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common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16

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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@6907 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2014-08-15 05:17:20 +00:00
parent a84b0a6dab
commit 1e3ef4b999
4 changed files with 6 additions and 9 deletions
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2
webrtc/common_audio/signal_processing/include/signal_processing_library.h
View File

@ -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) \

1
webrtc/common_audio/signal_processing/signal_processing_unittest.cc
View File

@ -48,7 +48,6 @@ TEST_F(SplTest, MacroTest) {
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;

8
webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
View File

@ -195,7 +195,7 @@ int16_t WebRtcIsacfix_DecHistBisectMulti(int16_t *data,
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 @@ int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t *data,
/* 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 @@ int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t *data,
}
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 @@ int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t *data,
}
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;

4
webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
View File

@ -150,9 +150,9 @@ int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData,
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;