ACM2/NetEq4 did not decode Opus in stereo

Two problems fixed in this CL:
- setting Opus decoder to stereo had no effect, and decoding always generated mono audio
- changing decoding setting from mono to stereo, or stereo to mono, for OPUS also had no effect (but required another change than the first one).

BUG=3082
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5754 4adac7df-926f-26a2-2b94-8c16560cd09d
Add AIMD option to BWE API.

TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10319005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5755 4adac7df-926f-26a2-2b94-8c16560cd09d
Add fir_filter to common_audio

It has 3 implementation:
* fir_filter_c with no optimization
* fir_filter_sse which outperforms the C version by a factor of 3x
* fir_filter_neon which outperforms the C version by a factor of 2x

R=andrew@webrtc.org, bjornv@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/9759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5756 4adac7df-926f-26a2-2b94-8c16560cd09d
VoE changes to allow forwarding of packets from VoE to ViE BWE.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
Disabled some of the remote bitrate estimator baseline tests.

These are disabled temporarily until updated.

R=solenberg@webrtc.org
TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5758 4adac7df-926f-26a2-2b94-8c16560cd09d
Add targetBitrate to VideoCodec struct.

To be used by a codec implementation. Could for instance be interpreted
as trying to fit as much as possible on one temporal layer and send
everything that doesn't fit within target bitrate on another one.

Prevents an existing hack where startBitrate is used by a codec
implementation to signify target bitrate. This hack forces a reset of
bitrate estimation to target bitrate which creates bitrate dips.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5759 4adac7df-926f-26a2-2b94-8c16560cd09d
Run Opus with lower complexity setting on Android, iOS and/or ARM

This CL includes a call to Opus to set a lower complexity figure, if we are compiling for Android, iOS, or ARM (e.g. ChromeOS on ARM), where we know the devices are not powerful enough to run on higher complexity setting.

BUG=3093
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5760 4adac7df-926f-26a2-2b94-8c16560cd09d
DelayEstimator: Updates delay_quality and adds soft reset.

These changes are currently not used in webrtc/ but helps in using the delay estimator.
* The last_delay_quality() is updated with respect to robust_validation and changed to return float.
* Tests are updated wtih respect to above.
* Adds the possibility to make a soft reset based on external circumstances like a known delay shift has been made.
* The soft reset change the lookahead dynamically. An API to ask for current lookahead has been added as well.

BUG=N/A
TESTED=trybots, modules_unittest
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5761 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 63560528-> 63648983

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5762 4adac7df-926f-26a2-2b94-8c16560cd09d
Have changes to REMB trigger RTCP to be sent immediately.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5763 4adac7df-926f-26a2-2b94-8c16560cd09d
PeerConnection(iOS): fix case in #import statements.

We've been skating by on OS/X's default case-insensitive filesystem, but this
is a bit silly.

This change brought to you by:
sed -i '' 's/\+internal\.h/+Internal.h/g' $(git grep -l '+internal.h')

BUG=3088
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5764 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5765 4adac7df-926f-26a2-2b94-8c16560cd09d
Disable flaky WebRtcVideoMediaChannelTests on memcheck and tsan.

BUG=3096
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5766 4adac7df-926f-26a2-2b94-8c16560cd09d
Add configuration for ability to use the encode usage measure for triggering overuse/underuse.

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5767 4adac7df-926f-26a2-2b94-8c16560cd09d
AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10

BUG=2168
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/9709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
Cleanups in libjingle to make it compile with chromium_code=1

Fixed all warnings that show up when compiling libjingle
in chromium with compiling with chromium_code=1.
chromium_code=1 enables various warnings that are off by
default. Most changes are for unused variables and consts.

R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5769 4adac7df-926f-26a2-2b94-8c16560cd09d
iOS VideoEngine: move video_{capture,render} to ARC.

Replaces ye olde timey explicit release with teh hotness of automatic
reference counting.

This is the webrtc/ half; the talk/ half is in https://webrtc-codereview.appspot.com/10499005/

BUG=3054,3055
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5770 4adac7df-926f-26a2-2b94-8c16560cd09d
PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios

- Removes a strong-reference cycle between RTCPeerConnection and
  RTCPeerConnectionObserver
- Gives RTCPeerConnectionObserver a virtual dtor
- Ensures RTCPeerConnectionTest tears down correctly
- Ensures AppRTCDemo tears down correctly

This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005

BUG=3054,3055,3100
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
iOS video_capture: start camera in the background.

Camera start is a blocking operation so never a good idea to do on a main
thread, but worse than that is that the guts of WebView appear to be
interacting with capture start in a bad way causing startup to pause for 10s
while a timeout expires.  This change eliminates that 10s delay.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5772 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding API for setting bandwidth estimation configurations.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5773 4adac7df-926f-26a2-2b94-8c16560cd09d