This changeset adds some additional tests to validate that the SSLAdapter
behaves as intended when used with a Custom SSL Certificate Verifier. It
validates the following scenarios:
1. Handshake succeeds on TLS handshakes if certificate verifier returns true.
2. Handshake fails on TLS handshakes if certificate verifier returns false.
3. Handshake succeeds on DTLS handshakes if certificate verifier returns true.
4. Handshake fails on DTLS handshakes if certificate verifier returns false.
5. Handshake succeeds on TLS transfers if certificate verifier returns true.
6. Handshake succeeds on DTLS transfers if certificate verifier returns true.
Bug: webrtc:9258
Change-Id: I48b72c9762a7023ece12d882ac4a05d9881bf9e6
Reviewed-on: https://webrtc-review.googlesource.com/75720
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23355}
End to end test for media sent over a TCP TURN server with both clients in relay
This test validates that media can be sent between two clients who are set up
to relay information with the server configured to use TCP instead of UDP.
Bug: webrtc:7668
Change-Id: I3efd04048589c144494f90f2cdf3df5f9f80300e
Reviewed-on: https://webrtc-review.googlesource.com/76507
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23354}
The Quic BBR implementation uses packet sequence numbers to keep track
of the time slots used for calculation of send receive rates. To avoid
protocol dependence the port were initially written to use send times
instead.
As there are issues with running BBR in WebRTC, it makes sense to
use an identical implementation as in Quic to ensure that there
aren't implementation issues causing bad behavior. This requires
providing sequence numbers.
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I2cd96bc6ffb88042bb2b91421bfe6cbf7c1ff8ac
Reviewed-on: https://webrtc-review.googlesource.com/76583
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23353}
Deletes the ALIGNP and RTC_ALIGNED_P macros from basictypes.h.
ALIGNP was used by MemoryStream, supposedly to make it more efficient.
If it really provided an efficiency improvement is unclear, and in any
case, MemoryStream is used for tests only, and doesn't need high
performance.
Bug: webrtc:6853
Change-Id: If835e881e3857dcc22c7a544491b92829a81d1b3
Reviewed-on: https://webrtc-review.googlesource.com/78021
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23350}
Unfortunately, pthread_condattr_setclock is lacking in the
versions of MacOS and iOS we support, and we have to stay
with gettimeofday on those platforms.
Bug: webrtc:9269
Change-Id: I8554e56496cc7b6948cb9b8a4c0bcf886c3544be
Reviewed-on: https://webrtc-review.googlesource.com/77122
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23349}
Reland of CL https://webrtc-review.googlesource.com/c/src/+/77421
Copied description:
--
Add presubmit check for changes in 3pp
Presubmit check will test will new changes be overriden by autoroll
or not. In more details presubmit will check:
1. Each dependency in third_party have to be specified in one of:
a. THIRD_PARTY_CHROMIUM_DEPS.json
b. THIRD_PARTY_WEBRTC_DEPS.json
2. Each dependency not specified in both files from #1
3. Changes won't be overriden by chromium third_party deps autoroll:
a. Changes were made in WebRTC owned dependency
b. Changes were addition of new Chromium owned dependency
--
Also if commit message contains tag NO_AUTOIMPORT_DEPS_CHECK equal
to True, than changes in chromium specific deps will be permitted.
It is required for autoroller to be able to commit its changes and
not to fail on presubmit check.
Bug: webrtc:8366
Change-Id: I545a4778445855cf3db7cf257ca0cb63753aac06
Reviewed-on: https://webrtc-review.googlesource.com/78042
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23348}
This CL implements a fix behind a field trial for a NetEq issue. NetEq restarts audio too quickly after a buffer underrun, which can quickly lead to another underrun in some circumstances. The fix changes NetEq's behavior to wait with restarting playback until sufficient audio is buffered.
Bug: webrtc:9289
Change-Id: I5968c9478ce8d84caf77f00b8d0a39156b47fc8d
Reviewed-on: https://webrtc-review.googlesource.com/77423
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23347}
These files are required when implementing tests based on the test fixture,
and should be exposed as part of the test api.
This CL also removes a usage of stringstream and fixes some chromium-style
lint issues.
Bug: webrtc:8982, webrtc:163
Change-Id: I132aea0da79a79587887f21897236fc9802b7574
Reviewed-on: https://webrtc-review.googlesource.com/74586
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23346}
This class will be used in upcoming VideoQualiyObserver.
Bug: none
Change-Id: I7d79a6caf3040a3f707ed8700842dea1de81e0a6
Reviewed-on: https://webrtc-review.googlesource.com/77724
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23343}
NetEq tapers down the audio produced through loss concealment when the
expansion has been going on for some time. When the audio packets starts
coming in again, there is a ramp-up that happens. This ramp-up could
before this change extend over more than one 10 ms block, which made
keeping track of the scaling factor necessary. With this change, we make
this ramp-up quicker in the rare cases when it lasted more than 10 ms,
so that it always ramps up to 100% within one block. This way, we can
remove the mute_factor_array.
