Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
If rate controller is trusted, we disable the frame dropper in the
media optimization module.
This is a re-land of
https://webrtc-review.googlesource.com/c/src/+/105020
Bug: webrtc:9890
Change-Id: I418e47a43a1a98cb2fd5295c03360b954f2288f2
Reviewed-on: https://webrtc-review.googlesource.com/c/109141
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25570}
Will be deleted as soon as downstream calls of
VideoCodingModule::Create are updated.
Tbr: sprang@webrtc.org # Trivial change in video/
Bug: webrtc:3380
Change-Id: Iaeb6da2fb68991225fe9086ddddd4a553e1620b4
Reviewed-on: https://webrtc-review.googlesource.com/c/107890
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25554}
This is a workaround for the case when there are no video frames in a
call for a very long time, such that RTP timestamps wraparound and
FrameBuffer can't figure out if the frame is older or newer.
Bug: webrtc:9974
Change-Id: Ie1eaa4938813dbbd637ddcbe7ff118ead2bfa4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/109882
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25548}
A preparation for deleting EventFactory and EventWrapper, to instead
use rtc::Event directly.
Bug: webrtc:3380
Change-Id: I4c40daca9268e57b06d506d91e09365091c42ad6
Reviewed-on: https://webrtc-review.googlesource.com/c/109880
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25545}
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.
This cleanup CL is related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
This utility class is needed in rtcp_rtp. Instead of reimplementing it
again, the existing class is moved to rtc_base, cleaned from unused
features and extended as required for the new usage.
Bug: webrtc:9914
Change-Id: I3b0d83d08d8fa5e1384b4721a93c6a90781948fd
Reviewed-on: https://webrtc-review.googlesource.com/c/109081
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25498}
Currently we send Nack as soon as we see packets out of order(a skip in packet sequence number). Sometimes this is not necessary because these "missing" packets just late for a couple of millisecond, or these packets can be recovered by FEC. This CL add a field trial parameter to configure a delay before sending Nack.
Bug: webrtc:9953
Change-Id: Ia8f5995d874f7c55a74091bc90d8395b9b88e66b
Reviewed-on: https://webrtc-review.googlesource.com/c/109080
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25488}
The frame time deltas are now capped based on the current noise.
This has been tested in various conditions using both screen content
and typical mobile video settings, to produce delays that are not overly
high screen content, and simultaneously not negatively affect mobile
calls on really bad network that may have high natural jitter.
Bug: webrtc:9898
Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48
Reviewed-on: https://webrtc-review.googlesource.com/c/107349
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25469}
This is a cleanup CL related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: I3d879196af83856ece1418fa786aab03a3dd3c8c
Reviewed-on: https://webrtc-review.googlesource.com/c/108820
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25466}
This is just a cleanup CL related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: I9a8347aa382bf44f3cd6c38d89bea0e9d68a50e0
Reviewed-on: https://webrtc-review.googlesource.com/c/108781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25464}
When FlexFEC is enabled, sometimes media packet will be recovered by FEC before the actual media packet's arrival. In current implementation this will be considered as packet out of order and nack will be sent, thus cause large increase in retransmit bitrate.
This fix:
1. Avoid sending nack for packet out of order caused by "early" recovered media packets.
2. Save recovered media packet in a set, and do not send nack for these packets.
Bug: None
Change-Id: I008ef4e33668bce6d2cb9ff52b4b5c8e3f349965
Reviewed-on: https://webrtc-review.googlesource.com/c/108090
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25444}
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705
Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
This deprecates the following methods in VideoEncoder:
virtual ScalingSettings GetScalingSettings() const;
virtual bool SupportsNativeHandle() const;
virtual const char* ImplementationName() const;
Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.
Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().
This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.
Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
This will allow downstream projects to use it to construct their own
injected codecs without pulling in dependencies on the software codecs.
Bug: webrtc:7925
Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8
Reviewed-on: https://webrtc-review.googlesource.com/c/107027
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25297}
This change prevents decoding corruption by not allowing keyframes with a
newer frame id but an older timestamp to be decoded. This does not handle
reordering well.
Bug: none
Change-Id: I4a67ca84ee86a782da74a10530c531d893d3bd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/107304
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25292}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the video send and receive path. If a FrameEncryptorInterface is set on an outgoing video RTPSender
then each outgoing video frame will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption. In addition to
this the new GenericFrameDescriptor will be added as additional data.
If a FrameDecryptorInterface is set on an incoming video RtpReceiver then each incoming
video payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
Bug: webrtc:9795
Change-Id: I9f743ce0cb63df0cf070f6144be7ada078b4e5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/103920
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25258}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
Bug: webrtc:9419
Change-Id: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94
Reviewed-on: https://webrtc-review.googlesource.com/c/106343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25207}
Only used for output filename nowadays. Previously, it was used for
selecting the codec implementation. That is now done by injecting
the appropriate codec factory.
