Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
a4d873786f10eedd72de25ad0d94ad7c53c1f68a
/
modules
/
audio_processing
/
audio_processing_unittest.cc
« Previous
eb3603b
Don't always downsample to 16kHz in the reverse stream in APM
by aluebs
· 8 years ago
3eeb2e8
Moved the audioprocessing unittest to the audio_processing folder
by peah
· 8 years ago
[Renamed from webrtc/modules/audio_processing/test/audio_processing_unittest.cc]
0bf612b
This CL is partially reverting the effects that
by peah
· 8 years ago
2a5609d
Increase kHasVoiceCountNear by one in audio_processing_unittest
by Alejandro Luebs
· 8 years ago
40cbec5
Fix the number of frames used when interleaving in AudioBuffer::InterleaveTo()
by Alejandro Luebs
· 8 years ago
b031955
Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API
by aluebs
· 8 years ago
df6416a
Dont always downsample to 16kHz in the reverse stream in APM
by aluebs
· 8 years ago
776593b
Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands
by aluebs
· 8 years ago
dfc2870
Revert of Drop the 16kHz sample rate restriction on AECM and zero out higher bands (patchset #3 id:40001 of https://codereview.webrtc.org/1774553002/ )
by perkj
· 8 years ago
f687d53
Drop the 16kHz sample rate restriction on AECM and zero out higher bands
by Alex Luebs
· 8 years ago
7b19b08
Reland "Calculating ERLE in AEC more properly."
by minyue
· 8 years ago
c9bbbe4
Revert "Calculating ERLE in AEC more properly."
by minyuel
· 8 years ago
944744b
Calculating ERLE in AEC more properly.
by minyuel
· 8 years ago
62eaacf
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/
by kwiberg
· 8 years ago
78ddd73
Update path for audioproc_debug proto output.
by kjellander
· 8 years ago
691b836
Using buffered signal to calculate the level of echo cancellation.
by minyue
· 8 years ago
d66b44d
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
by ivoc
· 8 years ago
6955870
Convert channel counts to size_t.
by Peter Kasting
· 8 years ago
25702cb
Misc. small cleanups.
by pkasting
· 9 years ago
e2976c8
Remove DISABLED_ON_ macros.
by Peter Boström
· 9 years ago
a4df27b
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
by ivoc
· 9 years ago
f4f5cb0
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
by ivoc
· 9 years ago
36d4c54
Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
by ivoc
· 9 years ago
ae2c5ad
Added option to specify a maximum file size when recording an AEC dump.
by ivoc
· 9 years ago
ff761fb
modules: more interface -> include renames
by Henrik Kjellander
· 9 years ago
f1104f6
Remove TODO referring to issue1981, which I just marked WontFix.
by Andrew MacDonald
· 9 years ago
98f5351
system_wrappers: rename interface -> include
by Henrik Kjellander
· 9 years ago
5aaa9b4
Removed unused API functions in AudioProcessing and AudioProcessingModule
by peah
· 9 years ago
dce40cf
Update a ton of audio code to use size_t more correctly and in general reduce
by Peter Kasting
· 9 years ago
60d9b33
Integrate Intelligibility with APM
by ekmeyerson
· 9 years ago
86c6d33
Allow more than 2 input channels in AudioProcessing.
by Michael Graczyk
· 9 years ago
64e753c
Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)
by magjed
· 9 years ago
c204754
Allow more than 2 input channels in AudioProcessing.
by Michael Graczyk
· 9 years ago
bb36fdf
Remove empty-string comparisons.
by pbos
· 9 years ago
0f133b9
Rename APM Config ReportedDelay to DelayAgnostic
by henrik.lundin
· 9 years ago
441f634
Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
by Henrik Lundin
· 9 years ago
3fbf3f8
Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
by Henrik Lundin
· 9 years ago
5f4b7e2
Rename APM Config DelayCorrection to ExtendedFilter
by Henrik Lundin
· 9 years ago
4774874
Enable AudioProcessing48kHzSupport by default
by Alejandro Luebs
· 9 years ago
fade179
Remove leaking aecdump testfiles.
by Peter Boström
· 9 years ago
cb05b72
Add WAV and arbitrary geometry support to nlbf test.
by Andrew MacDonald
· 9 years ago
0f663de
Rename Beamformer to NonlinearBeamformer.
by mgraczyk@chromium.org
· 9 years ago
00b8f6b
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
by kwiberg@webrtc.org
· 9 years ago
d35a5c3
Make ChannelBuffer aware of frequency bands
by aluebs@webrtc.org
· 9 years ago
200ac00
Remove temp files in audio_processing_unittest.cc.
by pbos@webrtc.org
· 9 years ago
b1786db
audio_processing: Added a new AEC delay metric value that gives the amount of poor delays
by bjornv@webrtc.org
· 9 years ago
f17ee9c
Add case to ApmTest.Process to test the extended filter mode
by aluebs@webrtc.org
· 9 years ago
d82f55d
Only adapt AGC when the desired signal is present
by aluebs@webrtc.org
· 9 years ago
d6e84d9
Always copy processed audio to output buffer in ProcessStream.
by mgraczyk@chromium.org
· 9 years ago
a525c98
Fix parallelizability in ApmTests.
by pbos@webrtc.org
· 9 years ago
bac0012
Extend delay estimation window in AEC to 500 ms on all platforms
by bjornv@webrtc.org
· 10 years ago
8789376
Move ChannelBuffer class to channel_buffer file
by aluebs@webrtc.org
· 10 years ago
8328e7c
Revert "Revert part of r7561, "Refactor audio conversion functions.""
