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gerrit-public.fairphone.software
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platform
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external
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webrtc
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3172c035d5dcf270e29f151d1d01d5cbdfa26c1f
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data
d661e9c
WebRTC: Replace ProjectRootPath by ResourcePath
by ehmaldonado
· 8 years ago
dedaf1c
Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath.
by ehmaldonado
· 8 years ago
8e56521
The output signal of the AEC needs to be buffered as the
by peah
· 8 years ago
2ace3f9
The audio processing module (APM) relies on two for
by peah
· 8 years ago
58530ed
Updating APM unittests on the echo metrics.
by minyue
· 8 years ago
53f7ada
Add OWNERS file to data/audio_processing
by henrik.lundin
· 8 years ago
b1fc54d
Corrected the delay agnostic AEC behavior during periods of silent farend signal.
by peah
· 8 years ago
eb3603b
Don't always downsample to 16kHz in the reverse stream in APM
by aluebs
· 8 years ago
0bf612b
This CL is partially reverting the effects that
by peah
· 8 years ago
40cbec5
Fix the number of frames used when interleaving in AudioBuffer::InterleaveTo()
by Alejandro Luebs
· 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
4774874
Enable AudioProcessing48kHzSupport by default
by Alejandro Luebs
· 9 years ago
b1786db
audio_processing: Added a new AEC delay metric value that gives the amount of poor delays
by bjornv@webrtc.org
· 10 years ago
f17ee9c
Add case to ApmTest.Process to test the extended filter mode
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
b6af428
Adjust speech probability in NS when echo
by aluebs@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
c98ce3b
modules/audio_processing: Updates output_data_fixed.pb test file
by bjornv@webrtc.org
· 10 years ago
12cd443
Noise suppression: Change signature to work on floats instead of ints
by kwiberg@webrtc.org
· 10 years ago
d5da250
Revert "Revert "Audio processing: Feed each processing step its choice
by mflodman@webrtc.org
· 10 years ago
b1a66d1
Revert "Audio processing: Feed each processing step its choice of int or float data"
by mflodman@webrtc.org
· 10 years ago
934a265
Audio processing: Feed each processing step its choice of int or float data
by kwiberg@webrtc.org
· 10 years ago
382c0c2
Allow the RTP level indicator computation to work at any sample rate.
by andrew@webrtc.org
· 10 years ago
bbd47fc
Enables robust delay validation in AEC delay logging.
by bjornv@webrtc.org
· 11 years ago
3994e03
ADM no longer reads PCM files from data/audio_device. Now uses the resource folder instead.
by henrika@webrtc.org
· 12 years ago
6f90983
Remove data files from data/audio_coding
by tina.legrand@webrtc.org
· 12 years ago
08329f4
Added API to port internal speech probability in NS.
by bjornv@webrtc.org
· 12 years ago
9dc45da
Move trunk/test/data -> trunk/data
by andrew@webrtc.org
· 12 years ago