Removing simulcast stream support as it was broken.
Bug: webrtc:9510
Change-Id: I42ba285bbea81e6ffd5b1d1a1aec4e5eb0990b1e
Reviewed-on: https://webrtc-review.googlesource.com/c/123040
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26684}
There was a name collision with downstream test frameworks.
Bug: webrtc:9510
Change-Id: I7e37a8a54701ef4a47c687aec51f37523759f1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/123044
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26683}
It doesn't do anything any more, so it should be removed.
Bug: webrtc:9586
Change-Id: I0b320b6ce4f480ff8cb59451db29bcc77b882b5f
Reviewed-on: https://webrtc-review.googlesource.com/c/120813
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26507}
Create audio stream instead of data channel to check compatibility of
network layer with PeerConnection. Replacement is done because there is
a data race inside data channel sctp transport. This CL will fix
bot behavior. Further data race investigation will be done in this
bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=10268
Bug: webrtc:10268, webrtc:10138
Change-Id: I4f7a1116c65dbf4a3508b7d81d654ccd320795f0
Reviewed-on: https://webrtc-review.googlesource.com/c/120807
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26495}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
This simplifies the design by making simulated network more self
sufficient. It also prepares for removing network node specific
configuration (The behavior implementation should be responsible
for handling any configuration.)
Bug: webrtc:9510
Change-Id: I218d70c0359774d9891178fbd8b1bbc729cbad92
Reviewed-on: https://webrtc-review.googlesource.com/c/120346
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26450}
It didn't have proper time to stabilize, making it sensitive to small
changes. This CL increases the stabilization period from 20 to 30s.
Also fixing some minor test suite bug found during investigation.
Bug: webrtc:9718
Change-Id: If56dba5383251ad3d3efe304eebcd880522afabe
Reviewed-on: https://webrtc-review.googlesource.com/c/119943
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26408}
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.
Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.
Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
This makes it possible to save log outputs from scenario tests to
either files or memory.
Bug: webrtc:9510
Change-Id: I883bd8240ab712d31d54118adf979041bd83481a
Reviewed-on: https://webrtc-review.googlesource.com/c/116321
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26284}
It is a step in the big refactoring to introduce new network emulation
layer for peer connection level e2e test, which will be based on system
sockets level injection.
Bug: webrtc:10138
Change-Id: Ie3854d22aa3eec289617bc432026ea670646556a
Reviewed-on: https://webrtc-review.googlesource.com/c/115943
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26137}
The return value is not used. This change prepares for future
refactoring by removing the requirement that TryDeliverPacket must be
synchronous. Also renaming to DeliverPacket as we no longer need to
indicate the meaning of the return value.
Bug: webrtc:9510
Change-Id: I78536434b198fa7bf4df88b10d6add23684767f1
Reviewed-on: https://webrtc-review.googlesource.com/c/115181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26066}
This makes it possible to test custom network controllers without
requiring update to test framework. Also updating BBR performance
test to use this feature.
Bug: webrtc:9510
Change-Id: I0446de0403fe9d1f6dc3710c1d114887a6c359c5
Reviewed-on: https://webrtc-review.googlesource.com/c/114640
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26046}
Preparation for replacing use of TestVideoCapturer as an interface,
instead using VideoSourceInterface.
Methods kept as non-virtual on the subclass FrameGeneratorCapturer,
but it's changed to be started on creation.
Bug: webrtc:6353
Change-Id: Iae1c9a0ee55d730d4992204f62227ef2f057d58e
Reviewed-on: https://webrtc-review.googlesource.com/c/114425
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26037}
Some tests had to be updated due to this change.
Bug: webrtc:9510
Change-Id: I79c4c0166d8ba5e8190a607d5d35b67dc30a3c14
Reviewed-on: https://webrtc-review.googlesource.com/c/113522
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25952}
The intention is to provide a bandwidth estimate that only updates if
the actual available bandwidth is known to have changed. This will be
used in media streams to avoid changing the configuration (such as
frame size, audio frame length etc), just because the control target
rate changed.
Bug: webrtc:9718
Change-Id: I17ba5a2f9e5bd408a71f89c690d45541655a68e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107726
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25772}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
This CL changes the behavior for RunFor and RunUntil so they do not
anymore restart the underlying streams every time they are called.
This has a side effect on the semantics of the calls. Previously,
both RunUntil and RunFor would restart the session and run until the
given time had passed. Now RunFor will still run for the provided
duration, however, to make the name of RunUntil more correct, it
will run until the time since start is equal to the max_duration
parameter. An extra overload of RunUntil was added to allow using
this behavior without providing an ending condition.
Bug: webrtc:9510
Change-Id: I9fe56a44116907fba3d102894b5c96af2ba6cffb
Reviewed-on: https://webrtc-review.googlesource.com/c/111502
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25726}
This prevents printing warning messages when the extensions aren't
found. The real parsing is done deeper in the stack and is unaffected.
Bug: webrtc:9510
Change-Id: Idf09f0e69c223bd4217be7044d21d1d0bbbdab92
Reviewed-on: https://webrtc-review.googlesource.com/c/110615
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25612}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
This makes it safer to reason about the common case where send
time information is available. We don't have to either assume that
it's available, or check it everywhere the PacketResult struct is used.
To achieve this, a new field is added to TransportPacketsFeedback
and a new interface is introduced to clearly separate which field is
used. A possible followup would be to introduce a separate struct.
That would complicate the signature of ProcessTransportFeedback.
Bug: webrtc:9934
Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15
Reviewed-on: https://webrtc-review.googlesource.com/c/107080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25465}
This makes the calculation more similar to the one in WebRTCVoiceEngine.
Bug: webrtc:9510
Change-Id: Ibca69842726e51c07b9cc9550ff9f15a24161e28
Reviewed-on: https://webrtc-review.googlesource.com/c/107653
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25448}
This CL consistently use:
* relative paths for WebRTC dependent targets (test_support)
* absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.
We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.
Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
