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34 Commits

Author SHA1 Message Date
pwestin@webrtc.org
cac787842c New attempt to cleanup TMMBR. 
Review URL: https://webrtc-codereview.appspot.com/472007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-05 08:30:10 +00:00
henrike@webrtc.org
0ad51862dc Revert 1961 - Clean up TMMBR handling. 
Review URL: https://webrtc-codereview.appspot.com/465001

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/473001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 16:54:13 +00:00
pwestin@webrtc.org
20f4440c73 Clean up TMMBR handling. 
Review URL: https://webrtc-codereview.appspot.com/465001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-30 11:40:15 +00:00
mflodman@webrtc.org
534a435751 Removed RTP Keepalive from RTP module. 
Review URL: https://webrtc-codereview.appspot.com/455007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-27 06:57:41 +00:00
stefan@webrtc.org
e0d6fa4c66 Adding classes for handling multi-frame FEC. 
The FEC behavior is unchanged with this commit, we will still be
limited to FEC over one frame for now.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/450006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-20 22:10:56 +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
f7b6078f6f Allow multiple send channels for REMB. Current implementation splits the remote estimate evenly between all senders. 
This CL will be followed by a CL adding support for several REMB groups.

TEST=video_engine_core_unittests

Review URL: https://webrtc-codereview.appspot.com/394002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 14:50:24 +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
henrike@webrtc.org
f5da4da409 Removes a global non POD instance from the RTP_RTCP module that was introduced in https://code.google.com/p/webrtc/source/detail?r=1076. 
Review URL: https://webrtc-codereview.appspot.com/314001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-15 23:54:59 +00:00
pwestin@webrtc.org
5e954814a8 Clanup handling of key frame requests and FIR. 
Review URL: https://webrtc-codereview.appspot.com/387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:13:12 +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
henrike@webrtc.org
567b99be5f Coverity report: fixes an issue where the returnvalue of a function is not checked. 
Review URL: https://webrtc-codereview.appspot.com/347013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 23:43:54 +00:00
pwestin@webrtc.org
f6bb77a6f0 Cleaning up all use of RTP_PAYLOAD_NAME_SIZE and RTCP_CNAME_SIZE also fixed the char handing in trace. 
Review URL: https://webrtc-codereview.appspot.com/358001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 17:16:59 +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
asapersson@webrtc.org
0b3c35a258 Review URL: http://webrtc-codereview.appspot.com/321011 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 11:06:31 +00:00
perkj@webrtc.org
ce5990cb0b Fix defect http://code.google.com/p/webrtc/issues/detail?id=222 
"ViE GetSentRTCPStatistics fails on a sending channel if it don't receive rtp video packets.

BUG=222
TEST= tested in loopback. No new test added yet.

Review URL: http://webrtc-codereview.appspot.com/343003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 13:00:08 +00:00
pwestin@webrtc.org
8281e7dd4a Added RTX to ViE. 
Review URL: http://webrtc-codereview.appspot.com/336001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 14:09:18 +00:00
pwestin@webrtc.org
3aa25de346 Bugfix OnNetworkChanged not triggered for RTCP compund messages if TMMBR is higher than last value. 
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/342001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-05 08:40:56 +00:00
pwestin@webrtc.org
c450a19669 Removed Version function from all modules. 
TBR=henrik_a
Review URL: http://webrtc-codereview.appspot.com/329023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 15:00:12 +00:00
stefan@webrtc.org
6a4bef4e65 Implements selective retransmissions. 
Default is set to not retransmit VP8 non-base layer packets or FEC packets.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 12:52:41 +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
asapersson@webrtc.org
5249cc8f77 Review URL: http://webrtc-codereview.appspot.com/295010 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 14:31:37 +00:00
henrike@webrtc.org
65573f2922 Removed usage of the deprecated critical section constructor in rtp_rtcp. 
Review URL: http://webrtc-codereview.appspot.com/315004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 19:17:27 +00:00
pwestin@webrtc.org
0644b1dc35 Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions 
A new RtpRtcpClock interface has been added to rtp_rtcp_defines.h
and provides time facilities used by an RTP/RTCP module. Also,
NTP constants have been made public in the
webrtc::ModuleRTPUtility namespace to make implementation of
external clocks easier.

An overloaded version of CreateRtpRtcp() accepts a clock argument. By
default, if no clock is provided, the module uses the system clock
(old ModuleRTPUtility implementation).

Throughout the RTP/RTCP module code, calls to TickTime and
ModuleRTPUtility time functions have been replaced with calls to time
methods on a clock object.

The following classes take a clock object in their constructor and
hold a _clock field (either directly, or inherited from a parent):

Bitrate
ModuleRtpRtcpImpl
RTCPReceiver
RTCPSender
RTPReceiver
RTPSender
RTPSenderAudio
RTPSenderVideo

Methods from other classes that do not derive any of those and
require a time take an additional nowMS parameter, that should be
the result of calling GetTimeInMS() on a clock object.
Review URL: http://webrtc-codereview.appspot.com/268017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:42:31 +00:00
mflodman@webrtc.org
26b9777e62 Only trigger one call to OnNetworkChanged for each incoming RTCP packet. 
Review URL: http://webrtc-codereview.appspot.com/289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 15:22:33 +00:00
mflodman@webrtc.org
7a4eb2837a Calculate the available bandwidth before sending a TMMBR 
Also changed the way TMMBR was processed since it did not match the new bandwidth estimator.

Review URL: http://webrtc-codereview.appspot.com/270003
Patch from pwestin1 <pwestin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 12:54:46 +00:00
stefan@webrtc.org
fbea4e555d Solves two bandwidth estimation issues and measures the sent video bitrate. 
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
   we reduced the rate relative the current estimate and not the actual
   rate sent.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/244011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:08:29 +00:00
stefan@webrtc.org
d0bdab0128 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. 
Also adding tests for this in vie_auto_test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/199001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 14:24:54 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine. 
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 15:19:55 +00:00
pwestin@webrtc.org
741da942ec Added support for new RTCP message REMB (remote estimated max bitrate) 
Review URL: http://webrtc-codereview.appspot.com/149001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 13:52:04 +00:00
pwestin@webrtc.org
e9f0e2eb20 Moved _rtpReceiver to protected 
Review URL: http://webrtc-codereview.appspot.com/132005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 13:16:52 +00:00
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
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
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