* Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter.
* Re-enable test CaptureNtpTimeWithNetworkJitter.
* Use 100ms as the threadhold as a FYI since this is a performance test.

BUG=3271
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6054 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove ALLOW_UNUSED.

Turns out Chromium won't be applying this to COMPILE_ASSERT. We don't
need it at all then.

R=thakis@chromium.org
TBR=thakis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6055 4adac7df-926f-26a2-2b94-8c16560cd09d
Allow the RTP level indicator computation to work at any sample rate.

Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.

We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.

This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:

[ RUN      ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27

BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 66138442-> 66236292

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6057 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove timestamp_extrapolator's dependency to Clock and vcm defines.

TEST=existing tests
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
Update svn:ignore for resources and third_party.

This prevents the following from getting wiped on each build:
resources/*.dat
third_party/binutils
third_party/libc++
third_party/libc++abi
third_party/openmax_dl



git-svn-id: http://webrtc.googlecode.com/svn/trunk@6059 4adac7df-926f-26a2-2b94-8c16560cd09d
Rolling new version of Opus.gyp

The new version enables optimizations on iOS.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6060 4adac7df-926f-26a2-2b94-8c16560cd09d