phoglund@webrtc.org
5c1948dfaf
Moved force_volume_max to its own gyp file to avoid a circular dependency.
...
BUG=
TBR=tlegrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4046 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 13:59:19 +00:00
phoglund@webrtc.org
61d3c552a1
Wrote a small portable tool for forcing the mic volume to 100%.
...
BUG=
R=henrika@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1477005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4045 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 13:10:00 +00:00
pbos@webrtc.org
29d5839233
New VideoEngine API implementation on top of old one, first steps.
...
BUG=1668
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1360004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4044 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 12:08:03 +00:00
stefan@webrtc.org
2038214c77
Log too long non-decodable duration events.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1488004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4043 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 11:39:06 +00:00
mflodman@webrtc.org
4dee30927a
Remove SetOverUseDetectorOptions and cleaned ViESharedData.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1486004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 11:13:18 +00:00
solenberg@webrtc.org
7ebbea14a9
Add handling of the absolute send time header extension to the rtp_rtcp module.
...
BUG=
R=asapersson@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1480004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 11:10:31 +00:00
vikasmarwaha@webrtc.org
59a06670b5
Updated apprtc demo to interop with firefox.
...
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1482004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4040 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 01:05:19 +00:00
vikasmarwaha@webrtc.org
40298d452c
Added webaudio-and-webtrc.html to the demos index.html.
...
R=dutton@google.com , henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1425005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4039 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 00:50:38 +00:00
fischman@webrtc.org
8c2e78b2de
Roll chromium_revision 193311:199267
...
This will fix static libraries will not be copied to product out dir issue on x86 Android
Remove third_party/WebKit/Tools/Scripts since it will not be used.
BUG=webrtc:1690
TEST=Trybots passing
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1457004
Patch from Jeremy Mao <yujie.mao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4038 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-15 22:50:23 +00:00
mikhal@webrtc.org
6cfa3907c8
Updating NACK RTX test
...
BUG=1513
R=holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1274006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4036 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-15 20:17:43 +00:00
mikhal@webrtc.org
cb20a5b2d7
VCM/JB: Bug fix in ExtractAndSetDecode
...
BUG=1771
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1466005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4035 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-15 17:10:44 +00:00
solenberg@webrtc.org
5add4ad09c
RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed.
...
BUG=
R=holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1481004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4034 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-15 13:49:57 +00:00
braveyao@webrtc.org
c93b1d038d
CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin
...
BUG=
TEST=voe_auto_test
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4033 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-15 10:14:56 +00:00
niklas.enbom@webrtc.org
e2a800644c
Linux support for typing detection
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1428006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4031 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 21:33:11 +00:00
turaj@webrtc.org
4ce838934c
Address sanitizer out of bounds read in iSAC
...
BUG=issue1770
TBR=tlegrand@google.com
Review URL: https://webrtc-codereview.appspot.com/1472006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4030 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 17:42:22 +00:00
pbos@webrtc.org
6bee05a4aa
Remove const for plain data types in common_video/
...
BUG=1644
R=holmer@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1464004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4028 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 14:27:15 +00:00
andresp@webrtc.org
29b2219914
Adding a factory to remote bitrate estimator and allow it to be set via config.
...
Additionally:
- clean api to set remote bitrate estimator mode.
- clean api to set over use detector options.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1448006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 12:10:58 +00:00
stefan@webrtc.org
1673481ed7
Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly.
...
BUG=1769
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1473004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4026 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 12:00:47 +00:00
phoglund@webrtc.org
736c6f775e
Fixed more perf expectations.
...
For Linux, the expectations just look a bit too tightly wound. On Windows there's a long-term increasing trend that we may want to have someone look at.
http://www.corp.google.com/~webrtc-cb/perf//linux-large-tests/vie_auto_test/report.html?history=1500&rev=-1&graph=total_delay_incl_network
http://www.corp.google.com/~webrtc-cb/perf//linux-large-tests/vie_auto_test/report.html?history=1500&rev=-1&graph=total_delay_incl_network
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1472005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4025 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 11:26:14 +00:00
phoglund@webrtc.org
80c7e3b606
Adjusted perf expectations for mac large tests.
...
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1472004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4024 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 10:51:13 +00:00
mflodman@webrtc.org
bb984f516e
Removed Mac capture crash and memory leak.
...
BUG=1697,1761
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1465005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4023 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 10:47:19 +00:00
kjellander@webrtc.org
a6ff84503e
Add script for comparing video quality
...
This script makes it easier to run a simple command line
comparison between a captured YUV file and a reference video.
BUG=none
TEST=command line invocation
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1320007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4022 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 09:43:04 +00:00
phoglund@webrtc.org
6d07ad9ccc
Added protoc_wrapper to blacklist, fixed tools/PRESUBMIT.py which was passing in the wrong args to CheckLongLines.
...
BUG=
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1470005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4021 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 09:42:39 +00:00
phoglund@webrtc.org
527f6c62fc
Reformatted FEC tables.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1400004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4020 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 09:25:01 +00:00
pbos@webrtc.org
8e3b594831
Remove const for plain data types in common_audio/
...
BUG=1644
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1464005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4019 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 09:24:49 +00:00
pbos@webrtc.org
9213521ea9
Remove const for plain data types in voice_engine/
...
