Much like https://bugs.chromium.org/p/chromium/issues/detail?id=855900,
the int32 gain table isn't always small enough for plain multiplication
with an int16.
This appears fixable through regular fixed-point arithmetic (multiply
out[i][n] by integer and fractional part of gain separately), but it's
less readable.
Bug: chromium:858989
Change-Id: Ie5aac25fd0cca4e51858cba69bde06c54a5d31bf
Reviewed-on: https://webrtc-review.googlesource.com/86602
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23815}
`gclient setdep` was changed in https://chromium-review.googlesource.com/1123940
to support any prefix as well, but note that that was a backwards incompatible
change, because it now requires the prefix to be passed. So we just stop stripping
the prefix in this CL.
Also clarify the error when a CIPD dep is present in WebRTC and missing in Chromium.
No-Try: True
TBR: phoglund@webrtc.org
Bug: webrtc:9470, chromium:858978
Change-Id: I5e42bbda04db37a628a0ac1de69667b9a30dd793
Reviewed-on: https://webrtc-review.googlesource.com/86280
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23814}
The ReceiveSendsFromThread function calls the OnMessage function.
However, instead we should be calling the Dispatch function which does the same thing as the OnMessage function except that it also does additional logging.
This logging is being missed for the cases where we call functions on a thread using the Invoke function.
Calling Dispatch fixes the issue and makes sure that this code path is consistent with other paths of posting to a thread like Post function which goes through Dispatch ultimately.
Bug: None
Change-Id: I75a5c8b464226cf4de60a3d19dff48f9e6197cca
Reviewed-on: https://webrtc-review.googlesource.com/85885
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23813}
This CL adds VP9 profile information in SDP. It adds the necessary fields and
enums to codec containers.
Additional profiles will be followed.
Bug: webrtc:9376
Change-Id: I78574714f06f8087262a71dd64c01f31a229dd54
Reviewed-on: https://webrtc-review.googlesource.com/81960
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23810}
When a TCP TURN port is destroyed, a TURN refresh request with zero
lifetime is first sent to release the TURN allocation at the server,
and the underlying TCP connection is closed afterwards.
The closing of the TCP connection is handled first by the
VirtualSocketServer in our test infrastructure, and the corresponding
server socket is asynchronously destroyed at the TURN server. The
refresh request is however still passed to this server socket and
further signaled to the TURN server, which fails a DCHECK. The
server implementation should disable any firing of signals from a
server socket to be destroyed.
The bug id is set to None since this is a one-liner CL.
TBR=pthatcher@webrtc.org
Bug: None
Change-Id: Ib457b3800511a322ef69d67c71f2de05f3d67967
Reviewed-on: https://webrtc-review.googlesource.com/86501
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23809}
ADAPTER_TYPE_ANY can be used to set the network ignore mask if an
application does not want candidates from the any address ports, the
underlying network interface types of which are not determined in
gathering. The ADAPTER_TYPE_ANY is also given the maximum network cost
so that when there are candidates from explicit network interfaces,
these candidates from the any address ports as backups, if they ever
surface, are not preferred if the other candidates have at least the
same network condition.
Bug: webrtc:9468
Change-Id: I20c3a40e9a75b8fb34fad741ba5f835ecc3b0d92
Reviewed-on: https://webrtc-review.googlesource.com/85880
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23807}
1. Adds support for Reset calls in AGC2. The AGC will be reset during
analog gain changes.
2. Allows AdaptiveModeLevelEstimator to return estimates > 0. This can
happen if the signal gain is too high. It's needed for letting the
analog AGC know that the gain is too high.
Bug: webrtc:7494
Change-Id: I38def17c21cc01c36aaea79a2401d8c2f289407b
Reviewed-on: https://webrtc-review.googlesource.com/79360
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23805}
Now that there is only one implementation of the decision logic, there
is no longer any need to have GetDecisionSpecialized being separate.
Bug: webrtc:9421
Change-Id: Id364ce09ac05d106652d749502058056f11bba27
Reviewed-on: https://webrtc-review.googlesource.com/86604
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23804}
Building NetEqEventLogInput requires protobuf support, while building
NetEqRtpDumpInput located in the same file does not. This makes both
classes unusable when protobuf support is lacking. With this CL, the
NetEqEventLogInput is broken out into separate files, to allow usage
of NetEqRtpDumpInput even when protobufs are not supported.
