Commit Graph

254 Commits

Author SHA1 Message Date
e519f38eaa Remove rtc::Location from SendTask test helper
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.

Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
2022-08-11 12:55:32 +00:00
b5b159d98c Update old TODO comments
Bug: None
Change-Id: I531ed648fe3d1f0dd1202f53c59ed023aed1ea7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267664
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37432}
2022-07-05 09:09:44 +00:00
3176ef79e9 Rename AudioReceiveStream to AudioReceiveStreamInterface
Bug: webrtc:7484
Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36965}
2022-05-23 08:44:26 +00:00
f6f4543304 Rename VideoReceiveStream to VideoReceiveStreamInterface
Bug: webrtc:7484
Change-Id: I653cfe46486e0396897dd333069a894d67e3c07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262769
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36958}
2022-05-22 10:54:38 +00:00
8139cda2ec Enable encoder frame drop for call tests and video quality tests
It was accidentally disabled by
https://webrtc-review.googlesource.com/c/src/+/262244, resulting in
lots of unintended changes in performance tests.

Bug: webrtc:6883, webrtc:14075
Change-Id: Ie414f729ec2248f0eef99922e9704f4c4a0b1aa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262813
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36931}
2022-05-19 14:55:46 +00:00
cf2c8915f4 Delete H264EncoderSpecificSettings
Production code always use the default settings.

Bug: webrtc:6883
Change-Id: I213fc6433bb1cd0a6623ad523fee2df1506588e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261903
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36926}
2022-05-18 13:53:20 +00:00
408a46c36c Don't disable frame dropping in screenshare tests
Sync test settings with changes made in prod in
https://webrtc-review.googlesource.com/c/src/+/242965

Bug: webrtc:9734
Change-Id: Id2055b125802d47d7df4fb10aa99b565d47dba8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247369
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35733}
2022-01-19 10:06:05 +00:00
d3251968d1 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.

Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
2021-11-15 21:44:59 +00:00
97597c0f51 Remove usage of INFO alias for LS_INFO in log messages
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
2021-11-04 13:46:17 +00:00
1c1f540487 Factor out common receive stream methods to a common interface.
Also including common Rtp config members.
Follow up changes will remove the ReceiveRtpConfig class in Call
and copy of extension headers, instead use the config directly
from the receive streams and not require stream recreation for changing
the headers.

Bug: webrtc:11993
Change-Id: I29ff3400d45d5bffddb3ad0a078403eb102afb65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221983
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34283}
2021-06-14 16:54:07 +00:00
c1d589146b Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
Bug: webrtc:12701
Change-Id: Ie50225374f811424faf20caf4cf454b2fd1c4dc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215930
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33818}
2021-04-23 12:04:39 +00:00
0bb354c540 Add and refactor functionality into rtc_base/win
This change moves ScopedComInitializer out of core_audio_utility and
into rtc_base/win so it can be reused elsewhere more easily.

It also adds HSTRING and GetActivationFactory functionality to
rtc_base/win. These two were heavily based on what is already present
base/win.

All of these are necessary for the new window capturer based on the
Windows.Graphics.Capture API. You can see how these APIs will be
used in this CL: 186603: Implement WgcCaptureSession |
https://webrtc-review.googlesource.com/c/src/+/186603

Bug: webrtc:9273
Change-Id: I0a36373aac98be779ccbabe1053bb8d6e234f6a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188523
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32522}
2020-10-29 20:39:10 +00:00
25a1334534 Updates how min bitrate is set for VP9 SVC in perf tests.
In the video quality tests the codec-level min bitrate is sometimes set
as if single-stream simulcast will be used. When VP9 spatial layers are
then generated the will get new appropriate min bitrate levels.
The encoder adjuster can however look at the codec level min bitrate
and incorrectly adjust the bitrate up if it is set too high.

This CL removes the codec-level min bitrate if svc is used.

Bug: webrtc:12080
Change-Id: I563a57f3031c90c116448f1d255d3b6711f4ee75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32509}
2020-10-27 16:24:32 +00:00
55c178693c Add support for NV12 frame generation for tests
This can be used in the future to test NV12 video frames with encoders, both
from unittests and from tools like video_loopback.