Bug: webrtc:9317
Change-Id: Ia2bf28f7df165fb65410ecd1f5d646ee6604e1be
Reviewed-on: https://webrtc-review.googlesource.com/c/106023
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25204}
Move MockVideoDecoder from
modules/video_coding/include/mock/mock_video_codec_interface.h
to
api/test/mock_video_decoder.h
The mock encoder has already moved:
https://webrtc-review.googlesource.com/c/src/+/105620
Keeping the old header until downstream projects have been updated.
Bug: webrtc:9722
Change-Id: I4bc849173a04813064212f17761876695ca3fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/105900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25170}
Also renaming it Vp8TemporalLayers to show that it is codec specific.
Bug: webrtc:9012
Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f
Reviewed-on: https://webrtc-review.googlesource.com/c/104643
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25137}
If rate controller is trusted, we disable the frame dropper in the
media optimization module.
Bug: webrtc:9722
Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/105020
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25107}
I don't think this has any impact, just wanted to have a first unit
test to play around with.
Bug: None
Change-Id: I892e2642f0243c5c9ee807cf71abcd96112b25f4
Reviewed-on: https://webrtc-review.googlesource.com/c/105000
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25089}
Set number of decode threads equal to number of available cores and
limit the maximum value to the maximum number of tiles possible for
HD resolution.
Bug: webrtc:9829, b/117291409
Change-Id: Ib5ccd5cc412011d4438258491efc060cdd050fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/104064
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25059}
Add FakeVp8Decoder that parse width and height from the payload.
Add unit test for testing that width and height is set when decoding frames.
Bug: none
Change-Id: Ifbfff4f62f99625309ce0ef21cf89c76448769d8
Reviewed-on: https://webrtc-review.googlesource.com/c/103140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25038}
- Added field trial to force issuing of key frame on deactivation of
spatial layer. This fixes video corruptions in VP9 K-SVC tests where
layers can be activated/deactivated on-fly due to bandwidth change.
- Added 100ms network delay to the test with restricted link capacity.
This fixes rapid drop of available bandwidth which happens when
bandwidth overuse is detected in the very beginning of call and several
feedback packets arrive without any delay. Also, this makes the test
more realistic.
- Disabled filtering of spatial layer in the test with restricted
link capacity. 1) We don't really need filtering in this test.
2) It appeared that in video quality tests filtering is done before
sending packets to network simulator. Filtering of high layers causes
channel underuse which is compensated by increase of sent bitrate.
This is why we got sent/media bitrates about 2Mbps in test where link
was limited to 1Mbps.
Bug: chromium:889017
Change-Id: I33ffcee0274523f6183c3bbd27d3d29395417d52
Reviewed-on: https://webrtc-review.googlesource.com/c/103520
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24988}
This refactoring merged PopulateCodecSpecific and FrameEncoded into a
single callback method. It also removes the FrameConfig parameter and
instead relies on the temporal layer to remember it internally.
Bug: webrtc:9012
Change-Id: I489b76821b534398ad452643f1322f411d3455b1
Reviewed-on: https://webrtc-review.googlesource.com/95681
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24957}
This CL makes the RtpGenericFrameDescriptor available in
RTPSenderVideo::SendVideo for encryption and in
RtpVideoStreamReceiver::OnReceivedFrame for decryption.
Bug: webrtc:9361
Change-Id: I5b6d10138c0874657862f103c8c9a2328e6d4a66
Reviewed-on: https://webrtc-review.googlesource.com/102720
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24929}
Use in VideoQualityTest replaced by creating a wrapper for the decoder,
similarly to https://webrtc-review.googlesource.com/94152 which
deleted the corresponding method on VideoSendStream.
Bug: webrtc:9106
Change-Id: I0a7798bc44704af8b36017655b9ffa34fa1423e6
Reviewed-on: https://webrtc-review.googlesource.com/97580
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24926}
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
used to determine if screenshare_layers or default_temporal_layers
should be used, and the number of temporal temporal layers to use.
Subsequent CLs will make further cleanup before attempting a move to api
Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
In the case a frame_object is kept for some time before it is deleted,
it may happend that a new frame is received with overlapping sequence
numbers. If the old frame_object is removed while receiving the new
frame there used to be a crash.
Bug: webrtc:9629
Change-Id: I270a8caa2b58b73c000542aa504c0ebe277d49c4
Reviewed-on: https://webrtc-review.googlesource.com/102683
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24914}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
The RTC_DCHECK is hit sometimes. This happens when there is no overlap
between the nack_list and frames in keyframes. The existing code
correctly handles this situation.
Bug: webrtc:9629
Change-Id: I7e3eed1b04781cd69974c5d3eb86e382e9587268
Reviewed-on: https://webrtc-review.googlesource.com/102340
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24860}
Mark all low spatial layer frames as references (not just frames of
active low spatial layers) if inter-layer prediction is enabled since
these frames are indirect references of high spatial layer, which can
later be enabled without key frame.
