This CL:
-Adds another config parameter that controls the duration of the initial
state.
-Adds reading of that parameter in audioproc_f from the json settings file.
-Adds missing reading of another parameter in audioproc_f from the json
settings file.
Bug: webrtc:8671
Change-Id: Ie6164c360492de5e6b0ade8838bbabe214560b5e
Reviewed-on: https://webrtc-review.googlesource.com/94621
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24360}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
Bug: none
Change-Id: I5a34c52e66bbd6c05859729ed14ae87ac26b5969
Reviewed-on: https://webrtc-review.googlesource.com/91861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24359}
This removes the reserved bitrate feature as it is not currently used.
This saves debugging time as there is less code to investigate. This
also makes it easier to compare the code with the task queue based
version which lacks this feature.
Bug: webrtc:9586
Change-Id: I207624ceb8d203c88c3d01bfc753d60523f99fe4
Reviewed-on: https://webrtc-review.googlesource.com/92622
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24357}
Change experimental max bitrate setting from 1.6Mbps to 1.25Mbps in
order to allow a larger fraction of participants to receive this layer.
Add a new field trial to allow setting the number of temporal layers for
the high-quality simulcast stream in screensharing separately from the
temporal layer count for regular video.
Bug: webrtc:9477
Change-Id: I1341b774f870c50710901da24963bd3ede96ffd8
Reviewed-on: https://webrtc-review.googlesource.com/95101
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24356}
- Allow use of flexible mode which was blocked in webrtc:9261 since it
only worked together with old screen sharing. Since webrtc:9244 flexible mode
works with both normal and screen coding modes.
- Add unit test that checks that reference list encoder writes into RTP
payload descriptor and the predefined one match.
Bug: webrtc:9585
Change-Id: I4a1bdc51cbf15e7224cc7c271af8b2e3d46657d1
Reviewed-on: https://webrtc-review.googlesource.com/94778
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24355}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
The was another definition of the --logs flag in full_stack_tests.cc.
The effect was that --logs set an unused bool, and the value of
FLAG_logs in test_main.cc was always false, regardless of actual
command line.
Tbr: sprang@webrtc.org
Bug: None
Change-Id: I073f8025dd897909c7e2b8d7c0ee080cb4b456ca
Reviewed-on: https://webrtc-review.googlesource.com/94900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24347}
Because DirectTransport is switched on SimulatedPacketReceiverInterface
we can't create it from some specific config in ctor, so all ctors,
that accept specific config are deprecated and you should pass concrete
implementation of underlying implememntation instead.
Bug: webrtc:9630
Change-Id: I7f241f310c993d8136b40898e55a6915924d61bd
Reviewed-on: https://webrtc-review.googlesource.com/94841
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24344}
It didn't account for implicit bitrate allocation, which is used in
some unit tests, when bitrate distribution is done by the encoder
wrapper.
Bug: none
Change-Id: I8fcf28e10f7a6c378580ef917221ad5c8d3869c9
Reviewed-on: https://webrtc-review.googlesource.com/94775
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24343}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I1be59a9277aad8f51765c26e34ab16b63bcaeb42
Reviewed-on: https://webrtc-review.googlesource.com/94774
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24340}
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.
This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.
Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
Linux has all hw webcams now, but it's tricky to stop invoking this
script just on Linux, so just make it do nothing for now.
It should be safe to turn on video_capture_tests on linux
after this lands.
Bug: webrtc:9292, webrtc:9636
Change-Id: I6e86716b4c7ca43244596f806ff904b7fdf9201a
Reviewed-on: https://webrtc-review.googlesource.com/94769
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24334}
This CL lowers the margins in the AEC3 conservative mode to increase
the transparency when there are audio buffer issues, and during call
startup.
In particular, this CL adjusts the parameters and thresholds to
-Make the requirements for filter divergence more strict, to minimize
the transparency loss during minor filter divergence.
-Decrease the echo power uncertainty used during initial filter
convergence, to increase transparency after audio buffer issues.
-Deactivate the enforcement of conservative suppressor gain after
audio buffer.
Bug: webrtc:9641,chromium:875611
Change-Id: Ie171bb411f17a1e8661c291118debd334f65c74f
Reviewed-on: https://webrtc-review.googlesource.com/94776
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24333}
This allows clients to stop using the old pointer-to-submodule interfaces
for enabling/disabling AEC2 and AECM.
