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Author SHA1 Message Date
xians@webrtc.org
8fff1f065e Merge r4394 from stable to trunk. 
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.

Fixed the AGC and interface problems on the new path.

In order to make the AGC work properly, we need to cache the volume value passed
by the callback, compare it with the value returned by
shared->transmit_mixer()->CaptureLevel(). If they are the same, we need to
return 0 to indicate no volume needs changing, otherwise return the new volume.
By doing this, we avoid setting the volume all the same, which allows the users
to change the volume manually.

This patch also fixes some minor issues with the interfaces too: make the int
channel[] const, and correct the order of the input params in
channel::Demultiplex.

R=tommi@webrtc.org

BUG=[2134]
TEST=compile && manual AGC test

Review URL: https://webrtc-codereview.appspot.com/1921004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:27:42 +00:00
xians@webrtc.org
2f84afad30 Merge r4326 from stable to trunk. 
r4326 was mistakenly committed to stable, so this is to re-merge back to trunk.

Add new interface to support multiple sources in webrtc.
CaptureData() will be called by chrome with a flag |need_audio_processing| to
indicate if the data needs to be processed by APM or not. Different from the old
interface that will send the data to all voe channels, the new interface will
specify a list of voe channels that the data is demultiplexing to.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:23:37 +00:00
turaj@webrtc.org
7126b38d8f Handel zero correlation if at the same time distortion is also zero. 
This is the conversation I had with Henrik Lundin regarding this problem.

Me:
In Expand::AnalyseSignal() we compute correlation and distortion, then calculate the ratio of correlation to distortion. There if distortion is zero we expect that correlation to be zero. Although in practice this might be true, I suppose we rarely hit into absolutely periodic signal, but in one of the tests the assertion in line 455 of expand.cc was triggered. The distortion is computed over a shorter length of the signal, while correlation is computed over longer segments. Therefore, I guess, if the signal has just enough zeros at the beginning we can end up in situation that distortion is zero but not the correlation. Do you agree? I didn't have time to attempt to solve this, but if my line of thought is correct, we should not have that assert. Perhaps, if correlation is zero we set the ratio to 0, otherwise, ratio would be the largest value of its own type. Any thoughts?

Henrik:
I agree with you. Go ahead with your solution.

R=minyue@google.com

Review URL: https://webrtc-codereview.appspot.com/1888006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:05:09 +00:00
pbos@webrtc.org
2d1a55caed Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ 
BUG=163
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1900004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:54:00 +00:00
pbos@webrtc.org
e72428442d Fix some chromium-style warnings in webrtc/modules/desktop_capture/ 
BUG=163
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1904004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:32:43 +00:00
pbos@webrtc.org
0193158634 Fix some chromium-style warnings in webrtc/modules/pacing/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1902005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:18:19 +00:00
pbos@webrtc.org
f3e4ceee47 Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1904005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:17:19 +00:00
pbos@webrtc.org
8f23df51d4 Fix some chromium-style warnings in webrtc/modules/remote_bitrate_estimator/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1905004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:16:52 +00:00
pbos@webrtc.org
4fac8a4699 Fix some chromium-style warnings in webrtc/modules/bitrate_controller/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1903004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:16:20 +00:00
phoglund@webrtc.org
a96d8771f2 Added libjingle_peerconnection_java_unittest to buildbot_tests.py 
The test apparently needs a custom LD_PRELOAD, so I made the script capable of handling custom environments.

