For each single experiment, a URL is defined by adding a specific anchor.
A URL can be copied using the button beneath the score of the experiment
one would like to share.
This CL also includes a few optimizations and fixes:
- JS and CSS are minified
- Dialog close event listener added, this fixes a small bug preventing
the play out audio to stop when pressing ESC instead of using the close
button
- Snackbar notifications added
- Simple unit test for the export module
BUG=webrtc:7218
Change-Id: Iad00ce69094a5968ee0520d105d59656cfafa4e2
TBR=
Change-Id: Iad00ce69094a5968ee0520d105d59656cfafa4e2
Reviewed-on: https://webrtc-review.googlesource.com/7960
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20266}
When the test was created, it was disabled for mobile platforms from
the beginning. This is likely a copy-paste from the related
NetEqDecodingTest.TestBitExactness which includes testing codecs not
supported on mobile platforms (e.g., iLBC). This restriction is not
needed for the Opus-only test.
The test remains disabled for iOS, since none of the bots actually run
the relevant test binary on actual iOS devices.
Bug: none
Change-Id: I9071e0e32c83b62c8c7af59ac1cb3e46227f8e8e
Reviewed-on: https://webrtc-review.googlesource.com/8561
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20264}
Add missing data dependencies and add it to gn_isolate_map.pyl
Bug: chromium:749648
Change-Id: I6b6c1bb2e4d647471a2747042788a691ce2e1e5d
Reviewed-on: https://webrtc-review.googlesource.com/8721
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20258}
This CL enables a factory method creating acoustic echo cancellers
that inherit EchoControl to be injected into the audio processing
module. AudioProcessing will call the factory method to create an
instance of the EchoControl subclass when needed. In the event of
sample rate changes, AudioProcessing will recreate the object using
the factory method.
Bug: webrtc:8346
Change-Id: I0c508b4d4cdb35569864cefaa0e3aea2555cc9b9
Reviewed-on: https://webrtc-review.googlesource.com/7742
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20251}
This reverts commit af721b72cc1bdc5d945629ad78fbea701b6f82b9.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
TBR=asapersson@webrtc.org,sprang@webrtc.org
Change-Id: Ic914f03ff7065ac410ae06b6f82b56a935399b66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8375
Reviewed-on: https://webrtc-review.googlesource.com/8480
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20248}
Temporary files created by AudioFormat tests in modules_unittest are
removed after each test case rather than after the whole suite is
finished. This saves disk space on the running device.
Bug: webrtc:8344
Change-Id: Iace3a7a62bb06e15fa596caf32da873944654c9a
Reviewed-on: https://webrtc-review.googlesource.com/8100
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20244}
Remove REMB accessor as used for debug checks only.
Merge SetRembData and SetRembStatus(true) eliminating
state 'remb can be send, but no data available yet'
Bug: None
Change-Id: I4c1c19435657e5cde02a17de90ec6de9f00b7daf
Reviewed-on: https://webrtc-review.googlesource.com/7983
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20240}
This CL changes the aggregation of the matched filter delay
estimates in AEC3 to using a histogram approach.
Bug: chromium:773541,webrtc:8379
Change-Id: I5322c65858188599397ef5716fecdebc34852e6a
Reviewed-on: https://webrtc-review.googlesource.com/8261
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20234}
This CL changes the tuning of AEC3 to increase the transparency.
In particular:
-The present parameters are re-tuned.
-An oversuppression factor is added in the newly added soft-knee in
the NLP gain. The purpose of this is to avoid fluctuations in the
residual echo.
-The dynamics of the computed gain are bounded to ensure that the
specified gain characteristics are realizable without echo leakage.
This also adds robustness against echo leakage in frequency regions
that are poorly estimated.
This change was needed to avoid echo leakage from the above
tunings.
Bug: chromium:773543,webrtc:8378
Change-Id: If8acc41c1423a6a2fa6f8c4daf2735c86f0b529a
Reviewed-on: https://webrtc-review.googlesource.com/8262
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20231}
Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
Store sent frame info in map to solve potential issue where sent framerate statistics could be
incorrect.
Bug: webrtc:8375
Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
Reviewed-on: https://webrtc-review.googlesource.com/7880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20225}
This CL adds a smooth rampup of the NLP gain in AEC3.
Bug: webrtc:8361
Change-Id: I49aa75904751ffe9150db1572271fe7a26232449
Reviewed-on: https://webrtc-review.googlesource.com/7740
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20213}
This CL separates the NLP gain computation for the different variants
of echo estimation. This simplifies the setting of tuning
parameters, with resulting transparency improvements and increased
echo removal performance.
Bug: webrtc:8359
Change-Id: I9b97064396fb6f6e2f418ce534573f68694390a1
Reviewed-on: https://webrtc-review.googlesource.com/7613
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20209}
This CL adds the ability to set a default echo path delay to use
in AEC3 when there is prior knowledge about the delay in the echo
path.
Bug: webrtc:8358
Change-Id: Ie368f9a6dec9f412e09bf0e095f89d84305045f9
Reviewed-on: https://webrtc-review.googlesource.com/7604
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20204}
Simple rename to reflect that any AEC implementing the EchoControl
interface could be used instead of EchoCanceller3.