BUG=1644
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1463004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 08:31:39 +00:00
andresp@webrtc.org
185bae4b6f
Replace ExtraCodecOptions with new Config class that supports multiple settings at once.
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1452004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4017 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 08:02:25 +00:00
fbarchard@google.com
c9cb4fffac
Fix typo in log statement. witdh should be width.
...
BUG=none
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1466004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4016 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 05:02:08 +00:00
justinlin@chromium.org
7bfb3a3227
Add more tracing for key frames.
...
R=mallinath@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1428004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 22:59:00 +00:00
vikasmarwaha@webrtc.org
941fcc5841
Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343.
...
TBR=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1463005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4014 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 20:28:23 +00:00
vikasmarwaha@webrtc.org
1993a559e8
Added Stereo url paramter to apprtc demo.
...
R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/1418004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4013 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 18:48:09 +00:00
elham@webrtc.org
52b3905ec8
Updated WebRTC version to 3.31
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1462004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4011 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 17:00:56 +00:00
phoglund@webrtc.org
43bf6ce322
Revert 4008 "Avoid resetting video encoder for similar configs."
...
> Avoid resetting video encoder for similar configs.
>
> BUG=1681
> R=holmer@google.com , mflodman@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1442006
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1431005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4010 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 15:39:26 +00:00
phoglund@webrtc.org
c53480fbcf
Disabled flaky codec test (RunsCodecTestWithoutErrors)
...
BUG=1734
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1460004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4009 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 15:10:02 +00:00
pbos@webrtc.org
aa4efd1535
Avoid resetting video encoder for similar configs.
...
BUG=1681
R=holmer@google.com , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1442006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4008 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 11:27:16 +00:00
andresp@webrtc.org
7707d060bb
Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1450008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4007 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 10:50:50 +00:00
henrika@webrtc.org
7a5615bc84
New WebAudio-WebRTC demo.
...
Capture microphone input and stream it out to a peer with a processing effect applied to the audio.
The audio stream is:
o Recorded using live-audio input.
o Filtered using an HP filter with fc=1500 Hz.
o Encoded using Opus.
o Transmitted (in loopback) to remote peer using RTCPeerConnection where it is decoded.
o Finally, the received remote stream is used as source to an <audio> tag and played out locally.
Press any key to add an effect to the transmitted audio while talking.
Please note that:
o Linux is currently not supported.
o Sample rate and channel configuration must be the same for input and output sides on Windows.
o Only the Default microphone device can be used for capturing.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1256004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4006 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 09:29:13 +00:00
pbos@webrtc.org
7ee822805d
Remove TEXT(x) for BUILDINFO macros.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1453004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4005 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 09:29:03 +00:00
andresp@webrtc.org
6b68c28cb1
Added a config class to ease passing a set of options across webrtc.
...
Its main design reason is to expose control of experimental webrtc features.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1450009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4004 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 08:06:36 +00:00
braveyao@webrtc.org
9ecd6861eb
Add svn:eol-style back which is lost in r3993 mistakenly.
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1428008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4003 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 05:38:13 +00:00
leozwang@webrtc.org
a404d1d8de
Change watchlist.
...
Watch all changes in webrtc.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1428012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4002 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-10 22:46:55 +00:00
tnakamura@webrtc.org
7311083ccc
Revert 3977
...
BUG=webrtc:1749
> Update protoc.gypi to match Chromium's latest.
>
> This is in preparation for enabling protobufs in Chromium. Requires
> syncing tools/protoc_wrapper.
>
> BUG=webrtc:830
> R=kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1426004
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1453005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4001 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-10 22:33:50 +00:00
elham@webrtc.org
05ea12f12e
Reverting r3978
...
BUG=webrtc:1749
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1454004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4000 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-10 17:04:59 +00:00
fischman@webrtc.org
d6ed000585
This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build.
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1444005
Patch from Jeremy Mao <yujie.mao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3999 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-10 16:34:01 +00:00
mikhal@webrtc.org
571b3369e7
Updating perf
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1428011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3997 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 20:03:47 +00:00
fbarchard@google.com
1e3c794688
Use 2 threads for HD, or 1 for VGA or less.
...
BUG=1739
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1438005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3996 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 18:43:38 +00:00
mikhal@webrtc.org
06806701f0
Updating perf
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1447004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3995 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 17:42:58 +00:00
fischman@webrtc.org
6a36f0e46f
Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation.
...
BUG=webrtc:1741
TEST=Build and run the Android WebRTC demo application
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1439006
Patch from Jeremy Mao <yujie.mao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3994 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 17:40:33 +00:00
braveyao@webrtc.org
e525309004
WebRTCDemo Android doesn't hangle activity recreation correctly.
...
Also optimize Statsview a little bit.
BUG=1740
TEST=Manual test with WebRTCDemo Android
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3993 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 08:52:50 +00:00
kjellander@webrtc.org
219762a68a
Drop Virtual webcam check script as moved into buildbot scripts.
...
Having this script being located in the WebRTC repo doesn't make sense
since it has no connection to the source code.
Updating this script should apply to all build configurations and since
this script will be used for Chromium builders, we'll end up with having
to wait for a new WebRTC revision to be rolled in DEPS before it's updated.
TEST=none
BUG=none
TBR=phoglund
Review URL: https://webrtc-codereview.appspot.com/1444006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3992 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 07:53:08 +00:00