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Author SHA1 Message Date
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
b73c3d1f5d Bugfix android build. 
Review URL: https://webrtc-codereview.appspot.com/374003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 15:25:30 +00:00
pwestin@webrtc.org
28a5cb29ab Bugfix receive side only packet loss estimate with NACK. 
Review URL: https://webrtc-codereview.appspot.com/373006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1529 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 14:34:58 +00:00
punyabrata@webrtc.org
6da8eeb946 Removing an assert for a case that can occur 
when corrupt packets are injected into voice engine.
Review URL: https://webrtc-codereview.appspot.com/373004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 00:48:36 +00:00
leozwang@webrtc.org
376be6c904 Fix compilation error 
Review URL: https://webrtc-codereview.appspot.com/358005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:46:38 +00:00
pwestin@webrtc.org
b30f0edce6 Bugfix buffer usage out of scope. 
Review URL: https://webrtc-codereview.appspot.com/372001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:23:31 +00:00
pwestin@webrtc.org
95cf47932d Remove list wrapper from FEC code. 
Review URL: https://webrtc-codereview.appspot.com/350013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 06:59:06 +00:00
pwestin@webrtc.org
0074187436 Removed map_wrapper from rtp_sender 
Review URL: https://webrtc-codereview.appspot.com/343014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:56:10 +00:00
pwestin@webrtc.org
3c9be1bc4d Removed list wrapper fromr overuse detector. 
Review URL: https://webrtc-codereview.appspot.com/353004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:55:54 +00:00
pwestin@webrtc.org
d4adc5b26f removed unused include from remote rate control. 
Review URL: https://webrtc-codereview.appspot.com/350015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:55:30 +00:00
pwestin@webrtc.org
af6f15c1bf Changed RTP reveivers to use stl map and list. 
Review URL: https://webrtc-codereview.appspot.com/349010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:53:59 +00:00
pwestin@webrtc.org
38f4816737 Removed unused include from rtp sender audio. 
Review URL: https://webrtc-codereview.appspot.com/348012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:53:33 +00:00
pwestin@webrtc.org
26f8d9c7f3 Removed list and map wrappers, for RTCP handling. 
Review URL: https://webrtc-codereview.appspot.com/349011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:53:09 +00:00
pwestin@webrtc.org
1da2327473 Changing header extension to use stl map. 
Review URL: https://webrtc-codereview.appspot.com/350014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 12:58:53 +00:00
andrew@webrtc.org
267ca3162b Fix comparison-always-true warning with -Wextra. 
TEST=build on Linux with -Wextra.

Review URL: https://webrtc-codereview.appspot.com/353005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 19:41:40 +00:00
henrik.lundin@webrtc.org
4407edc314 Bugfix in VP8 packetizer 
Handle the case with no small partitions in Vp8PartitionAggregator.
Also added a new unit test for the packetizer to verify that the
bug is fixed.

TEST=RtpFormatVp8Test.TestAggregateModeTwoLargePartitions
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/348011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 10:01:03 +00:00
henrik.lundin@webrtc.org
7f2c2a5db2 Adding optimized aggrgation to VP8 packetizer 
This change introduces a new algorithm for aggregating small
partitions into packets. The algorithm is based on a tree-search
to find an optimal allocation of the packets, such that the
difference in size between packets is minimized.

The new method is used when partition aggregation is allowed and
balanced packets are requested. Otherwise, the old method is used.

The new method is implemented using the new classes
Vp8PartitionAggregator and PartitionTreeNode. Both classes have
dedicated unit tests.

In order to facilitate the new algorithm, the packetizer was
redesigned to calculate all packet sizes when the first packet is
extracted. The information about all packets is stored in a packet
queue structure, which is then popped for each packet extracted.

Finally, a bug in the old packetizer algorithm was fixed. The bug
caused a +/-1 error in packet sizes when balanced packets were
produced. The unit test were updated accordingly.

TEST=rtp_rtcp_unittests: PartitionTreeNode.* Vp8PartitionAggregator.* RtpFormatVp8Test.*

Review URL: https://webrtc-codereview.appspot.com/345008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 08:21:15 +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
pwestin@webrtc.org
df9bd9bdbd Removed dead code. 
Review URL: https://webrtc-codereview.appspot.com/352010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 11:42:02 +00:00
pwestin@webrtc.org
aafa5a331c Coverty report: Unititialized members 
Review URL: http://webrtc-codereview.appspot.com/349007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 07:07:37 +00:00
asapersson@webrtc.org
43b8fc5c0d Review URL: http://webrtc-codereview.appspot.com/345011 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 13:49:04 +00:00
asapersson@webrtc.org
869ce2d441 Review URL: http://webrtc-codereview.appspot.com/353002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 11:58:36 +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
mflodman@webrtc.org
117c119501 Only update REMB value if there is a calid bitrate estimate. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/352005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 08:52:58 +00:00
stefan@webrtc.org
c8277db7c8 Fix selective retransmissions after corrupt merge in r1373. 
BUG=228
TEST=

