Commit Graph

220 Commits

Author SHA1 Message Date
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
fae05624ec Deprecate the static RtpRtcp::Create() method.
The method is being used externally to create instances
of the deprecated internal implementation.

Instead, I'm moving how we instantiate the internal implementation into
the implementation itself and move towards keeping the interface
separate from a single implementation.

Change-Id: I743aa86dc4c812b545699c546c253c104719260e
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31420}
2020-06-03 09:41:34 +00:00
3a5742c880 Add thread/sequence checks to ModuleRtpRtcpImpl.
This ended up with needing to fork the current implementation
in order to not break downstream projects that were inheriting
from it. While those get updated, we'll move on with the forked
class.

Bug: webrtc:11581,b/8278269
Change-Id: I05b596cbda71aa5b72894c31a7119d17d4761883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175500
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31334}
2020-05-20 15:45:21 +00:00
3c9bcc1f7a Reland of the test portion of:
https://webrtc-review.googlesource.com/c/src/+/172847

------------ original description --------------

Preparation for ReceiveStatisticsProxy lock reduction.

Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
2020-04-15 16:09:44 +00:00
16cc9efd54 Revert "Preparation for ReceiveStatisticsProxy lock reduction."
This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.

Reason for revert: Speculative revert: breaks downstream project

Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
> 
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
> 
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
> 
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
> 
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
> 
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
> 
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}

TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
2020-04-07 19:50:20 +00:00
24eed2735b Preparation for ReceiveStatisticsProxy lock reduction.
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
2020-04-06 14:34:38 +00:00
99eb20b513 StatsEndToEndTest: Configure bitrate via VideoEncoderConfig.
Configure bitrates via VideoEncoderConfig (and remove implementation of
VideoStreamFactoryInterface used to override the default bitrate configuration).

Bug: none
Change-Id: I935f27eaf0187f6c5dfb53a1af5406929867f209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169449
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30687}
2020-03-05 08:25:31 +00:00
ce515f7625 Add an integration test frame encryption works with DependencyDescriptor
Bug: webrtc:10342
Change-Id: I3a18c1fbe222eada7a484f8f62a0b5bad76eb073
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168888
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30595}
2020-02-24 16:01:04 +00:00
0f6bf75ab4 Make video engine tests aware of padding packets
Specifically do not try to parse them as rtx packets.

Bug: webrtc:11213, webrtc:11188
Change-Id: I3aa5929af433b1ada9fb26516618d11207f075a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162204
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30094}
2019-12-16 11:43:11 +00:00
ef3998ffd1 Add directive to make webrtc metrics optional.
Bug: webrtc:11144
Change-Id: I4e75e6aec033784685de3670e880bb9f2b6ee8d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30040}
2019-12-09 13:55:50 +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
c347585927 Use RtpPacket instead of legacy RtpHeaderParser in video/ tests
Bug: None
Change-Id: Ia35daa58aae51becef40910187006398d825c5b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161331
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30026}
2019-12-06 10:54:39 +00:00
014dd3c9f7 Trials should always be populated in call config.
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.

Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
2019-12-03 10:34:55 +00:00
5256d8bc4b Refactor FrameGenerator to return VideoFrameBuffer with VideoFrame::UpdateRect
Bug: webrtc:10138
Change-Id: I22079e2630bb1f3bb27472795fe923f9143b3401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161010
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29973}
2019-12-02 17:11:37 +00:00
0197887d71 Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in MultiStreamTester
Bug: webrtc:10933
Change-Id: I61ae0726fb197e5a779e036b5b1390c29ca96aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159714
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29829}
2019-11-19 10:52:12 +00:00
1242d9cc48 Reland Cleanup MultiStreamTester
Instead of taking TaskQueue from outside create one internally.
Detach MultiStreamTests from test::CallTest since that inheritance
only used for constants and for task_queue object.

Unlike original cleanup
keep using DEPRECATED_SingleThreadedTaskQueueForTesting for now.

Bug: webrtc:10933
Change-Id: Ife9143bfda0ebefd56a9199622296e64b14a7b20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159034
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#29744}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159280
Cr-Commit-Position: refs/heads/master@{#29782}
2019-11-13 08:53:22 +00:00
cae7f9f485 Revert "Cleanup MultiStreamTester"
This reverts commit d6b9b0a1f4132474c737b5e673e380c3d8e12e2c.

Reason for revert: breaks internal ios tests

Original change's description:
> Cleanup MultiStreamTester
> 
> Instead of taking TaskQueue from outside create one internally.
> Detach MultiStreamTests from test::CallTest since that inheritance
> only used for constants and (now unneeded) task_queue object.
> 
> Bug: webrtc:10933
> Change-Id: I7e30ddcf6faaa134ebcd9d53b578b40fdedf2a3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159034
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29744}

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

Change-Id: I0fe3d265fe12795ec96b420c21bdc934743c9c2f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10933
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159222
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29745}
2019-11-08 13:17:29 +00:00
d6b9b0a1f4 Cleanup MultiStreamTester
Instead of taking TaskQueue from outside create one internally.
Detach MultiStreamTests from test::CallTest since that inheritance
only used for constants and (now unneeded) task_queue object.

Bug: webrtc:10933
Change-Id: I7e30ddcf6faaa134ebcd9d53b578b40fdedf2a3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159034
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29744}
2019-11-08 12:22:45 +00:00
2f2049af23 Add blocking call in BandwidthStatsTest destructor
task_queue_ outlives the BandwidthStatsTest object, but Posted task
captures |this|. Blocking call in the destructor is a simple way to avoid
that race
(should work as long as posted task doesn't call virtual functions from |this|).

Bug: webrtc:10933
Change-Id: Id30badb711480af5ee737b96b9224c1a73e730ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158898
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29707}
2019-11-06 13:38:14 +00:00
8d65e9ab98 Fixes pacing interval dependency and race in BandwidthEndToEndTest
BandwidthEndToEndTest failed when I tested it with the new task-queue
based paced sender. This turned out to be issues with this test.
Problems fixed by this CL:

1. Send-side BWE not set up correctly. Caused probing to fail.
2. Test waited for non-zero pacer delay, but the new pacer will not
   generate any delay in this scenario.
3. Race condition during shutdown of test.

1) Is just a matter of configiuring the right header extension.
2) Set up test with high encoder bitrate to trigger pacer delay.
3) TaskQueue outlives the Call instances used in test, so make sure
   they are not referenced from lambda during teardown.

Bug: webrtc:10809
Change-Id: I6393975691dfa05eb5b25d9283e476062e23a876
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158722
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29669}
2019-10-31 15:27:21 +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
7c06777ab0 Cleanup includes in modules/include/module_common_types.h
Add missing includes to files that were transactivly depending on removed includes.

Bug: None
Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29396}
2019-10-07 16:06:26 +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
70bc753cc6 Add comments to MultiCodecReceiveTest.
Follow up to https://webrtc-review.googlesource.com/c/src/+/153880

Bug: none
Change-Id: If52e2ba638cc463f55330d5d5db1e1e566231562
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154349
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29294}
2019-09-25 08:15:20 +00:00
bf5ee00f8d Disable prerender smoothing in MultiCodecReceiveTest.
Avoids frame dropping in render queue.


Bug: webrtc:10828
Change-Id: I9e09fc2faee4626c8d60c152840b4208dbb89dc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153880
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29276}
2019-09-24 08:12:09 +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
7bf7a427bf Delete flag VideoReceiveStream::Config::Rtp::remb
This flag became unused in https://codereview.webrtc.org/2789843002;
it was set, but the setting had no effect.

Bug: webrtc:7135
Change-Id: I012a7c3600bc7a371c7a589695823b30ed5647a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29192}
2019-09-16 11:20:55 +00:00
0cd61b6e28 MultiCodecReceiveTest: fix for flaky test.
Bug: webrtc:10828
Change-Id: I0fb2f4cdf0481e6c0036ae4dba861c5fbd4b98e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152160
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29124}
2019-09-10 07:57:59 +00:00
6516f76f9b Deprecate SingleThreadedTaskQueueForTesting class.
This class doesn't strictly follow rtc::TaskQueue semantics,
which makes it surprising and hard to use correctly.
Please use TaskQueueForTest instead.

This CL follows usual deprecation process:

1/ Rename.
% for i in `git ls-files` ; sed -i "s:SingleThreadedTaskQueueForTesting:DEPRECATED_SingleThreadedTaskQueueForTesting:" $i

2/ Annotate old name for downstream users and accidental new uses.

Bug: webrtc:10933
Change-Id: I80b4ee5a48df1f63f63a43ed0efdb50eb7fb156a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150788
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29045}
2019-09-03 10:31:30 +00:00
25eb47ccf1 Make the RtpHeaderParserImpl available to tests and tools only.
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
  (a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
  See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
  sufficient for most production cases.

Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 15:56:40 +00:00
d77cc24f67 New const method StreamStatistician::GetStats
And a corresponding struct RtpReceiveStats. This is intended
to hold the information exposed via GetStats, which is quite
different from the stats reported to the peer via RTCP.

This is a preparation for moving ReceiveStatistics out of the
individual receive stream objects, and instead have a shared instance
owned by RtpStreamReceiverController or maybe Call.

Bug: webrtc:10679,chromium:677543
Change-Id: Ibb52ee769516ddc51da109b7f2319405693be5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148982
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28943}
2019-08-23 08:38:59 +00:00
54d5d2c75b Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.

The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.

Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
2019-08-21 09:45:21 +00:00
fdf38802a6 Make "WebRTC-BweAllocProbingOnlyInAlr/Enabled/" default and remove key.
Bug: chromium:951299
Change-Id: Idf612040e21f2962cc63d7de3dcb237bbf868034
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148985
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Konrad Hofbauer <hofbauer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28902}
2019-08-19 15:39:25 +00:00
e3a10e1b43 Remove usage of RtpRtcp::SetSSRC() in video/
That method is going away in favor in construction time setting.

Bug: webrtc:10774
Change-Id: I2aba5a2537e5846a3c9438a5b376b230e84c5f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149826
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28901}
2019-08-19 14:20:30 +00:00
891d393b80 Call Call::GetStats() from the correct thread in ProbingEndToEndTest.
Also removing the stop_event_ from the RampUpTester class, which I missed in review 148067.

Bug: webrtc:10847
Change-Id: I102cc75287503915b51e37ea4ee01dfcc2437699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148062
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28801}
2019-08-08 06:40:26 +00:00
83bbe91398 Delete deprecated rtc_event_log header
Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 10:58:17 +00:00
d23f67e6be Call Call::GetStats() from the correct thread in StatsEndToEndTest.
Bug: webrtc:10847
Change-Id: I8a82709073827f0eb901e20600f4e8bcf86d96a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148061
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28786}
2019-08-07 08:50:09 +00:00
e80885a89c Call Call::GetStats() from the correct thread in our bandwidth tests.
Bug: webrtc:10847
Change-Id: Ief8cdd72f9d5b600d5306c00c1d249c29fb20396
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148063
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28765}
2019-08-06 08:19:12 +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
53d45baa50 Make TaskQueueFactory required construction parameter for Call
Bug: webrtc:10284
Change-Id: I573ee0087c035e26918260c21b8b0213ddfe7ebc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143791
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28467}
2019-07-03 14:02:45 +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
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
fadb1811a8 Negotiate use of RTCP loss notification feedback (LNTF)
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.

Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
2019-05-24 12:44:14 +00:00
908e22e553 Use explicit TaskQueueFactory for FrameGeneratorCapturer in video/ tests
This replaces the implicit usage of GlobalTaskQueueFactory

Bug: webrtc:10284
Change-Id: Ic53254ee6d9a421be18bf2df62f1742c46812701
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133576
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27690}
2019-04-18 13:41:12 +00:00
16cb8f5d74 Reland "Replace usage of old SetRates/SetRateAllocation methods"
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org

Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
2019-04-12 13:37:32 +00:00
7ddef1af88 Revert "Replace usage of old SetRates/SetRateAllocation methods"
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
> 
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
> 
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
2019-04-11 10:50:29 +00:00
7ac0d5f348 Replace usage of old SetRates/SetRateAllocation methods
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.

Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
2019-04-11 07:46:09 +00:00