
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}
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}
Because of this (seemingly simple) change, I had to change the return
type of transport_name from `const std::string&` to `absl::string_view`
to handle the case when there's no transport assigned.
That in turn caused an avalanche of required updates.
Bug: webrtc:12230, webrtc:11993
Change-Id: I16ec6c6a5fc2f5f7c7de572355a3c6ca924bb9d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244084
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35617}
This makes things slightly simpler for the time being as surrounding
code is being refactored. This also removes a PostTask which has the
effect of shrinking the window between the Pending/Complete
notifications slightly since there's no additional async task
for the 'complete' step.
Bug: webrtc:11993
Change-Id: Ia86779b21c6f87301f37d763f89ace722e06e563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244081
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35609}
Removes a few constructors where similar ones existed.
Removes MediaConfig dependency from MediaChannel and fixes an iwyu.
Bug: none
Change-Id: I9e34a1da0852c3fb21222161fad315e70598db3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242966
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35608}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
The constants are being made private since no new code should use them.
However, the helper functions sill uses "AV1X" internally for backwards
compatibility.
Bug: webrtc:13166
Change-Id: I0a0cd46f31ca70bb7f395c9b1e9cdb202df11f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35289}
This field is unused within WebRTC, and doesn't seem to
be essential for any existing customers.
If this works well, it will be deprecated and removed.
Bug: none
Change-Id: I96d7485e4d094abfa6a8c3d1e6855834c13dedd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35263}
This change
- adds new type VideoTrackSourceConstraints expressing min/max FPS
constraints.
- adds new method VideoTrackSourceInterface::ProcessConstraints.
- adds new method VideoSinkInterface<>::OnConstraintsChanged.
- updates AdaptedVideoTrackSource and VideoBroadcaster to forward
the constraints to sinks.
- adds several unit tests for the added functionality.
- and finally, implements OnConstraintsChanged in VideoStreamEncoder.
Chromium will be updated in coming CLs to supply constraints set
through the MediaStream module.
go/rtc-0hz-present
Bug: chromium:1255737
No-Try: true
Change-Id: Iffef239217269c332a1aaa902ddeae2440929e22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235040
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35197}
along the lines of RTX handling but with limited support for missing
fmtp lines because of video/red.
BUG=webrtc:13178
Change-Id: Ia866c0e857da6da2ef1e4b81b51f90f534c7bb83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231948
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35107}
Setting different number of temporal layers is supported by SimulcastEncodeAdapter and LibvpxVp8Encoder will fallback to SimulcastEncoderAdapter if InitEncode fails.
Bug: none
Change-Id: I8a09ee1e6c70a0006317957c0802d019a0d28ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228642
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34785}
rtp_rtcp_format is lighter build target than rtc_media_base and
a more natural place to keep rtp parsing functions.
Bug: None
Change-Id: Ibcb5661cc65edbdc89a63f3e411d7ad1218353cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226330
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34504}
Remove unused functions GetRtpHeader/GetRtpHeaderLength
Replace usage of SetRtpHeader with webrtc::RtpPacket
Move SetRtpSsrc next to the only place it is used.
Bug: None
Change-Id: I3ecc244b1a2bdb2d68e0dbdb34dd60160a3101f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225547
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34447}
jitterBufferDelay and jitterBufferEmittedCount are defined
in RTCMediaStreamStats for both audio and video.
But for video, they were not populated in RTCInboundRtpStreamStats.
Bug: webrtc:12910
Change-Id: I135d473f055ecfb2c39b078ccf18c1bb9bc4f210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224280
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34398}
This reverts commit 096ad02c02b4bc6c046282b8793ef84d041dd0d8.
Reason for revert: Including a fix for the test issue.
Original change's description:
> Revert "Fix race between enabled() and set_enabled() in VideoTrack."
>
> This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a.
>
> Reason for revert: Breaks Chromium Android browser tests on fyi bots.
>
> Original change's description:
> > Fix race between enabled() and set_enabled() in VideoTrack.
> >
> > Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> > to VideoSourceBase except that it applies thread checks. I found that
> > it's easy to use VideoSourceBase incorrectly and in fact there appear
> > to be tests that do this.
> >
> > I made the source object const in VideoTrack, as it already was in
> > AudioTrack, and that allowed for making the GetSource() accessors
> > bypass the proxy thread hop and give the caller direct access.
> >
> > Bug: webrtc:12773, b/188139639, webrtc:12780
> > Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34096}
>
> TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12773
> Bug: b/188139639
> Bug: webrtc:12780
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#34101}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Change-Id: Ib35fe15a6c43de8f286d60aff02b19df1ab76925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219639
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34104}
This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a.
Reason for revert: Breaks Chromium Android browser tests on fyi bots.
Original change's description:
> Fix race between enabled() and set_enabled() in VideoTrack.
>
> Along the way I introduced VideoSourceBaseGuarded, which is equivalent
> to VideoSourceBase except that it applies thread checks. I found that
> it's easy to use VideoSourceBase incorrectly and in fact there appear
> to be tests that do this.
>
> I made the source object const in VideoTrack, as it already was in
> AudioTrack, and that allowed for making the GetSource() accessors
> bypass the proxy thread hop and give the caller direct access.
>
> Bug: webrtc:12773, b/188139639, webrtc:12780
> Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34096}
TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12773
Bug: b/188139639
Bug: webrtc:12780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34101}
Along the way I introduced VideoSourceBaseGuarded, which is equivalent
to VideoSourceBase except that it applies thread checks. I found that
it's easy to use VideoSourceBase incorrectly and in fact there appear
to be tests that do this.
I made the source object const in VideoTrack, as it already was in
AudioTrack, and that allowed for making the GetSource() accessors
bypass the proxy thread hop and give the caller direct access.
Bug: webrtc:12773, b/188139639, webrtc:12780
Change-Id: I022175c4239a1306ef54059c131d81411d5124fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34096}
This is to address flakiness of "DoubleThread" tests for the media
channel class. More investigation is in order though, so I'm adding
a TODO. The bug appears to be in test code only though, so this is
just to deflake the bots.
Bug: webrtc:12783
Change-Id: Ib6cf78927f2a9be9d2c6aa7f6915b1131a206e7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34049}
Xcode 12.5 triggers some warnings for -Wdeprecated-copy, and I believe
it is better to fix this problem than to suppress this warning.
Bug: webrtc:12749
Change-Id: I5ca5fd8fdcae18fe7d3941f78b3366b5f03b8c00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33990}
