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168 Commits

Author SHA1 Message Date
pwestin@webrtc.org
a070adbab2 Moved member RTPSender from private to protected. 
Review URL: http://webrtc-codereview.appspot.com/119006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 11:17:03 +00:00
andrew@webrtc.org
f81f9f8c2a Add -Werror and -Wextra to the Linux build. 
Includes all fixes required for -Wextra.
Review URL: http://webrtc-codereview.appspot.com/117006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 22:56:22 +00:00
hellner@google.com
977c2966fc Review URL: http://webrtc-codereview.appspot.com/109006 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 17:30:30 +00:00
mikhal@google.com
60873adc3e rtp_sender_video: Modify behavior on send video packet error. This issue was already updated in CL r217, and accidentally reverted in CL r231. 
Review URL: http://webrtc-codereview.appspot.com/106004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 22:30:00 +00:00
andrew@webrtc.org
8910f278c5 Switch to webrtc.org accounts (for those which exist). 
Review URL: http://webrtc-codereview.appspot.com/97010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-10 05:16:31 +00:00
xians@google.com
0b0665acc1 This CL changes all the freq relevant variables to be int type. So it will take away the VoE "comparison between signed and unsigned integer expressions" warnings. 
BR,
/SX
Review URL: http://webrtc-codereview.appspot.com/89014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 08:18:44 +00:00
leozwang@google.com
79835d1bd3 Clean up Android.mk 
Review URL: http://webrtc-codereview.appspot.com/92014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 21:01:02 +00:00
leozwang@google.com
d4e72f4ceb Add return value 
Review URL: http://webrtc-codereview.appspot.com/98004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-02 22:13:36 +00:00
marpan@google.com
5fc2dcd64a Change to make the VP8-RTP Fragmentation (FI bits) setting (in the payload header) 
agree with "draft-westin-payload-vp8-02" document.

This issue was raised in: http://code.google.com/p/webrtc/issues/detail?id=31 
Review URL: http://webrtc-codereview.appspot.com/92005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 21:47:46 +00:00
marpan@google.com
1b43b6d416 Changing the default VP8 packetization mode setting to kAggregate and balanced, from the previous settig of kStrict and balanced. 
The previous kStrict mode could generate very small packets when the encoded frame is smaller than MTU size. kAggregate will instead encapsulate whole frame into one packet if frame size is below MTU (and so will not generate too small packets), and otherwise it will separate out the first partition as in kStrict mode.

The balanced setting for kAggregate (from default of un-balanced) is also desirable, as equal size packets (for the first and remaining partition) should generally be more favorable for FEC.
Review URL: http://webrtc-codereview.appspot.com/89002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-21 16:49:54 +00:00
marpan@google.com
ade0c6ca28 Fix for numberFirstPartition setting: occurs when whole frame is packetized into one packet (0 was set instead of 1). 
Review URL: http://webrtc-codereview.appspot.com/88003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-20 20:54:55 +00:00
hlundin@google.com
7d3a2a3bca Set _numberFirstPartition when packetizing VP8 frames 
The variable _numberFirstPartition is now set in RTPSenderVideo::SendVP8.
The number of packets that contains data from the first partition
is not known until all packets have been packetized (at least all
first-partition packets). Therefore, the packetization loop in SendVP8
had to be broken up into two loops. The first loop gets all packets from
the VP8 packetizer (RtpFormatVp8) and puts them in a vector. The second
loop sends all packets from the vector to SendVideoPacket.
Review URL: http://webrtc-codereview.appspot.com/56004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-18 22:34:17 +00:00
marpan@google.com
80c5d7a80e Allow the setting of FEC-UEP feature on/off to be done in media_opt(VCM). 
Review URL: http://webrtc-codereview.appspot.com/71004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-15 21:32:40 +00:00
mikhal@google.com
b7540b0322 RTP: Changing the behavior in case of a send video packet error 
Review URL: http://webrtc-codereview.appspot.com/74005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-15 21:01:08 +00:00
hellner@google.com
1b627c72b5 Tests using the rtp_rtcp test data should now be run from inside trunk/test/data/rtp_rtcp. I.e. all test files were moved to the test folder. 
Review URL: http://webrtc-codereview.appspot.com/60006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-08 17:16:47 +00:00
niklase@google.com
0c3e855793 git-svn-id: http://webrtc.googlecode.com/svn/trunk@172 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 09:40:48 +00:00
niklase@google.com
9ad0cf1ae2 git-svn-id: http://webrtc.googlecode.com/svn/trunk@164 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:43:35 +00:00
niklase@google.com
470e71d364 git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00