d212ffcfc6
Remove unnecessary build message.
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6660 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 11:15:35 +00:00
4ef438e2de
Remove the send-side cname getter APIs from voice and video engine.
...
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
72491b9a90
Count total bytes sent in RTPSender::Bytes().
...
Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 16:24:54 +00:00
0422100818
Fix data race in VCMTiming::ResetDecodeTime.
...
Also thread annotating class.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6653 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 15:25:37 +00:00
8f1512140e
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 09:39:23 +00:00
5bde66e913
audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h
...
The change of definitions moved to aec_common.h was done in CL17839005.
BUG=3131
TBR=kwiberg@webrtc.org
TESTED=builds locally
Review URL: https://webrtc-codereview.appspot.com/16859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6648 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 08:09:50 +00:00
555fc78f27
Neon version of SubbandCoherence()
...
The performance gain on a Nexus 7 reported by audioproc is ~1.4%
The output is NOT bit exact. Any difference seen is +-1.
BUG=3131
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17839005
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6647 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 08:03:11 +00:00
ac800c8004
Neon version of rftbsub_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~4.5%
The output is bit exact.
BUG=3131
TESTED=trybots and manually
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19919005
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6646 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 07:53:13 +00:00
5ac876bae0
Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).
...
Reason breaks linux_memcheck.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 07:41:59 +00:00
47d1c98a4e
Remove remains of WEBRTC_NO_STL.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6641 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 20:18:28 +00:00
10ef8fe611
Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault.
...
BUG=crbug/385294
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6640 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 19:41:32 +00:00
b753762ce6
delay_estimator: Increases test coverage and makes input spectrum const
...
Noticed lack in tests verifying initial state is not left if we have zero input spectra. This CL adds such a test and change input spectra to const at affected places.
BUG=N/A
TESTED=trybots and manually
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6631 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 06:40:09 +00:00
12b4efefdd
Implement a work around for Chrome full-screen tab switch on Mac.
...
Chrome creates a new window in full-screen and minimizes the old window when a tab is switched to full-screen.
We try to find the new window to continue capturing for window sharing.
BUG=crbug/385294
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/19839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 22:05:24 +00:00
e55641d4f7
Neon version of rftfsub_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~3.3%
The output is bit exact.
BUG=3131
TESTED=trybots and manually on N7
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14819004
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6628 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 21:12:23 +00:00
d11bec40b2
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 14:32:58 +00:00
62bafae661
Some refactoring inside rtp_rtcp/.
...
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
241a9b0b65
Fixing compile error.
...
Made a mistake in https://webrtc-codereview.appspot.com/13849004/ ,
fixing that here.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6622 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 11:48:37 +00:00
22292df53b
Adding explicit check for using dummy file devices.
...
Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 11:39:19 +00:00
2bb1bdab8d
Preserve RTP states for restarted VideoSendStreams.
...
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 13:06:48 +00:00
73823cafa4
Add initial gn build files for video_coding and video_processing.
...
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
03c817e405
Fix pacer to accept duplicate sequence numbers on different SSRCs.
...
BUG=3550
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 10:20:35 +00:00
7832648824
Add missing break introduced in r6603.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6607 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 17:04:00 +00:00
bee164a214
Fix test issues and a win compile error introduced with r6605.
...
Also changes the name of a variable which has been hijacked by windef.h (included by windows.h), which forces #define near and #define far upon us. This issue was introduced via the following inclusion chain:
bwe_test_framework_unittest.cc includes
paced_sender.h
tick_util.h
windows.h
windef.h
And causes EXPECT_NEAR(foo, bar, near); to expand to EXPECT_NEAR(foo, bar,); generating a very confusing compile error.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 17:00:06 +00:00
875ad49dee
Revert conversion from TickTime to int64_t in paced sender.
...
Introduced with r6600, causing flakes in SuspendBelowMinBitrate. The reason for this flake is currently unknown.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 15:27:55 +00:00
b9f5453e29
Add boilerplate code for H.264.
...
R=mflodman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 12:42:07 +00:00
20c1f56992
Configure RTX send status on new modules.
...
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).
As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 10:58:12 +00:00
88e0dda475
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 09:20:42 +00:00
720964faac
Fix memcheck error in r6594.
...
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6596 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 21:14:07 +00:00
c8364539d3
Fix for FEC decoding with sequence number wrap-around.
...
BUG=3507
R=stefan@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6594 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 16:49:30 +00:00
69ef9911e4
delay_estimator: Allows dynamically used history sizes
...
Gives the user a possibility to dynamically change the history size. The main advantage is, for example, that you now can start with a wide delay range and over time decrease the search window to lower complexity.
Adds
- two new APIs.
- and updates unit tests.
- a history_size member variable to BinaryDelayEstimator.
- two help function re-allocating buffer memory.
