This CL also introduces NetEqFactory and NetEqControllerFactory
interfaces, as well as several convenience classes for working with
them: DefaultNetEqFactory, DefaultNetEqControllerFactory and
CustomNetEqFactory.
Bug: webrtc:11005
Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29671}
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}
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 fixes a minor bug in the event_log_visualizer where packets are
processed in RTP send time order rather than RTCP arrival time order.
The bug makes time appear to move backwards if RTCP feedback for a later
RTP packet arrives before the feedback of an earlier RTP packet.
Bug: None
Change-Id: I06e8a25d5c65602bedcfd9e4ea1d23874bee9318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156169
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29448}
The new target does not depend on libjingle_peerconnection_api, and to
do this, the named "audio" and "video" string literals had to be moved from
media_stream_interface.cc to media_types.cc.
In this cl, the dependency on libjingle_peerconnection_api can be
dropped from a few targets.
No-Presubmit: True
Bug: webrtc:8733
Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28998}
This tool is unused, this CL removes it in order to reduce the cost
of the maintenance (in the last 2 years only maintenance commits have
been landed in this directory).
Bug: None
Change-Id: Ieec113bc25c480405d32e284a0456572758352e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146204
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28619}
This is a trivial CL, updating rtp_generator.cc according to changes in
APIs in other places.
Bug: webrtc:10807
Change-Id: Ie85c6283f2d78dcf742979378db0b4fb0914c96c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145209
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28526}
All the usages of this flag library have been replaced by ABSL_FLAG.
Bug: webrtc:10616
Change-Id: I7b50772654d68e80055d79f46ef5bc0633e75508
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143482
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28401}
The IDs be more stable than the plot titles and could be used to identify specific graphs in scripts.
Change event_log_visualizer command line interface to control which plots are generated.
Old interface had one command line flag per plot as well as a set of 'profiles' that enabled
of disabled sets of plots. New interface has a command line flag
which takes a string of all the plot names or profiles that should be enabled.
In some cases, there are also slight naming changes for the plots.
For example, the former command
event_log_visualizer --plot_profile=sendside_bwe --plot_incoming_packet_sizes <filename> | python
is now
event_log_visualizer --plot=sendside_bwe,incoming_packet_sizes <filename> | python
The former command
event_log_visualizer --plot_profile=none --plot_incoming_packet_sizes <filename> | python
is now
event_log_visualizer --plot=incoming_packet_sizes <filename> | python
The former command
event_log_visualizer --plot_profile=all <filename> | python
is now
event_log_visualizer --plot=all <filename> | python
Bug: webrtc:10623
Change-Id: Ife432c1e51edfce64af565a769f1764a16655bb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140886
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28237}
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}
When running unpack_aecdump --full, unpack RuntimeSettings into files, on the format that can be imported into Audacity.
Output one file for each RuntimeSetting present in the aecdump. If outputting several WAV files, output file for each WAV file with corresponding time stamps.
Bug: webrtc:10643
Change-Id: If147e509d36207f5f838457354e2451df65549d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137426
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28007}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This CL adds a mode to simulate roughly what GoogCC could have been
doing during the recording of an rtc event log by using the logged
events as input to GoogCC and visualizing the resulting target rate.
This is similar to the existing simulated_sendside_bwe mode, but uses
the new NetworkControllerInterface to ensure more reliable GoogCC
simulation.
Bug: None
Change-Id: I57894aa666151efc8405407d928b5257fb9b7d61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123924
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27095}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
Import proto_library.gni when rtc_enable_protobuf is true instead of when
build_with_mozilla is false.
Makes it maybe easier to reason about the intention (e.g. intention is to not
compile any protobuf in, hence flag rtc_enable_protobuf)
The build file could not work if build_with_mozilla = true but
rtc_enable_protobuf = true.
