generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
2,622 Commits 1 Branch 0 Tags 108 MiB
ff33bdd9d7
Commit Graph

2622 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
leozwang@webrtc.org
2db85bcba7 Make webrtc build with audio device java impl and add an option to enable it 
BUG=
TEST=buildbots

This cl is to make audio device java implemenation build in webrtc, and add an
option in gyp so we can switch between opensl implementaiton and java
implementation.
Review URL: https://webrtc-codereview.appspot.com/801004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-18 20:19:00 +00:00
mikhal@webrtc.org
deea95f76f Fix Windows warnings (int/bool mismatch) 
BUG=

Review URL: https://webrtc-codereview.appspot.com/808004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-18 17:09:02 +00:00
mikhal@webrtc.org
043ed9ecbd Refactoring videoFrame - I420VideoFrame will eventually replace VideoFrame which is currently defined in modules_common_types. Main changes: the new class allows per plane pointers, stride and uses scoped_array. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/774004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-18 16:14:26 +00:00
fbarchard@google.com
2e7c22da7d Roll libyuv to match chrome and gtp roll. 
BUG=none
TEST=try bots pass
Review URL: https://webrtc-codereview.appspot.com/784012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-18 08:03:32 +00:00
kma@webrtc.org
f4ca522e9b Correct a file name in Android.mk in iSAC-fix (Android platform build). 
Review URL: https://webrtc-codereview.appspot.com/798010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-17 20:16:15 +00:00
leozwang@webrtc.org
f851802bd7 Change prebuilt libraries 
Because file struction was changed again

BUG=
TEST=local
Review URL: https://webrtc-codereview.appspot.com/785008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-17 00:03:41 +00:00
andrew@webrtc.org
0be1f234b6 Add merge_libs_dependencies and remove voice_engine_dependencies. 
TBR=wu,turaj
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/798006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-15 02:50:52 +00:00
elham@webrtc.org
19f200edf3 Updating version number to 3.12 
Review URL: https://webrtc-codereview.appspot.com/805004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-14 20:38:56 +00:00
andrew@webrtc.org
52b9c58fac Set prefer_fixed_point properly. 
It wasn't being set from within a variables scope.

Review URL: https://webrtc-codereview.appspot.com/784010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-14 17:28:32 +00:00
phoglund@webrtc.org
4c6c11553c Initial version of a peerconnection fuzzer. 
The fuzzer can either pass random utf-8 into the SDP parser or swap lines in the generated SDP offer or answer. I've tried to implement the fuzzer so that all random choices are coded into the javascript page so that the sole source of randomness is in the fuzzer program. I initially tried to load stored sample SDP offers and fuzz them in the fuzzer program, but it didn't work since the SDP message seems to contain some magic checksum that causes the parser to choke quickly.

There's a lot of ideas for follow up patches:
- Fuzz ALL input parameters to ALL functions, not just SDP
- Swap letters/words in SDP messages
- Insert random location.reload() anywhere in the call sequence
- Swap lines in the call sequence itself

BUG=

Review URL: https://webrtc-codereview.appspot.com/784004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-14 10:06:57 +00:00
wu@webrtc.org
5c01b18c75 Remove DD renderer, which we no longer maintain. 
Review URL: https://webrtc-codereview.appspot.com/796005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-13 22:43:10 +00:00
andrew@webrtc.org
75dbe9f600 Only use arm_neon when armv7==1. 
This corresponds to the Chromium understanding; from build/common.gypi:
# Set Neon compilation flags (only meaningful if armv7==1).
'arm_neon%': 1,

Review URL: https://webrtc-codereview.appspot.com/802004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-13 16:44:46 +00:00
stefan@webrtc.org
a36442db10 Roll libvpx to fixes for valgrind warnings. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/802005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-13 09:15:53 +00:00
mflodman@webrtc.org
0f27089e52 Refactored vie_autotest_simulcast.cc. This CL on changes the style and renames variables. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/787008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-13 08:12:32 +00:00
mallinath@webrtc.org
c2c509cce3 Fixing the coverity warnings in video capture module. 
Review URL: https://webrtc-codereview.appspot.com/783007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-13 00:30:04 +00:00
leozwang@webrtc.org
e00ef43903 Revert previous changes in audio_device.h 
Full path is required here, restore qualified paths in audio_device.h.

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/800004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 23:02:28 +00:00
leozwang@webrtc.org
d47585399d Make audio_device build in binary build and source build 
BUG=issue 819
TEST=local test

Message:
This cl is going to address issue 819.
The initial fix is https://code.google.com/p/webrtc/source/detail?r=2720.
The problem happens because include file path is conflict in binary build
and source build

Description:
I added stronger directive to work around this problem.

