This CL breaks up the CheckQp() operation into several steps managed
by the inner helper class CheckQpTask, making responding to high or
low QP an asynchronous operation. Why? Reconfiguring the stream in
response to QP overuse will in the future be handled on a separate
task queue. See Call-Level Adaptation Processing for more details:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
Instead of "bool AdaptDown()" when high QP is reported,
synchronously returning true or false depending on the result of
adaptation, this CL introduces
void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh(
rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>);
Where
QualityScalerQpUsageHandlerCallback::OnQpUsageHandled(
bool clear_qp_samples);
Instructs the QualityScaler whether to clear samples before
checking QP the next time or to increase the frequency of checking
(corresponding to AdaptDown's return value prior to this CL).
QualityScaler no longer using AdaptationObserverInterface, this class
is renamed and moved to overuse_frame_detector.h.
The dependency between CheckQpTasks is made explicit with
CheckQpTask::Result and variables like observed_enough_frames_,
adapt_called_ and adapt_failed_ are moved there and given more
descriptive names.
Bug: webrtc:11521
Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31140}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
with adjustments.
Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
Move definitions of mock classes to the only user, the unit tests for
the deprecated class vcm::VideoReceiver.
Bug: webrtc:7408
Change-Id: I05e38ed8ebbe615bb2db0b631ec914773fb0a520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168182
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30451}
Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to
move it to its own target and allow its dependant to avoid to take a
dependency rtc_base:on rtc_base_approved.
Bug: webrtc:9419
Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29548}
Rename file with tests to match code under test.
Rename fixture by moving 'Test' from prefix to suffix
Bug: None
Change-Id: I54c36d3b517bde7cdffa3a7e74528cc464ea7ad7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157301
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29532}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
This CL adds a field trial parameter WebRTC-SlowDownDecoder that is
used to simulate a slow decoder. The parameter specifies how many
extra ms it takes to decode each video frame. This must only be used
in manual testing.
Bug: None
Change-Id: Iad4079100d67b95c224277aaeaf572e38068717f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151911
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29153}
A followup cl will move VideoCodingModule and related code into this
target.
Bug: webrtc:7408
Change-Id: Iade572b597769456c9b8c76f584500e2bd9a58f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29122}
The vcm::VideoReceiver class is used by both VideoReceiveStream and
the legacy api VideoCodingModule. They have different requirements,
since the latter uses the old jitterbuffer and runs the code on a
ProcessThread.
By making a copy and trimming it down to what's actually used by
VideoReceiveStream, we can drop the dependency on the old
jitterbuffer, without breaking the legacy api. This should also make
it easier to do follow-up refactorings to trim down the class further,
and ultimately remove it.
Bug: webrtc:7408
Change-Id: Iec8a167fe5d0425114b0b67a5b4c2fd5fc4fa150
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151910
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29108}
Previously, FecControllerOverride was passed to
Vp8FrameBufferController::SetFecControllerOverride. Passing to
the factory is a more elegant way, since it's only used when
the controller is constructed.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28443}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
This CL makes it more flexible and easier to include/exclude H264 code
when using other build systems because it delegates the decision to
remove the code to the preprocessor instead of GN.
This CL should be a noop, and for WebRTC/Chromium the GN param
`rtc_use_h264` will still be the only thing to change in order to
include/exclude H264.
Moving code that requires ffmpeg or h264 out of the #ifdef/#endif
part should break the build since dependencies are only added if
`rtc_use_h264=true`.
Bug: webrtc:9213
Change-Id: Ibc04edc2f6b9e51489ffe638d5be4b32959cdca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137430
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28055}
Reland with fixes.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27930}
optional<int> min_frames: The minimum number frames to observe to make a
scaling decision.
Default: kMinFramesNeededToScale in quality_scaler.cc
optional<double> initial_scale_factor: The sample period scale factor.
Default: kSamplePeriodScaleFactor in quality_scaler.cc
optional<double> scale_factor: Option to use a reduced sampling interval when
last check did not result in an adaptation (if
unset the initial_scale_factor is used).
Bug: none
Change-Id: I3bb955d1f8d7d7d49bc118361614b5aa59605231
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135125
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27860}
8-bit H264 HDR content is not rendered correctly in Chrome on Windows.
This is a temporary fix that converts the 8-bit buffer to a 10-bit
buffer if the color space indicates that the buffer should be
rendered as HDR.
Bug: webrtc:10575
Change-Id: I106612ec489c6371fa774424a4cf07d9bad40fc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134040
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27766}