Bug: webrtc:9782
Change-Id: Iffa5039fab2673a5582e7cdc9be4a36d9e8deb63
Reviewed-on: https://webrtc-review.googlesource.com/102063
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24849}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
VP9 frame rate controller is supposed to be used in screen mode only
but it was partially enabled in normal video mode. This restricts use
of VP9 frame rate controller to screen mode.
Bug: chromium:884164
Change-Id: Ie2eaa31f3364a8abccbc4171007708cf7040fc38
Reviewed-on: https://webrtc-review.googlesource.com/100424
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24769}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
If the bandwidth is just on the edge of being able to enable a new
stream, the keyframe generated when it is enabled might be large enough
to trigger an overuse and force the stream off again.
To avoid toggling, this CL adds hysteresis so that the available
bandwidth needs to be above X% to start bitrate in order to enable the
stream. It will be shut down once available bitrate falls below the
original enabling bitrate.
For screen content, X defaults to 35.
For realtime content, X defaults to 0.
Both can be individually modified via field trials.
Bug: webrtc:9734
Change-Id: I941332d7be7f2a801d13d9202b2076d330e7df32
Reviewed-on: https://webrtc-review.googlesource.com/100308
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24745}
Today we use |is_first_packet_in_frame| to know when a frame begins and the
|markerBit| to know when it ends, but the markerbit does not actually mark the
end of a frame, it marks the end of a picture.
Bug: webrtc:9361
Change-Id: Icc70e6075590cdc31e875a4eb9d489868adbb67c
Reviewed-on: https://webrtc-review.googlesource.com/100160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24722}
Set target frame rate of spatial layer equal to minimum of two: maximum
frame rate of layer (SpatialLayer::maxFramerate) and maximum frame rate
of codec (VideoCodec::maxFramerate).
Bug: webrtc:9740, webrtc:9739, chromium:882358
Change-Id: I34f36e7fd2889f0417474347abab5327fa2d9d7c
Reviewed-on: https://webrtc-review.googlesource.com/99501
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24686}
These attributes were moved from CodecSpecificInfo to EncodedImage in
cl https://webrtc-review.googlesource.com/c/src/+/96780. This followup
deletes the old member variables, which were left temporarily to
transition downstream code.
Bug: webrtc:9378
Change-Id: I1b38ce404a005aec9d48916b73233cfbd7523cfe
Reviewed-on: https://webrtc-review.googlesource.com/97021
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24670}
This is a reland of ae9e188e67
Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}
TBR=sprang@webrtc.org
Bug: webrtc:9682
Change-Id: Idcce315890c79301da532f9ba4997e9606f73fb0
Reviewed-on: https://webrtc-review.googlesource.com/99340
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24669}
The declaration in common_types.h is probably a left-over from a
previous cleanup.
Bug: None
Change-Id: I3ee1bad2494ede0022c6aa8fdd106035471d50e2
Reviewed-on: https://webrtc-review.googlesource.com/99220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24666}
This reverts commit ae9e188e67.
Reason for revert: Verify if this causes chromium:882358.
Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}
TBR=sprang@webrtc.org,ssilkin@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9682, chromium:882358
Change-Id: Idc4051eef72104823038ed9139bb9c75018f7d86
Reviewed-on: https://webrtc-review.googlesource.com/99082
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24646}
Today, the internal frame dropper in libvpx vp8 encoder is enabled or
disabled based on video or screen content. This is then expected to
match up with screenshare vs default temporal layers implementation.
This cl makes libvpx query the temporal layers implementation as well,
breaking this implicit dependency and allows frames to be dropped if
default temporal layers is used with screen content.
Bug: webrtc:9734
Change-Id: If2523a211f4929f16e65a02fa7a6b4edf7328571
Reviewed-on: https://webrtc-review.googlesource.com/99062
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24645}
Assume that stream has single temporal layer if number of frames in GOF
is set to zero (valid case).
Bug: chromium:879584
Change-Id: I7ced082190e40c1bf4cc1468babfd98b0a61f0dd
Reviewed-on: https://webrtc-review.googlesource.com/98800
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24622}
This CL performs some cleanups on multiplex files:
- Adds more comments to factory about usage.
- Moves image packer outside /include as it doesn't need to be public.
- Other small lint issues.
Bug: webrtc:9632
Change-Id: I2e2e6929ea13645aee5483a3697199d1e6184b32
Reviewed-on: https://webrtc-review.googlesource.com/98700
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24615}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This allows VP9 encoder wrapper to control frame rate of each spatial
layer. The wrapper configures encoder to skip encoding spatial layer
when actual frame rate exceeds the target frame rate of that layer.
Target frame rate of high spatial layer is expected to be equal or
higher then that of low spatial layer. For now frame rate controller
is only enabled in screen sharing mode.