Bug: webrtc:9421
Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f
Reviewed-on: https://webrtc-review.googlesource.com/84587
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23803}
This CL adds two flags to audioproc_f. The flags control
AgcManagerDirect. The flags are
'--experimental_agc_agc2_level_estimator' and
'--experimental_agc_agc2_digital_adaptive'.
After this CL, the flags are be applied to AgcManagerDirect. The flags
have no effect in release-mode. They cause a crash in debug-mode.
In an upcoming CL, '--experimental_agc_agc2_level_estimator 1' will
replace the speech level estimation in ExperimentalAgc with that of
AGC2.
'--experimental_agc_agc2_digital_adaptive 1' will replace the digital
gain selection and application with that of AGC2.
These audioproc_f will activate both new settings:
./out/Target/audioproc_f --agc 1 --experimental_agc 1
--experimental_agc_agc2_digital_adaptive 1
--experimental_agc_agc2_level_estimator 1 --simulate_mic_gain 1
--simulated_mic_kind 2
See also https://webrtc-review.googlesource.com/c/src/+/79360
Bug: webrtc:7494
Change-Id: If0e65893dffdddb312e553787b8cedaf9a334323
Reviewed-on: https://webrtc-review.googlesource.com/86548
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23802}
The AGC submodule of APM changes analog gain. These gain changes are
typically ignored by the test tool audioproc_f.
There is an option of the test tool to take action on the gain
changes. It's the '--simulate_mic_gain' option. The option converts
the analog gain to a digital gain. The digital gain is applied to the
capture stream.
This change adds a new simulated microphone kind. The new microphone
has a gain curve defined by
modules/audio_processing/agc/gain_map_internal.h. That gain curve
defines how AGC1 expects a microphone to behave.
Bug: webrtc:7494
Change-Id: Ifb3f54a8c6f8c001a711fa977f39f32413069780
Reviewed-on: https://webrtc-review.googlesource.com/86128
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23801}
This is a reland of 80c4cca4915dbc6094a5bfae749f85f7371eadd1
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
This removes the legacy DelayBasedBwe to reduce code redundancy and
avoid the risk of applying changes on only one version.
Bug: webrtc:8415
Change-Id: I88aba03adbb77ee0ff0a97a8b3be6ddf028af48a
Reviewed-on: https://webrtc-review.googlesource.com/85364
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23798}
This replaces the old AlrDetector used by the pacer with the one in
GoogCC. This reduces the risk of accidentally changing only one version.
Note that the pacer instance will be removed when moving over to the
task queue based send side congestion controller.
Bug: webrtc:8415
Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5
Reviewed-on: https://webrtc-review.googlesource.com/85363
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23791}
In this work the performance of the linear filter is
estimated. The estimation aims at capture situations when the linear
filter is largely over-estimating the echo. In those circumstances,
the linear filter is scaled with the purpose of accelerating its
convergence.
Change-Id: I05ea3739d82838a6f08673432da92125c47943e0
Bug: webrtc:9466,chromium:857426
Reviewed-on: https://webrtc-review.googlesource.com/86133
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23789}
We don't want to maintain our own versions. This CL is step one in
getting rid of them.
Bug: webrtc:9473
Change-Id: Ib8a54288509f4768b482367b738224869a5af559
Reviewed-on: https://webrtc-review.googlesource.com/86282
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23786}
This reverts commit 5eb6045ce5754ce815929c54dd27ab0bf3ae62ba.
Reason for revert: Test breaks downstream.
Original change's description:
> Unit test for case where the number of active and configured spatial
> layers differ.
>
> Bug: webrtc:9472
> Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
> Reviewed-on: https://webrtc-review.googlesource.com/85644
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23782}
TBR=brandtr@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
Change-Id: Ib97cdb127e79ee969f7cb3f931cb7bd533f13af0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9472
Reviewed-on: https://webrtc-review.googlesource.com/86320
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23785}
To allow the AudioDeviceModule to be reinitialized on a different thread
after termination, detach AudioDeviceModule and the input/output devices
when Terminate is called. Also destroy the AudioDeviceBuffer.