TBR=kjellander@webrtc.org
BUG=1796

Review URL: https://webrtc-codereview.appspot.com/1916004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 10:50:30 +00:00
andrew@webrtc.org
0a4ca8f0bb Move internal aec_core defines out of header. 
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1915004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 08:13:08 +00:00
wu@webrtc.org
7446870a0f Suppress failing tests on Linux Memcheck bot. 
BUG=2153
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/1914004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 23:36:42 +00:00
wu@webrtc.org
9c9fc767b1 Fixing the memory check bots by suppressing some of the tests. 
BUG=1205,2078,2080
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1913004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 22:54:08 +00:00
wu@webrtc.org
933946ac55 Suppress libjingle_peerconnection_unittest failures on linux memcheck build bot. 
BUG=2153
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1912004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 22:28:29 +00:00
wu@webrtc.org
0342e65f8d Disable peerconnection tests that are failing on memcheck. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1910006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 22:28:14 +00:00
wu@webrtc.org
ae7bf1525b Disable p2p tests that are failing on memory test. 
BUG=1972
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1911004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 21:40:39 +00:00
fischman@webrtc.org
b59c6dd397 Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal). 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 19:34:07 +00:00
fischman@webrtc.org
85f07f59ee PeerConnectionTest.java: use java_home gyp var instead of hardcoding /usr. 
BUG=1796
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 18:11:07 +00:00
turaj@webrtc.org
fd7e3c52d8 Correcting Turaj's email. 
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1910004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 17:25:07 +00:00
fischman@webrtc.org
3d496fb046 Roll chromium_revision 205140:214260 to pick up build fixes for ninja iOS device build. 
TESTED=git try
BUG=2106
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1888005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 17:14:35 +00:00
henrike@webrtc.org
9638564340 Adds no parent to talk folder. 
BUG=1933
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1896004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 15:51:54 +00:00
pbos@webrtc.org
7f7162a003 Fix some chromium-style warnings in webrtc/modules/video_coding/ 
BUG=163
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1901005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 15:18:31 +00:00
pbos@webrtc.org
e6c3966530 Fix some chromium-style warnings in webrtc/test/ 
BUG=163
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1907004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 13:08:38 +00:00
pbos@webrtc.org
a6f56acc53 Fix some chromium-style warnings in webrtc/tools/ 
BUG=163
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1908004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 12:50:59 +00:00
pbos@webrtc.org
096515b070 Fix some chromium-style warnings in webrtc/modules/audio_device/ 
BUG=163
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1897005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 12:32:59 +00:00
braveyao@webrtc.org
10bbfeff5b Apprtc: add 'event' parameter to onkeydown event handler. 
BUG=
TEST=Manual test
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1898005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-30 09:27:49 +00:00
agalusza@google.com
d818dcb939 Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData. 
R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1841004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 21:48:11 +00:00
henrike@webrtc.org
a0b2f1794b Adds files still expected by the libjingle bots. 
BUG=2146
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1897004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 21:34:08 +00:00
fischman@webrtc.org
d6134c7cfd PeerConnectionTest.java: make the test work for the bots' v4l2loopback. 
- Make the test agnostic to the actual resolution used, since v4l2_file_player
  is playing a non-640x480 file (go/httfw)
- Teach DeviceInfoLinux::FillCapabilityMap() about I420 since that's what
  v4l2_file_player is feeding.

Requires https://gist.github.com/fischman/2e9a9b2efd2ad363ef82 be applied to the
v4l2loopback driver code.

BUG=1796
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1891004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 20:43:15 +00:00
fischman@webrtc.org
147d44a450 AppRTCDemo: replace the use of query-string parameters for pre-JB devices. 
Replaces the use of a query-string parameter with a (once-per-session)
JS-to-Java function call, because query-string parameters on file:// URLs are
busted on ICS and earlier Android releases
(https://code.google.com/p/android/issues/detail?id=17535).

Also added channel.html to the list of inputs to cause edits to it to cause a
rebuild of the .apk.

BUG=1949
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1890004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 19:07:33 +00:00
niklas.enbom@webrtc.org
7694562805 Land http://webrtc-codereview.appspot.com/1632005/ 
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1895004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 18:37:32 +00:00
henrike@webrtc.org
ea40bd0cc8 Presubmit script for preventing changes to protected files and add the full list of those files. 
BUG=2090
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1855004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 18:20:07 +00:00
elham@webrtc.org
c0aa29c98c Updated WebRTC version to 3.37 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1894004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 16:57:21 +00:00
phoglund@webrtc.org
8400246fce Improved error messages when binaries are missing. Also stderr = stdout now. 
Now that these scripts are called from browser tests, we need to print everything on stdout since the tests will throw away stderr when invoking programs. I chose to assign sys.stderr to sys.stdout. Otherwise I would have missed stuff like parser.error, which print to stderr.

The error message will get improved because the old code did not catch the case when the binary was missing, which lead to a very confusing error when that was the case. This gets fixed now.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1886004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 11:01:03 +00:00
kma@webrtc.org
f87177a757 To fix a bug in InverseFFTAndWindow() function in AECM. 
It's a bufer overwritting issue, and thus Android AppRTCDemo app was broken (reported by Ami).
Tested with audioproc offline test. Bit-exact.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 23:43:33 +00:00
henrike@webrtc.org
1e09a71126 Update talk folder to revision=49952949 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 19:17:59 +00:00
fischman@webrtc.org
367f640bea webrtc/.gitignore: add parts of talk/examples/android and third_party/llvm to the list. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1887004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 17:50:10 +00:00
kma@webrtc.org
b6a6a24fda Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()". 
Tested with audioproc. Bit exact.

