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gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
b7e5b2741f9cbd68e7be54e3445d28d266477c92
/
voice_engine
/
transmit_mixer.h
7f5e297
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
708ff4d
Resampler modifications in preparation for arbitrary audioproc rates.
by andrew@webrtc.org
· 10 years ago
f7c73b5
Consolidate audio conversion from Channel and TransmitMixer.
by andrew@webrtc.org
· 10 years ago
22c954a
Add an interface for accepting keypress signals to AudioProcessing.
by andrew@webrtc.org
· 10 years ago
c13a537
Move out typing detection to its own class.
by henrikg@webrtc.org
· 10 years ago
22470b5
Minor voice engine improvements around AGC.
by andrew@webrtc.org
· 11 years ago
1a6b274
Fix the include guard in transmit_mixer.h
by braveyao@webrtc.org
· 11 years ago
5d7992f
Fix the include guard in transmit_mixer.h
by braveyao@webrtc.org
· 11 years ago
39e22a1
Adds a new voice engine warning for the typing noise off state.
by jiayl@webrtc.org
· 11 years ago
0e6fa8c
Merge r4394 from stable to trunk.
by xians@webrtc.org
· 11 years ago
44f1239
Merge r4326 from stable to trunk.
by xians@webrtc.org
· 11 years ago
3b89e10
Proper spacing for end-of-namespace comments.
by pbos@webrtc.org
· 11 years ago
ca7a9a2
Remove const for plain data types in voice_engine/
by pbos@webrtc.org
· 11 years ago
28832e1
Refactoring for typing detection
by niklas.enbom@webrtc.org
· 11 years ago
4a68e95
Replace Resampler with PushResampler in transmit_mixer.
by andrew@webrtc.org
· 11 years ago
54f03bc
WebRtc_Word32 -> int32_t in voice_engine/
by pbos@webrtc.org
· 11 years ago
4de0a10
Don't upsample the capture signal early.
by andrew@webrtc.org
· 11 years ago
b563e5e
Properly error check calls to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
b015cbe
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago