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 72de680f2..ad95386fe 100644 --- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h +++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h @@ -89,7 +89,6 @@ // Shifting with negative numbers not allowed // We cannot do casting here due to signed/unsigned problem -#define WEBRTC_SPL_RSHIFT_W16(x, c) ((x) >> (c)) #define WEBRTC_SPL_RSHIFT_W32(x, c) ((x) >> (c)) #define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c)) diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc index cdfb01960..09518a9d2 100644 --- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc +++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc @@ -66,7 +66,6 @@ TEST_F(SplTest, MacroTest) { // Shifting with negative numbers not allowed // We cannot do casting here due to signed/unsigned problem - EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1)); EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1)); EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1)); diff --git a/webrtc/common_audio/vad/vad_core.c b/webrtc/common_audio/vad/vad_core.c index 98da6eaf0..6ebe65d86 100644 --- a/webrtc/common_audio/vad/vad_core.c +++ b/webrtc/common_audio/vad/vad_core.c @@ -639,10 +639,10 @@ int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame, // Downsample signal 32->16->8 before doing VAD WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]), frame_length); - len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1); + len = frame_length / 2; WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len); - len = WEBRTC_SPL_RSHIFT_W16(len, 1); + len /= 2; // Do VAD on an 8 kHz signal vad = WebRtcVad_CalcVad8khz(inst, speechNB, len); @@ -660,7 +660,7 @@ int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame, WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states, frame_length); - len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1); + len = frame_length / 2; vad = WebRtcVad_CalcVad8khz(inst, speechNB, len); return vad; diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c index c489731f8..0f1dd7c34 100644 --- a/webrtc/modules/audio_processing/aecm/aecm_core.c +++ b/webrtc/modules/audio_processing/aecm/aecm_core.c @@ -818,10 +818,8 @@ void WebRtcAecm_CalcEnergies(AecmCore_t * aecm, { if (aecm->farEnergyVAD > aecm->farLogEnergy) { - aecm->farEnergyVAD += WEBRTC_SPL_RSHIFT_W16(aecm->farLogEnergy + - tmp16 - - aecm->farEnergyVAD, - 6); + aecm->farEnergyVAD += + (aecm->farLogEnergy + tmp16 - aecm->farEnergyVAD) >> 6; aecm->vadUpdateCount = 0; } else {