Bug: webrtc:10338
Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597
Reviewed-on: https://webrtc-review.googlesource.com/c/123763
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26868}
This CL introduces a new rtp_generator tool that can be utilized to generate
.rtpdump files that can be replayed by the video_replayer. This allows
automated generation of corpus material for the new WebRTC RTP fuzzers in
addition to allowing anyone who is experimenting with a new RTP feature to
quickly debug issues.
It can be used as follows:
./rtp_generator --input_config=./rtc_tools/rtp_generator/configs/vp8.json --output_rtpdump=/tmp/vp8.rtpdump
./video_replay --config_file test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json --input_file /tmp/vp8.rtpdump
It works by generating squares randomly on the screen for a given duration. This
initial version is very limited and doesn't support FEC, RED and other
configurations. I plan to extend it to support these in future CLs.
Bug: webrtc:10117
Change-Id: I31d3dbb6fad73c727145ead4e7d085113d11fc51
Reviewed-on: https://webrtc-review.googlesource.com/c/119964
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26517}
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}
The WebRtcKeyValueConfig interface allows providing custom key value
configurations that changes per instance of GoogCcNetworkController.
Bug: webrtc:10009
Change-Id: I520fff030d1c3c755455ec8f67896fe8a6b4d970
Reviewed-on: https://webrtc-review.googlesource.com/c/116989
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26312}
This is done by creating a custom ReplacementAudioDecoderFactory.
Bug: webrtc:8396, webrtc:10080
Change-Id: Ie1cb614fec855b82d65c6ef86c3593f547254559
Reviewed-on: https://webrtc-review.googlesource.com/c/116795
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26217}
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}
This class adds logic for aligning what part of a test video has been
encoded from a reference video. It does that by cropping and zooming in
on a region of the reference video that most closely matches the test
video. A small cropping does not have much impact on human perception,
but it has a big impact on PSNR and SSIM calculations.
For example, if the test video is cropped with one row in the top and
bottom, adjusting for this improves average PSNR from 27.7146 to
29.3357 and average SSIM from 0.934891 to 0.95318 in an example test
video.
TBR=phoglund
Bug: webrtc:9642
Change-Id: I02cfe0e2261fb58df8cdb1e15ba93285e3dc4538
Reviewed-on: https://webrtc-review.googlesource.com/c/99480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25755}
This CL consistently use:
* relative paths for WebRTC dependent targets (test_support)
* absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.
We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.
Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
This class adds logic for aligning color space of a test video compared
to a reference video. If there is a color space mismatch, it typically
does not have much impact on human perception, but it has a big impact
on PSNR and SSIM calculations. For example, aligning a test run with VP8
improves PSNR and SSIM from:
Average PSNR: 29.142818, average SSIM: 0.946026
to:
Average PSNR: 38.146229, average SSIM: 0.965388.
The optiomal color transformation between the two videos were:
0.86 0.01 0.00 14.37
0.00 0.88 0.00 15.32
0.00 0.00 0.88 15.74
which is converting YUV full range to YUV limited range. There is
already a CL out for fixing this discrepancy here:
https://webrtc-review.googlesource.com/c/src/+/94543
After that, hopefully there is no color space mismatch when saving the
raw YUV values. It's good that the video quality tool is color space
agnostic anyway, and can compensate for differences when the test
video is obtained by e.g. filming a physical device screen.
Also, the linear least square logic will be used for compensating
geometric distorisions in a follow-up CL.
Bug: webrtc:9642
Change-Id: I499713960a0544d8e45c5d09886e68ec829b28a7
Reviewed-on: https://webrtc-review.googlesource.com/c/95950
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25193}
Extract functionality of test_main into separate library to be able to
reuse it if another main will be required.
Bug: webrtc:5996
Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd
Reviewed-on: https://webrtc-review.googlesource.com/c/105921
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25172}
This tool is no longer needed since we're deleting the AQ tests.
Bug: chromium:880074
Change-Id: I035d7b33c7c4feb5962cf9dafc8e7086a8dee440
Reviewed-on: https://webrtc-review.googlesource.com/c/105140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25162}
And also drop dependency on module_api, where possible. With this
change, common_video/ no longer depends on
libjingle_peerconnection_api.
Bug: None
Change-Id: Icc0648559bef5b7f549e81d58f2a5f97c0af3abf
Reviewed-on: https://webrtc-review.googlesource.com/c/103782
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24991}
After the removal of field_trial_default, metrics_default and
runtime_enabled_features_default, this build target doesn't build
anything and can be safely removed.
Bug: webrtc:9631
Change-Id: Iee1111e065ffefe0b4b9a695ee67a594e6d82caa
Reviewed-on: https://webrtc-review.googlesource.com/c/103702
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24976}
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 function checks the file extension to determine YUV or Y4M format.
Also adds a flag aligned_output_file to compare_videos.py, which allows
saving the aligned reference video to a file.
Bug: webrtc:9642
Change-Id: Ia59f5c123a1e41104756eb6b235b6581c4ffbd77
Reviewed-on: https://webrtc-review.googlesource.com/99503
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24787}
This is a reland of d8ff3f29ce.
See https://webrtc-review.googlesource.com/c/src/+/100681/1..4 for
the fix. Error "Failed to open video file for emulated camera" should
be addressed by that change.
Original change's description:
> Compile frame analyzer for the host machine on perf tests.
>
> Bug: webrtc:9665
> Change-Id: I05c01ee4bef0995556b1a679498b3d9132de7c26
> Reviewed-on: https://webrtc-review.googlesource.com/100360
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24756}
TBR=phoglund@webrtc.org, oprypin@webrtc.org
Bug: webrtc:9665
Change-Id: If6a4f2259dabf50718abf47c9cf303d143a1895a
Reviewed-on: https://webrtc-review.googlesource.com/100681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24762}
This is a reland of 9bb55fc09b
Original change's description:
> Reland "Update video_quality_analysis to align videos instead of using barcodes"
>
> This is a reland of d65e143801
>
> The binary for frame_analyzer.cpp is precompiled and stored in the cloud, so it
> won't automatically pick up change to the source file. Therefore, restore all
> old code to be backwards compatible.
>
> Original change's description:
> > Update video_quality_analysis to align videos instead of using barcodes
> >
> > This CL is a follow-up to the previous CL
> > https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> > logic for aligning videos. This will allow us to easily extend
> > video_quality_analysis with new sophisticated video quality metrics.
> > Also, we can use any kind of video that does not necessarily need to
> > contain bar codes. Removing the need to decode barcodes also leads to a
> > big speedup for the tests.
> >
> > Bug: webrtc:9642
> > Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> > Reviewed-on: https://webrtc-review.googlesource.com/94845
> > Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24423}
>
> TBR=phensman@webrtc.org,phoglund@webrtc.org
>
> Bug: webrtc:9642
> Change-Id: Id8d129ce103284504c67690f8363c03eaae3eee7
> Reviewed-on: https://webrtc-review.googlesource.com/96000
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24429}
TBR=phensman,phoglund
Bug: webrtc:9642
Change-Id: Ic248b7831ae148251a1a4ebeec5d154286f91a0a
Reviewed-on: https://webrtc-review.googlesource.com/98080
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24583}
This is a reland of b2c0e8f60f
Original change's description:
> Add tool for aliging video files
>
> This class adds logic for aligning a test video to a reference video
> by an algorithm that maximizes SSIM between them. Aligned videos will be
> easier to run video quality metrics on. This is a generic way of
> aligning videos and can be replace the intrusive barcode stamping that
> we currently use. This will be done in a follow-up CL.
>
> Change-Id: I71cf1e2179c0f1e03eff9e4d8fc492fd5cfbbb1c
> Bug: webrtc:9642
> Reviewed-on: https://webrtc-review.googlesource.com/94773
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24407}
TBR=phensman,phoglund
Bug: webrtc:9642
Change-Id: I35d6b0e598335b8d80fbfa37ba06d5c651bda4f6
Reviewed-on: https://webrtc-review.googlesource.com/98040
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24580}
This is a reland of 404be7f302
It adds support for reading .yuv files as well to not break anything.
