29 Commits

Author SHA1 Message Date
pwestin@webrtc.org
5c3a400fae Re-added ChangeUniqueId temporary for chrome.
Review URL: https://webrtc-codereview.appspot.com/594004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2287 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-24 09:52:19 +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
andrew@webrtc.org
e59a0aca6a Fix AudioFrame types.
volume_ is not set anywhere so I'm removing it.

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2196 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-08 17:12:40 +00:00
andrew@webrtc.org
63a509858d Rename AudioFrame members.
BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/542005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2164 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-02 23:56:37 +00:00
stefan@webrtc.org
af5ffd5bb9 Fixes for coverity warnings.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/461001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1933 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-23 16:01:15 +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
marpan@webrtc.org
9d76b4ea54 Updates for resolution adaptation:
1) code cleanup and some updates to selection logic for qm_select.
2) added unit test for the QmResolution class.
3) update codec frame size and reset/update frame rate in media-opt:
4) removed unused motion vector metrics and some related code of content metrics processing.
Review URL: https://webrtc-codereview.appspot.com/405008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1791 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-28 23:39:31 +00:00
marpan@webrtc.org
bd5648f2db Reverting 1718: failed linux video test.
TBR=stefan, andrew, marpan.
Review URL: https://webrtc-codereview.appspot.com/392018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1719 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 23:16:58 +00:00
marpan@webrtc.org
883e716304 Removed unused motion vector metrics from VideoContentMetrics;
also removed other related unused variables and code. 

Reset frame rate estimate in mediaOpt when frame rate reduction is decided.

Update content_metrics with frame rate and qm_resolution with frame size.
Review URL: https://webrtc-codereview.appspot.com/395007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1718 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 18:35:23 +00:00
andrew@webrtc.org
dde977ec83 AudioFrame payload shouldn't be mutable.
This requires making Mute() non-const, which is correct anyway.

BUG=
TEST=voe_auto_test on Linux

Review URL: https://webrtc-codereview.appspot.com/376001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1599 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-03 17:47:32 +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
kjellander@webrtc.org
5e1625ed2d Fixing Valgrind problem detected by video_processing_unittests.
Simple initialization of the allocated memory for the image buffer avoids reading uninitialized data in some special cases.

This fix is only intended for Linux, since the test is known to fail on Windows. But since we're currently only running Valgrind on Linux, this will give us improved control over memory issues.

BUG=
TEST=tools/valgrind-webrtc/webrtc_tests.sh -t cmdline out/Debug/video_processing_unittests

Review URL: http://webrtc-codereview.appspot.com/349012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1493 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-20 08:40:55 +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
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
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
henrik.lundin@webrtc.org
eda86dc76b Adding a LayerSync bit to VP8 RTP header
Updated RtpFormatVp8, ModuleRTPUtility, VP8Encoder and VP8Decoder
to support a new LayerSync ("Y") bit. Note, in VP8Encoder the bit
must be used together with a non-negative value for temporalIdx.
Fixing the plumbing between RTP module and and from VP8 wrapper.
Updating unit tests; all pass.

The new bit is yet to be used by the VP8 wrapper.

Review URL: http://webrtc-codereview.appspot.com/323008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1169 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-13 14:11:06 +00:00
henrik.lundin@webrtc.org
6f2c0168f0 Updating to VP8 RTP spec rev -02
Updating the VP8 packetizer class (RtpFormatVp8) and VP8 parser
(in class RTPPayloadParser) to follow the -02 revision of the spec.
See http://tools.ietf.org/html/draft-ietf-payload-vp8-02.

Updating the unit tests, too. Finally, updating the tests to
follow the recommendations from the test team; specifically
including the test code in the webrtc namespace, and omitting
the main function at the end of each test file.

Review URL: http://webrtc-codereview.appspot.com/296003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1013 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-24 12:52:40 +00:00
pwestin@webrtc.org
075e91fa27 Added parsing of width and height from VP8 header
Review URL: http://webrtc-codereview.appspot.com/241012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@875 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 23:14:58 +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
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
perkj@google.com
ac75cab618 Fix reference counting assert.
Change assert("teo") to assert(!"teo") so that the assert is actually triggered.
Review URL: http://webrtc-codereview.appspot.com/133018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@533 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-06 13:58:34 +00:00
perkj@google.com
ea72c34fb9 Temporary add dummy implementation to RefCountModule. The reason is so that ADM and VideoCapture implementations can change to refcounted versions before forcing them.
Review URL: http://webrtc-codereview.appspot.com/139014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@527 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-05 11:11:04 +00:00
perkj@google.com
ef04cf4b2e Adding reference counted version of the module interface.
The reason for this is that we would like to have reference counting on the modules you can register externally with ViE and VoE.
Currently we plan to use this on the ADM, VideoCapture module and VideoRenderModule.
Review URL: http://webrtc-codereview.appspot.com/138010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@517 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-02 09:47:28 +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
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
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