Handle the case if an unusually long peer name is provided in the peerconnection example.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove the VPM denoiser.

The VPM denoiser give bad results, is slow and has not been used in
practice. Instead we use the VP8 denoiser. Testing this denoiser takes
up a lot of runtime on linux_memcheck (about 4 minutes) which we can do
without.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6688 4adac7df-926f-26a2-2b94-8c16560cd09d
rtc::Fatal output: Print space between # and message

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6689 4adac7df-926f-26a2-2b94-8c16560cd09d
Eliminate unnecessary #include

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6690 4adac7df-926f-26a2-2b94-8c16560cd09d
Make RTCP sender report send media bytes.

r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix breakage introduced by r6691.

ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.

BUG=
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
Improvements to the pacer where it lost some budget due to truncation errors.

With this CL the resolution is increased to microseconds and proper rounding
is done in the Process() function. This means that we will be allowed to send
more than prior to r6664 as we previously truncated away parts of our budget.

We will also not lose budget due to inaccurate calculations in
TimeUntilNextProcess(), which was a regression in r6664.

BUG=cr/393950
TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6694 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove Thread::RunningForChannelManager().

I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
Minor refactoring of StatsCollector.
* Make GetTimeNow a static method in the cc file.
* Make GetTransportIdFromProxy a static method as well and not a class method.

The second change is in preparation of removing the proxy_to_transport_ member variable which isn't needed and is just a copy from the session stats.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6696 4adac7df-926f-26a2-2b94-8c16560cd09d
After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.

CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix an invalid memory access due to typo in win/cursor.cc.

BUG=crbug/391468
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6698 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement unittest for SetSendCodecsChangesExistingStreams.

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

Review URL: https://webrtc-codereview.appspot.com/19869004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6699 4adac7df-926f-26a2-2b94-8c16560cd09d
nrsh1 is written before tmp321 is read, so needs to be earlyclobber

Otherwise, the compiler is allowed to put them in the same register
under the assumption that all inputs are read before any
(non-earlyclobber) output is written, which in this case would result
in nrsh2 being corrupted.

BUG=3439
R=aluebs@webrtc.org, ljubomir.papuga@gmail.com

Review URL: https://webrtc-codereview.appspot.com/16089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6700 4adac7df-926f-26a2-2b94-8c16560cd09d
Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macros

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6701 4adac7df-926f-26a2-2b94-8c16560cd09d
int16<->float conversions: Use size_t for array length argument, not int

size_t is more appropriate for array lengths, since int might
theoretically be too small for a really large array. But more
importantly, if the caller's value is naturally of type size_t and the
function requires an int, VC++ will trigger warning C4267
(http://msdn.microsoft.com/en-us/library/6kck0s93.aspx) because the
implicit cast might be lossy, forcing the caller to do a manual cast.
Typing the function with size_t in the first place resolves the
problem.

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6702 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove old padding path in RTPSender.

Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically.

This is not an error case if for instance an external h.264 encoder is registered, but no internal implementation exists.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6704 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 71240799-> 71250251

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
Compile-time guard for iOS7 specific property.

BUG=3487
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6706 4adac7df-926f-26a2-2b94-8c16560cd09d
Add support of multiple STUN servers in UDPPort.
Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.

I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.

BUG=3310
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d