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Author SHA1 Message Date
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
mflodman@webrtc.org
65f995a3df New ViE interface. 
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/1113004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 12:02:52 +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
andrew@webrtc.org
ceaedc0014 Remove executable bit from dc1.html. 
Review URL: https://webrtc-codereview.appspot.com/1320010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 01:56:07 +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
kjellander@webrtc.org
f09016744d Revert "Updating test file contents to emmastjernloef" 
This reverts r3861

TEST=none
BUG=none
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/1317005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 14:59:28 +00:00
hta@webrtc.org
f1bf3a00b2 A device switcher code example, with fake. 
This demo shows the usage of the proposed getDeviceInfo call and its
associatied permissions model.

Review URL: https://webrtc-codereview.appspot.com/1320008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 14:24:21 +00:00
kjellander@webrtc.org
11959d3476 Updating test file contents to emmastjernloef 
This way, we honour Emma Stjernlöf in the Stockholm office.

TEST=test_support_unittests
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1320009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 13:58:54 +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
henrika@webrtc.org
4392d5f9f8 Fix for "RTP dynamic payload type 100 is reserved" 
TBR=perkj
BUG=227036 (in crbug.com)

TEST=out\Debug\voe_auto_test.exe --automated --gtest_filter=Dtmf* where I
manually modified the test and used 100 as new PT (which I first verified was
already used by CN, 48000).

BUG=

Review URL: https://webrtc-codereview.appspot.com/1319010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 07:34:25 +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
wu@webrtc.org
e44a064915 Make win_support_condition_variables_primitive global to aligned with |library| 
so that once we set it to true it will remain.
Review URL: https://webrtc-codereview.appspot.com/1319006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 18:06:42 +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
kjellander@webrtc.org
f806ad23db Roll chromium_revision 182149:193311 
This will among other things give us:
* Tons of updates to build/android stuff (needed to make Android NDK bots work
  in Chrome infra)
* Clang updated to 176256 (r187059)
* Support for fastbuild=2, which completely disables debug information (r191876)
* enable -Wstring-conversion when compiling with clang (r183998)
* Update ndk sysroot to API level 14 (r186254)

Detailed changelog:
http://build.chromium.org/f/chromium/perf/dashboard/ui/changelog.html?url=trunk%2Fsrc%2Fbuild&range=182149%3A193311&mode=html

build/common.gypi changes:
http://src.chromium.org/viewvc/chrome/trunk/src/build/common.gypi?r1=182149&r2=193311

TEST=trybots passing
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1317004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-15 05:52:57 +00:00
marpan@webrtc.org
c83b35661d Roll libvpx to 192165. 
-pick up libvpx roll to 3db60c8.

TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1321004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 21:43:48 +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
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