The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
The last run-time logic for selecting function pointers was removed in
May 2016, here: https://codereview.webrtc.org/1955413003
It would be even better if we could eliminate the function pointers
entirely and just have different implementations that we select at
compile time; I've left a TODO asking for this.
Bug: webrtc:9553
Change-Id: Ica71d71e19759da00967168f6479b7eb8b46c590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144053
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28414}
[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}
Only remaining user is WavReader. Demote its constructor
accepting a PlatformFile to private, to refactor implementation
in a later cl.
Bug: webrtc:6463
Change-Id: I7b950be6f02073cb135dd0fab1190b9dc0de1fba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144025
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28410}
In debug mode I hit the assert so this function can return 0. So
just skip the window in that case like for other desktop elements
Bug: None
Change-Id: I92abf2a1f450b677632f5eb4332ca218cfd850ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143860
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28408}
This is part of a series of CLs with the goal to make the network
state estimate fields more semantically useful.
Bug: webrtc:10742
Change-Id: I3185dc8c4574710a4dce44a4ffd6cefe0fb4e8cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144036
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28406}
Unlike TimeToSendPadding(), the new GeneratePadding() method will
generate RTP packets and put them in the pacer queue, which will be
responsible for actually sending them.
A slight difference from previous logic is that we do not use a lower
bound of 50bytes for getting payload packets, instead we always try and
then abort if the next padding packet is larger than the current
available budget.
Since we're not sending the packets immediately, we don't need to worry
about twcc sequence numbering or updating the stats, that will be
handled by the general SendPacket() codepath. We can also omit the
PacingInfo struct and the return value of bytes sent, as that will
be handled when taking the packets out of the queue.
Bug: webrtc:10633
Change-Id: I066c292805a0bf76c59f68e66c21ea23fdb56c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143794
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28403}
This is a reland of 0ded32d5a3d7acb9a00c3a1d9941e539aa94eee5
Original change's description:
> Reland "Change buffer level filter to store current level in number of samples."
>
> This is a reland of 87977dd06e702ed517f26235c12e37bd927527c7
>
> Original change's description:
> > Change buffer level filter to store current level in number of samples.
> >
> > The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> >
> > Bug: webrtc:10736
> > Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28368}
>
> Bug: webrtc:10736
> Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28393}
Bug: webrtc:10736
Change-Id: I251b8321e5a5fd870e018bc7c8083ec0a41de81b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144023
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28398}
This reverts commit 0ded32d5a3d7acb9a00c3a1d9941e539aa94eee5.
Reason for revert: breaks downstream projects.
Original change's description:
> Reland "Change buffer level filter to store current level in number of samples."
>
> This is a reland of 87977dd06e702ed517f26235c12e37bd927527c7
>
> Original change's description:
> > Change buffer level filter to store current level in number of samples.
> >
> > The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> >
> > Bug: webrtc:10736
> > Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28368}
>
> Bug: webrtc:10736
> Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28393}
TBR=henrik.lundin@webrtc.org,minyue@webrtc.org,jakobi@webrtc.org
Change-Id: I570c83ec3a88a24d7a1f883a351748dd71bea015
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144022
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28397}
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 CL enables to change surface_ice_candidates_on_ice_transport_type_changed
in RTCConfiguration via PeerConnection::SetConfiguration.
Bug: None
Change-Id: Ib7bc8a08bfc9bf59cf07fe217c6f57d0d63615f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143561
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28394}
This is a reland of 87977dd06e702ed517f26235c12e37bd927527c7
Original change's description:
> Change buffer level filter to store current level in number of samples.
>
> The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
>
> Bug: webrtc:10736
> Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28368}
Bug: webrtc:10736
Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28393}
This fixes a bug where NACK mode was not properly enabled
due to missing send side configuration.
Bug: webrtc:9510
Change-Id: I318fdf44f17e57d30589115a452f6a64f81ee973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28391}
When provided, these thresholds will be used instead of WebRTC default
limits specified in DropDueToSize() and GetMaxDefaultVideoBitrateKbps().
Bug: none
Change-Id: Ida45ea832041963b8b8475d69114b5c60a172fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142170
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28390}
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}
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28385}
Using relative paths for JNI includes is causing build failures in chromium.
WebRTC already uses full include paths for generated JNI headers, so this CL
just removes the "jni_package" parameter from WebRTC generate_jni() targets
and removes the "jni/" portion of includes. The "jni_package" variable will be
removed from the generate_jni() template shortly.
To fix includes:
find . -name *.cc -exec sed -i -E 's@(#include.+generated.+jni)/jni/(.+_jni.h)@\1/\2@' {} \;
See https://groups.google.com/a/chromium.org/forum/?#!topic/java/MEovGrAwbqI
for discussion on naming scheme.
No-Try: True
TBR: kwiberg@webrtc.org
Bug: chromium:964169
Change-Id: I758c1b41bf6f5005587e55b82f14065fe251baad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143521
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28380}
