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webrtc.googlesource.com
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src
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f9d303c042b13d4eaa9acfa901b7e2e9841c99fa
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modules
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audio_processing
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audio_processing_impl_unittest.cc
6f72f56
Change return types of refcount methods.
by Niels Möller
· 8 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 8 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 8 years ago
[Renamed from webrtc/modules/audio_processing/audio_processing_impl_unittest.cc]
56359be
Update thread annotiation macros in modules to use RTC_ prefix
by danilchap
· 8 years ago
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 8 years ago
890988c
Run the ClangTidy analyser on the AudioProcessing submodule of WebRTC.
by Alex Loiko
· 8 years ago
a9cc40b
Allow an external audio processing module to be used in WebRTC
by peah
· 8 years ago
4bdced5
Corrected the initialization of the AEC3
by Per Åhgren
· 8 years ago
ac9f876
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
by kwiberg
· 9 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 9 years ago
88ac853
The current scheme for setting parameters and specifying the
by peah
· 9 years ago
10f606d
Revert of Introduced new scheme for controlling the functionality inside the audio processing module (patchset #12 id:260001 of https://codereview.webrtc.org/2292863002/ )
by kjellander
· 9 years ago
2ace3f9
The audio processing module (APM) relies on two for
by peah
· 9 years ago
c8bbe3f
The current scheme for setting parameters and specifying the behavior
by peah
· 9 years ago
b031955
Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API
by aluebs
· 9 years ago
5b830fe
Drop the restriction on same forward and reverse sample rate on the AudioFrame interface of the APM
by Alex Luebs
· 9 years ago
ff761fb
modules: more interface -> include renames
by Henrik Kjellander
· 10 years ago
788acd1
Merge audio_processing changes.
by pbos@webrtc.org
· 11 years ago
12acd6e
Reorder includes in audio_processing_impl_unittest.
by andrew@webrtc.org
· 11 years ago
a8b9737
Add tests and modify tools for new float deinterleaved interface.
by andrew@webrtc.org
· 11 years ago
e84978f
Add a Config parameter to AudioProcessing::Create().
by andrew@webrtc.org
· 11 years ago
60730cf
Remove the requirement to call set_sample_rate_hz and friends.
by andrew@webrtc.org
· 12 years ago