by andrew@webrtc.org
· 10 years ago
bcfb4d0
Revert part of r7561, "Refactor audio conversion functions."
by kwiberg@webrtc.org
· 10 years ago
4fc4add
Refactor audio conversion functions.
by andrew@webrtc.org
· 10 years ago
30be827
Enable render downmixing to mono in AudioProcessing.
by andrew@webrtc.org
· 10 years ago
a0ce9fa
Call NS AnalyzeCaptureAudio before AEC
by aluebs@webrtc.org
· 10 years ago
dc0b37d
modules_unittests: Turned on ApmTest.Process test for Android
by bjornv@webrtc.org
· 10 years ago
8dd60cc
audio_processing_unittests: Enabled ApmTest.Process for all platforms but Android
by bjornv@webrtc.org
· 10 years ago
84f8ec1
Changes to tests and tools in audio_processing.
by bjornv@webrtc.org
· 10 years ago
5c3f4e3
Fixes and re-enables tests disabled on Android
by bjornv@webrtc.org
· 10 years ago
cb0ea43
audio_processing: Forces extended filter to be used in splitting filter test.
by bjornv@webrtc.org
· 10 years ago
b616e12
Disables some modules_unittests on Android.
by bjornv@webrtc.org
· 10 years ago
2812b59
Re-enables CommonFormats test for Android.
by bjornv@webrtc.org
· 10 years ago
be4ab99
Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8.
by stefan@webrtc.org
· 10 years ago
21299d4
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
by andrew@webrtc.org
· 10 years ago
103657b
Add keyboard channel support to AudioBuffer.
by andrew@webrtc.org
· 10 years ago
f26c9e8
Use unique filenames in AudioProcessingTests for parallelization.
by andrew@webrtc.org
· 10 years ago
ddbb8a2
Support arbitrary input/output rates and downmixing in AudioProcessing.
by andrew@webrtc.org
· 10 years ago
1092ea0
Add format specification to output file names
by henrik.lundin@webrtc.org
· 10 years ago
a8b9737
Add tests and modify tools for new float deinterleaved interface.
by andrew@webrtc.org
· 10 years ago
3e0b60f
Switch to correct interpretation of int and float input data in audio_processing_unittest
by bjornv@webrtc.org
· 10 years ago
17e4064
Add a deinterleaved float interface to AudioProcessing.
by andrew@webrtc.org
· 10 years ago
27c6980
Move the volume quantization workaround from VoE to AGC.
by andrew@webrtc.org
· 10 years ago
c9ee412
Re-enabling audio processing tests
by aluebs@webrtc.org
· 10 years ago
8bc4fcf
Temporarily disabling audio processing tests.
by aluebs@webrtc.org
· 10 years ago
bbd47fc
Enables robust delay validation in AEC delay logging.
by bjornv@webrtc.org
· 10 years ago
d335094
Init to 16 kHz in the fixed-point profile.
by andrew@webrtc.org
· 11 years ago
60730cf
Remove the requirement to call set_sample_rate_hz and friends.
by andrew@webrtc.org
· 11 years ago
f8be8df
audio_processing_unittest: unbreak clang compilation.
by fischman@webrtc.org
· 11 years ago
863b536
Allow opening an AEC dump from an existing file handle.
by henrikg@webrtc.org
· 11 years ago
3d9981d
Remove unused ThreadData struct.
by andrew@webrtc.org
· 11 years ago
d7696c4
Compile-out functions only used by the bit-exact test.
by andrew@webrtc.org
· 11 years ago
3555303
Roll chromium_revision 226126:228675 and fix clang warnings
by kjellander@webrtc.org
· 11 years ago
f3930e9
Small refactoring of AudioProcessing use in channel.cc.
by andrew@webrtc.org
· 11 years ago
a950300b
Disables unit tests that don't work on Android for Android.
by henrike@webrtc.org
· 11 years ago
c66aaaf
Rename unit_test.{cc,h} under module_unittest.
by pbos@webrtc.org
· 11 years ago
[Renamed from webrtc/modules/audio_processing/test/unit_test.cc]
83cebb2
Removes unused main function that is poluting the build.
by henrike@webrtc.org
· 11 years ago
8c34cee
Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD
by pbos@webrtc.org
· 11 years ago
7fad4b8
Include files from webrtc/.. paths in audio_processing/
by pbos@webrtc.org
· 11 years ago
91d11b3
Adds new AEC API to audio_processing.
by bjornv@webrtc.org
· 11 years ago
6be1e93
Properly error check calls to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
78693fe
Return an error when greater than 16 kHz is used with AECM.
by andrew@webrtc.org
· 11 years ago
3e10249
Added delay estimation test to audio processing unit tests.
by bjornv@webrtc.org
· 11 years ago
ae1a58b
Replace AudioFrame's operator= with CopyFrom().
by andrew@webrtc.org
· 11 years ago
8186534
Only reinitialize AudioProcessing when needed.
by andrew@webrtc.org
· 12 years ago
534e495
Qickly fixed android platform build breakage
by leozwang@webrtc.org
· 12 years ago
14b43be
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago
[Renamed from src/modules/audio_processing/test/unit_test.cc]
28d0140
Allow audioproc_unittest to be run with an absolute path.
by andrew@webrtc.org
· 12 years ago
8a7396f
Use TestSuite for startup in audioproc_unittest's custom main().
by andrew@webrtc.org
· 12 years ago
Next »