This should allow us to remove some SDP parsing in Chromium.
Bug: webrtc:12215
Change-Id: Ib85593d1c9226b29f2ec18617f945c76eca3b2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197806
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32840}
Adds ability to specify desired frame size separate from actual clip
resolution, as well as clip and desired fps.
This allows e.g. reading an HD clip but running benchmarks in VGA, and
to specify e.g. 60fps for the clip but 30for encoding where frame
dropping kicks in so that motion is actually correct rather than just
plaing the clip slowly.
Bug: webrtc:12229
Change-Id: I4ad4fcc335611a449dc2723ffafbec6731e89f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195324
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32839}
Two audio channels going into the AudioSource::Sink can either be
down-mixed to mono or encoded as stereo. This change enables WebRTC
users (such as Chromium) to query the number of audio channels actually
encoded. That information can in turn be used to tailor the audio
processing to the number of channels actually encoded.
This change fixes webrtc:8133 from a WebRTC perspective and will be
followed up with the necessary Chromium changes.
Bug: webrtc:8133
Change-Id: I8e8a08292002919784c05a5aacb21707918809c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197426
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32836}
- This CL also affects some return type handling in Android Voip demo
app due to changes in return type handling.
Bug: webrtc:12193
Change-Id: Id76faf7c871476ed1f2d08fb587211ae234ae8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196625
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32821}
Now that RtpVp9RefFinder sets an additional reference on the frame instead of marking it as inter_layer_predicted it is no longer used.
Bug: webrtc:12206
Change-Id: I10e0930336eafc32dc86feb2f690cb131e55be2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196740
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32814}
After the refined filter has been determined to perform better than
the coarse filter, and the coefficients of the coarse filters are
overwritten by the ones from the refined filter, at least 100 ms have
to pass before the adaptation of the refined filter is allowed to speed
up due to good coarse filter performance.
This change solves the vicious circle described in webrtc:12265, where
the coarse and refined filters can diverge over time.
This feature can be disabled remotely via a kill-switch. When disabled
the AEC output is bit-exact to before the change.
Bug: webrtc:12265,chromium:1155477
Change-Id: Iacd6e325e987dd8a475bb3e8163fee714c65b20a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196501
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32801}
This will soon become a compile-time error. Fix class hierarchies that
wrap StrictMock in a NiceMock or vice-versa by removing redundant
wrappings and removing inheritance from Nice/StrictMock and fixing the
call sites as appropriate.
Bug: b/173702213
Change-Id: Ic90b1f270c180f7308f40e52e358a8f6a6baad86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32783}
As requested on bugs.webrtc.org/12096#c2, this CL adds a Chromium
metric OWNERS in order to always have their review when WebRTC's UMA
metrics are updated.
Bug: webrtc:12096
Change-Id: Icd9ab7dda5f7a4ba6ac078f667c1fd39f3314123
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32728}
Needed in order to return different codes for different failures
in initialization.
Sideswipe: Check TURN URL hostnames for illegal characters.
Bug: webrtc:12238
Change-Id: I1af3a37b9654b83b268304f7356049f9f3786b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32710}
Instead of flushing all packets, it makes sense to flush down to the target level instead. This CL also initiates a flush when the packet buffer is a multiple of the target level, instead of waiting until it is completely full.
Bug: webrtc:12201
Change-Id: I8775147624536824eb88752f6e8ffe57ec6199cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193941
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32701}
This CL does not aim at cleaning up simulcast/SVC configuration, just to make it possible to set the scalability mode for AV1. Implementing a codec agnostic SVC/simulcast API is a (big) project on its own.
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
BUG: webrtc:11607
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192541
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32631}
This will cause encoding of a STUN message with an over-long
byte string attribute to fail.
Bug: chromium:1144646
Change-Id: I265174577376ce01439835c03f2d46700842d211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191322
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32603}
Starting from Android N, mobile app may not be able to access
microphone while in background where it fails the call.
In order to mitigate the issue, delay the ADM initialization
as late as possible.
Bug: webrtc:12120
Change-Id: I0fbf0300299b6c53413dfaaf88f748edc0a06bc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191100
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32598}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
Avoid excessive echo suppression in frequencies above 2 kHz when
there is a dominant nearend. Calls with clock drift will not be affected
by this change as they tend to have less accurate linear filters.
Bug: webrtc:11985
Change-Id: Iddc628da5e2ba572c1b47acd87dd3be35260dca1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188580
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32533}
Calculate quality metrics for dropped frames by comparing original
frame against last decoded one.
This feature makes comparison of encoders which do/don't drop frames
more fair.
The feature is controlled by analyze_quality_of_dropped_frames flag
and is disabled by default.
Bug: none
Change-Id: Ifab8df92d0b76e743ff3193c05d7c8dbd14921c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190660
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32518}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
This feature is active if and only if the RTP header extension
playout-delay is used with min playout delay=0 and max playout delay>0.
In this case, a maximum composition delay will be calculated and attached
to the video frame as a signal to use the low-latency renderer algorithm,
which is landed in a separate CL in Chromium.
The maximum composition delay is specified in number of frames and is
calculated based on the max playout delay.
The feature can be completetly disabled by specifying the field trial
WebRTC-LowLatencyRenderer/enabled:false/
Bug: chromium:1138888
Change-Id: I05f461982d0632bd6e09e5d7ec1a8985dccdc61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190141
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32493}
This allows making more members (including IsUnifiedPlan) const in a future CL.
Also revises the test for ReportUsageHistogram to use a configuration member
variable rather than a hook function in PeerConnectionFactory.
Bug: webrtc:12079
Change-Id: I6f1af7d6164c8a0d8466f76378a925d72d57d685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190280
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32485}
This is a reland of 239f92ecf7fc8ca27e0376dd192b33ce33377b3c
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
Bug: webrtc:3513
Change-Id: I48e338100f829f1df5b8165217c89b5ef860fe79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32457}
