Make RTPSender/RTPReceiver generic.

Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.

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

Review URL: https://webrtc-codereview.appspot.com/26399004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
Add ability to downscale content to improve quality.

BUG=3712
R=marpan@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7164 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix stack limit exceeded in http client.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7165 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix mac video_render implementation on cocoa.

Hit this while playing around with all compile possibilities for issue 3770.

BUG=3770
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7166 4adac7df-926f-26a2-2b94-8c16560cd09d
Disable MethodNotAllowedOnDifferentThreadInDebug.

BUG=3803
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7167 4adac7df-926f-26a2-2b94-8c16560cd09d
linux: remove stray libcrypto dependency

Followup to CL 20049004, which looks like it added an unneeded -lcrypto
on linux.

BUG=3625
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7168 4adac7df-926f-26a2-2b94-8c16560cd09d
Initialize restored_packet in nack_rtx_unittest.cc.

This is to get the DrMemory Full bots to go green, this was previously
suppressed. This fix is likely hiding a real bug that should be
investigated, but it's not a regression from before. The issue should
not be closed before we figure out why this is the case and revert this
"fix".

TBR=stefan@webrtc.org
BUG=3183

Review URL: https://webrtc-codereview.appspot.com/30369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7169 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."

Breaks other repos.

TBR=fbarchard@google.com
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/23639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7170 4adac7df-926f-26a2-2b94-8c16560cd09d
Temporary revert maximum video codec resolution back to 1080p.

BUG=3757, 3738
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7171 4adac7df-926f-26a2-2b94-8c16560cd09d
Add enable flag for Android device orientation change event.

There are reports (not reproducible with appRtcDemo) that
outstanding device orientation change event
OrientationEventListener.onOrientationChanged can be
triggered even after these events are disabled by
OrientationEventListener.disable() code.
Avoid calling native code in this case since underlying
C++ class may have already been deleted.

BUG=3564
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7172 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..."
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*

> Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
> 
> Breaks other repos.
> 
> TBR=fbarchard@google.com
> BUG=N/A
> 
> Review URL: https://webrtc-codereview.appspot.com/23639004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7173 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 75390072-> 75428737

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d
Set minimum SDK level to 10.7 for Mac and iOS.

This is needed since r7174 introduced a dependency
on AVFoundation, which is not present in the 10.6 SDK which is
still the default for Chromium.

BUG=
TESTED=Passing compile on trybots.
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7175 4adac7df-926f-26a2-2b94-8c16560cd09d
Stop building talk/xmllite since it is no longer used.

BUG=3379
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7176 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined

This requires two fixes:
1. Use DCHECK instead of assert in ThreadChecker's unittest.

2. Activate DCHECK when DCHECK_ALWAYS_ON in enabled.

Both these modifications are in line with Chromium's implementation.
The ThreadChecker unittest was changed to use assert instead of DCHECK
on the initial import (since WebRTC did not have a DCHECK back then).

BUG=3803
TEST=local out/{Debug,Release}/rtc_unittests built with and without DCHECK_ALWAYS_ON
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7178 4adac7df-926f-26a2-2b94-8c16560cd09d
Re-enable video send stream tests for android.

BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7179 4adac7df-926f-26a2-2b94-8c16560cd09d
Re-enable rampup_tests.cc for Android.

BUG=3770
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7180 4adac7df-926f-26a2-2b94-8c16560cd09d
Re-enable neteq_performance_unittest.cc for android.

BUG=3770
R=kjellander@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7181 4adac7df-926f-26a2-2b94-8c16560cd09d
audio_processing: Correct sample rate in aec_debug_dump

When writing to wav files in the low level flag aec_debug_dump incorrect sample rates were used for recordings using rates from 32 kHz and above. This since internally inside the AEC we process the data using 16 kHz. Any upper band is processed and combined later on.

This CL adds the correct sample rate to the recording.

BUG=3359
TESTED=locally on 44.1 kHz recordings on Linux
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7182 4adac7df-926f-26a2-2b94-8c16560cd09d
Make BW checks > 0 in peerconnection_unittest.cc.

These checks (> 40k) fail on LSan FYI bots and the purpose of them seem
to be that we're getting non-zero BW reported.

R=stefan@webrtc.org
TBR=jiayl@webrtc.org, solenberg@webrtc.org
BUG=3817,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/29479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7183 4adac7df-926f-26a2-2b94-8c16560cd09d