Add const qualifier to WebRtcPcm16b_Encode

BUG=909
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d
Change all system clock types to int64_t in bitrate_controller.

They are both compared to int64_t types inside the class, and is being called
with int64_t types. Could possibly cause bugs.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7832 4adac7df-926f-26a2-2b94-8c16560cd09d
Bugfix in AudioDecoderTest

When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
Add AudioEncoder::Max10MsFramesInAPacket

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7834 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove no longer used video codec test framework.

Moves one test to the vp8 unittests which might still be good to have.
Also does a bit of clean up in vp8 unittests.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7835 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix AppRTCDemo closing error for KK and JB Android devices.

- Do not allow connection output when sending http delete
request to ws server - this causes IOException for KK and JB devices.
- Avoid creating dialog box with error message when activity
has been already closed / paused -
this causes resource leak error message for KK devices.
- Plus some code clean up to support async http messages in
websocket channel wrapper and use Handler for running
peerconnection client funcitons on UI thread.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7836 4adac7df-926f-26a2-2b94-8c16560cd09d
Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper.

This makes it possible to clean up the recipes a lot.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7837 4adac7df-926f-26a2-2b94-8c16560cd09d
Add video send bitrates to histogram stats:
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")

Add retransmitted bytes to StreamDataCounters.

Change in UpdateRtpStats to also update counters for retransmitted packet.

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
Move the AudioDecoder interface out of NetEq

It belongs with the codecs, next to the AudioEncoder interface.

R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org

Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge VP8 changes.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding an codec interal CNG test in NetEq.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7843 4adac7df-926f-26a2-2b94-8c16560cd09d
Add empty implementation file that will hold statstypes.h implementation.
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding DTX to WebRTC Opus wrapper

This is a step toward adding Opus DTX support in WebRTC.

Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See

https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html

We transmit the first 1-byte packet to let decoder be in-sync

BUG=webrtc:1014
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
Add histograms for receive statistics:
- decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond")
- percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer")
- average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")

BUG=crbug/419657
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7847 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 7846 "Adding DTX to WebRTC Opus wrapper"

> Adding DTX to WebRTC Opus wrapper
> 
> This is a step toward adding Opus DTX support in WebRTC.
> 
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
> 
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
> 
> We transmit the first 1-byte packet to let decoder be in-sync
> 
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13219004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer

BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
Move implementation of types in statstypes. to its cc file.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix a comment.

R=juberti@webrtc.org, pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7851 4adac7df-926f-26a2-2b94-8c16560cd09d
iAppRTCDemo: WebSocket based signaling.

Updates the iOS code to use the new signaling model. Removes old Channel API
code. Note that this no longer logs messages to UI. UI update forthcoming.

BUG=
R=glaznev@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7852 4adac7df-926f-26a2-2b94-8c16560cd09d
Attempt to fix FYI bots.
The FYI bots went red after https://webrtc-codereview.appspot.com/32179004/ landed.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7853 4adac7df-926f-26a2-2b94-8c16560cd09d