Use flags set by the port allocator.

Currently, port allocator flags are ignored. This is inconvenient if you
want to have your own PortAllocatorFactory subclass.

BUG=webrtc:3958
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7524 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix ::~LogMessage to print as a string.

R=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7525 4adac7df-926f-26a2-2b94-8c16560cd09d
Increase max trace message size to 1024 characters.

A recent CL by pbos:
https://code.google.com/p/webrtc/source/detail?r=7518

added long log messages and triggered errors on the DrMemory bot due to
WEBRTC_TRACE. The trace mechanism _should_ truncate the log strings
but something appears to be going awry.

This sweeps the problem under the rug, but given that WEBRTC_TRACE
should die fairly soon, seems to be a reasonable tradeoff.

TEST=passing try on DrMemory.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27849004

Patch from Andrew MacDonald <andrew@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7526 4adac7df-926f-26a2-2b94-8c16560cd09d
isacfix: Refactor big-endian reading and writing

Make subroutines for encoding and decoding arrays of 16-bit big-endian
integers, and in the process fix a bug: When decoding an odd number of
bytes from be16, the least significant byte of the last int16 in the
array was properly taken to be zero instead of actually being read
(since it's outside the array). However, when encoding an odd number
of bytes, the least significant byte of the last int16 in the array
was written to the output as-is instead of being taken to be zero;
thus, we encoded one byte more than we should. This was probably not
harmful, and the value was dropped at decoding anyway; nevertheless,
writing a constant zero is the safe thing to do, and this patch does
so.

R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7527 4adac7df-926f-26a2-2b94-8c16560cd09d
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile

This CL contains some cleaning up and refactoring of
audio_decoder_test.cc. A new class ResampleInputAudioFile is created
and used in the tests.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7528 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement screencast settings for WebRtcVideoEngine2.

Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the SrtpFilter crash caused by two local offers.

BUG=http://crbug.com/421774
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7530 4adac7df-926f-26a2-2b94-8c16560cd09d
Upgrade our scoped_ptr copy to match Chromium's latest.

In particular add the move constructor and assignment operator.

Diff between our version and Chromium's:
https://paste.googleplex.com/4887047529562112

R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7531 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 78583324-> 78583691

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7532 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding setting screen to AppRTCDemo.

- Move server URL from connection screen
to the setting screen.
- Add setting for local video resolution.
- Auto save last entered room number.
- Use full screen mode in video renderer and fix
texture offsets recalculation when rendering type is
dynamically changed.

BUG=3935,3953
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7534 4adac7df-926f-26a2-2b94-8c16560cd09d
Move scoped_ptr "free" functions into the webrtc namespace.

Resolves a conflict with Chromium's scoped_ptr on the recently added
make_scoped_ptr().

TEST=local Chromium Linux build passes.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7535 4adac7df-926f-26a2-2b94-8c16560cd09d
Move (test) RtpFileReader to a lightweight target.

Moves RtpFileReader to rtp_packet_parser and renames it to
rtp_test_utils which is allowed to rely on rtp_rtcp.

R=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7536 4adac7df-926f-26a2-2b94-8c16560cd09d
Only configure the SSL library in one place.

Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.

This is to avoid colliding with Chromium's transition away from NSS.

This is a fixup of https://webrtc-codereview.appspot.com/29559004 to avoid
breaking use_legacy_ssl_defaults.

BUG=chromium:413497
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7537 4adac7df-926f-26a2-2b94-8c16560cd09d
Add a simple AudioConverter class.

This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.

The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.

BUG=chromium:405270
R=aluebs@webrtc.org, mgraczyk@chromium.org
TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/30779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 78616359-> 78642371

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7540 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix double backslashes in incoming_video_stream.cc

Originally uploaded in https://codereview.appspot.com/149160043/.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7541 4adac7df-926f-26a2-2b94-8c16560cd09d
Use neteq_unittest_tools in audio_decoder_unittests

With the recent move of RtpFileReader to the rtp_test_utils target
(in r7536), it is now possible to let audio_decoder_unittests depend
on neteq_unittest_tools without breaking the Android build.

BUG=2692
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7542 4adac7df-926f-26a2-2b94-8c16560cd09d