Added unit test which configures encoder to produce 3 spatial layers
with frame rates 10, 20 and 30fps and verifies that absolute delta of
final and target rate doesn't exceed 10%.
Bug: webrtc:9682
Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
Reviewed-on: https://webrtc-review.googlesource.com/96640
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24593}
This feature went to stable with M69. Switch is in M69 and M70 banches.
Since tot is now M71 and we have not seen any issues, let's clean this
up.
Bug: webrtc:9634
Change-Id: I708bab55b0443d0873b09dd5b71cdfad72397a7a
Reviewed-on: https://webrtc-review.googlesource.com/98002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24581}
It is called right after construction, so move the needed
implementation into the MediaOptimization constructor instead.
Bug: webrtc:9711
Change-Id: Ibca35670bf45a85538c34c8ead58ba855acc6b96
Reviewed-on: https://webrtc-review.googlesource.com/97325
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24540}
This is a reland of da0898dfae
Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}
Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
Replaces the VP8 simulcast index and VP9 spatial index formely part of
CodecSpecificInfo.
Bug: webrtc:9378
Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
Reviewed-on: https://webrtc-review.googlesource.com/83161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24485}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
It's reasonable to allow clients implementing their own VideoCodecTests
to decide wether they should run in real-time.
Removes the IsAsyncCodec helper, as the assumptions it made are outdated,
and it is no longer useful.
Bug: None
Change-Id: If766935d4947555af54f499a30cfe553bde4c1ab
Reviewed-on: https://webrtc-review.googlesource.com/95722
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24478}
This will allow us to configure VP9 encoder to produce spatial layers
with different frame rates.
Bug: webrtc:9650
Change-Id: I3a9c58072003b8a8da681d5291d8f7ede7f52fa4
Reviewed-on: https://webrtc-review.googlesource.com/95427
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24435}
Videosendstream can be created before capturer starts, so initially the frame resolution may be zero. Add a check to prevent test failure and undesired behavior.
Bug: webrtc:7737
Change-Id: I8f4402e866f45ea1eb112437f866170691a111f6
Reviewed-on: https://webrtc-review.googlesource.com/95102
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24404}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
This is reland after fixes for broken perf tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/91861
Bug: none
Change-Id: I6999a499408787be43a74a26a16b7826a0814a7b
Reviewed-on: https://webrtc-review.googlesource.com/95182
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24383}
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}
- 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}
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}
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}
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}
kAllBuffers in default_temporal_layers.cc introduces a static initializer,
that is banned in Chromium, and that blocks WebRTC roll into Chromium.
This CL removes it to unblock.
Bug: webrtc:9012
Change-Id: Ide181f63d85748dc2d09199024f1b80868d485fd
Reviewed-on: https://webrtc-review.googlesource.com/94460
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24307}
Multiplex encoder is now supporting attaching user-defined data to the video
frame. This data will be sent with the video frame and thus is guaranteed to
be synchronized. This is useful in cases where the data and video frame need
to by synchronized such as sending information about 3D objects or camera
tracking information with the video stream
Multiplex Encoder with data is implemented in a modular way. A new
VideoFrameBuffer type is created with the encoder. AugmentedVideoFrameBuffer
holds the video frame and the data. MultiplexVideoEncoder encodes both
the frame and data.
Change-Id: I23263f70d111f6f1783c070edec70bd11ebb9868
Bug: webrtc:9632
Reviewed-on: https://webrtc-review.googlesource.com/92642
Commit-Queue: Tarek Hefny <tarekh@google.com>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24297}
This CL introduces a few changes to the default VP8 temporal layers:
* The pattern is now reset on keyframes
* The sync flag is inferred rather than hard-coded
* Support is added for buffer search order
Bug: webrtc:9012
Change-Id: Ice19d32413d20982368a01a7d2540d155e185ad4
Reviewed-on: https://webrtc-review.googlesource.com/91863
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24288}
* Make shorter 4-frame pattern default if 2 temporal layers are used.
* Make DefaultTemporalLayers usable by upper simulcast stream with 2tl.
* If experimental settings are enable, bump the max bitrate for the top
stream. Since we're now using probing everywhere the rampup should be
less of an issue.
* Additionally, fixes an issue in full stack tests, where
ScopedFieldTrials in an experiment would override the
--force_fieldtrials specified at command line. Some trials added by
the test bots caused timeouts without this.
Bug: webrtc:9477
Change-Id: I42410605d416b51c4fbfe5b6b850997484af583c
Reviewed-on: https://webrtc-review.googlesource.com/92883
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24252}
This reverts commit fad2aa23b4.
Reason for revert: There seems to be a mismatch with Chrome's default for VP8.
Original change's description:
> Extract color space from Vp8 decoder
>
> Makes use of ColorSpace class to extract info from Vp8 stream.