Review URL: http://webrtc-codereview.appspot.com/345006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 15:38:50 +00:00
mflodman@webrtc.org
80d60420ff RTCPSender::_bitrate_observer not initialized. 
BUG=227
TEST=Valgrind

Review URL: http://webrtc-codereview.appspot.com/352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 14:28:53 +00:00
niklas.enbom@webrtc.org
553657b2f8 See http://codereview.chromium.org/9188022/ for details 
Review URL: http://webrtc-codereview.appspot.com/347009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 08:49:34 +00:00
mflodman@webrtc.org
04c18cb37a Update all child modules of with received bandwidth estimate. 
BUG=224

Review URL: http://webrtc-codereview.appspot.com/347007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 14:18:33 +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
stefan@webrtc.org
c5b73e3974 Further cleanup of OverUseDetector. Removed member no longer used. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/344007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 16:42:09 +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
asapersson@webrtc.org
c5a1cee73e Review URL: http://webrtc-codereview.appspot.com/348004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 12:54:44 +00:00
stefan@webrtc.org
727e1611ac Removes debug file writing. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/343006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 11:45:47 +00:00
stefan@webrtc.org
e21a8cf4d4 Fix issue 211: Make sure we always generate at least one FEC packet per frame if we need protection. 
BUG=211
TEST=

Review URL: http://webrtc-codereview.appspot.com/348002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 08:37:33 +00:00
pwestin@webrtc.org
12d97f6637 Made send pad data generic (audio and video) 
Review URL: http://webrtc-codereview.appspot.com/343001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-05 10:54:44 +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
6c1d41583a Fix for RTP extension audio level. 
Review URL: http://webrtc-codereview.appspot.com/339002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 17:04:51 +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
f6c6b1c5b5 Include the media packet FEC headers in the video bitrate. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/328014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-23 10:33:39 +00:00
mflodman@webrtc.org
1ce66e4dfb Don't report error when failing to send RTCP BYE. 
Review URL: http://webrtc-codereview.appspot.com/337002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 18:40:15 +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
henrik.lundin@webrtc.org
1e28d3c2e1 Change VP8 packetizer to use a single max payload size 
The packetizer class is changed so that the max payload size is
provided on construction of the class rather than for each packet.
The tests are re-written to comply with the new design.

Also fixing a few errors in the tests.

Review URL: http://webrtc-codereview.appspot.com/335010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 08:49:31 +00:00
pwestin@webrtc.org
8edb39db30 Prevent sending empty RTCP packet. 
Review URL: http://webrtc-codereview.appspot.com/331009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 07:40:33 +00:00
niklas.enbom@webrtc.org
6c9be123ef Letting strncpy do its job. Landing and extending http://webrtc-codereview.appspot.com/329010/ on behalf of tbreisacher. 
Review URL: http://webrtc-codereview.appspot.com/335009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:59:31 +00:00
henrik.lundin@webrtc.org
00e730730e Refactoring RtpFormatVp8Test 
This is the first change in a series of changes to get new functionality
into the VP8 packetizer.

This first refactors the RtpFormatVp8Test class, without changing the
operation of the tested RtpFormatVp8 class. A test helper class
RtpFormatVp8TestHelper is introduced to reduce code duplication.

Review URL: http://webrtc-codereview.appspot.com/304009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:51:36 +00:00
pwestin@webrtc.org
061fa5b828 Changed handling of padding data. 
Review URL: http://webrtc-codereview.appspot.com/331008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:56:17 +00:00
mikhal@webrtc.org
0e7d9d862a Adding layer info consideration when applying FEC protection. In this first version, we hard code protection zero for non-base layer frames. As a future enhancement, an array should be passed from mediaOpt to set the protection per layer. A TODO was added in MediaOpt. 
Review URL: http://webrtc-codereview.appspot.com/330005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 19:04:49 +00:00
henrika@webrtc.org
e32c08a5a6 Removes usage of default parameters and fixes a bug which was found 
using Clang on Linux.

BUG=none
TEST=none
TBR=pwestin

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:39:48 +00:00
andrew@webrtc.org
8a44259ea8 Move static consts out of class. 
Still causing a gtest error on non-Win platforms. This should fix it...

TBR=asapersson@webrtc.org
TEST=build on Linux

Review URL: http://webrtc-codereview.appspot.com/332006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 21:24:30 +00:00