Cname callback is used only on receive side, and statistics (soon)
only on the send side.
Bug: webrtc:10679
Change-Id: I122e9cafaea93cd0ba75dc955a652d9d4bddc379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147867
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28767}
This change removes the old `ContributingSources` class. It has been replaced by the new `SourceTracker`.
Bug: webrtc:10793
Change-Id: Ibd481cf6584837c46b229b9fc2a071362f07d361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147878
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28756}
optional<int> initial_bitrate_interval_ms: time interval since start of call
where fast adapt down is allowed.
optional<double> initial_bitrate_factor: try fast adapt down if bw estimate is
below initial bitrate * factor.
Bug: none
Change-Id: I63e1fdaac6556d8e9a961a42e11c925f9ecb9771
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147725
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28753}
We want to evaluate more data in order to make better choices in the
bitrate allocators.
In order to freely update the parameter list without
breaking the API many times for projects customizing them, we'll use a
struct instead.
Bug: webrtc:10126
Change-Id: I443f86781c5134950294cdd1e3197a47447cf973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141418
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28748}
Old way to produce this histogram was based on RtcpStatisticsCallback
reporting sent RTCP messages, with some additional processing by the
ReportBlockStats class. After this cl, to grand average fraction loss
is computed by StreamStatistician, queried by VideoReceiveStream when
the stream is closed down, and passed to ReceiveStatisticsProxy which
produces histograms.
This is a preparation for deleting the RtcpStatisticsCallback from
ReceiveStatistics.
Bug: webrtc:10679
Change-Id: Ie37062c1ae590fd92d3bd0f94c510e135ab93e8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147722
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28747}
This CL adds thread annotations and ensure that neither data races
nor deadlocks occur.
It prevents weird results and helps detecting other concurrency issues.
As a bonus, some dead code has been removed.
Bug: webrtc:10834
Change-Id: Ibd140db9e4dbf81b212044647e2d85bd18ef8d78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147278
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28737}
Use the newly added total_decode_time_ms to get an accurate value
for the average decode time. The sparsely sampled decode_ms is
sensitive to the sampling instance.
Bug: chromium:980853
Change-Id: I9b63c8d1053fa95f74918807b83d1edb5cd726fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147268
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28716}
This prepares for using VideoFrameBuffer::Type as
FrameGenerator::OutputType, which will reduce the
number of redundant enums in the code.
Bug: webrtc:9883
Change-Id: I253f5f1ea7181e02a5cf1a92925f51da8ada6aa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146982
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28696}
This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
This reverts commit 8b3e4e2d1166464f6b309f4fc533a29607d2771f.
Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.
Original change's description:
> Revert "Reland "Add ability to set RTCP sender ssrc at construction time""
>
> This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.
>
> Reason for revert: Speculative revert (some perf test are failing)
>
> Original change's description:
> > Reland "Add ability to set RTCP sender ssrc at construction time"
> >
> > This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
> >
> > Patch set 1 is the original CL.
> > Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> > if either current SSRC is 0 or if the SSRC is identical to the current
> > one. If so, don't schedule an early report.
> > This prevents a regression in which audio jitter became too low(?)
> >
> > Original change's description:
> > > Add ability to set RTCP sender ssrc at construction time
> > >
> > > Bug: webrtc:10774
> > > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28506}
> >
> > Bug: webrtc:10774
> > Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28520}
>
> TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10774
> Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28555}
TBR=mbonadei@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
Change-Id: I2e5c17e8edfd938424f901222158643baa04866e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28562}
This partially reverts these 2 CLs:
1) Reland "Copy video frames metadata between encoded and plain frames in one place"
https://webrtc.googlesource.com/src/+/2ebf5239782bf6b46d4aa812f34fa9f9e5a02be9
2) Don't copy video frame metadata in each encoder/decoder
https://webrtc.googlesource.com/src/+/ab62b2ee51e622be6d0aade15e87e927fa60e6f2
The problem with them were that ColorSpace was made to always be copied from the
EncodedImage in the GenericDecoder, which overwrote ColorSpace information from
the decoder.
If decoder applied color space transition or bitstream color space information
was different from the WebRTC signaled one, the incorrect color space data were
passed to the renderer.
This CL removes introduced change regarding color space data: GenericDecoder
doesn't copy or store it and software decoders are restored to copy it.
Relevant tests are also removed.
Bug: chromium:982486
Change-Id: I989e01476ff7f7df376c05578ab8f540b95a1dd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145323
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28556}
This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.