>
> Bug: webrtc:9522
> Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
> Reviewed-on: https://webrtc-review.googlesource.com/90183
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24086}
TBR=sprang@webrtc.org,emircan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9522
Change-Id: Ie589963159c9e7ccbc52bf3fdfcbc383656a4ca9
Reviewed-on: https://webrtc-review.googlesource.com/92500
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24191}
Downstream projects have been updated, so this can now be relanded.
This is a revert (and rebase) of: https://webrtc-review.googlesource.com/c/src/+/88820
Bug: none
Change-Id: I424664ddef7aeebd3c6c94ae67c7f70a342dc9a4
Reviewed-on: https://webrtc-review.googlesource.com/92082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24181}
This mitigates the long freeze issue caused by overflow of frame
buffer in RtpFrameReferenceFinder and subsequent removal of old, but
not yet decoded frames, from the buffer.
Bug: webrtc:9550
Change-Id: I03390bb58847688c6cb3f4868bf21269ad07073a
Reviewed-on: https://webrtc-review.googlesource.com/91124
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24152}
It was moved to api/video_codecs/video_encoder_config.h in cl
https://webrtc-review.googlesource.com/77683.
Bug: webrtc:8830
Change-Id: I197fd3270d3dea0a5ec98b22cc675c407c388e93
Reviewed-on: https://webrtc-review.googlesource.com/90243
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24092}
Makes use of ColorSpace class to extract info from Vp8 stream.
Bug: webrtc:9522
Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
Reviewed-on: https://webrtc-review.googlesource.com/90183
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24086}
Makes use of ColorSpace class to extract info from H264 stream.
Bug: webrtc:9522
Change-Id: I651d16707260bb2867b1eda95dd4956d62c47279
Reviewed-on: https://webrtc-review.googlesource.com/90180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24085}
In order to have public video bitrate allocator factory, the video bitrate allocator has be part of
the api.
Bug: webrtc:9513
Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2
Reviewed-on: https://webrtc-review.googlesource.com/88083
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#24073}
Move base64.h to the proper location and put redirect header into the
old place to be able to switch downstream users on new location.
Bug: webrtc:8366
Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291
Reviewed-on: https://webrtc-review.googlesource.com/88223
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24069}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
This CL is the first step for introducing color space information in webrtc.
- Add ColorSpace class listing color profiles.
- Add ColorSpace as a member of webrtc::VideoFrame.
- Make use of this class by extracting info from VP9 decoder.
Bug: webrtc:9522
Change-Id: I5e2514efee2a193bddb4459261387f2d40e936ad
Reviewed-on: https://webrtc-review.googlesource.com/88540
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23988}
We previously always set VCMNaluCompleteness to kNaluComplete for
kVideoCodecGeneric, which seems wrong. This CL fixes that and also
cleans up the code a bit. The logic for VP8, VP9, and H264
should be exactly preserved. This CL also updates the test to use
kVideoCodecGeneric instead of kVideoCodecUnknown. kVideoCodecUnknown
has no purpose and should be removed.
Bug: webrtc:9516
Change-Id: Ib8d2bf6a04d41b91c5774531f3a669edce3c6cb2
Reviewed-on: https://webrtc-review.googlesource.com/88181
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23933}
Value 10 seems to be too small for some implementations. Updates the
value to 20. This affects VideoCodecTestFixture.
Bug: None
Change-Id: Ibbeb7cb5ef23f8ac625d37aaa764c9d245f23e9d
Reviewed-on: https://webrtc-review.googlesource.com/87562
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23882}
This is a step toward simplifying the VideoCodec struct and removing the
targetBitrate. The hard-coded values now reside in
SimulcastRateAllocator.
A follow-up will do away with the field altogether.
Bug: webrtc:9504
Change-Id: I74d483682309d363048fbbbd31e0607d7242f504
Reviewed-on: https://webrtc-review.googlesource.com/87424
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23876}
This is to avoid clearing the |gof_info_| map when there are jumps in the
tl0 pic index.
Bug: chromium:855211
Change-Id: I762557070d65b3c535cb9a49498975bcd9c2c485
Reviewed-on: https://webrtc-review.googlesource.com/86943
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23872}
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant.
Bug: none
Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98
Reviewed-on: https://webrtc-review.googlesource.com/87261
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23856}
This reverts commit 425193b4a9.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Unit test for case where the number of active and configured spatial"
>
> This reverts commit 5eb6045ce5.
>
> Reason for revert: Test breaks downstream.
>
> Original change's description:
> > Unit test for case where the number of active and configured spatial
> > layers differ.
> >
> > Bug: webrtc:9472
> > Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
> > Reviewed-on: https://webrtc-review.googlesource.com/85644
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23782}
>
> TBR=brandtr@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
>
> Change-Id: Ib97cdb127e79ee969f7cb3f931cb7bd533f13af0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9472
> Reviewed-on: https://webrtc-review.googlesource.com/86320
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23785}
TBR=brandtr@webrtc.org,terelius@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9472
Change-Id: I796909c553702a0fa19e5e16e4586f915569b134
Reviewed-on: https://webrtc-review.googlesource.com/87220
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23847}
Since the frame is processed on the same thread as the decoding happens
on, keeping a reference to the frame may cause deadlocks on some
implementations.
