Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
73e1a8bb741e8e3956815f75b4977c99f0bba0db
/
engine_configurations.h
290c5a5
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 10 years ago
ae50521
Remove external encryption API for VoE.
by solenberg@webrtc.org
· 10 years ago
ddbd31e
Remove ViE external encryption API.
by solenberg@webrtc.org
· 10 years ago
e30fde1
Adds a new voice engine warning for the typing noise off state.
by jiayl@webrtc.org
· 11 years ago
a5b7b8c
Make PCM16 available in Chromium builds.
by andrew@webrtc.org
· 11 years ago
f4e4324
Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.
by andrew@webrtc.org
· 11 years ago
47e4f00
Remove WEBRTC_*_ENGINE_NETWORK_API use
by pwestin@webrtc.org
· 11 years ago
1ca9d42
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
by solenberg@webrtc.org
· 11 years ago
9a7b9f7
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004
by pwestin@webrtc.org
· 11 years ago
66ccc6e
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
by pwestin@webrtc.org
· 11 years ago
0cf911a
First pass of MediaCodecDecoder which uses Android MediaCodec API.
by dwkang@webrtc.org
· 12 years ago
a7b57da
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago