The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.
This cleanup CL is related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
And put codecs/cng/webrtc_cng.h in a non-public build target while
we're at it.
Bug: webrtc:8396
Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c
Reviewed-on: https://webrtc-review.googlesource.com/c/108782
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25486}
This makes it safer to reason about the common case where send
time information is available. We don't have to either assume that
it's available, or check it everywhere the PacketResult struct is used.
To achieve this, a new field is added to TransportPacketsFeedback
and a new interface is introduced to clearly separate which field is
used. A possible followup would be to introduce a separate struct.
That would complicate the signature of ProcessTransportFeedback.
Bug: webrtc:9934
Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15
Reviewed-on: https://webrtc-review.googlesource.com/c/107080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25465}
Additionally, AudioProcessing::GetStatistics(bool) is made pure
virtual and the default implementation in AudioProcessing is removed.
Deprecation PSA:
https://groups.google.com/forum/#!msg/discuss-webrtc/NgqEPvkNuDE/7HtwnMmADgAJ
Bug: webrtc:9947, webrtc:8572
Change-Id: I123402bf7d6c49f3613154c469b818109d8fad43
Reviewed-on: https://webrtc-review.googlesource.com/c/108783
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25463}
to Mdns.*.
MdnsResponderInterface now explicitly requires the reference counting
of created names to allow the coexistence of multiple users of the same
responder where one user would not remove identical names created by
others.
MDns.* is also renamed to Mdns.* per the style guide.
TBR=aleloi@webrtc.org
Bug: webrtc:9605
Change-Id: I047fc41f34de8d4e97c980409a7f373769c4c252
Reviewed-on: https://webrtc-review.googlesource.com/c/101921
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25458}
This makes the calculation more similar to the one in WebRTCVoiceEngine.
Bug: webrtc:9510
Change-Id: Ibca69842726e51c07b9cc9550ff9f15a24161e28
Reviewed-on: https://webrtc-review.googlesource.com/c/107653
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25448}
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 CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.
On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.
Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
Modified PressEnterToContinue() to run the Windows message loop in the
context of the SingleThreadedTaskQueueForTesting thread. The previous
PressEnterToContinue() was running the message loop in the context of
the main thread, but the "Local Preview" and "Loopback Video #0" are
created in the context of the SingleThreadedTaskQueueForTesting thread
and the message loop must be executed in the context of the thread that
created these windows in order for these windows to respond to any
event.
BUG=webrtc:9123
Change-Id: I2ec19f2569a940a510d3b2bd3881a89032d70332
Reviewed-on: https://webrtc-review.googlesource.com/c/67520
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25408}
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705
Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
Rather fail at compile time than at run-time.
Bug: chromium:898373
Bug: webrtc:9855
Change-Id: Iaae81e04e4a8135814c1226f82d3a994de75e9ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107886
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25364}
This deprecates the following methods in VideoEncoder:
virtual ScalingSettings GetScalingSettings() const;
virtual bool SupportsNativeHandle() const;
virtual const char* ImplementationName() const;
Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.
Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().
This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.
Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
Use absl::optional instead of special constant to show, that we failed
to get OutputPath in fileutils_override
Bug: webrtc:9792
Change-Id: Ice19a9bf425e88a747dd9b07e82dbb5bdc59685b
Reviewed-on: https://webrtc-review.googlesource.com/c/107630
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25331}
The fuzzer uses a bitmask to construct the field trials string.
Now that there's 33 relevant field trials it's no longer large enough, so switch to a 64-bit type.
Bug: chromium:898373
Change-Id: I1ea68d451ceadbd9b720079a577b573866293e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/107650
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25330}
Add AecDump to the list of fuzzed stuff. Attaches an AecDump to the
Audio Processing Module in the APM-fuzzer. The AecDump writes to
/dev/null.
Bug: webrtc:7820
Change-Id: I03916ce4d1c69906ca8bb7e6fbe29c11e4ea55e5
Reviewed-on: https://webrtc-review.googlesource.com/c/107622
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25321}
The old render buffering code has been replaced, but can still be
activated by a killswitch. This change enables fuzzer testing of
the old code path.
Bug: webrtc:9726
Change-Id: I6e91cd4b4a95388cc63d1a65dade21b3c44be71b
Reviewed-on: https://webrtc-review.googlesource.com/c/107562
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25303}
This CL makes it possible to change transport routes while running
a scenario based test.
To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.
Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d
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 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d.
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}
We want sanitizer bots to show failure only for sanitizer defects.
To do so, this CL forces exit code to 0 unconditionally.
Sanitized binaries will turn it to 66 if there is any defect with diagnostic.
Bug: webrtc:9849
Change-Id: I46b683dcae12b76f1be177603af59e3f34bff3a9
Reviewed-on: https://webrtc-review.googlesource.com/c/107060
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25273}
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}
These two files were using absl::make_unique without #including the
header that declares it.
Bug: None
Change-Id: I03019c9a7e06370631680b474d04dd33716b0fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/107041
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25266}
This is to avoid very long runs, resulting in time-outs.
NOTRY=True
Bug: chromium:895082
Change-Id: Iafdc3d10b3fb52f2d487547c954dca8ae7edb783
Reviewed-on: https://webrtc-review.googlesource.com/c/105960
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25175}
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}
Also fix retransmission video send stream tests to not depend on actual frames sizes
Also, reduce key-frame scaling factor in FakeEncoder to better reflect real encoders behavior.
Bug: none
Change-Id: I33118160f3fec67ae8e732d9a85f0e9ee0784b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/105642
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25165}
Also renaming it Vp8TemporalLayers to show that it is codec specific.
Bug: webrtc:9012
Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f
Reviewed-on: https://webrtc-review.googlesource.com/c/104643
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25137}
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."
This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch
It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
This CL adds flags to the PacketOptions and PacktInfo struct that are
intended to be used to indicate if the packet belongs to a media stream
that is part of bitrate allocation as well as if it is included in
transport wide packet feedback.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: Icdf3e1e13d3f119574ee1b2c574f2d3329a7e303
Reviewed-on: https://webrtc-review.googlesource.com/c/104920
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25069}
Avoid using post_encode_callback, instead add a new callback function
to ConfigurableFrameSizeEncoder. Intention is to delete
post_encode_callback and the EncodedFrameObserver class in a later cl.
Bug: None
Change-Id: I79c103adf11c8915878b3f7cacf24c9b02dd6373
Reviewed-on: https://webrtc-review.googlesource.com/c/104840
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25064}
