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31 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
marpan@webrtc.org
71707aaae8 Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. 
Review URL: https://webrtc-codereview.appspot.com/682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-13 16:27:51 +00:00
stefan@webrtc.org
ddfdfed3b5 Pass capture time (wallclock) to the RTP sender to compute transmission offset 
- Change how the transmission offset is calculated, to
  incorporate the time since the frame was captured.
- Break out RtpRtcpClock and move it to system_wrappers.
- Use RtpRtcpClock to set the capture time in ms in the capture module.
  We must use the same clock as in the RTP module to be able to measure
  the time from capture until transmission.
- Enables the RTP header extension for packet transmission time offsets.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/666006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-07-03 13:21:22 +00:00
mflodman@webrtc.org
f5e99db10b Made cpplint pass for vie_channel.* and vie_encoder.*. NOLINT is used for API changes, include guards and include files in WebRTC root. 
WebRTC types and webrtc:: will be removed in a follow up.

BUG=627
TEST=vie_auto_test + compiles

Review URL: https://webrtc-codereview.appspot.com/677005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-27 09:49:37 +00:00
mflodman@webrtc.org
8baed51f6e This CL is part of enabling cpplint check for video_engine uploads. 
BUG=627
TEST=vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/653006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-06-21 12:11:50 +00:00
pwestin@webrtc.org
2853dde520 Refactor the internal API to the rtp/rtcp module. 
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-11 11:08:54 +00:00
turaj@webrtc.org
3c383abd27 Revert 2211 - Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/563011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 23:01:04 +00:00
pwestin@webrtc.org
0774838f3d Refactor the internal API to the rtp/rtcp module. 
Review URL: https://webrtc-codereview.appspot.com/568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-10 12:33:50 +00:00
pwestin@webrtc.org
49888ce428 Breaking out send side bitrate controll cont. 
Review URL: https://webrtc-codereview.appspot.com/475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:25:53 +00:00
marpan@webrtc.org
efd01fd354 Removing unused code from QMVideoSettingsCallback. 
This code is not needed as of the change in r1950.
Review URL: https://webrtc-codereview.appspot.com/492010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-18 15:56:34 +00:00
stefan@webrtc.org
9784512bf3 Fix coverity warning and possible memory leak. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/493004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-13 07:47:05 +00:00
pwestin@webrtc.org
ce33035dee Cleanup encode call. 
Review URL: https://webrtc-codereview.appspot.com/491003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2011 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-12 06:59:14 +00:00
wu@webrtc.org
5d8c102899 Move those calls that may fail out of the ctor to Init function. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/491002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2003 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-10 16:54:05 +00:00
wu@webrtc.org
96b3017b33 When GetEncoder or SetSendCodec failed, we should clean up and return error. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/490001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-06 19:49:28 +00:00
marpan@webrtc.org
cf706c2b92 Removing the resetting/re-init of encoder from QMVideoSettingsCallback. 
This is not neeeded anymore as a change of frame size (down-sampling via QM callback) will trigger a new key frame in encoder.
Review URL: https://webrtc-codereview.appspot.com/456007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-27 21:04:13 +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
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
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
marpan@webrtc.org
79a99de8e4 Reverting 1680: valgrind memory leak reported. 
TBR=marpan
Review URL: https://webrtc-codereview.appspot.com/392011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 22:37:10 +00:00
marpan@webrtc.org
4e34dcbd62 Allow for spatial-downsampling without reinitializaing encoder. Change of frame size will automatically trigger new key frame in codec. This feature is set off in vie_encoder until we upgrade to the new libvpx. 
Also reset frame rate estimate in mediaOpt when frame rate reduction is decided.
Review URL: https://webrtc-codereview.appspot.com/390006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 17:26:24 +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
mflodman@webrtc.org
cdeb483c6a Fixed ignored return value. 
BUG=C-10011

Review URL: https://webrtc-codereview.appspot.com/353003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 09:00:04 +00:00
mflodman@webrtc.org
d32c44738a Changed constructor used for CriticalSectionScoped in ViE. 
Only changed:
- Name of some of the critsects.
- All critsects (but one) are now scoped_ptr.
- Use of ptr constructor of CriticalSectionScoped instead of reference version.

BUG=184
TEST=vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/330015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-22 14:17:53 +00:00
stefan@webrtc.org
f4c8286222 Pass NACK and FEC overhead rates through the ProtectionCallback to VCM. 
These overhead rates are used by the VCM to compensate the source
coding rate for NACK and FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 15:38:14 +00:00
mflodman@webrtc.org
626fbfd4cd Correcting vie_encoder nits. 
Review URL: http://webrtc-codereview.appspot.com/302004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 23:39:11 +00:00
mflodman@webrtc.org
c6182915a3 Fix vie_encoder.cc. 
TBR=ajm

Review URL: http://webrtc-codereview.appspot.com/301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:45:28 +00:00
mflodman@webrtc.org
84d17838ac Refactored ViEEncoder. 
Style changes + QT Metrics class from h-file to cc-file, type changes will be in another CL.

Review URL: http://webrtc-codereview.appspot.com/303001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:02:23 +00:00
mflodman@webrtc.org
94ea32ef60 Move video_engine/source* to video_engine/. No code changes except paths in gyp-files. 
Review URL: http://webrtc-codereview.appspot.com/283002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 14:49:31 +00:00