Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.
The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130.
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30654}
Before making it even more complicated that it is right now.
Bug: webrtc:10342
Change-Id: I54f67309b8832cd85b6c5213f9b090908814ebd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168766
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30557}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
with adjustments.
Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
This reverts commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
Reason for revert: breaks downstream tests
Original change's description:
> Do not propagate generic descriptor on receiving frame
>
> It was used only for the frame decryptor.
> Decryptor needs only raw representation that it can recreate
> in a way compatible with the new version of the descriptor.
>
> Bug: webrtc:10342
> Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30501}
TBR=danilchap@webrtc.org,sprang@webrtc.org,philipel@webrtc.org
Change-Id: I6634df06ee75aa8cdfda614994ab11f7a5845c70
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168488
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30502}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
Bug: webrtc:10342
Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30501}
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.
Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30404}
To allow to use the LossNotificationController with
an updated version of the frame descriptor extension
Bug: webrtc:10342
Change-Id: I5ac44dc5549dfcfc73bf81ad1e8eab8bd5dd136e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166166
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30369}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
instead of creating each time an object with RtpDepacketizer interface
this moves packet payload memcpy from RtpVideoStreamReceiver into
the depacketizers with possibility to remove it from there in follow ups.
Bug: webrtc:11152
Change-Id: If474207eb84d7e9d0207075bd395e60895f0d842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162185
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30095}
Together with RtpDepacketizer refactoring that would reduce
number of memcpy while handling an rtp packet
Bug: webrtc:11152
Change-Id: I6f4e09c93af5e2a9314967a15eac8ced57ec712e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161087
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29985}
it is easier to reduce and eliminate it when it is not bound to legacy video code
Bug: webrtc:10979
Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29821}
that brings RtpPacketReceived closer to the packet buffer
to allow strore original packets rather than VCMPacket in it.
Bug: webrtc:10979
Change-Id: Ia0fc0abf3551a843b19b0ee66ca0f20cae014479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157164
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29516}
In this CL:
- Moved critical section out of RtpFrameReferenceFinder.
- RtpFrameReferenceFinder can now assign picture ids with an offset.
- RtpVideoStreamReceiver will now reset the |reference_finder_| in case
of a codec switch.
Bug: webrtc:10795, webrtc:10828
Change-Id: I22631c121a465c434de24af5ce8be2a647fe3556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154353
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29317}
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.
Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29201}
The new target, modules/video_coding:video_coding_legacy, is not
depended upon by any webrtc non-test code.
Bug: webrtc:7408
Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29133}
Number of received FEC bytes is used for the
WebRTC.Video.FecBitrateReceivedInKbps UMA histogram. Before this cl,
that value is based on a FEC packet counter updated by
ReceiveStatistics::FecPacketReceived. This cl deletes that method, and
instead adds a byte count to the FecPacketCounter struct, which is
maintained by the UlpFecReceiver and used for other FEC-related stats.
Bug: webrtc:10917
Change-Id: I24bd494b6909a2fe109d28e2b71ca8f413d05911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150533
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28976}
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.
The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.
Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
That method is going away in favor in construction time setting.
Bug: webrtc:10774
Change-Id: I2aba5a2537e5846a3c9438a5b376b230e84c5f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149826
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28901}
Cname callback is used only on receive side, and statistics (soon)
only on the send side.
Bug: webrtc:10679
Change-Id: I122e9cafaea93cd0ba75dc955a652d9d4bddc379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147867
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@{#28767}
[1/2] - Make new version pure-virtual, and deprecated version non-pure.
This will allow deleting the deprecated version from downstream
projects.
[2/2] - Remove deprecated version.
TBR=stefan@webrtc.org
Bug: webrtc:10336
Change-Id: Ia132ef071b1f379fc74834178e75e981ca908125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144042
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28413}
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 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}
Loss notifications may either be sent immediately, or wait until another
RTCP feedback message is sent.
Bug: webrtc:10336
Change-Id: I40601d9fa1dec6c17b2ce905cb0c8cd2dcff7893
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28142}
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}
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}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
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}
