This CL allows for FEC protection of packets with VideoTimingExtension by
zero-ing out data, which is changed after FEC protection is generated (i.e
in the pacer or by the SFU).
Actual FEC protection of these packets would be enabled later, when all
modern receivers have this change.
Bug: webrtc:10750
Change-Id: If4785392204d68cb8527629727b5c062f9fb6600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143760
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28396}
This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the video side with the spec-compliant `SourceTracker`-implementation.
The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.
Bug: webrtc:10545
Change-Id: I895b5790280ac94c1501801d226c643633c67349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143177
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28386}
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
Some downstream clients have custom frame types that can't be converted.
The rest of EncodeVideoFrame is protected against these frames, but the
crop code assumes ToI420 always succeeds.
Bug: None
Change-Id: I8f4279e3975d3ae8cd1da59f7e84fafe0404fd15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141646
Commit-Queue: Noah Richards <noahric@chromium.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28256}
Before this change the max bitrate could be updated after it was passed
to rate allocator.
Bug: none
Change-Id: I742fca0f122bef3e95c1a768d6e844f8c28b6279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141661
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28253}
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
It's easy to make small errors when building field trial strings, and
those errors can cause all sorts of weird problems. This CL checks if
the FT string has an odd number of delimiters, duplicate
names or any trailing chars.
If so we'll log a error message. On debug builds we'll also crash.
Bug: webrtc:10729
Change-Id: Iebf7155d9b117a02d1e9cfe7f64408e11df2aec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140866
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28234}
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
In several places VideoFrame::Builder is used to create a new VideoFrame
when intent is to change only one or two fields of a const VideoFrame&.
This approach is bad because each and every metadata field have to be
added to all the places.
Instead, this CL adds missing setters and refactors the code to use
full copy of a VideoFrame and update required fields only.
Along the way few actual bugs are fixed, e.g. when ColorSpace isn't copied
when frame rotation or buffer is cropped or converted.
Bug: webrtc:10460
Change-Id: I2895a473ca938b150eed2916c689060bdf58cb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140102
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28170}
Some of the TODOs associated with webrtc:10336 which are
currently in the codebase have recently been resolved,
but not all relevant TODOs have been removed.
TBR=kwiberg@webrtc.org
Bug: webrtc:10336
Change-Id: Iff1d0fc94dee5bf49226f6ea3d9127fea77e9d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139902
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28167}
Replaced by separate methods
SendPictureLossIndication and SendFullIntraRequest.
The split SetKeyFrameRequestMethod/RequestKeyFrame implicitly
requires that the two methods are called on the same thread, to avoid a
data race. After downstream code is updated, both deprecated
methods and the member |ModuleRtpRtcpImpl::key_frame_req_method_| can
be deleted.
Bug: None
Change-Id: I454f6d16b667f2306cba0dec467ddc183ad449c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140043
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28163}
Reset() is called each time the encoder is reconfigured, but then it
happens the target bitrate isn't reset in encoder. So it might produce a
frame before next bitrate estimate is propagated to the metadata writer.
The incorrect zero bitrate would be treated as a paused encoder and would
cause metadata to be dropped.
Also, added unittest for that scenario at VideoStreamEncoder level.
Bug: webrtc:10460
Change-Id: I28024a527f1fb8474b172e2c5c2394fd38d69a07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140101
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28159}
Currently, if LNTF and NACK messages are both created, they will
be sent out in separate RTCP messages. This is wasteful.
This CL is the first of in a series of CLs that will ensure that
these feedback messages can be buffered together, without introducing
more of a delay than the CPU time required to process both messages.
Bug: webrtc:10336
Change-Id: I950324112ee346695a12a17d025483ea5e99c732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139112
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28136}
This is a partial revert of
https://webrtc-review.googlesource.com/c/src/+/130101.
The KeyFrameRequestSender argument is added back to the constructor of
RtpVideoStreamReceiver. It is optional; if a null pointer is passed,
key frame requests are sent via the internal RtpRtcp module, and this is
how the class is used by VideoReceiveStream. An injectable
KeyFrameRequestSender is useful for tests, for downstream applications
that want to route key frame requests elsewhere, and should also aid
later migration to RtcpTransciever.
Bug: None
Change-Id: Idf9baeed21570625ad74e9afbe38f7ea5bf79feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139107
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28102}
Currently used ':' is bad because it prevents from specifying absolute
path on windows (e.g. "C:\directory").
Now to specify path on windows, it can be passed unchanged in field trial:
"WebRTC-DecoderDataDumpDirectory/C:\path\on\windows/"
On linux ';' has to be used instead of '/':
"WebRTC-DecoderDataDumpDirectory/;path;on;linux/"
Bug: none
Change-Id: Ia46c94bdfab95385618dde4fd3f2857e9ddf2d1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138832
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28085}
LNTF (loss notifications) are no longer mutually exclusive with
NACK and key frames; a receiver may send both, and the sender would
be allowed to choose which to regard.
Bug: webrtc:10336
Change-Id: I1ae7d972f9f47b07fe2f493bb31c1916456d6a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138828
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28081}
Reland with fixes.
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: Id40a780d461e6b51cb44d275b8aa5d7b348d3586
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138215
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28054}
This reverts commit eb1754c5750dfcad23ac62b47aa3aa2176ae7be2.
Reason for revert: breaks downstream projects
Original change's description:
> VP9 screenshare: Don't base layers frame-rate on input frame-rate
>
> If input framerate is a little unstable, using it to cap layers will
> make output framerate even smaller for longer periods of time.
>
> Also, fix screenshare_loopback test for low-fps vp9 testing.
>
> Bug: webrtc:10257
> Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28050}
TBR=ilnik@webrtc.org,ssilkin@webrtc.org
Change-Id: I82bfbac58249cfe0da5ff565aa97a4745fd078ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138213
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28051}
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28050}
This is a standardized metric:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
We estimate the target frame size in bytes from the current encoder
target bitrate and encoder framerate.
We would expect that the average bytes produced by the encoder would
over time match the average target, which is calculated by polling
getStats() twice and dividing the delta totalEncodedBytesTarget with
the delta framesEncoded. This is meant to make googTargetEncBitrate
obsolete.
Bug: webrtc:10446
Change-Id: Ib10ce236476a2f965582d5c536f419952926d4e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137200
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28022}
RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.
Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28020}
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.
TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.
Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
RtpVideoStreamReceiver used to pass the PacketRouter when creating its
RtpRtcp module, but it's not needed for a receive-only module. Make the
PacketRouter optional to the constructor; it's used only for registering
the created RtpRtcp module as a candidate for sending rtcp feedback.
Bug: None
Change-Id: I371a0bdb9d68ac48b16f52e1d7939f8c177dc528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137429
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27984}