Original change's description:
> Add Y4mFileReader
>
> Encapsulate logic for reading .y4m video files in a single class. We
> currently have spread out logic for opening .y4m files with partial
> parsing. This CL consolidates this logic into a single class with a well
> defined interface.
>
> Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30
> Bug: webrtc:9642
> Reviewed-on: https://webrtc-review.googlesource.com/94772
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24398}
TBR=phensman,phoglund
Bug: webrtc:9642
Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615
Reviewed-on: https://webrtc-review.googlesource.com/97983
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24571}
This class adds logic for aligning a test video to a reference video
by an algorithm that maximizes SSIM between them. Aligned videos will be
easier to run video quality metrics on. This is a generic way of
aligning videos and can be replace the intrusive barcode stamping that
we currently use. This will be done in a follow-up CL.
Change-Id: I71cf1e2179c0f1e03eff9e4d8fc492fd5cfbbb1c
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/94773
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24407}
SequencedTaskChecker is not part of rtc_base_approved and will not work
in Chromium. This CL simply removes it since it was just a precaution
and is not necessary for the tool. The thread assumptions are stated in
the class comment.
TBR=phensman@webrtc.org
Bug: webrtc:9642
Change-Id: I871ac361975595d8ed07b2e2447e3581c9ba9968
Reviewed-on: https://webrtc-review.googlesource.com/95648
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24401}
Encapsulate logic for reading .y4m video files in a single class. We
currently have spread out logic for opening .y4m files with partial
parsing. This CL consolidates this logic into a single class with a well
defined interface.
Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/94772
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24398}
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: I9c26b6129db24263f1aada9561f477db64091049
Reviewed-on: https://webrtc-review.googlesource.com/89742
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24051}
Splits 'modules/audio_processing:audio_processing' target. The files
in modules/audio_processing/agc now are in targets in that folder.
Reason for doing this was to include
modules/audio_processing/agc/agc.h from another target in the
dependent CL https://webrtc-review.googlesource.com/c/src/+/86603
This could help reducing the binary size in the future.
Bug: webrtc:7494
Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887
Reviewed-on: https://webrtc-review.googlesource.com/87422
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23873}
This removes the legacy DelayBasedBwe to reduce code redundancy and
avoid the risk of applying changes on only one version.
Bug: webrtc:8415
Change-Id: I88aba03adbb77ee0ff0a97a8b3be6ddf028af48a
Reviewed-on: https://webrtc-review.googlesource.com/85364
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23798}
After [1], a manual dependency on exe_and_shlib_deps is no longer necessary
since it's automatically added. This CL removes all remaining manual references
to exe_and_shlib_deps.
[1] d7ed1f0a9c
BUG=chromium:845700
R=tommi@webrtc.org
Change-Id: I92942bc08c0e34c5c39df3c71f56f89476f8d95c
Reviewed-on: https://webrtc-review.googlesource.com/83061
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23573}
This CL replaces components in the congestion controller module
that are identical to equivalent components in the rtp and goog_cc
subfolder. Some redundant components are left as they were not
trivial to replace.
Bug: webrtc:8415
Change-Id: I86a1f164d7b100b8ec8ba7dbc1c9bda2128a4f37
Reviewed-on: https://webrtc-review.googlesource.com/78521
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23384}
The new API stores events gathered by event type. For example, it is
possible to ask for a list of all incoming RTCP messages or all audio
playout events.
The new API is experimental and may change over next few weeks. Once
it has stabilized and all unit tests and existing tools have been
ported to the new API, the old one will be removed.
This CL also updates the event_log_visualizer tool to use the new
parser API. This is not a funcional change except for:
- Incoming and outgoing audio level are now drawn in two separate plots.
- Incoming and outgoing timstamps are now drawn in two separate plots.
- RTCP count is no longer split into Video and Audio. It also counts
all RTCP packets rather than only specific message types.
- Slight timing difference in sendside BWE simulation due to only
iterating over transport feedbacks and not over all RTCP packets.
This timing changes are not visible in the plots.
Media type for RTCP messages might not be identified correctly by
rtc_event_log2text anymore. On the other hand, assigning a specific
media type to an RTCP packet was a bit hacky to begin with.
Bug: webrtc:8111
Change-Id: Ib244338c86a2c1a010c668a7aba440482023b512
Reviewed-on: https://webrtc-review.googlesource.com/73140
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23056}
This reverts commit 9e336ec0b8.
Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming.
Original change's description:
> Create new API for RtcEventLogParser.
>
> The new API stores events gathered by event type. For example, it is
> possible to ask fo a list of all incoming RTCP messages or all audio
> playout events.
>
> The new API is experimental and may change over next few weeks. Once
> it has stabilized and all unit tests and existing tools have been
> ported to the new API, the old one will be removed.
>
> This CL also updates the event_log_visualizer tool to use the new
> parser API. This is not a funcional change except for:
> - Incoming and outgoing audio level are now drawn in two separate plots.
> - Incoming and outgoing timstamps are now drawn in two separate plots.
> - RTCP count is no longer split into Video and Audio. It also counts
> all RTCP packets rather than only specific message types.
> - Slight timing difference in sendside BWE simulation due to only
> iterating over transport feedbacks and not over all RTCP packets.
> This timing changes are not visible in the plots.
>
>
> Media type for RTCP messages might not be identified correctly by
> rtc_event_log2text anymore. On the other hand, assigning a specific
> media type to an RTCP packet was a bit hacky to begin with.
>
> Bug: webrtc:8111
> Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
> Reviewed-on: https://webrtc-review.googlesource.com/60865
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23015}
TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org
Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/72500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23026}
The new API stores events gathered by event type. For example, it is
possible to ask fo a list of all incoming RTCP messages or all audio
playout events.
The new API is experimental and may change over next few weeks. Once
it has stabilized and all unit tests and existing tools have been
ported to the new API, the old one will be removed.
This CL also updates the event_log_visualizer tool to use the new
parser API. This is not a funcional change except for:
- Incoming and outgoing audio level are now drawn in two separate plots.
- Incoming and outgoing timstamps are now drawn in two separate plots.
- RTCP count is no longer split into Video and Audio. It also counts
all RTCP packets rather than only specific message types.
- Slight timing difference in sendside BWE simulation due to only
iterating over transport feedbacks and not over all RTCP packets.
This timing changes are not visible in the plots.
Media type for RTCP messages might not be identified correctly by
rtc_event_log2text anymore. On the other hand, assigning a specific
media type to an RTCP packet was a bit hacky to begin with.
Bug: webrtc:8111
Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
Reviewed-on: https://webrtc-review.googlesource.com/60865
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23015}
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.
This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).
Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
We currently have one build target containing everything for audio_device: the interfaces,
the "fine" audio buffer, and the actual implementations for each platform.
Since we are planning to move the Android implementation to the sdk/android folder,
we only want to depend on the interfaces and the "fine" audio buffer, not the other platform
specific implementations. This CL splits the audio_device target into three different targets:
the interfaces, the fine audio buffer, and the platform specific implementations. The default
audio_device target now points to the interfaces instead.
Bug: webrtc:7452
Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd
Reviewed-on: https://webrtc-review.googlesource.com/61321
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22452}
So that we can avoid dependency cycles.
Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
This cl prepares for a later CL introducing a new send side congestion
controller that will run on a task queue. It mostly consists of minor
fixes but adds some new interfaces that are unused in practice.