This change breaks bit-exactness, but careful listening could not reveal
an audible difference.
This change is a part of a larger refactoring of NetEq's PLC code.
Bug: webrtc:9180
Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e
Reviewed-on: https://webrtc-review.googlesource.com/77180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23342}
Force auto roll of chromium third_party into WebRTC to fail if any one
of required chromium-specific dependencies is missing in chromium
third_party repo of after checking it into WebRTC repo.
Also try to fix some flakes in autoroller.
Bug: webrtc:8366
Change-Id: I781cd4d4a4a230fb126cc32d8147310f70ab8b91
Reviewed-on: https://webrtc-review.googlesource.com/77722
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23340}
Follow up to https://webrtc-review.googlesource.com/c/src/+/39511,
which introduced a new Decode method, without the
RTPFragmentationHeader argument, and deprecated the old method.
Bug: webrtc:6471
Change-Id: Icd3c536ebedd4e3c2d57fdb4d6e078d6ff1de5b6
Reviewed-on: https://webrtc-review.googlesource.com/75180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23339}
This CL fixes the rounding of the estimated average call skew. Before it
was rounded down (toward INT_MIN). Now it is rounded to the nearest integer.
This avoids unnecessary fluctuations of the estimated call skew (and
unnecessary resets).
Bug: webrtc:9283,chromium:888042
Change-Id: Id5b3c593f812f5f9fd3dcdafb7e388a6ef1ac153
Reviewed-on: https://webrtc-review.googlesource.com/77684
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23338}
From those I removed, transport.h doesn't exist. For the others
I tried checking that the presubmit doesn't fire if I modify
all lines that include the previously +'d entry (for instance
call/rtp_config.h). I take this to mean that all callers of
for instance rtp_config.h now obtain checkdeps permission
elsewhere, closer to where they're located. This change should
not change checkdeps behaviour, therefore.
Bug: webrtc:4243
Change-Id: Ia909d13c5d79cb244f45b737142d2f47568ba77e
Reviewed-on: https://webrtc-review.googlesource.com/77801
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23337}
It was replaced be webrtc::DegradationPreference in this CL:
https://webrtc-review.googlesource.com/c/src/+/77024
But some downstream code was still referencing it.
Bug: webrtc:8830
Change-Id: Ibd0a3d15df7f13473c0f37a2493dd70cec6c0482
Reviewed-on: https://webrtc-review.googlesource.com/78082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23335}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
Instead of checking for an exact bitrate check that the bitrate is between
the min and max values.
Also relax a threshold in a bandwith adaptation test.
Bug: webrtc:9280
Change-Id: I465d785a53759f73242198ee1ccd7da1a26c48b7
Reviewed-on: https://webrtc-review.googlesource.com/78041
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23330}
The ERLE computation was improved by two means:
- The update function was always called and just parts of the internal code reacts to the converged filter flag
- When computing the ERLE, the ratio of energies is now computed using more points and, therefore, a more robust estimation is achieved.
Bug: webrtc:9284
Change-Id: Ie4f871f19cfad1a13741352ddd7b0a27ad6c3fb6
Reviewed-on: https://webrtc-review.googlesource.com/77767
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23329}
The overflow currently does not cause any problems, but it has been
found that it can cause crashes after a refactoring that is coming in
the near future.
Bug: webrtc:9180
Change-Id: Ia2c4e545c062c4f8ad13cbc47b8796c6e8a4e906
Reviewed-on: https://webrtc-review.googlesource.com/77667
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23327}
Enables us to do stuff like:
TEST(WindowsVersion, GetVersionGlobalScopeAccessor) {
if (GetVersion() < VERSION_WIN10) {
MethodNotSupportedOnWin10AndLater();
} else {
MethodSupportedOnWin10AndLater();
}
}
which is useful when working with Windows.
Note that, I also port a limited part of base::win::RegKey but only
those parts that are needed to implement OSInfo. Hence, I don't expose
any RegKey APIs.
NOTRY=TRUE
No-Presubmit: True
Bug: webrtc:9265
Change-Id: Ia2fc0963f24044ffaad954aa21d28df9c32b3ee7
Reviewed-on: https://webrtc-review.googlesource.com/77723
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23326}
The default implementation of the method is to return an empty list.
Clients should update their implementations before WebRTC starts calling
this method.
Also updates internal WebRTC implentations of this interface to
implement the method.
