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 72a5388fa4..2d9fff7cfc 100644 --- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h +++ b/webrtc/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/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc index 603294be3b..17d8d0371c 100644 --- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc +++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc @@ -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; diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c index c66be2e484..5311b3930e 100644 --- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c +++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c @@ -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; diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c index 9391fb3c1d..e57416580c 100644 --- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c +++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c @@ -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;