Set number of temporal layers for VideoSendStream.

Introduces a mapping between EncoderConfig and VideoCodec. More
specifically it also removes an assert that there should be no set
temporal layers in the new API, which is wrong and was temporary.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25619004

git-svn-id: http://webrtc.googlecode.com/svn@7256 4adac7df-926f-26a2-2b94-8c16560cd09d
Clang-format ns_core

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29539004

git-svn-id: http://webrtc.googlecode.com/svn@7257 4adac7df-926f-26a2-2b94-8c16560cd09d
Converting five tests to use new AudioCoding interface

The converted tests are:
AcmIsacMtTest
AcmReceiverBitExactness
AcmSenderBitExactness
AudioCodingModuleMtTest
AudioCodingModuleTest

In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old tests were copied and given the
suffix OldApi:
AcmIsacMtTestOldApi
AcmReceiverBitExactnessOldApi
AcmSenderBitExactnessOldApi
AudioCodingModuleMtTestOldApi
AudioCodingModuleTestOldApi

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23719004

git-svn-id: http://webrtc.googlecode.com/svn@7258 4adac7df-926f-26a2-2b94-8c16560cd09d
Convert AcmReceiverTest to new AudioCoding interface

In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old test was copied to
AcmReceiverTestOldApi.

Modified and extended AudioCoding and the implementation to make the
test compile and run.

Created a converter method from new to old config struct

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31409004

git-svn-id: http://webrtc.googlecode.com/svn@7259 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258).

Breaks android modules_unittests tests by crashing on AcmReceiverBitExactness.8kHzOutput
Was already visible on "git cl try" before submitting on https://webrtc-codereview.appspot.com/23719004/#

BUG=3520
R=kwiberg@webrtc.org, henrik.lundin@webrtc.org
TBR=kwiberg@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25629004

git-svn-id: http://webrtc.googlecode.com/svn@7260 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding webrtc_video_streaming test
This test is streaming video and audio between two bots using webrtc js api.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28469004

git-svn-id: http://webrtc.googlecode.com/svn@7261 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove Get/SetNetEQPlayoutMode APIs

These are not used anymore.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26569004

git-svn-id: http://webrtc.googlecode.com/svn@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding test file path as argument of the rtcBot run command's arguments.

The new command to run rtcBot is:-
node test.js <bot_type> <test_file_path>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31419004

git-svn-id: http://webrtc.googlecode.com/svn@7263 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert "Converting five tests to use new AudioCoding interface" (rev 7258).

This time reverts the Cl that actually broke the tests. Got the wrong rev before. :/

BUG=3520
TESTED=Locally with CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AcmReceiverBitExactness.8kHzOutput --verbose --isolate-file-path=webrtc/modules/modules_unittests.isolate
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26579004

git-svn-id: http://webrtc.googlecode.com/svn@7264 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove engine-level SetOptions.

Already removed in WebRtcVideoEngine.

R=andresp@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29549005

git-svn-id: http://webrtc.googlecode.com/svn@7265 4adac7df-926f-26a2-2b94-8c16560cd09d