Implement GetState() for channel's connectivity check state.

Previously, IceState is considered completed when there is only one connection (and the rest was trimmed). However, since the trimming logic is only done within the scope of network, when IPv6 and IPv4 both exist, the completion event is never fired.

This change adds the GetState() to each channel and it could decide what Completion means. The transport object then aggregates all channels before determining it's completed.

Each channel's IceState will be aggregrated at Transport level for overall Ice state

BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7804 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert "Implement GetState() for channel's connectivity check state."

This reverts commit ff72f9e692d0918b32646dadaf382aa4355d8437.

TBR=pthatcher@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/33469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7805 4adac7df-926f-26a2-2b94-8c16560cd09d
Redo the change of https://webrtc-codereview.appspot.com/30949004/

The previous change causes a build issue as there is subclass of TransportChannel in chromium. To break the circular dependency, a stub of implementation for GetState() is provided and will be removed once the jingle_glue::MockTransportChannel has the function defined.

TBR=pthatcher@webrtc.org

BUG=411086

Review URL: https://webrtc-codereview.appspot.com/34369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7806 4adac7df-926f-26a2-2b94-8c16560cd09d
Removing unused opus wrapper APIs.

WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().

WebRtcOpus_DecodePlcMaster/Slave() are also removed.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
Roll chromium_revision 24b4c73..f27c369

This enables 64-bit compilation for iOS.

Relevant changes:
* src/buildtools: 6ea835d..ded3294
* src/third_party/boringssl/src: 69a0160..00505ec
* src/third_party/libvpx: 429874c..64bec31
Details: https://chromium.googlesource.com/chromium/src/+/24b4c73..f27c369/DEPS

Clang version was not updated in this roll.

BUG=chromium:436831
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7808 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 7807 "Removing unused opus wrapper APIs."

> Removing unused opus wrapper APIs.
> 
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
> 
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
> 
> BUG=
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28139004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
Relanding r7807.

r7807 was reverted to be excluded from the cause of a failure.

It has been verified and can reland now.

BUG=

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge remote bitrate estimator changes.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7811 4adac7df-926f-26a2-2b94-8c16560cd09d
Add support for parsing header only RTP dumps with bwe_rtp_play.

Also adds support for printing the original_length in rtp_to_text.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d
Support new WebSocket signaling format.

- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.

BUG=3937,3995,4041
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.

BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d