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18 Commits

Author SHA1 Message Date
marpan@webrtc.org
5b88317820 Add VP9 codec to VCM and vie_auto_test. 
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.

This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-01 06:10:48 +00:00
henrik.lundin@webrtc.org
cfe3845b66 Enable G.722 for Chromium builds 
BUG=3909
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:32:44 +00:00
henrike@webrtc.org
b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 
BUG=3932
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/27779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 18:54:46 +00:00
marpan@webrtc.org
573c78e31c Add VP9 codec to VCM and vie_auto_test. 
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 16:44:47 +00:00
stefan@webrtc.org
b9f5453e29 Add boilerplate code for H.264. 
R=mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 12:42:07 +00:00
henrika@webrtc.org
1cec3957b8 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:19:19 +00:00
henrika@webrtc.org
b9309beea4 Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 14:12:50 +00:00
solenberg@webrtc.org
a07923339b Remove external encryption API for VoE. 
BUG=
R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 11:27:22 +00:00
solenberg@webrtc.org
fc320466d1 Remove ViE external encryption API. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 15:27:49 +00:00
jiayl@webrtc.org
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
andrew@webrtc.org
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
andrew@webrtc.org
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
pwestin@webrtc.org
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
solenberg@webrtc.org
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
pwestin@webrtc.org
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
pwestin@webrtc.org
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
dwkang@webrtc.org
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
andrew@webrtc.org
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