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162 Commits

Author SHA1 Message Date
jiayl@webrtc.org
e3cdd9959e Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." 
This reverts commit 56631a14bdae24aa0bfaceeb2b57df729fee1227.

TBR=henrike@webrtc.org
BUG=3235

Review URL: https://webrtc-codereview.appspot.com/19669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:32:57 +00:00
jiayl@webrtc.org
745a39cced Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. 
BUG=3235
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 19:24:02 +00:00
buildbot@webrtc.org
91c910469f (Auto)update libjingle 68701339-> 68703656 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 16:29:00 +00:00
buildbot@webrtc.org
7b6cbb3aa0 (Auto)update libjingle 68689052-> 68689059 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 10:54:08 +00:00
pbos@webrtc.org
6ae48c6609 Make VideoSendStream/VideoReceiveStream configs const. 
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.

CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.

This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.

R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260

Review URL: https://webrtc-codereview.appspot.com/20409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 10:49:19 +00:00
buildbot@webrtc.org
4b83a471de (Auto)update libjingle 68646004-> 68648993 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 21:11:28 +00:00
pbos@webrtc.org
0d523eea83 Remove static initializer from WebRtcVideoEngine2. 
BUG=
R=pliard@google.com, pthatcher@webrtc.org, pliard@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 09:10:55 +00:00
buildbot@webrtc.org
6f237769b3 (Auto)update libjingle 68507189-> 68543735 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 16:23:10 +00:00
buildbot@webrtc.org
40b45fc07a (Auto)update libjingle 68506654-> 68507189 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 14:48:33 +00:00
buildbot@webrtc.org
0cdcd23a03 (Auto)update libjingle 68501302-> 68506654 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 01:31:14 +00:00
buildbot@webrtc.org
af81b9bffd (Auto)update libjingle 68499439-> 68501302 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 00:08:54 +00:00
buildbot@webrtc.org
251fdf64cb (Auto)update libjingle 68495561-> 68499439 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 23:43:48 +00:00
buildbot@webrtc.org
53217848b2 (Auto)update libjingle 68465410-> 68487517 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 21:09:11 +00:00
pbos@webrtc.org
289a35c56d Add empty webrtcmediaengine.cc. 
Should contain CreateWebRtcMediaEngine as soon as
libjingle/libjingle.gyp in Chromium builds this file. This file is added
ahead of time to get a smoother rolling process.

BUG=1788
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 14:51:34 +00:00
buildbot@webrtc.org
b525a9d790 (Auto)update libjingle 68379861-> 68445177 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:42:15 +00:00
buildbot@webrtc.org
34a08b4fb8 (Auto)update libjingle 68275107-> 68379861 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:48:10 +00:00
buildbot@webrtc.org
1678db9df6 (Auto)update libjingle 68230113-> 68244456 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 14:02:09 +00:00
buildbot@webrtc.org
540a2251aa (Auto)update libjingle 68230011-> 68230113 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:35 +00:00
pbos@webrtc.org
35efb839ed Implement new-API test RecvStreamWithoutRtx. 
R=pthatcher@google.com, pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/20449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:04 +00:00
pbos@webrtc.org
c34bb3a886 Log default receive stream creation. 
Log when receiving a packet that doesn't have a receiver, this way you
can tell from logs where the AddRecvStream call came from.

R=pthatcher@google.com, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:38:43 +00:00
pbos@webrtc.org
198647473b Implement and fix new-API NackIsEnabled test. 
Required enabling NACK on receiver side which was apparently missed.

BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:35:47 +00:00
buildbot@webrtc.org
1d66be22c8 (Auto)update libjingle 68203780-> 68206793 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:54:24 +00:00
buildbot@webrtc.org
f9f1bfbdae (Auto)update libjingle 67686255-> 67689476 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 17:02:15 +00:00
buildbot@webrtc.org
ce4201df52 (Auto)update libjingle 67643194-> 67686255 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:22:51 +00:00
henrike@webrtc.org
000658a138 Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot. 
BUG=N/A
TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:01:13 +00:00
mcasas@webrtc.org
3b7e282caa Disabling systematically failing 
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga

TBR= pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 14:25:20 +00:00
pbos@webrtc.org
1e019d10b8 Fix delivery error-checking missed in r6151. 
Gets rid of quite a bit of false-warning logging in WebRtcVideoEngine2.

