Support arbitrary input/output rates and downmixing in AudioProcessing.

Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.

- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/gain_control_impl.cc

@@ -326,7 +326,7 @@ int GainControlImpl::InitializeHandle(void* handle) const {
                           minimum_capture_level_,
                           maximum_capture_level_,
                           MapSetting(mode_),
-                          apm_->sample_rate_hz());
+                          apm_->proc_sample_rate_hz());
 }
 
 int GainControlImpl::ConfigureHandle(void* handle) const {