Wire up bandwidth stats to the new API and webrtcvideoengine2.

Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove uses of build date/time.

Uses of __DATE__ and __TIME__ are blocking deterministic Chromium
builds. We're not really making use of these, and if anything they're
likely to be misleading as it's impossible to distinguish between a new
revision and a freshly-built old branch.

R=mflodman@webrtc.org, tnakamura@webrtc.org
BUG=3983

Review URL: https://webrtc-codereview.appspot.com/27039004

git-svn-id: http://webrtc.googlecode.com/svn@7635 4adac7df-926f-26a2-2b94-8c16560cd09d
This fixes a small memory leak (found using Xcode/Instruments on iOS) in
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient

BUG=3985
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28959004

Patch from Matthias Liebig <matthias.gcode@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28949004

git-svn-id: http://webrtc.googlecode.com/svn@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
Adjust parameter in vp9 rate control test.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25099004

git-svn-id: http://webrtc.googlecode.com/svn@7638 4adac7df-926f-26a2-2b94-8c16560cd09d
Exclude SendsAndReceivesVP9 for WinDrMemory.

https://code.google.com/p/webm/issues/detail?id=872

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28989004

git-svn-id: http://webrtc.googlecode.com/svn@7639 4adac7df-926f-26a2-2b94-8c16560cd09d
Change DrMemory exclusion to match changed test name.

Due to:
https://code.google.com/p/webrtc/source/detail?r=7574

TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27049004

git-svn-id: http://webrtc.googlecode.com/svn@7640 4adac7df-926f-26a2-2b94-8c16560cd09d
Exclude SendsAndReceivesVP9 for linux-memcheck.

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30899004

git-svn-id: http://webrtc.googlecode.com/svn@7641 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix memcheck and dr memory after flakiness dashboard deployment.

I deployed buildbot configuration for uploading data to the
flakiness dashboard but I didn't predict our Dr Memory and
Memcheck bots would fail due to the new flag passed to the test.
Adding the --gtest_output flag to the script will avoid the build
to fail as a workaround.

TBR=andrew@webrtc.org
TESTED=Passing test run using:
src/tools/valgrind-webrtc/webrtc_tests.sh --test audio_decoder_unittests --tool memcheck --target Release --build-dir src/out --gtest_output=xml:audio_decoder_unittests.xml

BUG=

Review URL: https://webrtc-codereview.appspot.com/28999004

git-svn-id: http://webrtc.googlecode.com/svn@7642 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 7642 "Fix memcheck and dr memory after flakiness dashboar..."

Reason: Turns out this does not solve the problem as the buildbots
run into another error later on during collecting gtest output.
The problem is solved by excluding these bots from flakiness dashboard
data instead, in https://codereview.chromium.org/705913002/

> Fix memcheck and dr memory after flakiness dashboard deployment.
> 
> I deployed buildbot configuration for uploading data to the
> flakiness dashboard but I didn't predict our Dr Memory and
> Memcheck bots would fail due to the new flag passed to the test.
> Adding the --gtest_output flag to the script will avoid the build
> to fail as a workaround.
> 
> TBR=andrew@webrtc.org
> TESTED=Passing test run using:
> src/tools/valgrind-webrtc/webrtc_tests.sh --test audio_decoder_unittests --tool memcheck --target Release --build-dir src/out --gtest_output=xml:audio_decoder_unittests.xml
> 
> BUG=
> 
> Review URL: https://webrtc-codereview.appspot.com/28999004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31969004

git-svn-id: http://webrtc.googlecode.com/svn@7643 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove the state_ member from AudioDecoder

The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.

Two small but not quite trivial cleanups are included because they
blocked the state_ removal:

  - AudioDecoderG722Stereo now inherits directly from AudioDecoder
    instead of being a subclass of AudioDecoderG722.

  - AudioDecoder now has a CngDecoderInstance member function, which
    is implemented only by AudioDecoderCng. This replaces the previous
    practice of calling AudioDecoder::state() and casting the result
    to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
    plainly visible in the AudioDecoder class declaration.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24169005

git-svn-id: http://webrtc.googlecode.com/svn@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
Advertise G722 as 8 kHz rather than 16 kHz

G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.

R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.

Review URL: https://webrtc-codereview.appspot.com/27879004

git-svn-id: http://webrtc.googlecode.com/svn@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation.

Also removed some unused "summary" ListPreference
fields.

The looks of it can be found in [1] (lowest row).

[1] https://drive.google.com/file/d/0By6DR2QIwc_ZQm9TMW5YVEpsMWc/view?usp=sharing

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27939004

git-svn-id: http://webrtc.googlecode.com/svn@7646 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 79285346-> 79320771

git-svn-id: http://webrtc.googlecode.com/svn@7647 4adac7df-926f-26a2-2b94-8c16560cd09d
Log formatting fix for VideoEncoderConfig.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30889004

git-svn-id: http://webrtc.googlecode.com/svn@7648 4adac7df-926f-26a2-2b94-8c16560cd09d
Removing unused method GetDefaultVideoEncoderConfig.

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

Review URL: https://webrtc-codereview.appspot.com/27959004

git-svn-id: http://webrtc.googlecode.com/svn@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove deprecated PeerConnection APIs.
Removes PeerConnectionObserver::OnError.
Removes MediaConstraints argument to PeerConnection::AddStream.
None of these have ever been implemented and have been removed from the spec.

R=tommi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24189004

git-svn-id: http://webrtc.googlecode.com/svn@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
Volume buttons in AppRTCDemo should affect output audio volume.

BUG=3279
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32399004

git-svn-id: http://webrtc.googlecode.com/svn@7651 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 79326895-> 79329222

git-svn-id: http://webrtc.googlecode.com/svn@7652 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert "Advertise G722 as 8 kHz rather than 16 kHz"

This reverts r7645.

TBR=pthatcher@webrtc.org
BUG=3951

Review URL: https://webrtc-codereview.appspot.com/24199004

git-svn-id: http://webrtc.googlecode.com/svn@7653 4adac7df-926f-26a2-2b94-8c16560cd09d