Longer term, we should probably move the frame processing to a separate
thread but this is an easy fix for now.
Bug: b/110246814
Change-Id: I251737e2188e1755d45b35165586d1b0daf14595
Reviewed-on: https://webrtc-review.googlesource.com/87104
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23843}
Also adjust to base-layer fraction for the shortened 3-tl pattern to be
60%, just like the 2-tl setting.
This CL removes direct use of the allocation matrix and moves it behind
a static getter.
Bug: webrtc:9477
Change-Id: Ifd7d1edffa0555024fd252834357b926997d13b5
Reviewed-on: https://webrtc-review.googlesource.com/86681
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23834}
This CL adds VP9 profile information in SDP. It adds the necessary fields and
enums to codec containers.
Additional profiles will be followed.
Bug: webrtc:9376
Change-Id: I78574714f06f8087262a71dd64c01f31a229dd54
Reviewed-on: https://webrtc-review.googlesource.com/81960
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23810}
This reverts commit 5eb6045ce5.
Reason for revert: Test breaks downstream.
Original change's description:
> Unit test for case where the number of active and configured spatial
> layers differ.
>
> Bug: webrtc:9472
> Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
> Reviewed-on: https://webrtc-review.googlesource.com/85644
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23782}
TBR=brandtr@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
Change-Id: Ib97cdb127e79ee969f7cb3f931cb7bd533f13af0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9472
Reviewed-on: https://webrtc-review.googlesource.com/86320
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23785}
The variable, num_active_spatial_layers, is used to construct ssData.
This CL reverts two instances of num_active_spatial_layers not
related to ssData construction.
Bug: None
Change-Id: I4d90d4578684dfdf8bd5a39c7a2fe778fce4414c
Reviewed-on: https://webrtc-review.googlesource.com/85643
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23756}
This removes call of av_register_all(), which is deprecated, and
related code.
Bug: webrtc:9352
Change-Id: Ib7de5931c900eaf1023ecf3046f560feaaeec8ef
Reviewed-on: https://webrtc-review.googlesource.com/85347
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23743}
When GetSvcConfig returned fewer spatial layers than the number
statically configured from the test, we would crash on a SIGFPE.
This is not a problem in the production code, since there we
reset the encoder with the correct number of spatial layers
whenever the resolution changes.
Bug: None
Change-Id: I339e4a3c0fa993c7c649533c0eae71e1314194e7
Reviewed-on: https://webrtc-review.googlesource.com/85374
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23741}
Added distinction between number of configured and number of actively
encoded spatial layers and include number of actively encoded spatial
layers in ssData. Modified layer_filtering_transport.cc test to
parse from the RTP header and use the number of actively encoded
spatial layers for filtering spatial video layers.
Bug: webrtc:9425
Change-Id: Ic9f8895ab08b0626f9bb53a75ec33d8e7eb8706e
Reviewed-on: https://webrtc-review.googlesource.com/84243
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23716}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org
Bug: webrtc:5840
Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8
Reviewed-on: https://webrtc-review.googlesource.com/84743
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23715}
This reverts commit 07efe436c9.
Reason for revert: Breaks downstream project.
cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).
Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
>
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
> under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
>
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}
TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com
Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
Bug: webrtc:5840
Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
Reviewed-on: https://webrtc-review.googlesource.com/64100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23705}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a followup to https://webrtc-review.googlesource.com/61640,
moving the responsibility for setting these values to the
PayloadRouter.
Bug: webrtc:8830
Change-Id: I8e5a02cf7bb7417166f04d5511aab7a778799bc1
Reviewed-on: https://webrtc-review.googlesource.com/83164
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23654}
Extract rtc_base/base64.h and rtc_base/base64.cc into separate target
to prepare to move them into third_party
Bug: webrtc:8366
Change-Id: I477e6da2b9d09307439b3272261f31042f479d74
Reviewed-on: https://webrtc-review.googlesource.com/83980
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23645}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
They have been disabled by default for years, and should have been made redundant by the event logs.
Bug: webrtc:8982
Change-Id: I491923cbc93378d28f5166d24756b335619d9c12
Reviewed-on: https://webrtc-review.googlesource.com/82800
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23598}
Need to depend on them from Chromium.
Bug: webrtc:7925
Change-Id: Iea1bb3b937c602920bfd87f885c87c790ac7bc17
Reviewed-on: https://webrtc-review.googlesource.com/82061
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23580}
This is a reland of efc71e565e
Differs from the original cl by not widening the type of
VideoCodec::width and VideoCodec::height.
Original change's description:
> Move class VideoCodec from common_types.h to its own api header file.
