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Author SHA1 Message Date
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
braveyao@webrtc.org
ebdfa8dcba Add fischman into OWNERS of WebRTCDemo Android. 
BUG=
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3991 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 07:30:38 +00:00
andrew@webrtc.org
d72262dc01 Fix compile errors in ViE with latest clang. 
Rolling to the latest Chromium picks up a new clang, which catches a fresh error:

error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
 VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
  VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
                              ^~~~~~~~~~~~~~~~
                              static_cast

This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).

Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
        AutoTestSleep(std::numeric_limits<long>::max());
        ~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

This fixes the errors and is required before stable can be rolled in Chromium.

TBR=mflodman,andresp

Review URL: https://webrtc-codereview.appspot.com/1450004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 02:12:07 +00:00
andrew@webrtc.org
c6a3755ada Update SincResampler with the latest Chromium code. 
* Brings in on-the-fly sample ratio updates (or varispeed) with minor modifications to build in webrtc.
* Moved SSE and NEON optimized functions into their own files to handle run-time detection properly. NEON optimizations now enabled.

TESTED=unit tests and ran voe_cmd_test loopback with both devices using 44.1 kHz to exercise SincResampler in real-time.
R=dalecurtis@chromium.org, kma@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1438004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 20:35:43 +00:00
andresp@webrtc.org
44272739c2 Clean creation of VideoEngine: 
- clean a static variable just used to debug and not so necessary IMO.
 - clean a really ugly reinterpret cast
 - clean a extern "C" code and loading of dlls which is no longer in use.

Review URL: https://webrtc-codereview.appspot.com/1385006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 19:20:23 +00:00
andrew@webrtc.org
6155be2c61 Add /tools/protoc_wrappers to .gitignore. 
TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/1444004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 18:51:07 +00:00
phoglund@webrtc.org
aeb7d8757d Tweaked webrtc_reformat. 
Fixed variable names such as maskByte and stuff within brackets.

Fixed bug where we would think that for instance foo_internal.h was the self include when the right answer was foo.h.

Removed comment conversion: it was doing more damage than good.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1442005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 13:56:23 +00:00
phoglund@webrtc.org
315d39866e Formatted dtmf_queue. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1398004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 10:04:06 +00:00
kjellander@webrtc.org
73a4d5ab12 Add script to ensure virtual webcam is running. 
This script will check that a webcam is running and start it if it's
not currently running.
It's tailored to the way our buildbots are currently configured.

TEST=local execution on Windows, Mac and Linux.
BUG=none
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1406005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 09:20:41 +00:00
pbos@webrtc.org
f6d67ae21f Disable clang C++11 warnings to permit OVERRIDE keyword. 
BUG=1623
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1431004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 08:34:34 +00:00
stefan@webrtc.org
d98e784f5f Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. 
BUG=1665
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1341004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 06:38:53 +00:00
andrew@webrtc.org
b55a12ad32 Enable protobuf use in Chromium. 
We might end up reverting this, but we need to get it committed and merged to
stable in order to test in a webrtc roll.

TBR=niklas.enbom
BUG=webrtc:830

Review URL: https://webrtc-codereview.appspot.com/1439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 00:03:30 +00:00
andrew@webrtc.org
e53084f837 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 23:19:58 +00:00
niklas.enbom@webrtc.org
3be565b502 Refactoring for typing detection 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 21:04:24 +00:00
stefan@webrtc.org
ef14488d03 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. 
BUG=1663
R=mikhal@webrtc.org, ronghuawu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 19:16:33 +00:00
mikhal@webrtc.org
8f86cc8712 VCM/Receiver: Return null when can't extract frame. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1435004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 18:05:21 +00:00
mikhal@webrtc.org
474e915272 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:55:03 +00:00
mikhal@webrtc.org
759b041019 Relanding r3952: VCM: Updating receiver logic 
BUG=r1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1433004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:36:00 +00:00
mikhal@webrtc.org
9c7685f9a6 VCM/JB: Break and skip to key if possible 
BUG=1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1421004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:07:52 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
stefan@webrtc.org
d3a1959678 Fix jitter buffer unittest. 
TBR=mflodman@webrtc.org
BUG=1737

Review URL: https://webrtc-codereview.appspot.com/1430005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:35:58 +00:00
stefan@webrtc.org
a5dee33639 Correctly add packets to nack list when sequence number wraps. 
BUG=1737
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1427004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 11:11:17 +00:00
pwestin@webrtc.org
0f29810288 Fix crash in pacer. 
BUG=1731
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1410006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 16:37:22 +00:00
stefan@webrtc.org
4ce19b1664 Revert r3952 "VCM: Updating receiver logic" 
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1410005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 13:16:51 +00:00
stefan@webrtc.org
273759048c Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." 
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1408005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 13:12:58 +00:00
xians@webrtc.org
233c58de47 Landing 1399004, Minor clean up on the un-used _measureDelay code 
Those code is/will never used, removing it makes the code better.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@3961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 11:52:47 +00:00
andrew@webrtc.org
59aaebc3cd Add an option to override the TestToStderr trace printout time. 
This is useful for offline file-based tests.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1407004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-05 19:44:19 +00:00
andrew@webrtc.org
f9c289bafe Consolidate all third party licenses in LICENSE_THIRD_PARTY. 
* Add the full license to all third party files.
* Correct some entries in LICENSE_THIRD_PARTY which were missing the full
license.
* Relicense all Chromium-licensed files under WebRTC.
* Remove third_party_mods/, which is now redundant.

R=jan.linden@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1396004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-05 18:54:10 +00:00
elham@webrtc.org
df3da84ec8 Updated WebRTC version number to 3.30 
R=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1404005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 23:11:37 +00:00
mikhal@webrtc.org
45f2da0920 VCM/JB: Porting jitter_buffer_test to gtest. 
Tests were not modified, but ported as is.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1391004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 22:22:46 +00:00
andrew@webrtc.org
a31c428307 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. 
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.

Our ALSA interface always requires 48 kHz, allowing ALSA to handle resampling.

This also removes WEBRTC_PA_GTALK which was not defined anywhere.

BUG=webrtc:1395
TESTED=Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good. Testing AEC was difficult as I can't find a way to change the sample rate of an individual device in PulseAudio. Using a webcam at 32 kHz, other problems were the overriding contribution to quality degradation (delay issues, possible clock drift from the camera). At least I verified that the quality got no worse with this patch.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 19:01:46 +00:00
andrew@webrtc.org
7cb766b016 Remove 44.1 kHz workaround from AudioDevice on WASAPI. 
We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.

BUG=webrtc:1395
TESTED=Set capture device to 44.1 and render device to 48 and vice versa and observed good AEC. The quality is considerably worse before this change. Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1383004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 18:56:38 +00:00