Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.
References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/https://stackoverflow.com/a/2524673
Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
- Make rtp_analyzer work with a single SSRC
- Simplify rtp_analyzer.sh (it allows to run the python script
from any directory)
- Update README.md (simplified, added missing dependency)
Bug: webrtc:10829
Change-Id: Idb82e7228918a973778762a39b732ce3b26b6bbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146711
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28668}
This is intended to by used for visualizing catagorical data, i.e. mapping
numerical enum values to string labels.
Bug: webrtc:10623
Change-Id: Ic9c3da9a3874f479c07412f394a774ae90fd3d7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145408
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28656}
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 the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
This is a temporary solution, as there are several other executables and
some tests in rtc_tools/BUILD.gn. Including all of them to default target
is not decided yet.
But as rtp_generator tends to be broken reguraly, It should be included
there at least for now.
Bug: webrtc:10807
Change-Id: I3acf5a93c74bf1e2474c6aaee35653efbb43d3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28595}
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}
Passing --stats_file_ref to frame_analyzer (which does not support
this flag anymore!) became an error with the switch to absl flags.
Bug: webrtc:10616
Change-Id: Ifc34001eafd9a92234ec1d12c3004d9f51a65f22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143783
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28370}
This is a reland of fa79081dca
It crashed due to inability to handle small timestamps in probe
estimator. This was fixed by moving history window check to avoid
subtracting from the timestamp.
Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}
Bug: webrtc:9510
Change-Id: I3bf91222068e4fbb6aa159bfeb7a73e00bb6a0d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143165
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28347}
This reverts commit fa79081dca.
Reason for revert: Breaks downstream project.
Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}
TBR=terelius@webrtc.org,srte@webrtc.org
Change-Id: I562365fc5d1da68326d603338ccc6371114d7e12
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9510
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143164
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28331}
As the send time congestion controller now has been removed,
we don't need the RTP related constructs anymore.
Bug: webrtc:9510
Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28330}
In https://chromium-review.googlesource.com/1650265 attributes like minSdkVersion were moved from AndroidManifest.xml to GN files. For WebRTC there were a few problems with that.
* We don't want to suppress UsesMinSdkAttributes lint but now there are these "invalid" manifest files that we can't exclude or discern. So disable this lint error.
https://chromium-review.googlesource.com/c/chromium/src/+/1650265/14/build/android/AndroidManifest.xml
* We should specify the versions in GN files, so I did that here (by exactly copying the versions that are already in the targets' corresponding XML files), but we never want to get rid of them in the XML files. For now this information will just be duplicated (without any synchronicity check!) so there should be followup to this.
Change log: 6ae0f0cd4c..bf62d746a4
Full diff: 6ae0f0cd4c..bf62d746a4
Changed dependencies
* src/base: 9e5e9332df..e5a1d1f652
* src/build: 5a031748ec..2ef566e990
* src/buildtools: 6ae683be2f..6f3775ad6e
* src/buildtools/linux64: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/mac: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/win: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/ios: 2f5c817266..7f1a97d593
* src/testing: 1d4247de57..b1b36ff0d4
* src/third_party: 6f7cbf7c46..42e96c4074
* src/third_party/android_sdk/public: ki7EDQRAiZAUYlnTWR1XmI6cJTk65fJ-DNZUU1zrtS8C..xhyuoquVvBTcJelgRjMKZeoBVSQRjB7pLVJPt5C9saIC
* src/third_party/android_sdk/public: iIwhhDox5E-mHgwUhCz8JACWQCpUjdqt5KTY9VLugKQC..ppQ4TnqDvBHQ3lXx5KPq97egzF5X2FFyOrVHkGmiTMQC
* src/third_party/android_sdk/public: 4Y2Cb2LGzoc-qt-oIUIlhySotJaKeE3ELFedSVe6Uk8C..MSnxgXN7IurL-MQs1RrTkSFSb8Xd1UtZjLArI8Ty1FgC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ed9fcf3f70..9e5dbd8b46
* src/tools: f58f33bca1..a9a4b8fc7b
DEPS diff: 6ae0f0cd4c..bf62d746a4/DEPS
No update to Clang.
Bug: chromium:891996
Change-Id: I773d6fa90e8083d934c84eecc1cb9d7d4496eca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142235
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28311}
This interface makes future refactoring difficult and is now in practice
only implemented by PacketRouter.
Bug: webrtc:10633
Change-Id: I3fcb8940781aa7431119649bde7594592a8c8851
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141669
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28251}
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}
It's easy to make small errors when building field trial strings, and
those errors can cause all sorts of weird problems. This CL checks if
the FT string has an odd number of delimiters, duplicate
names or any trailing chars.
If so we'll log a error message. On debug builds we'll also crash.
Bug: webrtc:10729
Change-Id: Iebf7155d9b117a02d1e9cfe7f64408e11df2aec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140866
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28234}
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}
The simulation currently doesn't set the transport sequence number before inserting
the packets into the send time history. This means that send times can't be looked up
when receiving feedback, essentially disabling BWE simulation.
Bug: None
Change-Id: I3f2789324eb81f784dd5a6c5a5a770767236a3fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138826
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28082}
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}
With this change, both the normal RTP and the transport-wide sequence
numbers are propagated with with AddPacket() call via a new
RtpPacketSendInfo struct, replacing the previous set of parameters.
The intent with this is that SendTimeHistory can hold a mapping from
transport-wide to rtp sequence numbers, and then via callbacks let the
RTP modules know when packets have been received by the remote end.
Bug: webrtc:8975
Change-Id: I6a24fc6282cbb041393752d39593c2867b242192
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133021
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27708}
This is because padding (sent as RTX) makes the plot unreadable.
TBR=terelius@webrtc.org
Bug: None
Change-Id: Iddf681eab6ec826c6f3c620aac65e2bd6f31b895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133182
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27641}
Moving the "$@" last in the invocation of git log enables line stats
restricted to specific files or directories, e.g.,
./rtc_tools/author_line_count.sh ... -- modules/foo
Bug: None
Change-Id: I6dc17a10f2b321beae452f5e2cc74bcba2f8aaf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130491
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27381}
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}
Both reference and tested videos were created
via a file whose path was fixed.
So, when tests were launched in parallel, race conditions ensued.
This CL creates an unique temporary filename for each video.
Bug: webrtc:10156
Change-Id: Ie3abf85abdfa95735cb86880bbd6a59393e609c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127883
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27219}
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 change adds incoming & outgoing packet rates to the
event_log_visualizer.
The outgoing packet rate is drawn on the graph with outgoing RTP rate,
because we want to see it together with bandwidth estimate and probe
clusters.
The incoming packet rate is drawn separately.
Bug: webrtc:9719
Change-Id: I32648d016359af110837440ed1a5f9c31c841ea7
Reviewed-on: https://webrtc-review.googlesource.com/c/122941
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26696}
This is not a functional change. I've verified that the event_log_visualizer outputs the same bytes before and after the CL.
Bug: webrtc:10102, webrtc:10312
Change-Id: I49c4c847926078cefc9b72fe57fbdaebf76423e9
Reviewed-on: https://webrtc-review.googlesource.com/c/122844
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26685}
Chromium's official builds set -D_FORTIFY_SOURCE=2, causing among other
things warnings about unused return values from stdlib functions.
We don't normally build "all" in that configuration, and so missed some
instances.
Bug: chromium:931227
Change-Id: I69820d4e639c5908e0092dded1dea39c51d45d6b
Reviewed-on: https://webrtc-review.googlesource.com/c/122560
Commit-Queue: Hans Wennborg <hans@chromium.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26657}
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}
This CL applies clang-tidy's performance-faster-string-find [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-faster-string-find.html
Bug: webrtc:10252
Change-Id: I4b8c0396836f3c325488e37d97037fa04742a5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/120047
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26423}
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}
Instead timestamps required for processing are provided explicitly.
This makes it easier to ensure correct usage in log processing
and simulation.
Bug: webrtc:10170
Change-Id: I724a6b9b94e83caa22b8e43b63ef4e6b46138e6a
Reviewed-on: https://webrtc-review.googlesource.com/c/118702
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26339}
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 will print out the major events during a NetEq simulation.
Bug: b/116685514
Change-Id: Iab172e9a9115695b42c67628d5523c727359bb89
Reviewed-on: https://webrtc-review.googlesource.com/c/114320
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26019}
This handles an unlikely corner case where you receive a RTCP feedback for a packet the same millisecond that you send it.
Bug: None
Change-Id: I77f460bef4073d4d9c5633c88f4d2dd8470f8577
Reviewed-on: https://webrtc-review.googlesource.com/c/113305
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25911}
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}
He works on video analysis code every now and then, and EngProd
isn't much help when reviewing here.
Bug: None
Change-Id: I30b5f12584305d17d4c6a9682790fd0eda67d867
Reviewed-on: https://webrtc-review.googlesource.com/c/111881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25783}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
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}
It's not currently used or maintained, so it shouldn't be a part of out API.
Bug: webrtc:9824
Change-Id: Ic44c5ea3a9eab8fb75e87a5005cbf6cdd4b1d4ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107645
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25593}
The RTC event log analyzer would previously only plot network latency
for incoming video streams. (The latency is computed from the capture
time in the RTP header, and the packet receive time.) This CL adds
support for audio packets, which requires estimating the RTP clock
frequency for the incoming packets.
Bug: None
Change-Id: Idf1ff9febfdd4097976b22a61f1c5679deb6068c
Reviewed-on: https://webrtc-review.googlesource.com/c/108784
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25580}
Replaced by a int64_t representing time in us. To aid transition of
downstream code, rtc::PacketTime is made an alias for int64_t.
Bug: webrtc:9584
Change-Id: Ic3a5ee87d6de2aad7712894906dab074f1443df9
Reviewed-on: https://webrtc-review.googlesource.com/c/91860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25503}
This is a preparation for deleting rtc::PacketTime. Next step, after
downstream code has been updated to not access the |timestamp| member,
is to make rtc::PacketTime an alias for int64_t.
Also delete the unused member rtc::PacketTime::not_before.
Bug: webrtc:9584
Change-Id: Iba9d2d55047d69565ad62b1beb525591fd432ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/108860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25468}
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 is a reland of ff292f30d9
I'm leaving empty .py files in place in order to not break downstream client builds.
Original change's description:
> Remove deprecated barcode scanning functionality
>
> This code is not used anymore, but it's not possible to land this CL
> until issue webrtc:9665 is fixed.
>
> Bug: webrtc:9642,webrtc:9665
> Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
> Reviewed-on: https://webrtc-review.googlesource.com/c/95951
> 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@{#25289}
TBR=phensman@webrtc.org,phoglund@webrtc.org
Bug: webrtc:9642, webrtc:9665
Change-Id: I248f8656b14c89b0b92e777f4408ee6a6dad41f9
Reviewed-on: https://webrtc-review.googlesource.com/c/107360
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25296}
This code is not used anymore, but it's not possible to land this CL
until issue webrtc:9665 is fixed.
Bug: webrtc:9642,webrtc:9665
Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
Reviewed-on: https://webrtc-review.googlesource.com/c/95951
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@{#25289}
This is a reland of 5ccdc1331f
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=kwiberg@webrtc.org
Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
This reverts commit 5ccdc1331f.
Reason for revert: Breaks downstream project.
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).
They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.
This CL adds the 'WEBRTC_' prefix to them.
