Fix constness of AudioBuffer accessors.

Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.

Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/

R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
Only clamp to 16 kHz when AECM is enabled.

Otherwise we could needlessly downsample to 16 kHz (rather than 32 kHz)
when HW AEC is used.

BUG=3259
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6033 4adac7df-926f-26a2-2b94-8c16560cd09d
Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels.

This fixes an issue where the user doesn't know which channels are "active" and therefore can't properly sum the estimates for all channels.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6041 4adac7df-926f-26a2-2b94-8c16560cd09d
Pointers were not dereferenced in GetRtpStatistics.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6042 4adac7df-926f-26a2-2b94-8c16560cd09d
Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test.

BUG=3289
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6043 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement the Windows screen capturer using the Magnification API.
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.

BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6048 4adac7df-926f-26a2-2b94-8c16560cd09d
Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest.

Fixes building with gcc 4.8.

TBR=fdegans@google.com
BUG=chromium:321833

Review URL: https://webrtc-codereview.appspot.com/12439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6050 4adac7df-926f-26a2-2b94-8c16560cd09d
WebRTCDemo: correct set trace filter operation.

BUG=3285
TEST=Manul Test
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6051 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 6048 "Implement the Windows screen capturer using the Mag..."

> Implement the Windows screen capturer using the Magnification API.
> The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
> 
> BUG=2789
> TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
> R=sergeyu@chromium.org, wez@chromium.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12149004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6052 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement the Windows screen capturer using the Magnification API.
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.

BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org, wez@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6048

Review URL: https://webrtc-codereview.appspot.com/12149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6053 4adac7df-926f-26a2-2b94-8c16560cd09d
* 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/webrtc@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/webrtc@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/webrtc@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
Update makefiles after merge of Chromium at 268379

This commit was generated by merge_from_chromium.py.

Change-Id: I0c7c73b002090ffaba1abc7940cb54b304976373
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/webrtc@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
video_render.gypi: clean up some libraries directives to be more specific.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6068 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix odd codes in video_capture on Mac.

BUG=3272
TEST=vie_auto_test
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6070 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 39d9fa5486157fb4b3ab28ae403aeaa6d651e92b

This commit was generated by merge_from_chromium.py.

Change-Id: I1cebd150247452024503e31c64ce6057c4b75b2a
Update makefiles after merge of Chromium at 268710

This commit was generated by merge_from_chromium.py.

Change-Id: I8ceb035a91083a340578d5f3a800eecad41a4e46
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 519f74e6b0c8a59db50e58158331f8922c4509b9

This commit was generated by merge_from_chromium.py.

Change-Id: I361f2e6793284ff89ccc93be3cebfb7959593e22