Resampler modifications in preparation for arbitrary audioproc rates.

- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.

This is a prerequisite of:
http://review.webrtc.org/9919004/

BUG=2894
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/acm2/acm_resampler.h

@@ -29,7 +29,7 @@ class ACMResampler {
                      int16_t* out_audio);
 
  private:
-  PushResampler resampler_;
+  PushResampler<int16_t> resampler_;
 };
 
 }  // namespace acm2

webrtc/modules/audio_coding/main/source/acm_resampler.h

@@ -15,7 +15,6 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-
 namespace acm1 {
 
 class ACMResampler {
@@ -30,11 +29,10 @@ class ACMResampler {
                          uint8_t num_audio_channels);
 
  private:
-  PushResampler resampler_;
+  PushResampler<int16_t> resampler_;
 };
 
 }  // namespace acm1
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_