Removing ASSERT for tcp candidate for port 0 and 9, as Android clients
may not be called with set_allow_tcp_listen(false).

This CL will also sends tcp candidate in RFC 6544 format.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3677
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6880 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.

BUG=3679
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6881 4adac7df-926f-26a2-2b94-8c16560cd09d
NetEq background noise generation off by default

This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable.

With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise.

Reference files are updated where needed.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
Update checksums for AcmReceiverBitExactness on android

This should have been a part of r6882.

BUG=3519
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6883 4adac7df-926f-26a2-2b94-8c16560cd09d
Removing TODOs related to AcmReceiverBitExactness checksums

Should have been part of r6883.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6884 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the audio source failure due to unsupported constraints.

Some constraints, like kEchoCancellation, kMediaStreamAudioDucking are supported in Chrome but not in Libjingle, if the users set it in mandatory, LocalAudioSource::Initialize() will fail the getUserMedia call.

This patch fixes the problem by fully initializing the LocalAudioSource even though some constraints are not supported in libjingle.

BUT=crbug/398080
TEST=manual test:
var constraints = {audio: { mandatory: { googEchoCancellation: true } }};
getUserMedia(constraints, gotStream, gotStreamFailed);
verify you get a gotStream callback

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

Review URL: https://webrtc-codereview.appspot.com/21049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6885 4adac7df-926f-26a2-2b94-8c16560cd09d
Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4).
Increased kStandardRampUpDelayMs (30 to 40s).

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6886 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 73072800 -> 73215194

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6887 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 73215194-> 73221069

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6888 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 73221069-> 73222930

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6889 4adac7df-926f-26a2-2b94-8c16560cd09d
Further DrMemory suppressions, likely from r6811

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6890 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 73222930-> 73226398

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixes failure triggered by include order re-ordering.

BUG=N/A
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6892 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove files from talk/PRESUBMIT.py blacklist.

Many files can now be submitted here and do not have to be rolled in.

BUG=
R=henrike@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6893 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove TODO saying to remove WebRtcVideoFrame.

Comment was added prematurely, there's no decision to get rid of this
type.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6894 4adac7df-926f-26a2-2b94-8c16560cd09d
Make sure that muting muted streams succeeds.

We don't want to report an error here, and PeerConnection relies on
being able to mute already-muted streams (I hit an assert when testing
manually).

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6895 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 73248599-> 73249894

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6896 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 6863 "Refactor StatsCollector and associated types."

Breaks chrome compilation on Mac:

/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
          _Tp __x_copy = __x;
              ^          ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
          _M_insert_aux(end(), __x);
          ^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
  reports.push_back(report1);
          ^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
  StatsReport() : timestamp(0) {}



> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
> 
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> 
> R=xians@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15119004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 73256845-> 73260148

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6898 4adac7df-926f-26a2-2b94-8c16560cd09d