Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
Because DirectTransport is switched on SimulatedPacketReceiverInterface
we can't create it from some specific config in ctor, so all ctors,
that accept specific config are deprecated and you should pass concrete
implementation of underlying implememntation instead.
Bug: webrtc:9630
Change-Id: I7f241f310c993d8136b40898e55a6915924d61bd
Reviewed-on: https://webrtc-review.googlesource.com/94841
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24344}
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.
This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.
Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/91840, which needed
transitional methods while updating downstream code. This cl completes
the deletion, and can be landed after downstream code is updated.
Bug: webtrc:9584
Change-Id: I4d3654748973a4757a8d79bb93f524c630a0eca3
Reviewed-on: https://webrtc-review.googlesource.com/93285
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24329}
Stop using of deprecated ctors of FakeNetworkPipe in most part of
webrtc codebase, except DirectTransport, where further refactoring will
be continued in future CLs.
Bug: webrtc:9630
Change-Id: I823404469e461601ddbc026eaeac668eeda8045f
Reviewed-on: https://webrtc-review.googlesource.com/94763
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24327}
Add replacements for all FakeNetworkPipe constructos, that will accept
instance of NetworkSimulationInterface instead of config to be able to
use any implmentation of network simulation.
Bug: webrtc:9630
Change-Id: Ifceb2f0d028faf255648891ce695b3742f866044
Reviewed-on: https://webrtc-review.googlesource.com/94541
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24320}
Switch from using FakeNetworkPipe::SetConfig on holding direct reference
on instances of NetworkSimulationInterface if you are required to
reconfigure it in the runtime.
Also add fake_network_unittests to built test targets.
Bug: webrtc:9630
Change-Id: I5fd6b33904367aa6dc00ca2e2f5f780f433acf35
Reviewed-on: https://webrtc-review.googlesource.com/94510
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24314}
Add FakeNetworkPipe ctor from NetworkSimulationInterface to be able to
inject different implementations of simulated networks.
If user requires to do something with injected simulation he can keep
raw reference, or introduce some proxy, that will hide shared ownership
from FakeNetworkPipe. Also FakeNetworkPipe guarantee to keep simulation
object alive while it is alive itself.
Bug: webrtc:9630
Change-Id: Ie00ea1d47bf7658b0c6433cf2efbf364f885c8a0
Reviewed-on: https://webrtc-review.googlesource.com/94153
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24312}
Switch on using NetworkSimulatedInterface in FakeNetwork pipe to be able
to inject different implementations in future.
Also temporary add SetConfig(...) method to NetworkSimulationInterface
to make it possible to use it in FakeNetworkPipe. This method will be
removed by futher refactoring.
Bug: webrtc:9630
Change-Id: I2ce2219f523b4121e46643699ab87b37da09d95b
Reviewed-on: https://webrtc-review.googlesource.com/94145
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24289}
This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.
Reason for revert: Breaks downstream project.
Original change's description:
> Add initial support for RtpEncodingParameters max_framerate.
>
> Add support to set the framerate to the maximum of |max_framerate|.
> Different framerates are currently not supported per stream for video.
>
> Bug: webrtc:9597
> Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> Reviewed-on: https://webrtc-review.googlesource.com/92392
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24270}
TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9597
Reviewed-on: https://webrtc-review.googlesource.com/94060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24277}
Add support to set the framerate to the maximum of |max_framerate|.
Different framerates are currently not supported per stream for video.
Bug: webrtc:9597
Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
Reviewed-on: https://webrtc-review.googlesource.com/92392
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24270}
It's useful to have one test without clock drift, to distinguish
between errors breaking handling of drift, and errors breaking A/V sync
generally.
Bug: None
Change-Id: Ibc1bdab142ef37cb37171b51c00c556907a5ba6e
Reviewed-on: https://webrtc-review.googlesource.com/93283
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24239}
When using the generic descriptor we want all simulcast streams to share one
frame id space (so that the SFU can switch stream without having to rewrite the
frame id). The state also needs to be restored when the RtpVideoSender is
recreated.
Note that |shared_simulcast_frame_id_| is only added, but not used in this CL.
Actually using it will be part of the next CL.
Bug: webrtc:9361
Change-Id: I7192a06d6ae4cab118ca5996ed99a56888ad1d97
Reviewed-on: https://webrtc-review.googlesource.com/92803
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24233}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
Downstream projects have been updated, so this can now be relanded.
This is a revert (and rebase) of: https://webrtc-review.googlesource.com/c/src/+/88820
Bug: none
Change-Id: I424664ddef7aeebd3c6c94ae67c7f70a342dc9a4
Reviewed-on: https://webrtc-review.googlesource.com/92082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24181}
Previously only alphanumeric characters were allowed.
