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18 Commits

Author SHA1 Message Date
mikhal@webrtc.org
e41bbdfecc Adding an API that allows recording of video data 
removing vie_codec from cl

Moving debug call from Codec to File impl.

Updating cl following review

Updating file name

Updating cl following review.

Updating CL following review.

Adding an API that allows recording of video data

updating cl

Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 16:15:16 +00:00
henrike@webrtc.org
f7884f9900 Revert 2660 - updating cl 
Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751005

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/752007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-25 02:00:19 +00:00
mikhal@webrtc.org
6a6121c0b1 updating cl 
Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-24 22:08:25 +00:00
mflodman@webrtc.org
90071dd647 Added API to set RTP timestamp offset extension. 
BUG=745

Review URL: https://webrtc-codereview.appspot.com/710011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-13 17:13:27 +00:00
astor@webrtc.org
c0496e66f6 Expose a function for setting bandwidth estimation parameters in ViERTP_RTCP. 
Review URL: https://webrtc-codereview.appspot.com/678007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 10:14:43 +00:00
mflodman@webrtc.org
3e820e5109 Remove RTP Keep-alive from VoE and ViE. The RTP module functionality will be removed in a follow-up CL shortly. 
TEST=VoE autotest and ViE autotest

Review URL: https://webrtc-codereview.appspot.com/458002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-23 09:41:44 +00:00
leozwang@webrtc.org
0975d2158c Cleanup messy data type of unknown_payload_type 
BUG=322
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/430002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 20:59:13 +00:00
mflodman@webrtc.org
9ec883e8bd Allow multiple REMB groups and introduce receive channels. 
BUG=312
TEST=ViE standard autotest and API test.

Review URL: https://webrtc-codereview.appspot.com/432005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 17:12:41 +00:00
wu@webrtc.org
69f8be3875 Change the ExternalRenderer to provide both rtp timestamp and the render time. 
Review URL: https://webrtc-codereview.appspot.com/394006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 18:32:02 +00:00
stefan@webrtc.org
439be29445 Add APIs for getting receive-side estimated bandwidth and codec target rate. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 14:45:37 +00:00
mflodman@webrtc.org
8224e19dd9 Fixed incorrect packet loss reported to encoder. 
BUG=275

Review URL: https://webrtc-codereview.appspot.com/394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:41:57 +00:00
stefan@webrtc.org
07b45a5dc4 Added API for getting the send-side estimated bandwidth. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/372006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-02 08:37:48 +00:00
mflodman@webrtc.org
6cf529d082 Changed REMB return value to int instead of bool. 
Review URL: https://webrtc-codereview.appspot.com/366001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 06:16:16 +00:00
pwestin@webrtc.org
5621057956 Removing unused code. 
Review URL: https://webrtc-codereview.appspot.com/349008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 12:45:47 +00:00
mflodman@webrtc.org
e5297d2aaa Big parameter passed as argument. 
BUG=C-10503, C-10504, C-10505

Review URL: https://webrtc-codereview.appspot.com/343011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 12:44:41 +00:00
mflodman@webrtc.org
d5a4d9bce6 First refactoring of ViE interface. 
Review URL: http://webrtc-codereview.appspot.com/337003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-02 13:04:05 +00:00
mflodman@webrtc.org
84dc3d134d Add REMB functionality to ViE. 
This CL only adds support for encoding one stream, but receiving multiple streams.

BUG=
TEST=video_engine_core_unittest + auto_test/loopback

Review URL: http://webrtc-codereview.appspot.com/333011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 10:26:13 +00:00
mflodman@webrtc.org
a4863dbdf0 Moved video_engine/main/interface to video_engine/include. 
Only changed include paths in files, gyp-files and Android.mk.

TEST=vie_auto_test and peerconnection_client builds.

Review URL: http://webrtc-codereview.appspot.com/330017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:51:52 +00:00