Disabling SendsSetSimulcastSsrcs.

Disabling as bots are turning red. This should be because
VideoSendStream::ReconfigureVideoCodec caps video_codec.startBitrate to
max bitrates and as the start bitrate is just enough to transmit there
might be some rounding errors here causing the top stream not to be
sent. Since no REMB is received (send-side test) this remains as the
transmit bitrate.

I need some more time to figure out if this is the case so I'm disabling
these for now to avoid reverting the big CL. VideoSendStreams aren't
used in production yet.

TBR=mflodman@webrtc.org
BUG=3078

Review URL: https://webrtc-codereview.appspot.com/10229005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5727 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixes RTX related bugs.

- An RTX packet with no payload should be dropped prior to parsing RTX header since it doesn't have an RTX header. This can for example happen when sending padding-only packets over the RTX stream.
- The retransmit code path when the pacer is disabled doesn't properly update the abs-send-time and ts-offset header extensions.

TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5728 4adac7df-926f-26a2-2b94-8c16560cd09d
Properly account for retransmitted packets when not using the pacer.

This regression was introduced in r5728.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5729 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 63352036-> 63363208

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5731 4adac7df-926f-26a2-2b94-8c16560cd09d
Adjust the captured window rect when the window is maximized.
GetWindowRect includes the window frames for maximized window even they are off screen, causing content outside the window being captured falsely. The fix is to remove the left/right/bottom window frame from the captured rect. Mouse capturing is adjusted accordingly as well.

BUG=3076
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5732 4adac7df-926f-26a2-2b94-8c16560cd09d
Extend perf tests to perform rampup on single stream.

R=kjellander@webrtc.org, stefan@webrtc.org
BUG=3065

Review URL: https://webrtc-codereview.appspot.com/10049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5733 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered.

BUG=none
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5734 4adac7df-926f-26a2-2b94-8c16560cd09d
Prevent playout delay wrap-around in VoiceEngine

In the case where a network glitch causes a packet to arrive so late
that the jitter buffer has gone into expand mode, the playout timestamp
could have been increased to a value that is larger than the RTP
timestamp of the late packet when it finally arrives. This causes
the difference to be negative, and would make the value wrap (unsigned).

With this fix, the difference is set to zero when the playout
timestamp is ahead of the incoming RTP timestamp. Further down in the
method, a zero-value will lead to the averaging filter not being updated.

BUG=3080
R=henrika@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5735 4adac7df-926f-26a2-2b94-8c16560cd09d
Add ability to configure cpu overuse options via an API.

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9299006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5736 4adac7df-926f-26a2-2b94-8c16560cd09d
Add system wrapper dependency to libjingle targets.
This is necessary to handle usage of STR_CASE_CMP in
common_types.h ( as in https://webrtc-codereview.appspot.com/10099005/)

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5737 4adac7df-926f-26a2-2b94-8c16560cd09d
Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets

This is a relanding of r5725, now with a fix for the failing tests.

BUG=2935
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5738 4adac7df-926f-26a2-2b94-8c16560cd09d
Use codec width/height as the encoded_image width/height.

The raw_->w and raw_->h which are the stored image width/height may not be the encoded image size in the case when the incoming frame has a odd size.

R=marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5739 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 63363208-> 63493960

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5740 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert "(Auto)update libjingle 63363208-> 63493960" (r5740)

BUG=N/A
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5741 4adac7df-926f-26a2-2b94-8c16560cd09d
ARGBScale fix for bilinear down sampling overread when source size is odd.
BUG=chromium:352592
TESTED=drmemory on libyuv new unittest scaling unittest.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5742 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 63493960-> 63503990

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5743 4adac7df-926f-26a2-2b94-8c16560cd09d