The tests are now all gated behind a singe WEBRTC_HAVE_SCTP flag,
including the invocations, so the issue of configs that vary which
tests are included should have gone away.
This CL also clears out the remains of
DataChannelIntegrationTestWithFakeClock, which is no longer used.
Bug: webrtc:12525
Change-Id: Ie0133e3e84358e9238b8e7465200f000519b8765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219783
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34085}
The retransmission timeout (RTO) value is updated on every measured
RTT and is a function of the RTT value and its stability. In reality,
the RTT is never constant - it fluctuates, which makes the RTO become
much larger than the RTT. But for extremely stable RTTs, which we get
in simulations, the RTO value can become the same as the RTT, and that
makes expiration timers be scheduled to the RTT value, and will race
with packets that are expected to stop the expiration timer. And that
race should be avoided in simulations.
So ensuring that the RTO value is always greater, if only be a single
millisecond, will work fine in these simulations.
Bug: webrtc:12614
Change-Id: I30cf9c97e50449849ab35de52696c618d8498128
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219680
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34084}
This allows callers to use timestamps generated from their own clocks
without worrying about converting to webrtc time.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:11327
Change-Id: I7b1935654a2b23cf844c7b3622ed68763ced9da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219785
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34076}
WebRTC is going to remove cricket::DtlsTransportState and switch to
webrtc::DtlsTransportState. This CL introduces a set of new methods to
allow to subscribe to DTLS transport state updates while WebRTC
internally migrates away from cricket::DtlsTransportState.
No-Try: True
Bug: webrtc:12762
Change-Id: I79cf48e2f7122cc76fde6e4fd1541382fd096d53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218607
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34075}
This change prepares for a later change in Chromium that makes it
stop depending on headers exposed by WebRTC that require inclusion of
api/proxy.h.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:12787
Change-Id: I628424fe49e873027595b80336be2b821c22245e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219688
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34072}
The timestamps returned by the clocks do not have an epoch. Each clock
should be able to convert a timestamp it returns to an NTP time.
The default implementation for querying for an NTP time is converting
the current timestamp.
This is favored over returning the offset between the relative and the
NTP time because there is a field trial that makes the real clock revert
to using system dependent methods for getting the NTP time.
Bug: webrtc:11327
Change-Id: Ia139b2744b407cae94420bf9112212ec577efb16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219687
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#34071}
`CurrentTime` and `CurrentNtpTime`. Make all other methods non-virtual.
Bug: webrtc:11327
Change-Id: I391d9eaec1ba27ec4f8e1901498c68c28a7ec4ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219466
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#34065}
These tests document existing behavior; there are notes about where
this behavior is either surprising or wrong.
Bug: chromium:1148951
Change-Id: If9875fb744c44c129ff9949d1bab3d3d99f17b81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217520
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34058}
This change enables the use of two different adaptation speeds of the
matched filter of the delay estimator of AEC3.
One speed is used when no delay has been found, and one is used after a
reliable delay has been found. The purpose is to use a slower adaptation
speed to reduce the risk of divergence during double-talk without
slowing down the search for the initial delay.
The CL prepares for experimentation by adding field trials for
controlling the two adaptation speeds.
Bug: webrtc:12775
Change-Id: I817a1ab5ded0f78d20de45edcf04c708290173fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219083
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34055}
Reduced the level so that the library can be run with INFO level without
a lot of spam. VERBOSE is still reserved for frequent logs.
Also, using WARNING for logs that are not fatal and which can easily
be triggered by the user.
Bug: webrtc:12614
Change-Id: If09c302b2b5bfc002471f86a8aeb74ba1172c705
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219465
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34054}
Reporting the duplicate TSNs is a SHOULD in the RFC, and using the
duplicate TNSs is a MAY, and in reality I haven't seen an implementation
use it yet. However, it's good for debugging and for stats generation.
Bug: webrtc:12614
Change-Id: I1cc3f86961a8d289708cbf50d98dedfd25077955
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219462
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34053}
This is to address flakiness of "DoubleThread" tests for the media
channel class. More investigation is in order though, so I'm adding
a TODO. The bug appears to be in test code only though, so this is
just to deflake the bots.
Bug: webrtc:12783
Change-Id: Ib6cf78927f2a9be9d2c6aa7f6915b1131a206e7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34049}
In CL https://webrtc-review.googlesource.com/c/src/+/216323 we fixed
the issue where I420 and I420A not being equal would result in dropping
frames in release builds.
But we forgot to update the corresponding DCHECK, meaning the I420 not
being the same as I420A issue still causes crashes on debug builds.
(I must have been running a release build not to catch this before?)
This CL replaces the DCHECK_EQ with an RTC_NOTREACHED inside the
IsCompatibleVideoFrameBufferType check.
Because this only affects debug builds, this CL does not need to be
backmerged anywhere.
Bug: chromium:1203206
Change-Id: I101823e8bca293e94d0f7ce507fe78cedff3ea1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219281
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34048}
This is a follow-up to r34019. It adds checks for when padding can be
sent before media - and how timestamps are set on RTX padding.
Bug: webrtc:11340
Change-Id: I46fbd3c3eff9e308b5c65220718df749f2d9c46b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219162
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34041}
This change adds the field trial "WebRTC-TransientSuppressorForcedOff"
that can be used to disable the transient suppressor (removal of
keyboard typing sounds). The field trial can be enabled by users via
command-line or via experimentation.
Bug: chromium:1186705
Change-Id: I7272df6a20fbbee24a7ba0904502c76bd775d275
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219282
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34038}
Specifically, use reference instead of pointer for out parameter
and place the out parameter last, for the following methods
ReadUInt8
ReadUInt16
ReadUInt32
ReadBits
PeekBits
ReadNonSymmetric
ReadSignedExponentialGolomb
ReadExponentialGolomb
Bug: webrtc:11933
Change-Id: I3f1efe3e29155985277b0cd18700ddea25fe7914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218504
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34037}