Bug: webrtc:7452
Change-Id: I50ef77c531f33d4efa0567d0475dd8280337bed9
Reviewed-on: https://webrtc-review.googlesource.com/86127
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23784}
Remove unused member noise_var from RateControlInput struct.
Rename incoming_bitrate to estimated_throughput_bps to reflect
that the AimdRateControl may be running on either the send side
or the receive side.
Bug: webrtc:9411
Change-Id: Ie1ae0c29dc9559ef389993144e69fcd41684f1a4
Reviewed-on: https://webrtc-review.googlesource.com/83728
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Anastasia Koloskova <koloskova@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23783}
This CL removes the constraint that freezes the filter adaptation
whenever the estimated echo or the prediction error is saturated. This
allows for much more rapid filter recovery in cases where the echo path
gain for some reason changes, such as when the analog AGC gain is
adjusted or the loudspeaker volume is changed.
TBR: devicentepena@webrtc.org
Bug: webrtc:9466,chromium:857426
Change-Id: Ic0b3b03f41f12e9a607aaadd2ee91cbaa16cac52
Reviewed-on: https://webrtc-review.googlesource.com/86124
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23775}
The Copy() function previously did not copy the logging timestamp.
To be able to use Copy() in this test, we add private copy
constructors for RtcEvents which the Copy() can use to copy
everything including the timestamp.
Also adds missing test for RtcEventAlrState,
RtcEventIceCandidatePairConfig and RtcEventIceCandidatePair.
Bug: webrtc:8111
Change-Id: I3901231735baa4e671173c921eada0a4be6de7c9
Reviewed-on: https://webrtc-review.googlesource.com/86042
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23774}
This CL adds a field trial that disables the feature that the pacer will
ignore the pacing rate and send extra fast to drain the queues if the
pacer queue starts to fill up. BBR assumes that the pacing rate will be
respected and sending more increase the risk of overestimating the
bandwidth.
Bug: webrtc:8415
Change-Id: Ibba315360dafef1c317d14a83199172f9f8cc6aa
Reviewed-on: https://webrtc-review.googlesource.com/80964
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23773}
This CL changes the behavior when the main filter diverges.
Instead of entering non-linear mode, the AEC continues to operate in
linear mode but estimates the residual echo differently. R2 is S2
scaled by a factor of 10.
Bug: chromium:857018,webrtc:9462
Change-Id: I41212efe164ad319cf38a163cdf9d3ea151e0997
Reviewed-on: https://webrtc-review.googlesource.com/85981
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23772}
Also adds tests, and adds a bit of logging in ParseIceServers.
Bug: chromium:718508
Change-Id: Id41ccb7cccbdab5af76e380b32b4d8ba0c4a0a72
Reviewed-on: https://webrtc-review.googlesource.com/86121
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23769}
Those are static functions in the spec, so implemented as member functions
of the PeerConnectionFactory instead.
Bug: webrtc:7577, webrtc:9441
Change-Id: Iccb24180e096e713d24e7e25ecfd5d7bbd7638f9
Reviewed-on: https://webrtc-review.googlesource.com/85341
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23768}
The behavior of division-by-zero is undefined, so the DivisionByZeroFails test isn't correct. As we don't need any specific behavior on division-by-zero we leave the current code untouched.
Additionally, since the DivisionFailsOnLargeSize EXPECT_DEATH checks rely on DCHECKs, we only run those when DCHECKs are enabled.
Bug: webrtc:9443
Change-Id: I0fdd7be55a7bc76b4203b2f6d5cd0ed8ac5cc688
Reviewed-on: https://webrtc-review.googlesource.com/85362
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23767}
Both incoming and outgoing datachannels should cause
the DATA_ADDED flag to be set.
This CL also moves all tests into their own file, and
improves scaffolding.
Bug: chromium:718508
Change-Id: I5c4c257ccb6f26799f7593bce8b27ebf59015b1e
Reviewed-on: https://webrtc-review.googlesource.com/85348
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23766}