Bug: webrtc:9537
Change-Id: I3fd793ad88520b25ecd884efe3a698f2f0af4639
Reviewed-on: https://webrtc-review.googlesource.com/89388
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24167}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: Ia58a3b4f3becf9e620d3991da8451d81f32f8ad0
Reviewed-on: https://webrtc-review.googlesource.com/90406
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24118}
It was moved to api/video_codecs/video_encoder_config.h in cl
https://webrtc-review.googlesource.com/77683.
Bug: webrtc:8830
Change-Id: I197fd3270d3dea0a5ec98b22cc675c407c388e93
Reviewed-on: https://webrtc-review.googlesource.com/90243
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24092}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd.
Reason for revert: Reland by removing the conflict with the broken CL.
Original change's description:
> Revert "Move allocation and rtp conversion logic out of payload router."
>
> This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
>
> Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
>
> This causes a merge conflict. So need to revert this first.
>
> Original change's description:
> > Move allocation and rtp conversion logic out of payload router.
> >
> > Makes it easier to write tests, and allows for moving rtp module
> > ownership into the payload router in the future.
> >
> > The RtpPayloadParams class is split into declaration and definition and
> > moved into separate files.
> >
> > Bug: webrtc:9517
> > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> > Reviewed-on: https://webrtc-review.googlesource.com/88564
> > Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23983}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9517
> Reviewed-on: https://webrtc-review.googlesource.com/88821
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23991}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org
Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/89020
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23996}
This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
This causes a merge conflict. So need to revert this first.
Original change's description:
> Move allocation and rtp conversion logic out of payload router.
>
> Makes it easier to write tests, and allows for moving rtp module
> ownership into the payload router in the future.
>
> The RtpPayloadParams class is split into declaration and definition and
> moved into separate files.
>
> Bug: webrtc:9517
> Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> Reviewed-on: https://webrtc-review.googlesource.com/88564
> Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23983}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/88821
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23991}
Makes it easier to write tests, and allows for moving rtp module
ownership into the payload router in the future.
The RtpPayloadParams class is split into declaration and definition and
moved into separate files.
Bug: webrtc:9517
Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
Reviewed-on: https://webrtc-review.googlesource.com/88564
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23983}
This ensures the event logs in CallTest will be used by default.
Bug: webrtc:9510
Change-Id: I9df82b5ef39e1b2cba2789f8c5c7a9fed3c4c2f6
Reviewed-on: https://webrtc-review.googlesource.com/88562
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23970}
This prepares for replacing single instance members with vectors in a
follow up CL.
Bug: webrtc:9510
Change-Id: Ie05436ec89a0af9ce9fe9cece9842a39227246ec
Reviewed-on: https://webrtc-review.googlesource.com/88180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23968}
This makes it possible to use them in VideoQualityTest and prepares for
allowing saving logs in other tests as well.
Bug: webrtc:9510
Change-Id: I3b1cc187d88e4f17745414433c2f96efd836a302
Reviewed-on: https://webrtc-review.googlesource.com/88561
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23964}
This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend.
Bug: webrtc:9517
Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7
Reviewed-on: https://webrtc-review.googlesource.com/88241
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23936}
Removes checks that are not relevant to the particular tests. The checks
create dependencies on the CallTest base class. This prepares for
further refactoring in CallTest.
Bug: webrtc:9510
Change-Id: Ie6b0093a8fcb8a152ca58f421727c4b085b60a87
Reviewed-on: https://webrtc-review.googlesource.com/87845
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23930}
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.
Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
The purpose is to make the fixture reusable in downstream
projects. The CL adds the following things to API:
- api/test/video_quality_test_fixture.h
- api/test/create_video_quality_test_fixture.h
The following things are moved to API:
- call/bitrate_constraints.h (api/bitrate_constraints.h)
- call/simulated_network.h (api/test/simulated_network.h)
- call/media_type.h (api/mediatypes.h)
These are required by the params struct passed to the
fixture. I didn't attempt to split the params struct into
an internal-only and public version in this CL, and as
a result we need to pull in the above things. They are
quite harmless though, so I think it's worth it in order
to avoid splitting up the test config struct.
This CL doesn't solve all the problems we need to
implement downstream tests; we probably need to upstream
tracing variants of FakeNetworkPipe for instance, but
that will come later. This puts in place the basic
structure for now.
Bug: None
Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911
Reviewed-on: https://webrtc-review.googlesource.com/69601
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23714}
Only user was the replay.cc tool, when dumping frames to a file. It is
changed to instead inject a special decoder.
Bug: None
Change-Id: I521fbba1a0ef440cff7d786f6f4c6397e33f764f
Reviewed-on: https://webrtc-review.googlesource.com/83121
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23675}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This experiment is now wired up inside of chrome using field trial and
this passthrough is now obsolete.
Bug: chromium:794608
Change-Id: I1407e391d39c7e8696add9f656f059e7d8a27a08
Reviewed-on: https://webrtc-review.googlesource.com/82780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23625}