Tested using video_loopback with generator NV12.

Bug: webrtc:11978
Change-Id: I0d24ae3ebab2267f076703cbda81e99cec465ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185045
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32206}
2020-09-28 09:48:08 +00:00
2549f174b5 Remove RTPFragmentationHeader creation and propagation through webrtc
Bug: webrtc:6471
Change-Id: I5cb1e10088aaecb5981888082b87ae9957bbaaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31955}
2020-08-17 16:37:33 +00:00
059f4f74df Cleanup setting VideoQualityTestFixtureInterface::Param struct
put default values next to the (named) members.
Remove constructor
declared as VideoQualityTestFixtureInterface::Param::Param,
but defined as VideoQualityTest::Param::Param

avoid braced initialisation for structures with lot's of fields that change over time

Bug: None
Change-Id: I647e3ace47ae981c515d6939274cb2744dd1b866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178811
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31674}
2020-07-08 13:53:57 +00:00
636865e05d Delete field trial WebRTC-GenericDescriptor
this trial is by default on for three months since
https://webrtc-review.googlesource.com/c/src/+/168661

Bug: webrtc:11503
Change-Id: I8f2e0996fd1c77113715628198a409f12a525d51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176242
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31424}
2020-06-03 13:00:30 +00:00
a54ba4c02e Make video_loopback work with av1
Bug: webrtc:11404
Change-Id: Id4fb4ac7e545df2e4f0a0d91b3531074ff77c9f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172340
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31210}
2020-05-11 12:20:09 +00:00
9e46cf5cc5 Introduce a RunLoop class that supports the TaskQueue interface
on the current thread.

This simplifies writing async tests that use TaskQueue and doesn't
require spinning up a new thread for simple things. The implementation
is currently based on rtc::Thread, which could also be useful in
some circumstances while migrating code over to TQ.

Remove PressEnterToContinue from the test_common files since
it's very specific and only used from one file.

Bug: none
Change-Id: I8b2c6c40809271a109ec17cf7e1120847645d58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31160}
2020-05-04 18:10:00 +00:00
8ec11b8312 Do not register generic frame descriptor v1 in integration tests
Bug: webrtc:11358
Change-Id: I2fb42198d760ba95c5cddc4abb73e58b427aefca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173585
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31078}
2020-04-15 18:45:43 +00:00
06c7095bc7 Make video quality tests to always take a fixed duration
It was possible before if an input fps dropped due to cpu adaptation

Also, this CL removes occasional test failure (it could've happened if
input framerate got very low)

Bug: webrtc:11432,webrtc:11426
Change-Id: Id1a4df23302f7b8ab6781f1e7cca5112bfcfe9ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170469
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30802}
2020-03-16 15:38:27 +00:00
eac08bfe23 Reland "Wire up internal libvpx VP9 scaler to statistics proxy"
This reverts commit a2cb93d8b9659292f7ec73db53421d481f84c22c.

Reason for revert: Reland with no changes after downstream projects are
updated.

Original change's description:
> Revert "Wire up internal libvpx VP9 scaler to statistics proxy"
> 
> This reverts commit 50327a51007c3e25bc3bcd35b5d0945fe0f27d05.
> 
> Reason for revert: Breaks downstream tests
> 
> Original change's description:
> > Wire up internal libvpx VP9 scaler to statistics proxy
> > 
> > Bug: webrtc:11396
> > Change-Id: I5ac69208b00cc75d4e5dbb3ab86f234b3e1f29f8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169922
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30725}
> 
> TBR=ilnik@webrtc.org,hbos@webrtc.org,nisse@webrtc.org
> 
> Change-Id: I53dcb41bdf8f8dccfcd43b717509ec047f590648
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11396
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170102
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30734}

TBR=ilnik@webrtc.org,hbos@webrtc.org,nisse@webrtc.org,srte@webrtc.org

Change-Id: Ie47df4aec199701256c1dba8fa64176683becabc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11396
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170105
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30738}
2020-03-10 11:15:51 +00:00
a2cb93d8b9 Revert "Wire up internal libvpx VP9 scaler to statistics proxy"
This reverts commit 50327a51007c3e25bc3bcd35b5d0945fe0f27d05.