Bug: webrtc:7925
Change-Id: I258de2f09f6d4cc5dd9f4657e5d54e8411f8f5d8
Reviewed-on: https://webrtc-review.googlesource.com/77641
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23325}
Previous code had a FakePeriodicVideoSource and a
VideoTrackSource, where the latter is reference counted and
outlives the former. That results in potential races when
RemoveSink is called on the VideoTrackSource after the
FakePeriodicVideoSource is destroyed, with a complicated sequence
to do correct shutdown.
The new class, FakePeriodicVideoTrackSource, owns a
FakePeriodicVideoSource, and they get the same lifetime.
Bug: webrtc:6353
Change-Id: Ic33b393e00a31fa28893dce2018948d3f90e0a9e
Reviewed-on: https://webrtc-review.googlesource.com/76961
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23320}
Following a change in libvpx, FullStackTest.VP9SVC_3SL_High has
become flaky. It will be disabled until the libvpx issue is fixed.
Bug: webrtc:9293
NOTRY: true
Change-Id: Ib375363bdefdbb4104130a1f0f02ea34dc26e7f9
Reviewed-on: https://webrtc-review.googlesource.com/77663
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23319}
This is a reland of 79ded653fee7183d5c0d94c5addf570bcfb29c9e
Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}
TBR=henrik.lundin@webrtc.org
Bug: webrtc:9280
Change-Id: I6bfcd1c5e1d5298543024a0faa6a695026434df3
Reviewed-on: https://webrtc-review.googlesource.com/77980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23318}
- Limit framerate by dropping frames before encoding.
- The max framerate at screen sharing is set to 5fps.
Bug: webrtc:9261
Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760
Reviewed-on: https://webrtc-review.googlesource.com/76921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23316}
We have a number of RtpParameters that aren't implemented. If a client
is setting these values it creates unexpected results when the value
doesn't do anything for them. This change incorporates returning the
correct error if the parameter is unimplemented.
It also changes the scale_resolution_down_by and scale_framerate_down_by
RtpEncodingParameters to rtc::Optionals because they aren't implemented.
This change is part of the effort to ship get/setParameters in Chrome.
Bug: webrtc:8772
Change-Id: I9797695e5116e6aeb3c02afddbf460b2a0d7d5ab
Reviewed-on: https://webrtc-review.googlesource.com/75421
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23314}
It seems convenient since the EngProd team make repo-wide changes
every now and then.
Bug: None
Change-Id: If429fc8ed503a3c24c912ac2e8d120f93edc4823
Reviewed-on: https://webrtc-review.googlesource.com/77760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23312}
- Remove referencing control from encoder wrapper. Use fixed temporal
prediction structure.
- Remove flexible mode from encoder wrapper. It only worked with
referencing control which this CL removes.
- Remove external framerate/bitrate controller. Keep codec's internal
frame dropping enabled at screen sharing.
- Use GetSvcConfig() to configure layering.
Bug: webrtc:9261
Change-Id: I355baa6aab7b98ac5028b3851d1f8ccc82a308e0
Reviewed-on: https://webrtc-review.googlesource.com/76801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23311}
Any injected loggable or NativeLogger would be deleted if PCFactory
was reinitialized without calling setInjectableLogger. Now native
logging is not implemented as a Loggable, so it will remain active
unless a Loggable is injected.
This is a reland of 59216ec4a4151b1ba5478c8f2b5c9f01f4683d7f
Original change's description:
> Injectable logging
>
> Allows passing a Loggable to PCFactory.initializationOptions, which
> is then injected to Logging.java and logging.h. Future log messages
> in both Java and native will then be passed to this Loggable.
>
> Bug: webrtc:9225
> Change-Id: I2ff693380639448301a78a93dc11d3a0106f0967
> Reviewed-on: https://webrtc-review.googlesource.com/73243
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23241}
Bug: webrtc:9225
Change-Id: I2fe3fbc8c323814284bb62e43fe1870bdab581ee
TBR: kwiberg
Reviewed-on: https://webrtc-review.googlesource.com/77140
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23310}
This CL applies a high pass filter to the delay estimator signals which
improves the adaptation of the matched filters in noisy environments.
This results in faster delay estimation.
Bug: webrtc:9288
Change-Id: I8ffe5442eab7ac2f10a7ba236b08a0f07ec90645
Reviewed-on: https://webrtc-review.googlesource.com/77725
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23308}
This reverts commit 79ded653fee7183d5c0d94c5addf570bcfb29c9e.
Reason for revert: Different repos have different Opus
Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}
TBR=henrik.lundin@webrtc.org,gustaf@webrtc.org
Change-Id: I3c18db2d6052c4049d836c3e595b00189aebcbc8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9280
Reviewed-on: https://webrtc-review.googlesource.com/77800
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23307}
Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
CL re-enables recently disabled unittests and updates the expected bitrates.
Bug: webrtc:9280
Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
Reviewed-on: https://webrtc-review.googlesource.com/77766
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23306}