Bug: webrtc:8415
Change-Id: I1b58d0180a18eb15320d18733dac0dfe2e0f902a
Reviewed-on: https://webrtc-review.googlesource.com/53321
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22099}
This reverts commit 4e849f6925.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Reland "Moved congestion controller to task queue.""
>
> This reverts commit 57daeb7ac7.
>
> Reason for revert: Cause increased congestion and deadlocks in downstream project
>
> Original change's description:
> > Reland "Moved congestion controller to task queue."
> >
> > This is a reland of 0cbcba7ea0.
> >
> > Original change's description:
> > > Moved congestion controller to task queue.
> > >
> > > The goal of this work is to make it easier to experiment with the
> > > bandwidth estimation implementation. For this reason network control
> > > functionality is moved from SendSideCongestionController(SSCC),
> > > PacedSender and BitrateController to the newly created
> > > GoogCcNetworkController which implements the newly created
> > > NetworkControllerInterface. This allows the implementation to be
> > > replaced at runtime in the future.
> > >
> > > This is the first part of a split of a larger CL, see:
> > > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > > For further explanations.
> > >
> > > Bug: webrtc:8415
> > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21868}
> >
> > Bug: webrtc:8415
> > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> > Reviewed-on: https://webrtc-review.googlesource.com/48000
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21899}
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:8415
> Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
> Reviewed-on: https://webrtc-review.googlesource.com/52980
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22017}
TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/53262
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22023}
This reverts commit 57daeb7ac7.
Reason for revert: Cause increased congestion and deadlocks in downstream project
Original change's description:
> Reland "Moved congestion controller to task queue."
>
> This is a reland of 0cbcba7ea0.
>
> Original change's description:
> > Moved congestion controller to task queue.
> >
> > The goal of this work is to make it easier to experiment with the
> > bandwidth estimation implementation. For this reason network control
> > functionality is moved from SendSideCongestionController(SSCC),
> > PacedSender and BitrateController to the newly created
> > GoogCcNetworkController which implements the newly created
> > NetworkControllerInterface. This allows the implementation to be
> > replaced at runtime in the future.
> >
> > This is the first part of a split of a larger CL, see:
> > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > For further explanations.
> >
> > Bug: webrtc:8415
> > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21868}
>
> Bug: webrtc:8415
> Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> Reviewed-on: https://webrtc-review.googlesource.com/48000
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21899}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8415
Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
Reviewed-on: https://webrtc-review.googlesource.com/52980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22017}
This is a reland of 0cbcba7ea0.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
Bug: webrtc:8415
Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
Reviewed-on: https://webrtc-review.googlesource.com/48000
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21899}
WHAT: made a BUILD.gn with library and tests in the Audio Processing
Module Voice Activity Detector directory. Updated depending
code. Fixed a Clang warning.
WHY: to make it possible for a target to depend on just the VAD and
not the whole APM. There are other benefits:
* Sometimes faster compilation.
* The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making
a peerconnection shared object file without the VAD has to be done in
steps. The first step is a custom target for the VAD. Hence this Cl.
Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af
BUG: webrtc:5716, webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/47521
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21893}
This reverts commit 0cbcba7ea0.
Reason for revert: Major regressions on perf bots.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ia8a273eb9e92b7d0d960c49658c228208170962d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/47560
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21877}
The goal of this work is to make it easier to experiment with the
bandwidth estimation implementation. For this reason network control
functionality is moved from SendSideCongestionController(SSCC),
PacedSender and BitrateController to the newly created
GoogCcNetworkController which implements the newly created
NetworkControllerInterface. This allows the implementation to be
replaced at runtime in the future.
This is the first part of a split of a larger CL, see:
https://webrtc-review.googlesource.com/c/src/+/39788/8
For further explanations.
Bug: webrtc:8415
Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
Reviewed-on: https://webrtc-review.googlesource.com/43840
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21868}
This is a reland of 001546da95
Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
>
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
>
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org
Bug: None
Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647
Reviewed-on: https://webrtc-review.googlesource.com/46900
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21862}
