common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND

This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation.

BUG=3348,3353
TESTED=trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
index 3db793b..ce84ec8 100644
--- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -99,8 +99,6 @@
 #define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
     ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t) \
                                   (((int32_t)1) << ((c) - 1)))) >> (c))
-#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \
-    ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t) (1 << 14))) >> 15)
 
 // C + the 32 most significant bits of A * B
 #define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
index 7b1f7fd..90e116f 100644
--- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -68,7 +68,6 @@
 
     EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
     EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
 
     EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
     EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
index 0dc8174..deba0d5 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
@@ -834,13 +834,15 @@
 
     /* bandwidth expansion */
     for (n = 1; n <= ORDERLO; n++) {
-      a_LOQ11[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecLo[n-1], a_LOQ11[n]);
+      a_LOQ11[n] = (int16_t) ((WEBRTC_SPL_MUL_16_16(
+          kPolyVecLo[n-1], a_LOQ11[n]) + ((int32_t) (1 << 14))) >> 15);
     }
 
 
     polyHI[0] = a_HIQ12[0];
     for (n = 1; n <= ORDERHI; n++) {
-      a_HIQ12[n] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecHi[n-1], a_HIQ12[n]);
+      a_HIQ12[n] = (int16_t) ((WEBRTC_SPL_MUL_16_16(
+          kPolyVecHi[n-1], a_HIQ12[n]) + ((int32_t) (1 << 14))) >> 15);
       polyHI[n] = a_HIQ12[n];
     }