Minor refactoring of the API, to put optional arguments last. Also
changed internal structures to reflect that order, for consistency.
Also reduced size of Item from 88 to 72 bytes, by packing fields better.
Bug: webrtc:5696
Change-Id: I1b9d50831a8e9a358224682d06a782a3269b8416
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264123
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37403}
This flag has gone unused for a long time, time to clean it up.
While we're here, convert NackRequester to use unit types.
Bug: webrtc:8624
Change-Id: I1f314f9b5b6771d4f9c351a7a9a887130b86907c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267408
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37400}
Let the OutgoingStream reference the parent instead of passing
references to individual items it needs, as follow-up CLs will add even
more items.
No functional change - pure refactoring.
Bug: webrtc:5696
Change-Id: I914e590c0d90e898d7d230a16170cf4faff2338c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264142
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37398}
Return the bitrate estimate as DataRate type
Remove list of affected ssrcs as unused
Bug: None
Change-Id: Ie31dce591d861624736d834194f90eb6c93f70f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37397}
This CL adds the API to enable message lifecycle events to be generated.
Those can in turn be used to generate metrics, e.g. latency metrics
tracking the time to send a message, the time until it's acknowledged,
and metrics tracking how often messages are expired.
This will be used to validate that message interleaving really improves
latency for high priority data channels.
The actual implementation of the API will be provided in follow-up CLs.
Bug: webrtc:5696
Change-Id: Ic06f8244d1c79a336975e35479130521dff17519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37396}
For this I added a header called no_cfi_icall.h and use it.
Also, some files use the gio header, but if the //base dependency is
not used, compilation errors occur. So I added an explicit dependency
on gio.
Bug: webrtc:13662
Change-Id: If732ede202dd413be6702bf06bf024cd203fdae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267340
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37395}
which is useful for validating flexfec recovers frames correctly.
This can be tested by passing a keyframe covered by flexfec along
with the fec packets and removing one packet from the frame.
BUG=None
Change-Id: Icd73eca138f62b9387bf850a6efbd7db03b4b569
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261956
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37392}
This is an error that puts the PC into an inconsistent state, so
causing a crash is the right thing to do.
Bug: chromium:1341043
Change-Id: Ie1eb89400ad87f0c83634b7073236b07e92ec7ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267281
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37391}
which is already supported in modules/video_coding/utility/ivf_file_reader.cc
BUG=None
Change-Id: I7b00659dc460d372312dff2eb53837a321ab16af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37390}
There was also some refactoring to create the TCB at the same time,
to ensure the metric is always set.
Bug: webrtc:13052, webrtc:5696
Change-Id: I5557ad5f0fc4a0520de1eaaafa15459b3200c4f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262259
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37388}
Before this CL, some components, e.g. the SendQueue, was first created
and then later restored from handover state, while some were created from
the handover state, as an optional parameter to their constructors.
This CL will make it consistent, by always creating the components in a
pristine state, and then modifying it when restoring them from handover
state. The name "RestoreFromState" was used to be consistent with SendQueue
and the socket.
This is just refactoring.
Bug: None
Change-Id: Ifad2d2e84a74a12a93abbfb0fe1027ebb9580e73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267006
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37384}
Allow to use AnyInvocable in webrtc.
Demonstrate how AnyInvocable can make interface clearer to read and implement
Demonstrate that AnyInvocable can reduce binary size
Bug: None
Change-Id: I33cf33fac6ed3bf4c5e46077d1cd984ca0f253a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267165
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37380}
since the fec packets are initialized to 0 there is no need
to special-case the first packet since
A XOR 0
is the identity operator.
BUG=None
Change-Id: I0cb55283ecdca06f8e3a7b5856ec1f9fbbad1ffb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251522
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37378}
Currently, a default max bitrate is determined within WebRtcVideoEngine,
which maxes out at 2.5Mbps - and that limits the max bitrate deteremined
by SvcConfig for resolutions above 720p.
This does not affect simulcast, as WebRtcVideoEngine already knows to
trust the rate allocation in simulcast.cc instead.
Bug: webrtc:14017
Change-Id: I0c310a6fd496e9e5a10eae45838900068aa1ae2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267160
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37370}
This reverts commit 915d419eb94782ce424ae59c07d8b4366e88be74.
Reason for revert: The bots which can run this tests is now working
Original change's description:
> Temporarily disable video_capture_tests on linux
>
> No bots can run linux video_capture_tests for now.
>
> Bug: webrtc:14220
> Change-Id: I5dbdefdd104befff92648a0693bee331d2b50e88
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266660
> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#37312}
Bug: webrtc:14220
Change-Id: I0a7a82cf64b0692be62a38f3df6416a33917c3d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267105
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37369}
Look for first echo (and not only the strongest one) on the same matched
filter.
This change is bit exact with previous version when `pre_echo` is false.
Author: Jesús de Vicente Peña <devicentepena@webrtc.org>
Bug: webrtc:14205
Change-Id: I6782eaa1d690b0df78d00f6d425a85c951b2ca9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266321
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37360}
In H264, reordered packets can cause a frame following padding to become stuck in the packet buffer.
A minimal example:
_, P, 1 - padding packet p and frame 1. Frame 1 has not been returned because of missing packet 0
0, P, 1 - when packet 0 arrives, FindFrames will stop incrementing i when it sees padding packet P, and frame 1 will never be returned
Bug: webrtc:14216
Change-Id: I78b76df9709fa8593c5025d647e2b868af3749f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266465
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37357}
This adds support to enable message interleaving in the stream scheduler
from the socket, proxied by the send queue.
It also adds socket unit tests to ensure that prioritization and
interleaving works. Also, send queue test has been added to validate the
integration of the stream scheduler. But the actual scheduling parts of
it will be tested in the stream scheduler unit tests.
Bug: webrtc:5696
Change-Id: Ic7d3d2dc28405c77a107f0148f0096882961eec7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262248
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37355}
