Add a deinterleaved float interface to AudioProcessing.

This is mainly to support the native audio format in Chrome. Although
this implementation just moves the float->int conversion under the hood,
we will transition AudioProcessing towards supporting this format
throughout.

- Add a test which verifies we get identical output with the float and
int interfaces.
- The float and int wrappers are tasked with conversion to the
AudioBuffer format. A new shared Process/Analyze method does most of
the work.
- Add a new field to the debug.proto to hold deinterleaved data.
- Add helpers to audio_utils.cc, and start using numeric_limits.
- Note that there was no performance difference between numeric_limits
and a literal value when measured on Linux using gcc or clang.

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

Review URL: https://webrtc-codereview.appspot.com/9179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/debug.proto

@@ -10,17 +10,31 @@ message Init {
   optional int32 num_reverse_channels = 5;
 }
 
+// May contain interleaved or deinterleaved data, but don't store both formats.
 message ReverseStream {
+  // int16 interleaved data.
   optional bytes data = 1;
+
+  // float deinterleaved data, where each repeated element points to a single
+  // channel buffer of data.
+  repeated bytes channel = 2;
 }
 
+// May contain interleaved or deinterleaved data, but don't store both formats.
 message Stream {
+  // int16 interleaved data.
   optional bytes input_data = 1;
   optional bytes output_data = 2;
+
   optional int32 delay = 3;
   optional sint32 drift = 4;
   optional int32 level = 5;
   optional bool keypress = 6;
+
+  // float deinterleaved data, where each repeated element points to a single
+  // channel buffer of data.
+  repeated bytes input_channel = 7;
+  repeated bytes output_channel = 8;
 }
 
 message Event {