Reason for revert: Breaks downstream tests

Original change's description:
> Wire up internal libvpx VP9 scaler to statistics proxy
> 
> Bug: webrtc:11396
> Change-Id: I5ac69208b00cc75d4e5dbb3ab86f234b3e1f29f8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169922
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30725}

TBR=ilnik@webrtc.org,hbos@webrtc.org,nisse@webrtc.org

Change-Id: I53dcb41bdf8f8dccfcd43b717509ec047f590648
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11396
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170102
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30734}
2020-03-10 08:09:50 +00:00
50327a5100 Wire up internal libvpx VP9 scaler to statistics proxy
Bug: webrtc:11396
Change-Id: I5ac69208b00cc75d4e5dbb3ab86f234b3e1f29f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169922
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30725}
2020-03-09 13:47:25 +00:00
420ad1af1e Fix video_loopback crash when selecting all streams
When selecting all streams there was an index out of bounds
checking the selected temporal layer, which is -1 so was irrelevant.

My fix is to prevent selecting a temporal layer and all streams
at the same time.

Bug: webrtc:11402
Change-Id: I0641b926cba35878945b866f2c59b4b0281f0852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169720
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30679}
2020-03-04 10:25:06 +00:00
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
7a9a092708 Delete media transport integration.
MediaTransport is deprecated and the code is unused.

No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
2019-11-26 19:19:36 +00:00
d15a0283d1 Hide deprecated SingleThreadedTaskQueueForTest behind an accessor
this change is intentionally noop.
Goal is to minimize change that would replace the
SingleThreadedTaskQueueForTest with a regular task queue.

Bug: webrtc:10933
Change-Id: I6da768941af048de3716af13e41b8f0f1ccd4cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157892
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29569}
2019-10-22 11:57:49 +00:00
85a10001a5 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue
Bug: webrtc:10933
Change-Id: I749ecd9cedb6798f1640ce663c6ebb6679889b67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29565}
2019-10-22 08:34:57 +00:00
82a3f0ad7f Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::SendTask
Bug: webrtc:10933
Change-Id: I60738434b46e77b4644173ad168bc0efa58459b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156001
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29551}
2019-10-21 08:45:02 +00:00
3b819f3d8b Move video_sources_.clear() call to CallTest::DestroyStreams
When one of the sources is a FrameGeneratorCapturer, this implies that
its TaskQueue is stopped. Before this change, the FrameGeneratorCapturer
was destroyed later, by the CallTest destructor, which led to a
use-after-free race on the Clock object passed to the capturer.

Bug: webrtc:11018
Change-Id: I3e53f95a725b6fb53b13e182ecd2caf03ea15bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156170
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29443}
2019-10-11 07:56:52 +00:00
44db436e87 Propagate task queue to create test::DirectTransport by TaskQueueBase interface
actual task queue implementation for these tests is intentionally unchanged for now.

while at it, change return type of created transports to unique_ptr to note passing ownership.

Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
2019-09-30 03:23:07 +00:00
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
65f17ca6b4 Move MediaTransportInterface out of the libjingle_peerconnection_api target
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.

Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-13 10:49:56 +00:00
1c2f6372f6 Simplify the VideoFrameDumpingDecoder API.
This CL changes the VideoFrameDumpingDecoder API to only expose a
factory function creating the wrapper instead of the full class.

Bug: webrtc:10902
Change-Id: I1e7e3a60accea1a7c48207d4262ed4bacacab4a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150040
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28924}
2019-08-21 09:49:02 +00:00
ed0febf573 Add k prefix to FrameGenerator::OutputType enum values
This prepares for using VideoFrameBuffer::Type as
FrameGenerator::OutputType, which will reduce the
number of redundant enums in the code.

