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gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
11e96c736207147b6c79963073f89112a8dbefb4
/
voice_engine
/
transmit_mixer.cc
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
c1878ac
Only clamp to 16 kHz when AECM is enabled.
by andrew@webrtc.org
· 10 years ago
ba47616
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 10 years ago
692224a
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
f7c73b5
Consolidate audio conversion from Channel and TransmitMixer.
by andrew@webrtc.org
· 10 years ago
120c725
Extends max sample rate from 96kHz to 192kHz on the input side.
by henrika@webrtc.org
· 10 years ago
df08c5d
Resolves TSan v2 warnings in voe_auto_test.
by henrika@webrtc.org
· 10 years ago
22c954a
Add an interface for accepting keypress signals to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
c13a537
Move out typing detection to its own class.
by henrikg@webrtc.org
· 11 years ago
22470b5
Minor voice engine improvements around AGC.
by andrew@webrtc.org
· 11 years ago
e95dc25
Remove the requirement to call set_sample_rate_hz and friends.
by andrew@webrtc.org
· 11 years ago
e06943f
Clean up AudioProcessing defaults and errors.
by andrew@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
b3ada15
Ref-counted rewrite of ChannelManager.
by pbos@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
45ce6a8
TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC
by henrika@webrtc.org
· 11 years ago
4de0a10
Don't upsample the capture signal early.
by andrew@webrtc.org
· 12 years ago
b563e5e
Properly error check calls to AudioProcessing.
by andrew@webrtc.org
· 12 years ago
d468236
Replace AudioFrame's operator= with CopyFrom().
by andrew@webrtc.org
· 12 years ago
422ea9d
Use %d for signed value in trace.
by andrew@webrtc.org
· 12 years ago
e5ac24f
Move capture level computation after all processing.
by andrew@webrtc.org
· 12 years ago
b015cbe
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago