If the first simulcast stream is disabled, but the second one is enabled,
FrameEncodeMetadataWriter would fail to store frame metadata for all
streams and later fail to restore it for encoded frames.
Bug: none
Change-Id: Ib0d257abb863716ea94e56730f7caabef6ebeb64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162480
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30106}
If for some reason capture timestamp is unset, the default value of 0 would be
passed to RtcpSender. This will cause rtp timestamps to grow at double the rate
in Sender Reports because it has time since the last frame capture as a term.
Bug: none
Change-Id: I2fe09dabef6b0957fb504deaa06393dedc4a9e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30105}
The network emulation framework does not support creation sockets that
receive from all addresses (e.g. 0.0.0.0) but would instead crash at
runtime. This CL explicitly ensures that we don't provide such networks.
Bug: webrtc:9883
Change-Id: I1d77df0f2c68f878eace30e4b037ebc7eb9f1aa6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162482
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30104}
The test would flake using a real clock since time may pass between
calls to PostDelayedTask which would result in the tasks running
out of the expected order.
Bug: webrtc:11208, webrtc:11219
Change-Id: Ice5fe6ec4e9bf2ce89f00c6de7ed06b89dbe88cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162100
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30101}
This CL propagates a "closed" signal from DTLS up to the
SCTP section of the data channel controller, where it causes
closing of all open datachannels.
Bug: chromium:1030631, webrtc:10360
Change-Id: I88bb9e1aff5c25f330edfd092ef609d4fcc3a9f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162206
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30099}
instead of creating each time an object with RtpDepacketizer interface
this moves packet payload memcpy from RtpVideoStreamReceiver into
the depacketizers with possibility to remove it from there in follow ups.
Bug: webrtc:11152
Change-Id: If474207eb84d7e9d0207075bd395e60895f0d842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162185
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30095}
It's not great to use asserts in util functions like this because it
breaks the arrange-act-assert rule, but using checks is worse because
they will crash the test process on failure (= no other tests get run
after that).
Bug: b/143587130
Change-Id: If4d085311de0792b9fca1584db299fd24199e72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30093}
This is in preparation for writing tests that verify that
they close, and that they close at the right times.
Bug: chromium:1030631, webrtc:10360
Change-Id: I8129a9fc9731c1bfe1a660e82e23c1aeff1e5087
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162181
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30091}
It was found that generator can be destroyed on another thread
comparing to the one, from which frame were generated. It can happen
because generator injected into PC though scoped_ref object and the
last pointer to that object can be destroyed on different thread
depending on machine load.
To fix this sequence checker is replaced with lock. It is required
to ensure that generator won't be destroyed while it is reading frame,
because otherwise it can catch SIGSEGV.
Bug: webrtc:10138
Change-Id: Ia3488bd8ae396c209b90977469593784bb82114b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162182
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30086}
to return time of the last receieved packet of a key frame rather than
last received first packet of a key frame.
To match VideoReceiveStream expectation and prevent requesting
a new key frame if a large key frame is currently on the way.
Bug: None
Change-Id: I443a60872a3580d324f050080a9868f7b90d71a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161730
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30084}
This log should print removed elements in 'networks' which pointer to
vector<Network*> really. But it was printing just tailing elements of
the vector. For example, assume that there are 4 elements in 'networks',
and 1st and 3rd elements are removed. Then 'networks' will be changed
like this.
<Before>
[0] id = 1
[1] id = 2
[2] id = 3
[3] id = 4
<After>
[0] id = 2
[1] id = 4
Then this log should print the elements with id=1, id=3 which removed.
But currently it is printing the elements with id=3 and id=4 which
tailing 2 elements of the vector. It's related with how std::remove_if
works. So I replaced it with std::partition.
Bug: none
Change-Id: Idfdae04f2d321212310bddb4d8742ba2dccc4db9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159060
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30082}
USE_X11 is magically set by the toolchain. Let's tie this to
whether X11 extensions is on or not. WEBRTC_USE_X11 is used
in the audio device module so let's be consistent with that.
Bug: b/143587130
Change-Id: I15b1744c9acb534dbdf2119deb2dc5d17aa04184
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161901
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30080}
This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
Reason for revert: Downstream tests have been updated.
Original change's description:
> Revert "Distinguish between send and receive video codecs"
>
> This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
>
> Reason for revert: Breaks downstream test.
>
> Original change's description:
> > Distinguish between send and receive video codecs
> >
> > Even though send and receive codecs are the same,
> > they might have different support in HW.
> > Distinguish between send and receive codecs to be able to keep
> > track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30041}
>
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
>
> Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30042}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30078}
from LS_INFO to LS_VERBOSE.
By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.
Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
This prepares for moving TransportFeedbackAdapter to TaskQueue.
Bug: webrtc:9883
Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30076}
This is used to allow using a pre-configured builders as arguments to
fixture code.
Bug: webrtc:9510
Change-Id: I7837d284580fdbc926535ce5b2d8f582056534ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161948
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30070}
Earlier, rtc::Thread::PostTask was added as a convenient
alternative to MessageHandlers. This CL additionally adds support
for posting delayed tasks in a similar manner.
Bug: webrtc:10294
Change-Id: I0957b59ca2133a882c980bd2ad109fa03d701a16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161740
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30069}
Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.
This CL works around that by preferring to send padding on video SSRCs.
Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30066}
ScalingSettings has some public constructors. These should be
able to be called from exteranl code. However, a linker fails
on windows because ScalingSettings doesn't have RTC_EXPORT.
Bug: chromium:1031965
Test: build crrev.com/c/1949841
Change-Id: Iddaea77f87c52edbe8f77551322d7aa198bc0aeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30063}
This patch introduces a new type of STUN ping,
GOOG_PING_REQUEST/RESPONSE which is similar
to a STUN_BINDING but does not transmit any values.
The Connection class automatically sends these if
no STUN attributes has changed since last call to Connection::Ping()
if the remote peer has signaled that it supports it.
BUG=webrtc:11100
Change-Id: Ib1b590f0b90ca6cb56f2eb07cd62f976e246bc8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159961
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30062}
This reverts commit 60ec3703cd1f87081c0e4becde5d9ef210a6d44a.
Reason for revert: Needs back-end test before always enabling.
Original change's description:
> Ensure loss-based controller is always enabled.
>
> The new default parameters are the ones that were used in the Chrome
> Finch trial. The deleted unit test is invalidated by these changes.
>
> Bug: chromium:941413
> Change-Id: I597f4b0defaebe5bb3a6710b071fae2ee5c6f461
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160652
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30049}
TBR=srte@webrtc.org,crodbro@webrtc.org,jonasolsson@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:941413
Change-Id: I5da4676ad8be2569ad7eed99e954e0d0b624110b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161902
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30061}
This CL removes the use of absl::InlineVector in the PushResampler which
causes real-time reallocations for setups with more than 8 channels.
As part of the CL, it also removes one dependency on absl for the
common_audio module.
Bug: webrtc:11197
Change-Id: I0788ee9a0f3d08b91bb18caa65f660fb52368a97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161729
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30059}
