Commit Graph

9 Commits

Author SHA1 Message Date
bf00740c92 Adds a new voice engine warning for the typing noise off state.
The old VE_TYPING_NOISE_WARNING is unchanged and fired whenever typing noise is detected.
The new VE_TYPING_NOISE_OFF_WARNING is fired when typing noise was detected and is gone now.
This is necessary for converting the typing state to a PeerConnection stats.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4770 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17 18:09:20 +00:00
8fa03a15ab Make PCM16 available in Chromium builds.
PCM16 can be useful for unit tests in Chromium. In particular Mikhal
would like to use it for ChromeCast.

This currently (r222592) has no impact on Chrome binary size, presumably
because PCM16 is unused and the linker strips the symbols.

To measure the potential impact, I looked at the size (bytes) of
out/Release/vie_auto_test on Linux with various codecs removed:
r4724    : 4567384
No PCM16 : 4565936
No ILBC  : 4500424
No G722  : 4555800
No RED   : 4565880

Giving the following size increases of adding each codec:
PCM16 :  1.4 kB (0.03%)
ILBC  : 70.0 kB (1.49%)
G722  : 11.6 kB (0.25%)
RED   :  1.5 kB (0.03%)

R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2195005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4732 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:30:30 +00:00
0851df8d60 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls
where it actually is supported.
* No error to call GetTypingDetectionStatus.
* Consolidate typing detection disablement to reduce boilerplate.

R=niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1683004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-19 17:03:47 +00:00
b9e402d99f Remove WEBRTC_*_ENGINE_NETWORK_API use
Review URL: https://webrtc-codereview.appspot.com/1203009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-04 19:51:42 +00:00
a442d4d983 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.
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-28 09:14:36 +00:00
684f0577fb Revert r3667 and r3665
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 23:20:57 +00:00
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 17:52:42 +00:00
6bd737a714 First pass of MediaCodecDecoder which uses Android MediaCodec API.
Background:
As of now, MediaCodec API is the only public interface which enables us
to access low level HW resource in Android. ViEMediaCodecDecoder will be
used for further experiments/exploration.

TODO:
  To fix known issues. (detaching thread from VM and frequent GC)
Review URL: https://webrtc-codereview.appspot.com/933033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3233 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 06:38:19 +00:00
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00