This change removes a memory leak from a faulty test.
It's unknown why this leak doesn't trigger the leak
checker, see the referred bug for further info.
Bug: webrtc:11605
Change-Id: I05979a696352ed2bc90cd029e462788c0912e02d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175919
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31348}
This reduces the number of threads allocated per PeerConnection when
more than one PC is needed.
Bug: webrtc:11598
Change-Id: I3c1fd71705f90c4b4bbb1bc3f0f659c94016e69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31347}
This only replaces the methods with the signatures VerifyX(wants,
value). This ensures that failing expectations give the corrent line,
where previously it gave the line in the helper methods.
Bug: None
Change-Id: I89e883ccd15b3ea5ce364d2c16b0c3ac219f139c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175912
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31345}
Since RateStatistics is in practice always used with increasing
timestamps, and is often sparesely populated, replace the pre-allocated
ring buffer with a simple deque where each element tracks which time it
represents.
Bug: webrtc:11600
Change-Id: I866d7cfa607228c35452f0f19575825d2e694f75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175906
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31344}
This change adds an optional delay to NetEq's output. Note, this is not
equivalent to increasing the jitter buffer with the same extra length.
Bug: b/156734419
Change-Id: I8b70b6b3bffcfd3da296ccf29853864baa03d6bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175110
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31343}
Since locking model has been cleaned up, PacingController can now call
PacketRouter directly - without having to go via PacedSender or
TaskQueuePacedSender.
Bug: webrtc:10809
Change-Id: I181f04167d677c35395286f8b246aefb4c3e7ec7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175909
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31342}
This will help show what the restrictions are before
and after they transform into sink wants.
These will not be so spammy as adaptations do not happen
very often.
Bug: None
Change-Id: Ib72b313f68c6934d7833d8a3f14ce00df602832b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175800
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31341}
Use speed 6 for better quality for low resolution, speed 8 for HD for better speed.
This will better balance speed and quality.
Change-Id: I3d8dbd45533471ce58d53c1ac26f92c7b1106259
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175281
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31336}
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}
This is a reland of 6b9c60b06d04bc519195fca1f621b10accfeb46b
Original change's description:
> Removes lock release in PacedSender callback.
>
> The PacedSender currently has logic to temporarily release its internal
> lock while sending or asking for padding.
> This creates some tricky situations in the pacing controller where we
> need to consider if some thread can enter while we the process thread is
> actually processing, just temporarily busy sending.
>
> Since the pacing call stack is no longer cyclic, we can actually remove
> this lock-release now.
>
> Bug: webrtc:10809
> Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31206}
Bug: webrtc:10809
Change-Id: Id39fc49b0a038e7ae3a0d9818fb0806c33ae0ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175656
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31332}
With an optional parameter this allows the task-queue based paced
sender to mimic the old behavior and coalesce sending of packets in
order to reduce thread wakeups and provide opportunity for batching.
This is done by simply overriding the minimum time the thread should
sleep. The pacing controller will already handle the "late wakup" case
and send any packets as if it had been woken at the optimal time.
Bug: webrtc:10809
Change-Id: Iceea00693a4e87d39b0e0ee8bdabca081dff2cba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175648
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31328}
Some VideoStreamEncoderTests such as
AdaptsFramerateForLowQuality_MaintainResolutionMode had been observed
to be flaky. In a local run, this test failed 1/12000 times, but on
bots it may have failed more often.
All tests could sometimes fail due to expecting something to be the
case without waiting for that thing to happen, which was made evident
when adding SleepMs() at strategic points in the code and running the
tests locally.
With this CL, the tests should no longer be flaky after having added
waiting for adaptation to be applied or EXPECT_TRUE_WAIT for sink wants
to have the appropriate values.
Bug: webrtc:11586
Change-Id: I60e76c742d9ccc8305feb14a834a4f61a60a62a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175654
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31327}
Remove android_manifest_for_lint from BUILD.gn files
The Chromium Roll into WebRTC isn't flowing
The first CL that caused the problem is https://webrtc-review.googlesource.com/c/src/+/175140/
The error is: ERROR at //examples/BUILD.gn:104:33: Assignment had no effect. android_manifest_for_lint = "androidapp/AndroidManifest.xml"
android_manifest_for_lint has ben removed so update BUILD files that use that feature to reflect this.
BUG=None
Change-Id: If526d9a4dd80cddca7f2c9dd7f67ba9efe3f1a84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175661
Commit-Queue: Courtney Edwards <courtneyfe@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31325}
This CL extends the WebRTC testing API to allow audioproc_f -based
testing using a pre-created AudioProcessing object. This is an
important feature to allow testing any AudioProcessing objects
that are injected into WebRTC.
Beyond adding this, the CL also changes the simulation code to
operate on a scoped_refptr<AudioProcessing> object instead of a
std::unique<AudioProcessing> object
Bug: webrtc:5298
Change-Id: I70179f19518fc583ad0101bd59c038478a3cc23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175568
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31319}
- Moves AnalyzerConfig and helper functions IsAudioSsrc, IsVideoSsrc, IsRtxSsrc, GetStreamNam and GetLayerName to analyzer_common.h
- Moves log_segments() code to rtc_event_log_parser.h
- Moves TriageAlert/Notification code to a new file with a couple of minor fixes to make it less spammy.
Bug: webrtc:11566
Change-Id: Ib33941d8185f7382fc72ed65768e46015e0320de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174824
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31318}
...for the NullDataDumper, WrongCaptureBlockSize and
DISABLED_WrongRenderBlockSize tests. This is to avoid creation
of additional threads on Mac, which can cause issues on asan bots.
Bug: webrtc:11577
Change-Id: I4e6a64d47ec3b0a0e0018b19a0486208ba7e6ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175600
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31307}
This will improve support for tests that use Clock as well as offer
a way to remove use of Sleep() in the tests.
Bug: none
Change-Id: I25fd0c6fc1b52ec0c917e56fae6807b136213d8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175566
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31305}
This reverts commit 576ad5d510894040d7bbc041d5c86745c67f30f8.
Reason for revert: Causes compile error in Chrome.
Original change's description:
> Make TransformableVideoFrameInterface::GetMetadata pure virtual.
>
> GetMetadata() has been implemented downstream and can be made pure
> virtual.
>
> Bug: chromium:1069295
> Change-Id: I62a3be6106552d2d82d8c413c6f523d31626b0d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175001
> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31281}
TBR=hta@webrtc.org,marinaciocea@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1069295
Change-Id: I5915270d5b8dab9fc30a07f22fddedb29beca01a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175620
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31304}
Logging of content name (mid) is valuable to debug issues
in scenarios with multiple m= line sections in SDP.
For example, video conferencing applications which
uses SFU and Unified Plan SDPs will likely to leverage
from more detailed logs when issues need to be debugged.
Bug: webrtc:10139
Change-Id: Id52ba3ad54af5caa0f8c03daaa51bdb0caf9fe67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175115
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31302}
The Chromium Roll into WebRTC isn't flowing
The first CL that caused the problem is https://webrtc-review.googlesource.com/c/src/+/175140/
The error is: ERROR at //examples/BUILD.gn:104:33: Assignment had no effect. android_manifest_for_lint = "androidapp/AndroidManifest.xml"
android_manifest_for_lint has ben removed so update BUILD files that use that feature to reflect this.
BUG=None
Change-Id: Ic3eb16eab8e4a4ab87daac9998d7a07373fec493
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175569
Commit-Queue: Courtney Edwards <courtneyfe@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31299}
