Disable a test in libjingle_peerconnection_unittest for DrMemory

BUG=3453
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6334 4adac7df-926f-26a2-2b94-8c16560cd09d
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.

This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND

This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation.

BUG=3348,3353
TESTED=trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
The correct fix of workaround in r6261.

The CL also includes same changes to filterbanks.c in iSAC fix and aecm_core_c.c

BUG=3370,3395,3439
TESTED=trybots
R=fdegans@chromium.org, glaznev@webrtc.org, kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6337 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove static initializer from WebRtcVideoEngine2.

BUG=
R=pliard@google.com, pthatcher@webrtc.org, pliard@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding back platform specific renderer to video loopback test.

BUG=3039
TEST=locally on Mac and Win, video_loopback test
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6339 4adac7df-926f-26a2-2b94-8c16560cd09d
NetEq: Add thread annotation to const scoped_ptrs

Since the objects pointed to are not const, only the pointer to them,
they too must be accessed under lock.

Move the crit_sect to above the variables it is protecting.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12679006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6340 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixes and enables SystemDelayTests.

The root cause for failure was that the delay handling of reported delays was bypassed on Android, whereas the tests assumes that part of AEC to be run.
This CL checks if it is in use.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6341 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove remaining samples (AppRTC) since moved to Github

In r5871 the samples directory was removed since they've now
moved to GitHub at https://github.com/GoogleChrome/webrtc

AppRTC needed to be kept in here (restored in r5873) since
automated tests in Chromium pulled AppRTC.
Now that a Chromium mirror has been setup for the GitHub repo
and that the automated tests have been updated, we can remove
this once and for all.

BUG=chromium:362483
TEST=None, but the automated tests have been verified syncing
the new location.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6342 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 6341 "Fixes and enables SystemDelayTests."

> Fixes and enables SystemDelayTests.
> 
> The root cause for failure was that the delay handling of reported delays was bypassed on Android, whereas the tests assumes that part of AEC to be run.
> This CL checks if it is in use.
> 
> BUG=3445
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12689005

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6343 4adac7df-926f-26a2-2b94-8c16560cd09d
Opus send rate overflows if over 65 kbps

The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.

I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.

BUG=3267
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d
AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct.

BUG=3407
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16619006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6345 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.

Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.

Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.

BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc

Review URL: https://webrtc-codereview.appspot.com/14559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
Rebase webrtc/base with r6345 version of talk/base:
cd webrtc/base
svn diff -r 6249:6300 http://webrtc.googlecode.com/svn/trunk/talk/base >
6300.diff
patch -p0 -i 6300.diff
ls genericslot* | xargs rm
cp ../../talk/base/sigslottester* .
manual edits of sigslottester* to get rid of talk and talk_base.

BUG=3379
TBR=jiayang

Review URL: https://webrtc-codereview.appspot.com/19649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6347 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 68646004-> 68648993

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6348 4adac7df-926f-26a2-2b94-8c16560cd09d