diff options
5 files changed, 5 insertions, 7 deletions
diff --git a/common_audio/signal_processing/include/signal_processing_library.h b/common_audio/signal_processing/include/signal_processing_library.h index 3966c2f5..69023b26 100644 --- a/common_audio/signal_processing/include/signal_processing_library.h +++ b/common_audio/signal_processing/include/signal_processing_library.h @@ -123,7 +123,6 @@ #define WEBRTC_SPL_RSHIFT_W32(x, c) ((x) >> (c)) #define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c)) -#define WEBRTC_SPL_RSHIFT_U16(x, c) ((uint16_t)(x) >> (c)) #define WEBRTC_SPL_LSHIFT_U16(x, c) ((uint16_t)(x) << (c)) #define WEBRTC_SPL_RSHIFT_U32(x, c) ((uint32_t)(x) >> (c)) #define WEBRTC_SPL_LSHIFT_U32(x, c) ((uint32_t)(x) << (c)) diff --git a/common_audio/signal_processing/signal_processing_unittest.cc b/common_audio/signal_processing/signal_processing_unittest.cc index 26225b5f..48f6eb3a 100644 --- a/common_audio/signal_processing/signal_processing_unittest.cc +++ b/common_audio/signal_processing/signal_processing_unittest.cc @@ -89,7 +89,6 @@ TEST_F(SplTest, MacroTest) { EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1)); EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1)); - EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1)); EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1)); EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1)); EXPECT_EQ(32766u, WEBRTC_SPL_LSHIFT_U32(a, 1)); 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 8de74752..9391fb3c 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 @@ -383,7 +383,7 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, streamData->full = 1; } else { streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) | - WEBRTC_SPL_RSHIFT_U16(*streamPtr, 8); + ((*streamPtr) >> 8); streamData->full = 0; } } else { diff --git a/modules/audio_processing/agc/digital_agc.c b/modules/audio_processing/agc/digital_agc.c index faef9141..d3acc1f3 100644 --- a/modules/audio_processing/agc/digital_agc.c +++ b/modules/audio_processing/agc/digital_agc.c @@ -118,7 +118,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16 limiterLvlX = analogTarget - limiterOffset; limiterIdx = 2 + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((int32_t)limiterLvlX, 13), - WEBRTC_SPL_RSHIFT_U16(kLog10_2, 1)); + (kLog10_2 / 2)); tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio); limiterLvl = targetLevelDbfs + tmp16no1; diff --git a/modules/audio_processing/ns/nsx_core.c b/modules/audio_processing/ns/nsx_core.c index e627a2ef..4993321d 100644 --- a/modules/audio_processing/ns/nsx_core.c +++ b/modules/audio_processing/ns/nsx_core.c @@ -1409,7 +1409,7 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU // Shift the largest value of sum_log_i and tmp32no3 before multiplication tmp_u16 = WEBRTC_SPL_LSHIFT_U16((uint16_t)sum_log_i, 1); // Q6 if ((uint32_t)sum_log_i > tmpU32no1) { - tmp_u16 = WEBRTC_SPL_RSHIFT_U16(tmp_u16, zeros); + tmp_u16 >>= zeros; } else { tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros); } @@ -2071,8 +2071,8 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram tmpU16no1 += nonSpeechProbFinal[i]; // Q8 tmpU32no1 += (uint32_t)(inst->noiseSupFilter[i]); // Q14 } - avgProbSpeechHB = (int16_t)(4096 - - WEBRTC_SPL_RSHIFT_U16(tmpU16no1, inst->stages - 7)); // Q12 + assert(inst->stages >= 7); + avgProbSpeechHB = (4096 - (tmpU16no1 >> (inst->stages - 7))); // Q12 avgFilterGainHB = (int16_t)WEBRTC_SPL_RSHIFT_U32( tmpU32no1, inst->stages - 3); // Q14 |