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path: root/engine_configurations.h
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2014-11-01Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
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/webrtc@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29Enable G.722 for Chromium buildshenrik.lundin@webrtc.org
BUG=3909 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7555 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."henrike@webrtc.org
BUG=3932 R=marpan@google.com Review URL: https://webrtc-codereview.appspot.com/27779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
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/webrtc@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04Add boilerplate code for H.264.stefan@webrtc.org
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/webrtc@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up ↵henrika@webrtc.org
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/webrtc@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14Removes VoECallReport sub API as part of a clean-up operation where the goal ↵henrika@webrtc.org
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/webrtc@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18Remove external encryption API for VoE.solenberg@webrtc.org
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/webrtc@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11Remove ViE external encryption API.solenberg@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8079005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17Adds a new voice engine warning for the typing noise off state.jiayl@webrtc.org
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/webrtc@4770 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12Make PCM16 available in Chromium builds.andrew@webrtc.org
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/webrtc@4732 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-19Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.andrew@webrtc.org
* 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/webrtc@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-04Remove WEBRTC_*_ENGINE_NETWORK_API usepwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1203009 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-28Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE ↵solenberg@webrtc.org
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/webrtc@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13Revert r3667 and r3665pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13Removed the engine API:s related to transport such as SetSendDestination, ↵pwestin@webrtc.org
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/webrtc@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04First pass of MediaCodecDecoder which uses Android MediaCodec API.dwkang@webrtc.org
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/webrtc@3233 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22Move src/ -> webrtc/andrew@webrtc.org
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
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