common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16
Macros should in general be avoided. WEBRTC_SPL_UMUL_32_16_RSFT16 is only used in iSAC fixed point as part of multiplying with LSB and MSB. A better approach is to have one function for that complete operation in iSAC.
This CL removes the macro and replace the operation locally.
BUG=3148, 3353
TESTED=locally on Linux and trybots
R=tina.legrand@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6907 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/common_audio/signal_processing/include/signal_processing_library.h b/common_audio/signal_processing/include/signal_processing_library.h
index 72a5388..2d9fff7 100644
--- a/common_audio/signal_processing/include/signal_processing_library.h
+++ b/common_audio/signal_processing/include/signal_processing_library.h
@@ -57,8 +57,6 @@
((uint32_t) (uint16_t)(a) * (uint16_t)(b))
#define WEBRTC_SPL_UMUL_32_16(a, b) \
((uint32_t) ((uint32_t)(a) * (uint16_t)(b)))
-#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b) \
- ((uint32_t) ((uint32_t)(a) * (uint16_t)(b)) >> 16)
#define WEBRTC_SPL_MUL_16_U16(a, b) \
((int32_t)(int16_t)(a) * (uint16_t)(b))
#define WEBRTC_SPL_DIV(a, b) \
diff --git a/common_audio/signal_processing/signal_processing_unittest.cc b/common_audio/signal_processing/signal_processing_unittest.cc
index 603294b..17d8d03 100644
--- a/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/common_audio/signal_processing/signal_processing_unittest.cc
@@ -48,7 +48,6 @@
b = WEBRTC_SPL_WORD16_MAX >> 1;
EXPECT_EQ(1073627139u, WEBRTC_SPL_UMUL_16_16(a, b));
EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
- EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
a = b;
diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
index c66be2e..5311b39 100644
--- a/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
+++ b/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
@@ -195,7 +195,7 @@
for ( ;; )
{
W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+ W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16;
sizeTmp = WEBRTC_SPL_RSHIFT_W16(sizeTmp, 1);
if (sizeTmp == 0) {
break;
@@ -325,7 +325,7 @@
/* start at the specified table entry */
cdfPtr = *cdf + (*initIndex++);
W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+ W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16;
if (streamval > W_tmp)
{
@@ -339,7 +339,7 @@
}
W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *++cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+ W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16;
if (streamval <= W_tmp) {
break;
@@ -359,7 +359,7 @@
}
W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
+ W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16;
if (streamval > W_tmp) {
break;
diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
index 9391fb3..e574165 100644
--- a/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
+++ b/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
@@ -150,9 +150,9 @@
W_upper_LSB = (uint16_t)W_upper;
W_upper_MSB = (uint16_t)WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
W_lower = WEBRTC_SPL_UMUL_32_16(cdfLo, W_upper_MSB);
- W_lower += WEBRTC_SPL_UMUL_32_16_RSFT16(cdfLo, W_upper_LSB);
+ W_lower += (cdfLo * W_upper_LSB) >> 16;
W_upper = WEBRTC_SPL_UMUL_32_16(cdfHi, W_upper_MSB);
- W_upper += WEBRTC_SPL_UMUL_32_16_RSFT16(cdfHi, W_upper_LSB);
+ W_upper += (cdfHi * W_upper_LSB) >> 16;
/* shift interval such that it begins at zero */
W_upper -= ++W_lower;
