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Author SHA1 Message Date
fischman@webrtc.org
5a602d7996 Enable WebRTC demo application on x86 Android 
Steps to build the demo application for x86 Android:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug
cd webrtc/video_engine/test/android
ndk-build APP_ABI=x86
ant debug

Commmitted as https://code.google.com/p/webrtc/source/detail?r=4053

R=fischman@webrtc.org, leozwang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1478004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 17:20:04 +00:00
pbos@webrtc.org
21632124dd Include gflags properly and X11 include order in VideoEngine. 
BUG=

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1500004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 14:25:02 +00:00
pbos@webrtc.org
f5d4cb1958 Include files from webrtc/.. paths in video_engine/ 
BUG=1662
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1492004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 13:44:48 +00:00
stefan@webrtc.org
9f557c140e Improve wraparound handling in the render time extrapolator. 
This was actually working as intended, but as r3970 changed when render timestamps were extrapolated to when a frame was taken out for decoding, the wraparound could have happened in the Update() step before it had happened in the ExtrapolateLocalTime() step. This causes render timestamps to be generated 13 hours into the future.

TEST=trybots
BUG=1787
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1497004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 12:55:07 +00:00
phoglund@webrtc.org
14d7700d00 Moved command line parsing to internal tools and moved back the mic volume thingie. 
BUG=
R=henrika@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1491004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 11:52:08 +00:00
fischman@webrtc.org
e874a8f24b Enable WebRTC demo application on x86 Android 
Steps to build the demo application for x86 Android:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug
cd webrtc/video_engine/test/android
ndk-build APP_ABI=x86
ant debug

R=fischman@webrtc.org, leozwang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1478004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 05:41:07 +00:00
turaj@webrtc.org
8630cfe016 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. 
BUG=issue1770
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1485004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 23:54:54 +00:00
hclam@chromium.org
fe307e1332 Add one unit test for NACKing a key frame 
Adding a test case that wasn't covered. This new test is passing.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 21:19:59 +00:00
hclam@chromium.org
b3e5acfb66 Cleanup traces in WebRTC 
Remove some unused traces and add a trace counter for encoded video size.

R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1476004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 21:13:02 +00:00
pbos@webrtc.org
b9bb3d1e7d Avoid resetting encoder on identical settings. 
BUG=1681
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1481005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 18:40:48 +00:00
marpan@webrtc.org
890f6092e6 Bugfix: VCM would report wrong sentBitrate 
issue: https://code.google.com/p/webrtc/issues/detail?id=1755

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1484004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 15:38:44 +00:00
phoglund@webrtc.org
9919ad5caf Formatted FEC stuff. 
Unfortunately I had to pull in quite a bit of stuff due to use of unencapsulated public member variables.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1401004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 15:06:28 +00:00
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