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4673 Commits

Author SHA1 Message Date
turaj@webrtc.org
28d54ab18f Improve AV-sync when initial delay is set and NetEq has long buffer. 
Review URL: https://webrtc-codereview.appspot.com/1324006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 18:53:35 +00:00
kjellander@webrtc.org
1b427719dc emove desktop_capture.gypi from modules.gyp 
When adding this in
we started getting linking problems on the mac_asan bot due to
the empty list of source files for the library target.

Please re-add it into modules.gyp when the library has source files
to compile.

BUG=none
TEST=Passing mac_asan trybot.

Review URL: https://webrtc-codereview.appspot.com/1313009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 13:13:49 +00:00
mikhal@webrtc.org
e0e029e8cb Revert 3876 
Review URL: https://webrtc-codereview.appspot.com/1341005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 21:11:41 +00:00
mikhal@webrtc.org
ee184b9520 VCM/Receiver: Only update render time when decoding 
Review URL: https://webrtc-codereview.appspot.com/1336004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 19:15:47 +00:00
mikhal@webrtc.org
a73d52ca52 revert r3871 
TBR= solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1331004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 20:26:28 +00:00
solenberg@webrtc.org
9756017717 - Replace the BWE_MIN and BWE_MAX macros with std::min and std::max 
- Add 'virtual' to a bunch of overridden methods of RemoteBitrateEstimatorMultiStream and RemoteBitrateEstimatorSingleStream.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1324005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 19:12:42 +00:00
solenberg@webrtc.org
d26457fdeb Apply Chromium C++ style to BitRateStats. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1325006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 12:25:32 +00:00
braveyao@webrtc.org
c14b728b71 Add lock to prevent possible rare race condition in Win coreAudio capture implementation. 
BUG = 
TEST = voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1320011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 09:32:07 +00:00
sergeyu@chromium.org
a0cd9182aa Add desktop_capture directory for screen and window capturers. 
The screen captures will be moved from chromium to WebRTC to make it easy
to share this code with other projects. This CL adds a new directory where
the current screen capturer code will be moved to.

Review URL: https://webrtc-codereview.appspot.com/1297005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3866 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 18:21:42 +00:00
mikhal@webrtc.org
dbd6a6d653 Updating delay for first value 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1327005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 16:23:22 +00:00
andresp@webrtc.org
48c5882f2a Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. 
R=stefan@webrtc.org,marpan@webrtc.org
BUG=201

Review URL: https://webrtc-codereview.appspot.com/1323005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 15:31:40 +00:00
tina.legrand@webrtc.org
db11fab49e Adding Opus unit test 
This CL adds a unit test for Opus, as well as new APIs for true stereo decoding (skipping master/slave approach).

BUG=

Review URL: https://webrtc-codereview.appspot.com/1222006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 10:39:41 +00:00
turaj@webrtc.org
f1a3b4bc0c Issue 1647. Avoid unsequenced modification. 
issue=1647
test=trybots,manual

Review URL: https://webrtc-codereview.appspot.com/1327004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 17:01:35 +00:00
pbos@webrtc.org
6e788df19e Remove vim/emacs modelines from .gypi files 
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 12:40:34 +00:00
solenberg@webrtc.org
56b5f77a2b Add support for multiple streams to RtpPlayer: 
- Tests video_rtp_play.cc, video_rtp_play_mt.cc, decode_from_storage.cc rewritten
 - rtp_player.cc/.h rewritten; added interfaces for externally setting up sinks
 - Support for reading .rtp files pulled out into rtp_file_reader namespace
 - Added support for reading .pcap (libpcap/wireshark/tcpdump) files, see pcap_file_reader

BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1201009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 10:31:56 +00:00
stefan@webrtc.org
885cd13356 Start NACKing as soon as we have the first packet of a key frame. 
BUG=1605

Review URL: https://webrtc-codereview.appspot.com/1307007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 09:38:26 +00:00
stefan@webrtc.org
bdb9b971be Change receive statistics bitrate to be provided in bps instead of kbps. 
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1326004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 09:02:03 +00:00
turaj@webrtc.org
92d1f07551 Elevate NetEq short-term activity statistics to ACM level for logging. 
Review URL: https://webrtc-codereview.appspot.com/1313004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 16:52:04 +00:00
kjellander@webrtc.org
4b8de90dce Disable -Wunsequenced warning in audio_coding_module 
BUG=1647
TEST=Compile locally on Linux with clang enabled.
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1316005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 06:38:56 +00:00
mikhal@webrtc.org
c2a3aa7926 Partial revert of r3844 
Review URL: https://webrtc-codereview.appspot.com/1320004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 19:53:30 +00:00
mikhal@webrtc.org
d6bd7cd2b1 removing redundant calls to cleanframes 
Review URL: https://webrtc-codereview.appspot.com/1318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 17:09:51 +00:00
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
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
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
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
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
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
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
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
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
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