This CL paves the way to making FrameBufferController injectable.
LibvpxVp8Encoder can manage multiple streams. Prior to this CL,
each stream had its own frame buffer controller, all of them held
in a vector by LibvpxVp8Encoder. This complicated the code and
produced some code duplication (cf. SetupTemporalLayers).
This CL:
1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs
will make this factory injectable.)
2. Makes LibvpxVp8Encoder use a single controller. This single
controller will, in the case of multiple streams, delegate
its work to multiple controllers, but that fact is not visible
to LibvpxVp8Encoder.
This CL also squashes CL #126046 (Send notifications of RTT and
PLR changes to Vp8FrameBufferController) into it.
Bug: webrtc:10382
Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27206}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
Vp8FrameBufferController is currently just a renamed Vp8TemporalLayers,
but subsequent CLs will modify Vp8FrameBufferController in ways that are
not relevant for Vp8TemporalLayers. Namely:
1. Loss notifications will be added.
2. Packet-loss rate will be tracked.
3. RTT will be tracked.
4. Vp8FrameBufferController will be made injectable.
Vp8TemporalLayers is retained in order to:
1. Avoid needlessly changing api/.
2. Place for code shared between DefaultTemporalLayers and ScreenshareLayers.
We can remove it in the future (with a proper public announcement).
Bug: webrtc:10382
Change-Id: I49ad1b9bc1954d51bb0b5e60361985f1eb12ae9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126045
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27009}
If minQP is reached and encoder undershoot consistently, we consider the
quality good enough and throttle encode frame rate.
Bug: webrtc:10310
Change-Id: Ifd07280040dd67ef6e544efdd4619d47bff951e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125461
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27003}
In this CL:
- Updated Vp8TemporalLayers::OnEncodeDone to take a CodecSpecificInfo
instead of a CodecSpecificInfoVP8, so that both the VP8 specific and
generic information can be populated.
- Added structs to represent the GFD template structure.
- Added code to generate templates for video/screensharing.
Bug: webrtc:10342
Change-Id: I978f9d708597a6f86bbdc494e62acf7a7b400db3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123422
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26987}
there's no easy way to inject the Clock in ScreenshareLayers under
normal use. To allow faking the clock, rtc::TimeMillis is used instead.
Bug: webrtc:10365
Change-Id: I46c7f76514672190a0f0f5816a2c858bc6c76fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/125189
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26946}
Simulcast screenshare appears broken due to unrelated changes. It
implicitly relied on SimulcastEncoderAdapter fallback, which happened before
if streams had same resolution. It's not the case anymore. Thus, this CL
adds checks for different frame-rate in simulcast streams.
FullStackTests are also updated to use actual parameters.
Bug: none
Change-Id: I2c1ddb1b39edb96464a0915dfcb9cb4e18844187
Reviewed-on: https://webrtc-review.googlesource.com/c/124494
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26869}
This quality boost means that we sometimes drop a _lot_ of frames in the
base layer. It also interacts poorly with the bitrate adjuster since
even if frames are dropped they are often over-sized.
The setting still leaves the current behavior as default, but can be
changed using the WebRTC-VideoRateControl field trial.
Bug: webrtc:10155
Change-Id: I1a92ec69bab61b5148fe9d8bc391ac5ee1019367
Reviewed-on: https://webrtc-review.googlesource.com/c/122840
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26659}
This CL continues the work began by CL #119958, extending it
to ScreenshareLayers.
Bug: webrtc:10249
Change-Id: I59d0c062a93b288007977e00aa3a2e0929509e0c
Reviewed-on: https://webrtc-review.googlesource.com/c/120042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26526}
Prior to this CL, when software VP8 encoding was done with one temporal
layer, instead of only predicting from the latest frame, the code
allowed the encoder to reference the latest key frame as well.
This improves quality for the few frames immediately after
the key frame, but is not useful for later frames, which diverge
significantly from the key frame. However, the cost of producing
the prediction from more than one reference is incurred by all frames.