Bug: webrtc:9883
Change-Id: I253f5f1ea7181e02a5cf1a92925f51da8ada6aa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146982
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28696}
2019-07-29 09:41:31 +00:00
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
8f01c4e1b6 Define FecControllerOverride and plumb it down to VideoEncoder
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}
2019-06-28 15:57:22 +00:00
5894b6aad4 Add kPayloadTypeGeneric to CallTest and use it in VideoQualityTest.
- Add CallTest::kPayloadTypeGeneric
- Allow for unrecognized codec names in VideoQualityTest.
  Generic packetization is implicitly enabled for these codecs.

Tested: autoninja -C out/Debug && out/Debug/video_loopback
Bug: webrtc:10738
Change-Id: I57001be997db2f0eed9197eb40801b5ad936d222
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141864
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28270}
2019-06-13 15:45:00 +00:00
3c396e52da Add injectable video encoder and decoder to video quality test.
Bug: webrtc:10738
Change-Id: Ia5180cf0252ecd1c58a2080e3954fcb886b066e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141667
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28263}
2019-06-13 13:03:05 +00:00
08fa953711 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
This reverts commit fd5166c305068772d00ad7edf50151bba215400b.

Reason for revert: Stop using CreateTestAudioDeviceModule in downstream

Original change's description:
> Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
> 
> This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2.
> 
> Reason for revert: Breaks downstream importer.
> 
> Original change's description:
> > Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> > 
> > Bug: webrtc:10284
> > Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28227}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10284
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28228}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,philipel@webrtc.org

Change-Id: I42bc19793d48350ca45b751d7e1b26124ac7fbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141670
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28254}
2019-06-12 14:44:01 +00:00
370f93a34a Reland "Inform VideoEncoder of negotiated capabilities"
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4

Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org

Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
2019-06-11 14:49:37 +00:00
fd5166c305 Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2.

Reason for revert: Breaks downstream importer.

Original change's description:
> Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> 
> Bug: webrtc:10284
> Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28227}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org

Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28228}
2019-06-11 12:32:23 +00:00
fc961357a7 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28227}
2019-06-11 12:15:44 +00:00
49d661a7d3 Revert "Inform VideoEncoder of negotiated capabilities"
This reverts commit 11dfff0878c949f2e19d95a0ddc209cdad94b3b4.

Reason for revert: Downstream import failure.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
> 
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
> 
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org

Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
2019-06-11 11:56:04 +00:00
11dfff0878 Inform VideoEncoder of negotiated capabilities
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().

Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
2019-06-11 11:32:13 +00:00
5c18a5ff5e Reland "VP9 screenshare: Don't base layers frame-rate on input frame-rate"
Reland with fixes.

If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.

Also, fix screenshare_loopback test for low-fps vp9 testing.

Bug: webrtc:10257
Change-Id: Id40a780d461e6b51cb44d275b8aa5d7b348d3586
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138215
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28054}
2019-05-24 11:05:04 +00:00
ce723234ba Revert "VP9 screenshare: Don't base layers frame-rate on input frame-rate"
This reverts commit eb1754c5750dfcad23ac62b47aa3aa2176ae7be2.

Reason for revert: breaks downstream projects

Original change's description:
> VP9 screenshare: Don't base layers frame-rate on input frame-rate
> 
> If input framerate is a little unstable, using it to cap layers will
> make output framerate even smaller for longer periods of time.
> 
> Also, fix screenshare_loopback test for low-fps vp9 testing.
> 
> Bug: webrtc:10257
> Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28050}

TBR=ilnik@webrtc.org,ssilkin@webrtc.org

Change-Id: I82bfbac58249cfe0da5ff565aa97a4745fd078ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138213
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28051}
2019-05-24 09:31:56 +00:00
eb1754c575 VP9 screenshare: Don't base layers frame-rate on input frame-rate
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.

Also, fix screenshare_loopback test for low-fps vp9 testing.

Bug: webrtc:10257
Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28050}
2019-05-24 09:04:51 +00:00
b32f2c7f57 Publish rtc event log api and default factory for it in api/
Bug: webrtc:10206
Change-Id: I34194ddb6fd2b0a3d7c553fadc9ddc1ea9740da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28023}
2019-05-22 13:38:25 +00:00