Mike and Stefan, please check if it will break other builds.
Review URL: https://webrtc-codereview.appspot.com/795007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 21:43:47 +00:00
wu@webrtc.org
d58deb3df2 Fix coverity issues on video render. 
Review URL: https://webrtc-codereview.appspot.com/788007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 18:53:54 +00:00
phoglund@webrtc.org
0df21d01f0 snprintf doesn't exist on windows. 
TBR=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/792005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 17:02:10 +00:00
phoglund@webrtc.org
54d7faa5e3 Fixed release error. 
TBR=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/785007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 16:36:25 +00:00
phoglund@webrtc.org
db81d5b8f6 Fixed errors from last patch. 
BUG=
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/793007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2759 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 16:24:20 +00:00
phoglund@webrtc.org
f72943dadc Rewrote menu handling for vie custom call. 
The intended trajectory of this patch is to abstract out all i/o for custom_call.
The reason is that kjellander@ will need to be able to configure custom calls using
flags, and using the same framework to gather all input gathering to a single place
will make this a lot easier.

This patch focuses on choices. The next will focus on field entries, like "enter
ip address" or "enter port number."

BUG=
TEST=Manually tested all menus in custom call, ran new unit tests.

Review URL: https://webrtc-codereview.appspot.com/757005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2758 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 15:59:22 +00:00
stefan@webrtc.org
ed2e2eecae Revert removal of libjpeg include paths. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/791006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 14:43:00 +00:00
mflodman@webrtc.org
5a7507f26a Add API for transmission smotthening. 
BUG=818
TEST=Only API tests added now.

Review URL: https://webrtc-codereview.appspot.com/787009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 13:47:06 +00:00
wu@webrtc.org
633a6fa815 Fixing coverity warnings for render module. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/791005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 03:51:27 +00:00
leozwang@webrtc.org
430e31c2c0 Change VQE settings 
Change some VQE settings and make iSAC (item 0 in the list) to be the
default

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/790005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 03:47:08 +00:00
kma@webrtc.org
587073cbb2 Added calling WebRtcSpl_Init() in fixed point NS, and removed it from floating point NS. 
Review URL: https://webrtc-codereview.appspot.com/781009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-12 00:23:40 +00:00
leozwang@webrtc.org
08a8ff496e Change macro define 
This is correctionof r2720, should use HAVE_WEBRTC_VOICE when
build 3rd party library

BUG=None
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/781006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 22:08:14 +00:00
marpan@webrtc.org
cb8050c166 Coverity fix 10325: uninitialized scalar field. 
Review URL: https://webrtc-codereview.appspot.com/788005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 20:34:15 +00:00
sjlee@webrtc.org
414fa7f0c4 Change MAC_IPHONE to WEBRTC_IOS. 
Review URL: https://webrtc-codereview.appspot.com/788004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 17:25:46 +00:00
stefan@webrtc.org
2578300eee Fix GCC 4.6 build error. 
TBR=kjellander

BUG=

Review URL: https://webrtc-codereview.appspot.com/794004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 14:25:05 +00:00
stefan@webrtc.org
c58be0d217 Tune for faster ramp-up. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 14:11:58 +00:00
tina.legrand@webrtc.org
1617f65eec Coverity warnings in audio codecs 
Coverity reported errors/warnings in iSAC and iLBC. Seven of the warnings should be fixed with this CL.

Review URL: https://webrtc-codereview.appspot.com/793004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 14:06:27 +00:00
henrik.lundin@webrtc.org
f0effa12d9 Removing dead code in NetEQ 
This CL should resolve Coverity DefectIds 14004, 14017, 14018, 14019.
Mostly harmless dead code, but also a bug in WebRtcNetEQ_RedundancySplit.
(The bug would only trigger if more than 2 redundancy payloads were sent
in the same packet.)

BUG=Coverity DefectIds 14004, 14017, 14018, 14019.
TEST=trybots, neteq_unittests, audio_coding_module_test.

Review URL: https://webrtc-codereview.appspot.com/783006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 12:44:06 +00:00
kjellander@webrtc.org
b43f85ffd3 Support for being executed from runtests.py 
This is also needed to make it possible to run unit tests easily using Chromium's buildbot source code.