BUG=3228
R=perkj@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 11:38:45 +00:00
buildbot@webrtc.org
ef5a752c29 (Auto)update libjingle 67043374-> 67044055 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 21:35:19 +00:00
pbos@webrtc.org
4e545cc244 Update webrtcvideoengine2.cc to use DeliveryStatus. 
talk/ changes corresponding to https://review.webrtc.org/12289005/.

BUG=3228
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 13:58:13 +00:00
pbos@webrtc.org
b5a22b1464 Revert r6110 and r6109. 
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.

BUG=
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 11:07:01 +00:00
buildbot@webrtc.org
eaf2bd916b (Auto)update libjingle 66813165-> 66836233 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 23:12:19 +00:00
buildbot@webrtc.org
17911dca80 (Auto)update libjingle 66798415-> 66813165 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:42:49 +00:00
henrike@webrtc.org
0df2ea064f Rollback of r6108 
BUG=N/A
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:41:12 +00:00
pbos@webrtc.org
a7f70a487f Initialize bitrates in ValidateCodecFormat. 
Attempt to un-break a Visual Studio build (unknown version) that
incorrectly reports that these are potentially uninitialized.

BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 18:20:40 +00:00
pbos@webrtc.org
d266a2020f Initial wiring of new webrtc API in libjingle. 
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
TBR=juberti@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 14:32:01 +00:00
buildbot@webrtc.org
073dfdd10a (Auto)update libjingle 66539128-> 66540208 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:36:21 +00:00
buildbot@webrtc.org
d1ae89fae1 (Auto)update libjingle 66524760-> 66539128 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 19:19:26 +00:00
jiayl@webrtc.org
f7026cd7c8 Check SCTP_EWOULDBLOCK instead of EWOULDBLOCK in SctpDataMediaChannel. 
usrsctp.h redefines EWOULDBLOCK to WSAEWOULDBLOCK on Windows, but usrsctp_sendv still returns the BSD EWOULDBLOCK (i.e. SCTP_EWOURLBLOCK) when sending data fails due to congestion.
We will need to revert this change when usersctp is fixed.

BUG=2866
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 16:02:23 +00:00
buildbot@webrtc.org
2219037e5e (Auto)update libjingle 66406192-> 66424806 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 17:52:33 +00:00
buildbot@webrtc.org
dd4742a9ef (Auto)update libjingle 66388864-> 66406192 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 14:50:35 +00:00
buildbot@webrtc.org
ed97bb0eb4 (Auto)update libjingle 66340694-> 66388864 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 11:15:20 +00:00
buildbot@webrtc.org
861d4b0de9 (Auto)update libjingle 66322380-> 66326258 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 22:11:02 +00:00
buildbot@webrtc.org
0581f0ba0a (Auto)update libjingle 66303009-> 66322380 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 21:36:31 +00:00
buildbot@webrtc.org
a18b4c96af (Auto)update libjingle 66301332-> 66303009 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:48:14 +00:00
buildbot@webrtc.org
e65c9a6e67 (Auto)update libjingle 66299810-> 66301332 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:28:28 +00:00
buildbot@webrtc.org
0b53bd29af (Auto)update libjingle 66294299-> 66299810 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:12:36 +00:00
buildbot@webrtc.org
150835ea34 (Auto)update libjingle 66236292-> 66294299 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 15:54:38 +00:00
buildbot@webrtc.org
13d6776c46 (Auto)update libjingle 66098243-> 66100938 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 17:33:29 +00:00
buildbot@webrtc.org
0d34f1446a (Auto)update libjingle 66033941-> 66098243 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 16:54:25 +00:00
jiayl@webrtc.org
9c16c39e61 Sets the SCTP port codec in the native SessionDescription. 
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.

BUG=3141
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 18:30:30 +00:00