>
> Bug: webrtc:7660
> Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
> Reviewed-on: https://webrtc-review.googlesource.com/79881
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23544}
Bug: webrtc:7660
Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8
Reviewed-on: https://webrtc-review.googlesource.com/82140
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23569}
And update most internal calls to use it.
Bug: webrtc:5740, webrtc:9372
Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f
Reviewed-on: https://webrtc-review.googlesource.com/81743
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23567}
This reverts commit 16e28d143a.
Reason for revert: Fix has supposedly landed upstream.
Original change's description:
> Disabling VeryLowBitrateVP9 to unblock roll.
>
> This should be re-enabled very soon since the libvpx thinks this
> is fixed upstream and is only waiting for merge.
>
> TBR=marpan@google.com
>
> Bug: webrtc:9292
> Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
> Reviewed-on: https://webrtc-review.googlesource.com/81660
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23525}
TBR=phoglund@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9292
Change-Id: I995953070536e8ee3540e7c30bc11dc1200e0463
Reviewed-on: https://webrtc-review.googlesource.com/82200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23552}
This should be re-enabled very soon since the libvpx thinks this
is fixed upstream and is only waiting for merge.
TBR=marpan@google.com
Bug: webrtc:9292
Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
Reviewed-on: https://webrtc-review.googlesource.com/81660
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23525}
webrtc:8745 is closed and it was about unblocking the Chromium roll,
webrtc:9352 is the new bug to keep track of the removal of ffmpeg
deprecated functions.
Bug: webrtc:9352
Change-Id: I2818dba804f3d611d4df80559a635e7cf1ee5338
No-Try: True
TBR: phoglund@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/80882
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23506}
This is a reland of 3409cfa378
Needed to change RtpVideoStreamReceiver to stop deregistering a payload
type if two payload types refer to the same codec (which now happens,
with the packetization mode 0/1 payload types). It's not clear why this
was being done in the first place.
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
Bug: chromium:600254
Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259
Reviewed-on: https://webrtc-review.googlesource.com/78399
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23494}
This will allow exposing the interface to downstream users that
want to test VP8 simulcast. No functional changes to the tests
themselves are expected.
Bug: webrtc:9281
Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da
Reviewed-on: https://webrtc-review.googlesource.com/77361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23469}
Use new frame dropping mode - FULL_SUPERFRAME_DROP - in VP9 encoder and
configure it to drop entire superframe if any layer is overshooting.
Bug: none
Change-Id: Ie22ed5c175e530bcce365d40cba0d10cb608ad4f
Reviewed-on: https://webrtc-review.googlesource.com/79622
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23447}
It should be the responsibility of the fixture user to provide the exact
codecs that should be tested instead. This reduces the coupling between
the test fixture and the codec instantiation.
Bug: webrtc:9317
Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b
Reviewed-on: https://webrtc-review.googlesource.com/79147
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23425}
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.
Artificial Sdp parameter is added to the sdp format if the flag is set.
Additionally, sdp format is propagated in vp8 simulcast adapters.
Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
In flexible mode, use VP9E_GET_SVC_REF_FRAME_CONFIG to get indices of
reference frame buffers and buffers update by encoded frame.
Set inter_pic_predicted to true only if encoder actually used temporal
prediction.
Bug: webrtc:9244, webrtc:9270
Change-Id: I4e439abeab9e063d50abdcefc59bf58d6596ea6c
Reviewed-on: https://webrtc-review.googlesource.com/74780
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/master@{#23410}
Thus we don't need to initialize new members added to the structure
in the future.
Bug: None
Change-Id: Id9f5b127c224660f3016973261045b4231a617c1
Reviewed-on: https://webrtc-review.googlesource.com/79080
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23404}
All decoders are injectable, no need to create built-in codecs from
there.
Bug: webrtc:7925
Change-Id: Iabf3d59a8e4d721ad29386acbf138b7e5992ce5e
Reviewed-on: https://webrtc-review.googlesource.com/72441
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23397}
This is needed for downstream users of the impl, as we currently pull
in Chromium specifics in the android_codec_factory_helper. Further,
the downstream users should explicitly supply their own factories
if they do not want to use the internal ones.
Bug: None
Change-Id: Ia7b01a66aadaba3d5accf44e5ca38e1a319e4e34
Reviewed-on: https://webrtc-review.googlesource.com/78420
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23390}
This reverts commit 3409cfa378.
Reason for revert: Broke WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsH264 on Windows 7/10 bots
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,sprang@webrtc.org
Change-Id: I2f2a2b4ca20ba883764cd5265911e1453d3df66e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:600254
Reviewed-on: https://webrtc-review.googlesource.com/78398
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23374}
The work was already done to support it, but it wasn't being negotiated
in SDP.
This means we'll now see 8 H264 payload types instead of 4; one for each
combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
This could be problematic in the future, since we're starting to run
out of dynamic payload types (using 25 of 32).