BUG=None
TEST=tools/valgrind-webrtc/webrtc_tests.sh --test test_support_unittests --build_dir=out/Debug

Review URL: https://webrtc-codereview.appspot.com/784007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 11:22:45 +00:00
kjellander@webrtc.org
b764d78831 Adding support for --test argument 
BUG=Cannot use this from Chrome buildbot scripts.
TEST=Ran tools/valgrind-webrtc/webrtc_tests.sh --test out/Debug/test_support_unittests

Review URL: https://webrtc-codereview.appspot.com/784006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 09:34:51 +00:00
henrik.lundin@webrtc.org
1e100a9a5e Removing WebRtcNetEQ_GetVersion 
We no longer use version functions.
This should resolve Coverity DefectId 14108

BUG=Coverity DefectId 14108
TEST=trybots, neteq_unittests, audio_coding_unittests, audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/792004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 09:23:32 +00:00
henrik.lundin@webrtc.org
a63b3db614 Fixing uninitialized member variables in RtpFormatVp8TestHelper 
This should fix Coverity DefectId 14374.

BUG=Coverity DefectId 14374
TEST=trybots, rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/791004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 09:20:07 +00:00
xians@webrtc.org
8890b3b5b2 fixing the opensles problem. 
Review URL: https://webrtc-codereview.appspot.com/737005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 08:12:53 +00:00
stefan@webrtc.org
7c3523c1a4 Change audio/video sync to be based on mapping RTP timestamps to NTP. 
Video Engine:
- Instead compensate for video capture delay by modifying RTP timestamps.
- Calculate the relative offset between audio and video by converting
  RTP timestamps to NTP and comparing receive time.

RTP/RTCP module:
- Removes the awkward video modification of NTP to compensate
  for video capture delay.
- Adjust RTCP RTP timestamp generation in rtcp_sender to have the same offset
  as packets being sent from rtp_sender.

BUG=
TEST=trybots,steam_synchronization_unittest

Review URL: https://webrtc-codereview.appspot.com/669010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 07:00:42 +00:00
andrew@webrtc.org
fa418ac0af Consolidate common_video targets to improve gyp run time. 
Not sure if this change is measurable; perhaps a 1% savings.

BUG=webrtc:34

Review URL: https://webrtc-codereview.appspot.com/785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-11 01:34:21 +00:00
fbarchard@google.com
852fd687e0 Fix for clang ia32 on linux which does not support optimize attribute. 
BUG=libyuv bug 83
TEST=./build/gyp_chromium --depth=. -Dtarget_arch=ia32 -Dclang=1 webrtc.gyp && make libyuv
Review URL: https://webrtc-codereview.appspot.com/790004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 22:01:32 +00:00
fbarchard@google.com
1594e84ca0 Roll to libyuv r341 to fix unittest failures on build bots. Version number updated so unittests pass version check. Disabled 4 tests that failed valgrind - libyuv bugs 79,80,81,82 opened for followup. 
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/787005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 20:54:05 +00:00
sjlee@webrtc.org
4b42508cc0 This CL is WebRTC VoiceEngine for iOS and is from CL713004. 
After patching this, first comments some video related lines in webrtc.gyp and src/module/module.gyp
And then do the below command.

$> ./build/gyp_chromium --depth=.  -DOS=ios -Dtarget_arch=armv7 -Dinclude_tests=0 -Denable_protobuf=0 -Denable_video=0 webrtc.gyp
$> xcodebuild -sdk iphoneos [-configuration Release]
Review URL: https://webrtc-codereview.appspot.com/768009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 17:58:21 +00:00
andrew@webrtc.org
c732ca697b Return audio_processing_tests.gypi to module.gyp. 
TBR=xians

Review URL: https://webrtc-codereview.appspot.com/781005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 16:33:43 +00:00
phoglund@webrtc.org
4a2639a210 Disabled one more test on Linux that was probably flaky. 42 remaining. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/786004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 13:28:39 +00:00
stefan@webrtc.org
e37ecc6f81 Adding test for relaying all simulcast streams to different receive channels. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/776007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 13:27:47 +00:00
mflodman@webrtc.org
deaf685b66 Fix gcc 4.6 compilation for video_engine_unittest 
TEST=Manually built using gcc 4.6.

Review URL: https://webrtc-codereview.appspot.com/787004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-10 13:19:08 +00:00
leozwang@webrtc.org
a96f8d9584 Change audio_processing libraries because of r2723 
Buildbot will be ready soon, so such problem will hopefully not happen again.

BUG=None
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/782004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-08 23:26:16 +00:00
andrew@webrtc.org
8c4696cd76 Consolidate audio_processing targets to improve gyp time. 
Saves about 5% on a standalone Mac-XCode gyp run.

BUG=issue34

Review URL: https://webrtc-codereview.appspot.com/781004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-08 19:27:24 +00:00