My measurements of the effect of this show an improvement
in CPU utilization of 5%-13% when this is not done.
foreman_352x288, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(566.187, 570.012, 575.665) = 570.621
send_avg_qp: 45.36
send_avg_psnr: 37.13
Post-change:
send_enc_speed_fps: avg(633.188, 604.694, 623.232) = 620.371
send_avg_qp: 45.88
send_avg_psnr: 37.0749
Improvement in send_enc_speed_fps: 8.71%
foreman_480x272, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(481.244, 486.971, 487.322) = 485.179
send_avg_qp: 48.9
send_avg_psnr: 37.6217
Post-change:
send_enc_speed_fps: avg(521.651, 499.416, 511.551) = 510.872
send_avg_qp: 48.88
send_avg_psnr: 37.6094
Improvement in send_enc_speed_fps: 5.29%
news_352x288, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(699.407, 697.837, 699.49) = 698.9113333
send_avg_qp: 24.15
send_avg_psnr: 40.9551
Post-change:
send_enc_speed_fps: avg(758.526, 768.104, 757.232) = 761.2873333
send_avg_qp: 23.9833
send_avg_psnr: 40.9697
Improvement in send_enc_speed_fps: 8.92%
Bridge_180x320_15 (video of brandtr@ from Google), 15fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(454.757, 450.399, 446.812) = 450.656
send_avg_qp: 17.6771
send_avg_psnr: 39.9267
Post-change:
send_enc_speed_fps: avg(500.014, 513.316, 513.613) = 508.981
send_avg_qp: 17.6837
send_avg_psnr: 39.9137
Improvement in send_enc_speed_fps: 12.94%
Bug: webrtc:10281
Change-Id: If02736e1535c5f46689fd42b657e35a1e1f64d6d
Reviewed-on: https://webrtc-review.googlesource.com/c/120904
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26511}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
FrameConfig is not specific to temporal layers. Anything that
can control referenced/updated buffers could potentially use it.
Bug: webrtc:10259
Change-Id: I04ed177ee884693798c3b69e35fd4255ce1e9062
Reviewed-on: https://webrtc-review.googlesource.com/c/120355
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26448}
Prior to this CL, RtpPayloadParams had code that assumed
dependency patterns in VP8, in order to write that information
into the [Generic Frame Descriptor] RTP extension.
This CL starts moving that code out of RtpPayloadParams.
Upcoming CLs will migrate additional encoder-wrappers to
the new scheme, then remove the deprecated code.
Bug: webrtc:10249
Change-Id: I5fc84aedf8e11f79d52b989ff8b7ce9568b6cf32
Reviewed-on: https://webrtc-review.googlesource.com/c/119958
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26438}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
This CL add new data to the VideoEncoder::EncoderInfo struct, indicating
how the encoder intends to allocate frames across spatial and temporal
layers.
This metadata will be used in upcoming CLs to control how the encoder's
rate controller performs.
Bug: webrtc:10155
Change-Id: Id56fae04bae5f230d1a985171097d7ca83a3be8a
Reviewed-on: https://webrtc-review.googlesource.com/c/117900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26300}
Use size() accessor function. Also replace most nearby uses of _buffer
with data().
Bug: webrtc:9378
Change-Id: I1ac3459612f7c6151bd057d05448da1c4e1c6e3d
Reviewed-on: https://webrtc-review.googlesource.com/c/116783
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26273}
Without the added preprocessor check, iOS device will be using the desktop logic to determine the number of thread. This put iPhone 8 and iPhone X to use 3 threads and all other iPhones after iPhone 5 to use a single thread.
This CL added a preprocessor for WEBRTC_IOS to have it own thread number calculation logic. In which, the maximum number of thread is fetched from a field_trial and capped by the number of CPU available on the device.
Bug: webrtc:10005
Change-Id: I8c6257fcbf85b07bc986b5f733dbabb3feee37f7
Reviewed-on: https://webrtc-review.googlesource.com/c/110941
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25997}
Read using capacity() method, write using set_buffer() method. This is
a preparation for making the member private, and renaming it to
capacity_.
Bug: webrtc:9378
Change-Id: I2f96679d052a83fe81be40301bd9863c87074640
Reviewed-on: https://webrtc-review.googlesource.com/c/113520
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25934}
This is the first step in moving the metadata and eventually replacing
VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo.
Bug: webrtc:10065
Change-Id: If925b895718e1b1225d2cf49bede1adb3ff281b8
Reviewed-on: https://webrtc-review.googlesource.com/c/112285
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25856}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
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}
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 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}
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}