Reason for revert: Speculative revert (some perf test are failing)
Original change's description:
> Reland "Add ability to set RTCP sender ssrc at construction time"
>
> This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
>
> Patch set 1 is the original CL.
> Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> if either current SSRC is 0 or if the SSRC is identical to the current
> one. If so, don't schedule an early report.
> This prevents a regression in which audio jitter became too low(?)
>
> Original change's description:
> > Add ability to set RTCP sender ssrc at construction time
> >
> > Bug: webrtc:10774
> > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28506}
>
> Bug: webrtc:10774
> Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28520}
TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10774
Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28555}
As per the spec, you should be able to use simulcast with screenshare.
We remove the field trial for it and keep the old behavior only for
screenshare sources with conference flag on.
Bug: webrtc:8785
Change-Id: I1d6d4e18256fb5cfe0195620706de068f25b8d9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144785
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28543}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
Patch set 1 is the original CL.
Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
if either current SSRC is 0 or if the SSRC is identical to the current
one. If so, don't schedule an early report.
This prevents a regression in which audio jitter became too low(?)
Original change's description:
> Add ability to set RTCP sender ssrc at construction time
>
> Bug: webrtc:10774
> Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28506}
Bug: webrtc:10774
Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28520}
If VideoEncoderConfig::max_bitrate_bps is unset then max bitrate of
video stream is set equal to max bitrate value recommended by encoder
for given resolution via encoder capabilities (if available).
Bug: webrtc:10796
Change-Id: I7fce9afc476b794a16956e694e891faee110048e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144526
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28515}
In this CL:
- Added WEBRTC_VIDEO_CODEC_ENCODER_FAILURE return code that can
be returned by the encoder wrapper in case of a broken encoder.
- Added EncoderFailureCallback interface that can be called
to request encoder fallback to be performed. Implemented by
WebRtcVideoChannel and called from the VideoStreamEncoder.
- Updated SelectSendVideoCodec to select all compatible codecs instead
of just one.
Bug: webrtc:10795
Change-Id: I87a83fd02e48c40493c930471c06c3d0941031ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140888
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28462}
Before this change, an attempt to recreate video encoder would fail if
video encoder factory supports only single instance of an encoder.
Added tracking of max number of existed simultaneously encoder
instances to VideoEncoderProxyFactory.
Bug: webrtc:10776
Change-Id: I317cbdf1af94dfb4c72bf99c5cd4ce7b454188fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144044
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28457}
- Don't reset encoder if max/min bitrate changed.
- Removed min/max bitrate DCHECKs from encoder wrappers.
- Reset encoder if start_bitrate changed. Only do this if encoding
has not yet started.
- Updated ReconfigureBitratesSetsEncoderBitratesCorrectly test.
- Removed EncoderSetupPropagatesCommonEncoderConfigValues test since it
was a subset of ReconfigureBitratesSetsEncoderBitratesCorrectly.
Bug: webrtc:10773
Change-Id: Id9cbb2ea229232fd95967819e2a937b26948de9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144028
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28446}
Previously, FecControllerOverride was passed to
Vp8FrameBufferController::SetFecControllerOverride. Passing to
the factory is a more elegant way, since it's only used when
the controller is constructed.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28443}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
[1/2] - Make new version pure-virtual, and deprecated version non-pure.
This will allow deleting the deprecated version from downstream
projects.
[2/2] - Remove deprecated version.
TBR=stefan@webrtc.org
Bug: webrtc:10336
Change-Id: Ia132ef071b1f379fc74834178e75e981ca908125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144042
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28413}
Only remaining user is WavReader. Demote its constructor
accepting a PlatformFile to private, to refactor implementation
in a later cl.
Bug: webrtc:6463
Change-Id: I7b950be6f02073cb135dd0fab1190b9dc0de1fba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144025
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28410}
This CL allows for FEC protection of packets with VideoTimingExtension by
zero-ing out data, which is changed after FEC protection is generated (i.e
in the pacer or by the SFU).
Actual FEC protection of these packets would be enabled later, when all
modern receivers have this change.
Bug: webrtc:10750
Change-Id: If4785392204d68cb8527629727b5c062f9fb6600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143760
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28396}
This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the video side with the spec-compliant `SourceTracker`-implementation.
The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.
Bug: webrtc:10545
Change-Id: I895b5790280ac94c1501801d226c643633c67349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143177
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28386}
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
The rewriter updates video signal parameters in VUI such that they
match to given webrtc::ColorSpace.
Bug: webrtc:10723
Change-Id: I8d0593e3cb727bfee7eb00e3f9ff0b41b93b78bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140881
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28306}
