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/webrtc@5943 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 6 insertions, 6 deletions

diff --git a/voice_engine/utility_unittest.cc b/voice_engine/utility_unittest.cc
index a5d0bcd8..8f7efa87 100644
--- a/voice_engine/utility_unittest.cc
+++ b/voice_engine/utility_unittest.cc
@@ -39,7 +39,7 @@ class UtilityTest : public ::testing::Test {
                        int dst_channels, int dst_sample_rate_hz,
                        FunctionToTest function);
 
-  PushResampler resampler_;
+  PushResampler<int16_t> resampler_;
   AudioFrame src_frame_;
   AudioFrame dst_frame_;
   AudioFrame golden_frame_;
@@ -127,11 +127,11 @@ void VerifyFramesAreEqual(const AudioFrame& ref_frame,
 }
 
 void UtilityTest::RunResampleTest(int src_channels,
-                                      int src_sample_rate_hz,
-                                      int dst_channels,
-                                      int dst_sample_rate_hz,
-                                      FunctionToTest function) {
-  PushResampler resampler;  // Create a new one with every test.
+                                  int src_sample_rate_hz,
+                                  int dst_channels,
+                                  int dst_sample_rate_hz,
+                                  FunctionToTest function) {
+  PushResampler<int16_t> resampler;  // Create a new one with every test.
   const int16_t kSrcLeft = 30;  // Shouldn't overflow for any used sample rate.
   const int16_t kSrcRight = 15;
   const float resampling_factor = (1.0 * src_sample_rate_hz) /