R=andrew@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/1859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 16:24:34 +00:00
braveyao@webrtc.org
b6433b7a1e Access receiving_ under receive_cs critical section 
Note: InsertRTPPacket/InsertRTCPPacket could be merged into 
ReceivedRTPPacket, as there are no other callers.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 09:02:46 +00:00
sergeyu@chromium.org
abab1d8456 Don't set clang_use_chrome_plugins in common.gypi 
This caused a failure on chrome os ASAN bots (where that flag is disabled):
http://build.chromium.org/p/chromium.memory/builders/Chromium%20OS%20%28x86%29%20ASAN/builds/5491

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1882004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 00:55:46 +00:00
henrike@webrtc.org
14c966c706 Fixes resources and data path in modules_unittests.isolate. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 22:44:04 +00:00
andrew@webrtc.org
b86fbaf1d4 Downstream latest Chromium SincResampler changes. 
Replace the BlockSize() workaround we were using previously to support
the push wrapper with the upstream request_frames interface. This
requires a bit of a trick to ensure we don't add more delay than
necessary. On the first pass we use a dummy Resample() call in order to
prime the buffer such that all later calls only require a single input
request through Run().

Notably, this brings in an optimized loop condition, improving
performance by ~2% - 3% on tested platforms and avoids a 20% performance
hit with clang. This addresses issue2041.

Only negligible changes to the PushSincResamplerTest SNR thresholds, due
to a fractional sample adjustment in output delay.

This still retains the per-instance CPU detection, as webrtc lacks a
LazyInstance helper for static initialization.

Ideally, we would adopt SetRatio() in PushSincResampler's
InitializeIfNeeded() for on-the-fly changes, but this will require a way
to update request_frames.

The diff against Chromium upstream is available here:
https://codereview.chromium.org/19470003

BUG=2041
TESTED=unit tests, voe_cmd_test in loopback running through all codecs
with 44.1 kHz and 48 kHz device formats using a stereo mic.

R=dalecurtis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1838004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 22:04:30 +00:00
fischman@webrtc.org
e691b4f952 Roll libvpx 211873:212975 to pick up build fixes for ninja iOS device build. 
(this originally landed in r4391 and was reverted in r4399 on suspicion of
breaking the mac bots; relanding just libvpx without rolling chromium to isolate
the problem).

BUG=2106
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/1877004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 21:22:41 +00:00
fischman@webrtc.org
6439afce5b Revert 4403 "Roll chromium_revision 205140:212975 to pick up bui..." 
Broke bot: http://chromegw.corp.google.com/i/client.webrtc/builders/Mac32%20Release/builds/312

> Roll chromium_revision 205140:212975 to pick up build fixes for ninja iOS device build.
> 
> (this originally landed in r4391 and was reverted in r4399 on suspicion of
> breaking the mac bots; relanding just the chromium roll without rolling libvpx
> to isolate the problem).
> 
> BUG=2106
> R=marpan@google.com
> 
> Review URL: https://webrtc-codereview.appspot.com/1878004

TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 21:02:21 +00:00
fischman@webrtc.org
60e4b0e472 Roll chromium_revision 205140:212975 to pick up build fixes for ninja iOS device build. 
(this originally landed in r4391 and was reverted in r4399 on suspicion of
breaking the mac bots; relanding just the chromium roll without rolling libvpx
to isolate the problem).

BUG=2106
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/1878004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 20:49:59 +00:00
sergeyu@chromium.org
099b8c9e8e Update include paths in device_info_external.cc 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1875004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 18:41:43 +00:00
andrew@webrtc.org
61e596fc49 Add a Config class interface to AudioProcessing for passing options. 
Pass the Config down to all AudioProcessing components.

Also add an EchoCancellationImplWrapper to optionally create different
EchoCancellationImpls.

BUG=2117
TBR=turaj@webrtc.org
TESTED=git try

Review URL: https://webrtc-codereview.appspot.com/1843004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 18:28:29 +00:00
turaj@webrtc.org
c46967dc53 Revert 4391 "Roll chromium 205140:212975 to support ninja iOS ar..." 
r4391 results in Mac Release Bot fail: http://chromegw/i/internal.client.webrtc/builders/Mac32%20Release/builds/334/steps/modules_integrationtests


> Roll chromium 205140:212975 to support ninja iOS armv7 build.
> 
> In particular, picks up new clang, libvpx, libsrtp, yasm, and gyp.
> 
> TESTED=git try on patchset #1
> BUG=2106
> R=henrike@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1849005

TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1874004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 17:48:24 +00:00
niklas.enbom@webrtc.org
8e3bbedacd Fix include path in video_capture_external.cc 
Fix build error introduced in r4337

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1873004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 16:55:58 +00:00
fischman@webrtc.org
33584f942c Introduce a bit of sanity to talk/PRESUBMIT.py's license checking. 
The comma this allows is a very common variant of the license header (3:1
preferred over the no-comma variant in talk/).

Also pacify pylint a bit, and correct a flagrantly incorrect header I happened
to come across.

BUG=2098,2133
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1866004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 16:43:30 +00:00