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41 Commits

Author SHA1 Message Date
stefan@webrtc.org
5eb64f06be Fix BitrateSent() API when having a default RTP module. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 13:42:50 +00:00
stefan@webrtc.org
c4d1983b7b Changes in rtp_format_vp8_unittest to match the changes in CL 774. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/241006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 08:19:34 +00:00
henrike@webrtc.org
509c9c5d09 operator + is evaluated before ?: 
Parenthesis ensures the intended behavior.
Review URL: http://webrtc-codereview.appspot.com/239003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 18:31:01 +00:00
stefan@webrtc.org
ffd28f95c5 Request key frames to battle error propagation. 
The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).

For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 15:55:39 +00:00
stefan@webrtc.org
5b15cfc6dd Fix BWE unit test build issue 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 07:22:33 +00:00
wu@webrtc.org
76aea651ff When _audioConfigured, should not try to use the _video. 
Review URL: http://webrtc-codereview.appspot.com/224004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@758 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 21:40:32 +00:00
marpan@webrtc.org
14aaaf116a Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming. 
Review URL: http://webrtc-codereview.appspot.com/231001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 16:28:02 +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
marpan@webrtc.org
5a3e20f678 Removed unused variables (build error) for test_fec. 
Review URL: http://webrtc-codereview.appspot.com/223001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 16:59:24 +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
henrike@webrtc.org
bf54ef9bb7 Removed code under a non-existing define. 
Review URL: http://webrtc-codereview.appspot.com/193006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 18:14:25 +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
henrik.lundin@webrtc.org
4c36d3b424 Fixing windows warnings in rtp_utility 
Adding explicit casting to bool to avoid warnings when compiling
in windows.

Review URL: http://webrtc-codereview.appspot.com/140002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 08:16:20 +00:00
andrew@webrtc.org
2cef36fa98 Fix Windows gyp run. 
On Windows, gyp seems to require valid source files. The matlab_plotting_test target was missing its one source file, so I removed the target.

Also moving bwe_standalone.gypi to the test include list.
Review URL: http://webrtc-codereview.appspot.com/143001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:03:15 +00:00
xians@google.com
d3185fe219 refactor the gyp file to gypi file. 
Basically, the gypi file is a copy of gyp file, but has some difference on the
path of the dependencies.
Review URL: http://webrtc-codereview.appspot.com/137020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 12:24:39 +00:00
stefan@webrtc.org
9e812fca9f Adding missing parts related to VP8 partitions 
Review URL: http://webrtc-codereview.appspot.com/131017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 10:11:24 +00:00
stefan@webrtc.org
269f8a14c6 Undoing change committed in r514 since it broke bandwidth estimation 
Review URL: http://webrtc-codereview.appspot.com/132011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@531 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 09:51:59 +00:00
andrew@webrtc.org
4d905f88c6 Fix clang warnings in rtp. 
Review URL: http://webrtc-codereview.appspot.com/132006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 19:22:27 +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
henrik.lundin@webrtc.org
8571af7be6 Updating to new VP8 rtp format 
The VP8 packetizer and tests have been updated to the new
RTP draft (http://tools.ietf.org/html/draft-ietf-payload-vp8-01).
The receive-side parser is also updated, and a new unit test
is implemented for it. Finally, some data traversing work to
get the parsed information into the decoder.
Review URL: http://webrtc-codereview.appspot.com/116011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 15:37:12 +00:00
andrew@webrtc.org
4f390000dd Fix warnings on Ubuntu 11.04 (gcc 4.5) 
http://code.google.com/p/webrtc/issues/detail?id=63
Review URL: http://webrtc-codereview.appspot.com/125004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 20:35:35 +00:00
hellner@google.com
a386fc0a8b Fixes build warnings due to unused variables. 
Code directly from http://code.google.com/p/webrtc/issues/detail?id=58.
Review URL: http://webrtc-codereview.appspot.com/119007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 21:26:09 +00:00
perkj@google.com
12f1fc4fe5 Fix initialization defect in constructor webrtc::ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(WebRtc_Word32, bool) initialization list. 
Review URL: http://webrtc-codereview.appspot.com/125002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 14:26:33 +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
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