Bug: webrtc:8346
Change-Id: Id9abdc15bf3e0b30197077b8c11e20891a7463b3
Reviewed-on: https://webrtc-review.googlesource.com/7611
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20203}
This CL adds some general AEC3 transparency improvements.
Specifically:
-A minimum for how the nearend is masking echo is added.
-A temporal smoothing constant is increased to increase the transparency.
-Parameters are surfaced to the parameter config struct.
Bug: webrtc:8360
Change-Id: I2a4881eb40f4fab53ad740c4001925f0af86bbec
Reviewed-on: https://webrtc-review.googlesource.com/7605
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20200}
Remove redundant null pointer checks.
move header fields validation when they passed in rather when used.
Validate all used fields from the header.
Bug: webrtc:8335
Change-Id: I20b132c6fb8966e49a5414fe757f74d504f4a61a
Reviewed-on: https://webrtc-review.googlesource.com/7400
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20194}
Care should be taken when landing this, because it will affect users of
WebRTC. I'm thinking primarily of Chromium. Chromium will start to
support High profile and Baseline profile using SW codecs with this CL.
Clients who do SDP munging without looking at the H264 profile might
switch from Constrained Baseline to High profile with this change.
Bug: webrtc:8317
Change-Id: Idca3a6b761a66d9e521b913b850c6ae14381f1f4
Reviewed-on: https://webrtc-review.googlesource.com/6341
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20190}
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to
not rely in the indirect include.
Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
These field trials can be set with a string similar to:
WebRTC-BweWindowSizeInPackets/Enabled-150/WebRTC-BweBackOffFactor/Enabled-0.95/
BweWindowSizeInPackets
Number of packets which the delay-based BWE window is based on. A larger value means lower delay-sensitivity.
Default in WebRTC: 20
Reasonable values for streaming: 50-150
BweBackOffFactor
How far the BWE will back off when the delay increases. A value closer to 1.0 means smaller back-off.
Range: > 0.0, < 1.0
Default in WebRTC: 0.85
Reasonable values for streaming: 0.85-0.95
Bug: webrtc:8212
Change-Id: I61f0883788b689847a43273b63cef663042f4d42
Reviewed-on: https://webrtc-review.googlesource.com/6764
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20172}
This CL adds the possibility to specify a custom path for the noise tracks to use with
the addivitve noise test data generator (formerly called environmental noise generator).
It also includes a minor refactoring of ApmModuleSimulator to allow injection and remove
all the parameters that were forwarded to its dependencies.
Bug: webrtc:7494
Change-Id: I07bc359913c375a51bd3692822814d3ce8437268
Reviewed-on: https://webrtc-review.googlesource.com/5982
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20163}
This is expected to result in a slight loss of overall quality, but
should offset by quicker switching between temporal layers with flaky
connections.
Bug: webrtc:7694
Change-Id: Ib605802bb59f12773652324ac66cdf4774ae6bb6
Reviewed-on: https://webrtc-review.googlesource.com/6881
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20160}
A script for producing boxplots by parsing data generated by the
apm_quality_assessment.py tool.
The script groups data by the values of one or several audioproc_f
parameters. For every such subgroup it draws a boxplot. All boxplots
are shown next to each other with the parameter value as the x axis.
It is similar to this matplotlib example:
https://matplotlib.org/mpl_examples/pylab_examples/boxplot_demo_06.png
The script
1. reads config file names from the pandas dataframe generated by
quality_assurance.collect_data
2. parses the (JSON) config files to read the parameter values
3. groups data with matching param values together
4. draws a boxplot for each group using matplotlib
TBR=alessiob@webrtc.org # reviewed already in old gerrit https://chromium-review.googlesource.com/c/external/webrtc/+/660559
BUG: webrtc:7218
Change-Id: I380a1363d26721feb975fad1506835c622e9d926
Reviewed-on: https://webrtc-review.googlesource.com/6340
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20139}
This is a reland of 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30
Original change's description:
> Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
>
> (Re-upload of https://codereview.webrtc.org/3020493002/)
>
> Bug: webrtc:4690, webrtc:7306
> Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
> Reviewed-on: https://webrtc-review.googlesource.com/5360
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20083}
Bug: webrtc:4690, webrtc:7306
Change-Id: Ib019439fe6ab0e6b759819e1e9bd320ba1d983bd
Reviewed-on: https://webrtc-review.googlesource.com/6300
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20137}
This is needed in the general case, now that we aim to support codecs
other than those built-in to WebRTC.
BUG=webrtc:8159
Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706
Reviewed-on: https://webrtc-review.googlesource.com/5380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20136}
SignalProcessingUtils.MixSignals() now allows different padding options.
This CL also adds more unit tests for SignalProcessingUtils.MixSignals().
Bug: webrtc:7494
Change-Id: Id62fe9998e512c275cb6399e0aedf11f23a9f36e
Reviewed-on: https://webrtc-review.googlesource.com/5780
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20122}
