Revert "Revert part of r7561, "Refactor audio conversion functions.""
This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/28899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index 61edd8f..a99f342 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <math.h>
#include <limits>
#include "webrtc/audio_processing/debug.pb.h"
@@ -153,4 +154,26 @@
return msg->ParseFromArray(bytes.get(), size);
}
+template <typename T>
+float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
+ float mse = 0;
+ float mean = 0;
+ *variance = 0;
+ for (int i = 0; i < length; ++i) {
+ T error = ref[i] - test[i];
+ mse += error * error;
+ *variance += ref[i] * ref[i];
+ mean += ref[i];
+ }
+ mse /= length;
+ *variance /= length;
+ mean /= length;
+ *variance -= mean * mean;
+
+ float snr = 100; // We assign 100 dB to the zero-error case.
+ if (mse > 0)
+ snr = 10 * log10(*variance / mse);
+ return snr;
+}
+
} // namespace webrtc