The probe configuration is currently a single field trial. To allow
multiple experiments with non-overlapping subsets of these keys I've
added a few extra keys that override different subsets of the config.
Bug: webrtc:10394
Change-Id: I54ffd1105129794fcdae4cce314910acaa4074af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138274
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28076}
The imports of Packet, ReceivedPacket from ForwardErrorCorrection::
collides with other usages of the names introduced in a followup CL.
Bug: webrtc:9883
Change-Id: Ib042c9091ad8e350cbdf01b837af09c820dbff33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138279
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28075}
This implements the essentials of RTCRemoteInboundRtpStreamStats. This
includes:
- ssrc
- transportId
- codecId
- packetsLost
- jitter
- localId
- roundTripTime
https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
The following members are not implemented because they require more
work...
- From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded,
packetsRepaired, burstPacketsLost, burstPacketsDiscarded,
burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate,
gapLossRate and gapDiscardRate.
- From RTCRemoteInboundRtpStreamStats: fractionLost.
Bug: webrtc:10455, webrtc:10456
Change-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28073}
RtpTransportInternal does not need to expose these. They are only used
by tests and for setting options. Instead, it can expose a SetRtpOption
and SetRtcpOption to set options relevant to each of its transports.
Also updates tests to work around no longer having access to internals.
This will simplify the composite needed during negotiation of different
RTP transport types, as we no longer need to have composites of both
RtpTransport and PacketTransport.
Bug: webrtc:9719
Change-Id: I91bfa6e95b7aa384d10497f47e7d2483c2e0bef2
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138282
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28066}
Because of a low bitrate target, base layer has drops much more frequently
than other layers. But it reduces overall framerate, especially then
input framerate is low (5 fps).
This CL allows pre-layer drops and disables droppoing on higher spatial
layers for screenshare, solving the issue.
Additional care have to be taken then new spatial layers are enabled
dynamically to not create non-compatible with RTP references.
Bug: webrtc:10257
Change-Id: Ie056484c99a3f35ff4405ef71337dc2d034db8bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138262
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28063}
Using simple IsEnabled/IsDisabled instead of the parser for Enabled/
Disabled flags to improve readability.
Bug: webrtc:9883
Change-Id: I3dbf906d49f99269f73a8ced6b3f042181228f3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138078
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28062}
It seems unnecessary to lock it if not actually reinitializing.
Bug: webrtc:10205
Change-Id: Ib3292e1d640a92a7df77400aebe9583cf877f824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/115460
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28060}
This CL adds error handling for sysconf, which can return -1 and
adds an RTC_CHECK_GT to ensure the value returned is always greater
than 0.
On top of that CpuInfo::DetectNumberOfCores is made thread safe because
the static local variable is initialized with the right values istead
of 0.
Bug: None
Change-Id: I294684e7380b12cda55ec8d6c7a90e132dc3af85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138210
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28057}
This CL makes it more flexible and easier to include/exclude H264 code
when using other build systems because it delegates the decision to
remove the code to the preprocessor instead of GN.
This CL should be a noop, and for WebRTC/Chromium the GN param
`rtc_use_h264` will still be the only thing to change in order to
include/exclude H264.
Moving code that requires ffmpeg or h264 out of the #ifdef/#endif
part should break the build since dependencies are only added if
`rtc_use_h264=true`.
Bug: webrtc:9213
Change-Id: Ibc04edc2f6b9e51489ffe638d5be4b32959cdca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137430
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28055}
Reland with fixes.
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: Id40a780d461e6b51cb44d275b8aa5d7b348d3586
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138215
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28054}
This is a reland of 74b373f04a69b279e45b0792d86c594cb33d22c1
Original change's description:
> Delete STACK_ARRAY macro, and use of alloca
>
> Refactor the few uses of STACK_ARRAY to avoid an extra copy
> on the stack.
>
> Bug: webrtc:6424
> Change-Id: I5c8f3c0381523db0ead31207d949df9a286c76ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137806
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28038}
Bug: webrtc:6424
Change-Id: Id635ccdfae12157cbb3ab9089c5e4a9f77f742ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138211
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28052}
This reverts commit eb1754c5750dfcad23ac62b47aa3aa2176ae7be2.
Reason for revert: breaks downstream projects
Original change's description:
> VP9 screenshare: Don't base layers frame-rate on input frame-rate
>
> If input framerate is a little unstable, using it to cap layers will
> make output framerate even smaller for longer periods of time.
>
> Also, fix screenshare_loopback test for low-fps vp9 testing.
>
> Bug: webrtc:10257
> Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28050}
TBR=ilnik@webrtc.org,ssilkin@webrtc.org
Change-Id: I82bfbac58249cfe0da5ff565aa97a4745fd078ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138213
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28051}
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28050}
This reverts commit 890bc3069cbababa19b40ec02684253d60e051b2.
Reason for revert: Div by zero.
Original change's description:
> Cleanup of video packet overhead calculation.
>
> This CL updates the video packet overhead calculation to make it more
> clear. This prepares for future work on improving the accuracy of the
> calculation.
