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Author SHA1 Message Date
mflodman@webrtc.org
9f5ebb5251 Adding a payload type for RTX. 
BUG=736
TEST=Modified RTP unittests.

Review URL: https://webrtc-codereview.appspot.com/1278004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 14:55:46 +00:00
stefan@webrtc.org
b8e7f4cc97 Change capture interface to use NTP capture time. 
Move NTP functionality to Clock.

BUG=1563
TEST=trybots and vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/1313005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 11:56:23 +00:00
kjellander@webrtc.org
d35dff7664 Move to Chrome infra try server. 
I'm not sure how critical it is to have the android+android_ndk try bots, as they're not yet up at http://build.chromium.org/p/tryserver.webrtc/waterfall
I have a CL for android_ndk (https://codereview.chromium.org/11896066), but I don't have a good solution for the android platform build yet.

After this is submitted, developers can still send jobs to the old try server (assuming we keep those bots over there) with:
git try -H webrtc-cb-linux-master.cbf.corp.google.com -P 9018 --bot=android,android_ndk

The default (and the only option for public users) will however be the new Chromium try server (via the SVN queue).

BUG=chromium:174594
TEST=successfully submitted a try job that was built at http://build.chromium.org/p/tryserver.webrtc/waterfall

Review URL: https://webrtc-codereview.appspot.com/1213004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 07:40:33 +00:00
pwestin@webrtc.org
1de01354e6 Adding playout buffer status to the voe video sync 
Review URL: https://webrtc-codereview.appspot.com/1311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 20:23:35 +00:00
mikhal@webrtc.org
9da751715f VCM/JB:Removing hybrid and setting a decodable state. 
Review URL: https://webrtc-codereview.appspot.com/1283004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 18:49:13 +00:00
stefan@webrtc.org
7bc465bd21 Fix issues with incorrect wrap checks when having big buffers and high bitrate. 
Introduces shared functions for timestamp and sequence number wrap checks.

BUG=1607
TESTS=trybots

Review URL: https://webrtc-codereview.appspot.com/1291005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 17:48:02 +00:00
stefan@webrtc.org
122d209e67 Fixes an issue where the start bitrate is stored in kbps instead of bps. 
BUG=1638
TEST=trybots and vie_auto_test loopback with nack.

Review URL: https://webrtc-codereview.appspot.com/1312004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 17:21:40 +00:00
wu@webrtc.org
eac36b8561 Fix -Wstring-conversion warnings. 
Review URL: https://webrtc-codereview.appspot.com/1299007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 15:37:46 +00:00
andresp@webrtc.org
523f93729b Re-write the build of the nacklist. 
Review URL: https://webrtc-codereview.appspot.com/1304008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 11:30:39 +00:00
fischman@webrtc.org
f2a97fc2b4 WebRTCDemo: handle stride!=width from first frame. 
Previously only mid-stream frames handled stride!=width correctly.

BUG=1615

Review URL: https://webrtc-codereview.appspot.com/1304009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 23:21:10 +00:00
marpan@webrtc.org
d40e404be4 Revert r3815 
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1301006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 21:37:03 +00:00
elham@webrtc.org
1b2a6e0be0 Updated WebRTC version number to 3.29 
TBR=mallinath1 
Review URL: https://webrtc-codereview.appspot.com/1305005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 21:28:33 +00:00
fischman@webrtc.org
6f41ca9fd2 WebRTCDemo: Enable making multiple calls. 
Previously after the first call subsequent attempts to bind the RTP/RTCP ports would fail, since r3754.

BUG=1618

Review URL: https://webrtc-codereview.appspot.com/1302007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 20:33:27 +00:00
kjellander@webrtc.org
59d8889704 Add OWNERS file for channel_transport 
Readding the OWNERS file that used to be located in
webrtc/modules/udp_transport before it was dropped in r3788
(should have been added in r3701).

BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/1310006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 20:16:26 +00:00
marpan@webrtc.org
6bfcbcda13 Roll libvpx to 192165. 
-pick up libvpx roll to 3db60c8.

TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1307006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 20:08:53 +00:00
pbos@webrtc.org
e4b6064f8e Replace legacy G_CONST with const. 
BUG=1608

Review URL: https://webrtc-codereview.appspot.com/1310005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 18:06:57 +00:00
pbos@webrtc.org
ab9202b673 Removing remaining WebRtc_Word32 not in typedefs.h 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1306006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 17:59:17 +00:00
fischman@webrtc.org
77d59fe408 WebRTCDemo: no-op out instead of NPEing on destroyed camera. 
BUG=1617

Review URL: https://webrtc-codereview.appspot.com/1310004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 17:11:51 +00:00
pbos@webrtc.org
dfc5bb9c97 WebRtc_Word32 -> int32_t in video_capture/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1298005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 08:23:13 +00:00
pbos@webrtc.org
ddf94e71e5 WebRtc_Word32 -> int32_t in video_render/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1304006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 08:09:04 +00:00
pbos@webrtc.org
b7192b8247 WebRtc_Word32 -> int32_t in audio_processing/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 07:50:54 +00:00
marpan@webrtc.org
557e92515d Reapply the reverted r3747. 
https://code.google.com/p/webrtc/source/detail?r=3747

r3747 timed-out on a tsan test. Verified that it passes
the test and reduced the execution time of that test (r3782).

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 21:21:32 +00:00
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00
stefan@webrtc.org
4d2f5de67a Improve how NACK lists are generated before a frame has been decoded. 
BUG=1598

Review URL: https://webrtc-codereview.appspot.com/1295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 18:24:41 +00:00
pbos@webrtc.org
ac891627c6 WebRtc_Word32 -> int32_t in audio_conference_mixer/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1306004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 17:40:15 +00:00
pbos@webrtc.org
b09130763b WebRtc_Word32 -> int32_t in common_audio/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 16:40:28 +00:00
stefan@webrtc.org
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
pbos@webrtc.org
b238d1210b WebRtc_Word32 -> int32_t in video_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:41:51 +00:00
pbos@webrtc.org
1ab45f6dd5 WebRtc_Word32 -> int32_t in video_processing/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1297006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:38:10 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
fd2bfc8fca WebRtc_Word32 -> int32_t in common_video. 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1300004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:36:08 +00:00
pbos@webrtc.org
c75102eba7 WebRtc_Word32 -> int32_t in utility/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1307005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:32:55 +00:00
pbos@webrtc.org
0ea11c1768 WebRtc_Word32 -> int32_t in media_file/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:31:37 +00:00
hta@webrtc.org
a701c0ed03 Fixing the flakiness of ThreadWakesTwice. 
TESTED=ran the test 10.000 times with machine load.
BUG=1270

Review URL: https://webrtc-codereview.appspot.com/1303004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 12:36:28 +00:00
pbos@webrtc.org
a5f1787f63 WebRtc_Word32 -> int32_t in test/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 11:10:21 +00:00
pbos@webrtc.org
2550988baa WebRtc_Word32 -> int32_t in audio_device/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:30:35 +00:00
pbos@webrtc.org
6141e13873 WebRtc_Word32 -> int32_t in voice_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:09:10 +00:00
pbos@webrtc.org
046deb9b20 WebRtc_Word32 -> int32_t in system_wrappers 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 09:06:11 +00:00
pbos@webrtc.org
29758de9b6 Always set render delay in ViEChannel::RegisterExternalDecoder. 
BUG=1523

Review URL: https://webrtc-codereview.appspot.com/1219007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:34:42 +00:00
pbos@webrtc.org
0946a56023 WebRtc_Word32 => int32_t etc. in audio_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:28:06 +00:00
pwestin@webrtc.org
6faf71d27b Remove the old unused udp_transport 
Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 23:25:25 +00:00
vikasmarwaha@webrtc.org
4c44fe0561 Updated pranswer, dtmf demos & deleted pc1-deprecated.html. 
Review URL: https://webrtc-codereview.appspot.com/1287007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 21:23:58 +00:00
marpan@webrtc.org
6ff76c7404 Reduce execution time of rate control test. 
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1289005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 20:32:48 +00:00
kma@webrtc.org
cf8e108158 Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array. 
BUG=227286
Review URL: https://webrtc-codereview.appspot.com/1293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 16:37:53 +00:00
pbos@webrtc.org
b4a0623e43 Fix of lint script errors in apprtc.py 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1285007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 15:59:24 +00:00
pbos@webrtc.org
f2e7bc6b6a Added maxlen=80 to CheckLongLines() call in PRESUBMIT.py 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1285006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 15:46:07 +00:00
pbos@webrtc.org
034f004a4f WebRtc_Word32 => int32_t in video_coding/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1203008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:13:29 +00:00
pbos@webrtc.org
2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:08:41 +00:00
mflodman@webrtc.org
367804cce2 Clean packets on the network when closing + made loopback test actually run again. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1290006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:42:50 +00:00
pbos@webrtc.org
ff7e1303e8 WebRtc_Word32 => int32_t remote_bitrate_estimator/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1275009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:04:37 +00:00