One thing that makes this a little complicated is that you are allowed to have multiple delay estimators with the same reference, so changing the buffer sizes at one place will automatically give you a mismatch at other places.
BUG=3532, 3504
TESTED=trybots and manually
R=aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6593 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 14:59:03 +00:00
224a140339
Make experimental NS API not purely virtual
...
Because not all subclasses will want to bother overriding these methods.
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 13:41:39 +00:00
c0ba4392f1
common_audio: Removes macro WEBRTC_SPL_SHIFT_W16
...
We should avoid macros in general (see style guide). This shift macro is not a severe one, since there is a check for negativity.
BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 13:38:53 +00:00
38214d53db
EchoCancellationImpl::ProcessRenderAudio: Use float samples directly
...
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.
Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.
R=aluebs@webrtc.org , andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 09:47:33 +00:00
93426cd2ff
Implement BUILD.gn for desktop_capture.
...
BUG=3441
R=brettw@chromium.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6584 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-02 15:47:12 +00:00
74bf7a6523
Add tkchin@ to OWNERS.
...
Adding myself to OWNERS of subdirectories containing iOS bits. Added niklas.enbom@ for audio_device and wu@ for everything else.
R=niklas.enbom@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 18:38:28 +00:00
9138eb649b
Fix compile error introduced with r6571.
...
TBR=mflodman@webrtc.org
BUG=3527
Review URL: https://webrtc-codereview.appspot.com/20799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6572 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 12:44:05 +00:00
5779ca478d
Fixes a potential BWE clock mismatch bug.
...
Since libjingle provides a packet arrival timestamp to webrtc, and the clock in remote bitrate estimator and the clock used for packet arrival timestamp can be different. This can cause the bandwidth estimator to malfunction.
This CL changes the remote bitrate estimator so that packet arrival timestamps never are compared to the time taken from the internal clock.
BUG=3527
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 12:14:49 +00:00
6d21ddca5f
audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations
...
Puts functionality necessary to calculate sub-band coherences into a function.
BUG=3131
TESTED=trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6570 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 10:03:42 +00:00
59adb1dcd7
Neon version of cftmdl_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~2.3
The output is bit exact.
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19829004
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6568 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-30 19:34:33 +00:00
9825afc3bd
Add ExperimentalNs support in Config
...
R=andrew@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6567 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-30 17:39:53 +00:00
cd9b90ab53
Neon version of cft1st_128()
...
The performance gain on a Nexus 7 reported by audioproc is ~2%
See comments regarding the output.
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21679004
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-30 12:05:18 +00:00
ae7cfd7bc8
Make MediaOptimization thread-safe.
...
HW encoder posts the encode callback to libjingle worker
thread. It accesses MediaOptimization and is not protected
by the critial section of VideoSender. Make MediaOptimization
thread-safe to fix it.
BUG=chromium:367691
TEST=Run apprtc loopback with SW or HW encoders.
Run module_unittests.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-30 08:01:47 +00:00
c8e98187d1
Receiver bit-exactness test for AudioCoding Module
...
This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.
The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-26 19:07:04 +00:00
aa0e56e8e8
Fixes a bug causing NACKs to be dropped excessively at the send-side.
...
This was introduced in r6472 where the target bitrate was changed to be stored in bits/s instead of kbits/s, but the storage type was accidentally left as uint16_t. This caused the bitrate to be truncated, which at times causes NACKs to be dropped due to insufficient bitrate available.
BUG=3518
TEST=Tested in Chrome, trybots and verified that it fixes the bug in vie_auto_test loopback test.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-26 11:44:49 +00:00
fe526ff10f
fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions.
...
BUG=N/A
R=pwestin@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 20:59:51 +00:00
3b84b3a58c
Add RTCP packet types to packet builder:
...
REMB, TMMBR, TMMBN and
extended reports: RRTR, DLRR, VoIP metric.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9299005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 12:22:17 +00:00
6568e97d10
This is to compare NetEq with various codecs under a shared packet loss pattern.
...
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 12:17:41 +00:00
d5075bdbb5
Neon version of FilterFar()
...
The performance gain on a Nexus 7 reported by audioproc is ~3.5%.
The output is bit exact.
BUG=3131
TESTED=verified performance manually, passed trybots
R=bjornv@webrtc.org , cd@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12799005
Patch from Scott LaVarnway <slavarnw@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 12:15:46 +00:00
1ed1af9b31
Remove payload duplication in AudioDecoderTest
...
This hack was made to come around issue 845. Now that is solved, and
the test code can be cleaned up.
BUG=845
R=kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6534 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25 07:59:40 +00:00
eecf5e6ba7
Removing neteq decode lock and friends
...
NetEq is thread-safe by virtue of it's own lock, and in r6404 the
ACMISAC class was made thread-safe. Therefore, the neteq decode lock
is no longer needed.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-24 13:11:22 +00:00