Generated with:
for x in DECLARE DEFINE; do
for y in bool int float string FLAG; do
git grep -l "\b$x\_$y\b" | \
xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
done
done
git cl format
Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
This CL avoids that unpack_aecdump produces an empty callorder.char file
regardless of it not writing any data to that file
Bug: webrtc:5298
Change-Id: I15b01764a0dc16045346dd680e9bd4c1869c0d2c
Reviewed-on: https://webrtc-review.googlesource.com/c/98340
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25214}
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}
compare_videos.py will now print the VMAF score for each frame.
The CL also removes some stale comments.
Bug: webrtc:9642
Change-Id: I5623588580dea06dd487d7763dc3a2511bd2cd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/105103
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25092}
SetExecutablePath isn't used anymore.
Nobody was using the fancy select-per-platform functionality, and the
documentation was wrong anyway. In the cases somebody needed an
override per platform, they were using defines in their own test
instead. I think that is more verbose but more predictable and easy
to understand (see how it's done in audio_processing_unittest.cc
when loading output_data_mac, for instance).
Bug: webrtc:9792
Change-Id: I7289bf5883fe43852638922d7c7583eae0c08601
Reviewed-on: https://webrtc-review.googlesource.com/c/104482
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25062}
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 is a reland of e307d56bd7
options.yuv_directory would be unset if vmaf was not used.
It now gets set to None.
Also adds a try-finally around the temp directory for YUV files.
Original change's description:
> Add option to call VMAF in compare_videos.py.
>
> VMAF compares videos on several metrics and produces a unified score.
>
> Calling it from compare_videos required passing in a path to a VMAF
> executable and a model.
>
> VMAF needs to compare aligned videos in YUV format, so two videos
> (ref and test) will be saved by frame_analyzer after it has aligned
> them.
>
> Bug: webrtc:9642
> Change-Id: Idddfcf6b1b235e7f925696ffc38938fb84c4ff9e
> Reviewed-on: https://webrtc-review.googlesource.com/102140
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24876}
Bug: webrtc:9642
Change-Id: I1d04a56090e68df47dc3e6b7e710384244470d0c
TBR: phoglund
Reviewed-on: https://webrtc-review.googlesource.com/102544
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24896}
This reverts commit e307d56bd7.
Reason for revert:
Breaks client.webrtc.perf bots. Example failure:
https://ci.chromium.org/buildbot/client.webrtc.perf/Android32%20Tests%20(L%20Nexus7.2)/8635
AttributeError: Values instance has no attribute 'yuv_directory'
Original change's description:
> Add option to call VMAF in compare_videos.py.
>
> VMAF compares videos on several metrics and produces a unified score.
>
> Calling it from compare_videos required passing in a path to a VMAF
> directory, where there should be a C++ wrapper executable and a model.
> For now, the relative paths to those are constant.
>
> VMAF needs to compare aligned videos in YUV format, so two videos
> (ref and test) will be saved by frame_analyzer after it has aligned
> them.
>
> Bug: webrtc:9642
> Change-Id: Idddfcf6b1b235e7f925696ffc38938fb84c4ff9e
> Reviewed-on: https://webrtc-review.googlesource.com/102140
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24876}
TBR=phoglund@webrtc.org,sakal@webrtc.org,phensman@webrtc.org
Change-Id: I3e1dc98d7dfc0309ee2934cb3a978eecf274c477
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/102561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24883}
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}
VMAF compares videos on several metrics and produces a unified score.
Calling it from compare_videos required passing in a path to a VMAF
directory, where there should be a C++ wrapper executable and a model.
For now, the relative paths to those are constant.
VMAF needs to compare aligned videos in YUV format, so two videos
(ref and test) will be saved by frame_analyzer after it has aligned
them.
Bug: webrtc:9642
Change-Id: Idddfcf6b1b235e7f925696ffc38938fb84c4ff9e
Reviewed-on: https://webrtc-review.googlesource.com/102140
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24876}
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}
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}