The legacy suppression level flag for AEC2 is not yet activated.
NoTry=TRUE
Bug: webrtc:9535
Change-Id: Ie2c3378d832a6b393aec656d96597f85e299f500
Reviewed-on: https://webrtc-review.googlesource.com/94771
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24328}
Stop using of deprecated ctors of FakeNetworkPipe in most part of
webrtc codebase, except DirectTransport, where further refactoring will
be continued in future CLs.
Bug: webrtc:9630
Change-Id: I823404469e461601ddbc026eaeac668eeda8045f
Reviewed-on: https://webrtc-review.googlesource.com/94763
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24327}
This CL is a minimum effort/low risk fix.
Later CLs take a more thorough approach.
Bug: webrtc:9634
Change-Id: I728a061a4e71b38a559ee438646de83cd2cb3517
Reviewed-on: https://webrtc-review.googlesource.com/94760
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24326}
Making video_capture_tests have the non_parallel_console_test_launcher
type will pass --workers=1 to gtest-parallel, which ensures the tests
execute sequentially. This is necessary now that we're accessing a
real physical webcam, which is a system-wide resource that doesn't
work well if several processes access it concurrently.
Follow-ups:
1) get video_capture_test back up on Linux
2) drop sw webcams for Mac and Win
3) remove ensure_webcam_is_running.pu and surrounding machinery
Bug: webrtc:9292
Change-Id: I5e3347ad234f6b942de736513075097d79c0fd36
Reviewed-on: https://webrtc-review.googlesource.com/94761
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24325}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
Bug: webrtc:8396
Change-Id: I032b12f3813af6ac3ea0dfb688006899dffe4855
Reviewed-on: https://webrtc-review.googlesource.com/94150
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24323}
Add replacements for all FakeNetworkPipe constructos, that will accept
instance of NetworkSimulationInterface instead of config to be able to
use any implmentation of network simulation.
Bug: webrtc:9630
Change-Id: Ifceb2f0d028faf255648891ce695b3742f866044
Reviewed-on: https://webrtc-review.googlesource.com/94541
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24320}
Some changes need access to both the APM interface and the AECs,
hence we can't make the changes inside the AECs themselves.
The proxies also make it easy to drop support for individual parts of the
interfaces one at a time.
Bug: webrtc:9535
Change-Id: I3398e1182157f7d8b1e4c455060b830b61c20dd9
Reviewed-on: https://webrtc-review.googlesource.com/94500
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24317}
There is no difference between how we handle "generic" and "unkown" codecs,
so we don't need to represent both.
Bug: webrtc:8136
Change-Id: I42b0dbc8a0bae67cc21742303c963c8dd5bde1f6
Reviewed-on: https://webrtc-review.googlesource.com/92086
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24316}
Switch from using FakeNetworkPipe::SetConfig on holding direct reference
on instances of NetworkSimulationInterface if you are required to
reconfigure it in the runtime.
Also add fake_network_unittests to built test targets.
Bug: webrtc:9630
Change-Id: I5fd6b33904367aa6dc00ca2e2f5f780f433acf35
Reviewed-on: https://webrtc-review.googlesource.com/94510
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24314}
The other modes are little-tested and nigh-unsupported.
Surrounding APM code is tuned for high suppression.
Both WebRtcVoiceEngine and Chrome default all usage to high
suppression.
Bug: webrtc:9535
Change-Id: Ic1a6bd90b86a994338addfef7f473132ab43092a
Reviewed-on: https://webrtc-review.googlesource.com/91865
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24313}
Add FakeNetworkPipe ctor from NetworkSimulationInterface to be able to
inject different implementations of simulated networks.
If user requires to do something with injected simulation he can keep
raw reference, or introduce some proxy, that will hide shared ownership
from FakeNetworkPipe. Also FakeNetworkPipe guarantee to keep simulation
object alive while it is alive itself.
Bug: webrtc:9630
Change-Id: Ie00ea1d47bf7658b0c6433cf2efbf364f885c8a0
Reviewed-on: https://webrtc-review.googlesource.com/94153
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24312}
Make SimulatedNetwrokConfig configuration of default implementation of
NetworkSimulationInterface, that will be used by WebRTC in case of
network simulation.
Bug: webrtc:9630
Change-Id: Ib7c3d0c69fc09627f3d8694e61ac8409101e8392
Reviewed-on: https://webrtc-review.googlesource.com/94154
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24311}