>
> Bug: webrtc:9883
> Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28040}
TBR=sprang@webrtc.org,srte@webrtc.org
Change-Id: Icbdfc7b9252f8f9aa8e7e97b85b04171a27935e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138212
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28049}
- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports.
- Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor.
- Propagate maximum datagram size to video encoder via MediaTransportConfig.
TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport.
Bug: webrtc:9719
Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28047}
Prior to this CL, only the mline index of an ice candidate was used to
look up contents. However, due to recent changes, it is possible that
no mline index is specified, but that only a mid is specified.
No mline index is indicated with a -1 value.
This CL makes sure the mid is used if no mline index is given.
Bug: chromium:965483
Change-Id: I8962e71acb386f7b50349802f27358ba24c11921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138075
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28045}
This reverts commit 74b373f04a69b279e45b0792d86c594cb33d22c1.
Reason for revert: This breaks chromium, blocking webrtc from rolling.
...
In file included from ../../third_party/webrtc\rtc_base/strings/string_builder.h:23:
../../third_party/webrtc\rtc_base/string_utils.h(54,28): error: implicit conversion loses integer precision: 'std::__1::basic_string<wchar_t, std::__1::char_traits<wchar_t>, std::__1::allocator<wchar_t> >::size_type' (aka 'unsigned long long') to 'int' [-Werror,-Wshorten-64-to-32]
ws.size());
See https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8912652299012991936/+/steps/compile__with_patch_/0/stdout
Original change's description:
> Delete STACK_ARRAY macro, and use of alloca
>
> Refactor the few uses of STACK_ARRAY to avoid an extra copy
> on the stack.
>
> Bug: webrtc:6424
> Change-Id: I5c8f3c0381523db0ead31207d949df9a286c76ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137806
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28038}
TBR=kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I223fceab60855dde363cc9ce8375e8f5cca43c02
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6424
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138209
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28043}
The following private methods needlessly took a reference to the
RtpConfig on which they had worked, which was itself a member.
* ConfigureProtection
* ConfigureSsrcs
* ConfigureRids
Bug: None
Change-Id: I013ca438915336d1b8f3477fe8b9f1bf87f514f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138205
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28041}
This CL updates the video packet overhead calculation to make it more
clear. This prepares for future work on improving the accuracy of the
calculation.
Bug: webrtc:9883
Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28040}
Change the way the forget factor converge to the steady state so that we don't overemphasize the first packets received.
The logic is controlled by the delay histogram field trial which has an added parameter to control if emphasis should be even (c=1, default) or put on later packets (c>1) until we reach our steady state forget factor.
Bug: webrtc:10411
Change-Id: Ia5d46c22d1a4a66994652f71c8cde664362bfacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137050
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28039}
Refactor the few uses of STACK_ARRAY to avoid an extra copy
on the stack.
Bug: webrtc:6424
Change-Id: I5c8f3c0381523db0ead31207d949df9a286c76ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137806
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28038}
WebRTC combines VP9 SVC spatial layer frames into superframe and passes
it to a decoder. The chromium HW VP9 decoder (wrapper) needs to know
location of each spatial layer frame in the frame buffer. To provide
decoder with such information this CL:
- Adds Set/SpatialLayerFrameSize methods to EncodedImage.
- Sets size of each spatial layer frame on superframe at assembly stage.
Bug: webrtc:10495
Change-Id: I68c3c0d668c67dfa1740e004059d860dd98f67f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28032}
The gathering of host candidates with mDNS names is asynchronous and its
completion can happen after a srflx candidate is gathered by the same
underlying socket. We have a broken check in UDPPort::CreateConnection()
that assumes the gathering of host and srflx candidates is sequential.
This CL also does minor refactoring and clean-up.
Bug: chromium:944577
Change-Id: Ic28136a9515081f40b232a22fcbf4209814ed33a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138043
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28030}
This implements RTCAudioSourceStats and RTCVideoSourceStats, both
inheriting from abstract dictionary RTCMediaSourceStats:
https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
All members are implemented except for the total "frames" counter:
- trackIdentifier
- kind
- width
- height
- framesPerSecond
This means to make googFrameWidthInput, googFrameHeightInput and
googFrameRateInput obsolete.
Implemented using the same code path as the goog stats, there are
some minor bugs that should be fixed in the future, but not this CL:
1. We create media-source objects on a per-track attachment basis.
If the same track is attached multiple times this results in
multiple media-source objects, but the spec says it should be on a
per-source basis.
2. framesPerSecond is only calculated after connecting (when we have a
sender with SSRC), but if collected on a per-source basis the source
should be able to tell us the FPS whether or not we are sending it.
Bug: webrtc:10453
Change-Id: I23705a79f15075dca2536275934af1904a7f0d39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137804
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28028}
Scenario tests runs all its activities on task queues. This is not
allowed by the default event log writer, causing a DCHECK failure.
This CL makes it possible to stop the event asynchronously,
thereby avoiding the need for the DCHECK.
Bug: webrtc:10365
Change-Id: I1206982b29fd609ac85b4ce30ae9291cbec52041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136685
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@{#28027}