Bug: chromium:600254
Change-Id: Ief2340db77c796f12980445b547b87e939170fae
Reviewed-on: https://webrtc-review.googlesource.com/77264
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23372}
These files are required when implementing tests based on the test fixture,
and should be exposed as part of the test api.
This CL also removes a usage of stringstream and fixes some chromium-style
lint issues.
Bug: webrtc:8982, webrtc:163
Change-Id: I132aea0da79a79587887f21897236fc9802b7574
Reviewed-on: https://webrtc-review.googlesource.com/74586
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23346}
Follow up to https://webrtc-review.googlesource.com/c/src/+/39511,
which introduced a new Decode method, without the
RTPFragmentationHeader argument, and deprecated the old method.
Bug: webrtc:6471
Change-Id: Icd3c536ebedd4e3c2d57fdb4d6e078d6ff1de5b6
Reviewed-on: https://webrtc-review.googlesource.com/75180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23339}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
- Limit framerate by dropping frames before encoding.
- The max framerate at screen sharing is set to 5fps.
Bug: webrtc:9261
Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760
Reviewed-on: https://webrtc-review.googlesource.com/76921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23316}
- Remove referencing control from encoder wrapper. Use fixed temporal
prediction structure.
- Remove flexible mode from encoder wrapper. It only worked with
referencing control which this CL removes.
- Remove external framerate/bitrate controller. Keep codec's internal
frame dropping enabled at screen sharing.
- Use GetSvcConfig() to configure layering.
Bug: webrtc:9261
Change-Id: I355baa6aab7b98ac5028b3851d1f8ccc82a308e0
Reviewed-on: https://webrtc-review.googlesource.com/76801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23311}
In preparation for landing https://webrtc-review.googlesource.com/c/src/+/72441
a downstream project that uses the VideoCodingModule needs to be able to
inject a decoder object created from the outside, just like how encoders
are possible to inject.
Bug: webrtc:7925
Change-Id: Ibaeffda55f84410436d79f75730e7352e298b9f0
Reviewed-on: https://webrtc-review.googlesource.com/77160
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23297}
After https://webrtc-review.googlesource.com/c/src/+/70740, we are
creating/destroying the codecs on a task queue in the VideoStreamEncoder. This
CL updates the VideoCodecTest to do the same.
Also, this CL switches from manually Wait()'ing on the task queue to using
TaskQueueForTest::SendTask.
Bug: None
Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e
Reviewed-on: https://webrtc-review.googlesource.com/76800
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23257}
This CL simply renames the test cases that were not renamed in
prior CLs.
Bug: None
Change-Id: If616eb823e1453bc92ba037722b77a219d54409c
Reviewed-on: https://webrtc-review.googlesource.com/76780
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23240}
- Two quality layers (same resolution, different bitrate).
- Max bitrate of low layer is limited to 200kbps. The choice of the
limit is driven by VP8 screen sharing which limits max bitrate of low
temporal layer to 200kbps. Using the same value for VP9 guarantees
that there will be no regressions for participants with limited
bandwidth.
- Max bitrate of high layer is limited to 500kbps. According to test
results this value is enough to get up to +5dB higher PSNR than VP8
SS provides on 1.2Mbps (max total bitrate for VP8 SS) link.
- Max total sent bitrate is limited to 700kbps. It is 500kbps lower
than that in VP8 SS (1200kbps).
Bug: webrtc:9261
Change-Id: I7919cc3933064664567c39e380a44cad0c65f1e8
Reviewed-on: https://webrtc-review.googlesource.com/76380
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23226}
Changed minimum bit rate threshold formula to raise the minimum bit rate
at which 720p video is presented in VP9 SVC to ensure that the video
quality for VP9 SVC is the same or better than VP8 SIM. The minimum bit
rate threshold values for lower resolutions remain largely unchanged.
Also changed maximum bit rate threshold formula to lower the maximum
bit rate for low resolutions (e.g., 180p) in order to ensure higher
frame rates when downlink bit rates are very low (e.g., < 100 kbps).
Bug: webrtc:9242
Change-Id: I8f9c76c9188b98f3fd40a608551b576b0c3b8f34
Reviewed-on: https://webrtc-review.googlesource.com/75244
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23218}
The previous effort of building WebRTC without SW codecs stopped when
libjingle_peerconnection was possible to build. In order to make the
group("default") target build, this basically updates a bunch of
tests to explicitly depend on the built-in software video codecs.
Bug: webrtc:7925
Change-Id: I2715414770c197fca01cb8dbde173a21f4434500
Reviewed-on: https://webrtc-review.googlesource.com/70503
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23216}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
This removes second reference for frame 3 in GOF predefined for 3
temporal layers since encoder never use that reference.
Bug: webrtc:9245
Change-Id: I6fbdbe7d3c753dda7fbcfcbd05f3530f70f80728
Reviewed-on: https://webrtc-review.googlesource.com/74705
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23193}
