There aren't any tests for this and the code isn't currently
active except for the fact that it adds complexity to the Call
class, synchronization into the active code path and makes future
improvements to the class more complex or impossible.
Bug: webrtc:9719
Change-Id: Ia41af0b2186b8a36ca70a07858990b6af7f3a5c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148078
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28807}
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.
The new stat will not be reported unless it is GT 0.
Reporting of decoding_codec_plc events
Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
A malformed session description can assign the same codec to
different payload types which would hit a DCHECK in the
WebRtcVideoEngine. This changes the video engine to just ignore
the duplicate payload type instead of failing.
Bug: chromium:987598
Change-Id: I2034dd11d315ef05448630c860c7ca3f69ef700b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147943
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28796}
This CL removes code related to the usage of the delay agnostic and
extended filter modes in AEC2.
Bug: webrtc:8671
Change-Id: I1a2c7a9eba54b03f5a015df3adb617785f52a939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133912
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28789}
When using simulcast, if the source is too small, it might end up generating
layer sizes that are problematic for hardware encoders.
We can temporarily restore the old behavior that adapts the layer count to the source size until we fix the HW encoder behavior.
to fix HW encoder issues
Bug: webrtc:10849, chromium:990823
Change-Id: Ie1486c9209b408c797c92d1b319d4116fe77171b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148069
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28761}
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}
This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358
Original change's description:
> Only include payload in bytes sent/received.
>
> According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
> https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
> statistic should not include headers or padding.
>
> Similarly, according to
> https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
> received are calculated the same way as bytes sent (eg. not including
> padding or headers).
>
> This change stops adding padding and headers to these statistics.
>
> Bug: webrtc:8516,webrtc:10525
> Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28647}
Bug: webrtc:8516, webrtc:10525
Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28731}
This reverts commit 74a1b4b1321b426392d4c32e4a02361226ad5358.
Reason for revert: requested by chromium
Original change's description:
> Only include payload in bytes sent/received.
>
> According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
> https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
> statistic should not include headers or padding.
>
> Similarly, according to
> https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
> received are calculated the same way as bytes sent (eg. not including
> padding or headers).
>
> This change stops adding padding and headers to these statistics.
>
> Bug: webrtc:8516,webrtc:10525
> Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28647}
TBR=steveanton@webrtc.org,ilnik@webrtc.org,hbos@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,mellem@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8516, webrtc:10525
Change-Id: Ibd31a8264c19f0c6f57d8deb3974593d198046ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147400
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28701}
Some applications are having issues controlling how many active simulcast
layers they have and configuring them when the source track changes size
and rely on the old behavior automatically reducing the layer count.
Using the WebRTC-LegacySimulcastLayerLimit field trial, they can get back
the old behavior until they transition to the newer API.
Bug: webrtc:8785
Change-Id: I92d4dcd62b79a483a6a8867f97c5f502c6aa4db7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146709
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28666}
According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
statistic should not include headers or padding.
Similarly, according to
https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
received are calculated the same way as bytes sent (eg. not including
padding or headers).
This change stops adding padding and headers to these statistics.
Bug: webrtc:8516,webrtc:10525
Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28647}
The GetImplementations function is similar to the GetSupportedFormats function, but instead of providing one SdpVideoFormat per codec it provides one per codec implementation. These SdpVideoFormats can then be tagged so that a certain implementation can be instantiated when CreateVideoEncoder is called.
Bug: webrtc:10795
Change-Id: I79f2380aa03d75d5f9f36138625abf3543c2339d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145215
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28553}
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}
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}
- 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}
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}
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}
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28385}
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 11dfff0878c949f2e19d95a0ddc209cdad94b3b4.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
In several places VideoFrame::Builder is used to create a new VideoFrame
when intent is to change only one or two fields of a const VideoFrame&.
This approach is bad because each and every metadata field have to be
added to all the places.
Instead, this CL adds missing setters and refactors the code to use
full copy of a VideoFrame and update required fields only.
Along the way few actual bugs are fixed, e.g. when ColorSpace isn't copied
when frame rotation or buffer is cropped or converted.
Bug: webrtc:10460
Change-Id: I2895a473ca938b150eed2916c689060bdf58cb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140102
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28170}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
H264 requires that the packetization modes are the same in order to
consider the code the same. This logic was added to VideoCodec::Matches
but was not reflected in IsSameCodec. This could manifest itself when a
remote description with an unsupported packetization mode is set.
Bug: webrtc:10693
Change-Id: Icda07f7d56a464895d2267a41cc0f2fd9d5f42ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138983
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28126}
pc/ depends on media/, so the media/ tests should not have circular
dependencies on pc/.
Bug: None
Change-Id: I849cefecd91e9cd11415bbd93465a98dead735d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139361
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28115}
