From b38b009d216bc6fc45934d8dafa9e7489ec8e91a Mon Sep 17 00:00:00 2001 From: "bjornv@webrtc.org" Date: Tue, 10 Mar 2015 06:40:02 +0000 Subject: [PATCH] Refactor audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16_RSFT The macro is defined as #define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \ (WEBRTC_SPL_MUL_16_16(a, b) >> (c)) where the latter macro is in C defined as #define WEBRTC_SPL_MUL_16_16(a, b) \ ((int32_t) (((int16_t)(a)) * ((int16_t)(b)))) (For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h) The replacement consists of - avoiding casts to int16_t if inputs already are int16_t - adding explicit cast to if result is assigned to (other than int or int32_t) - minor cleanups like remove of unnecessary parentheses and style changes In addition an implicit cast from int32_t to int16_t was removed, which was a bug. BUG=3348,3353 TESTED=Locally on Mac and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41169004 Cr-Commit-Position: refs/heads/master@{#8665} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8665 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../modules/audio_processing/aecm/aecm_core.c | 7 ++-- .../audio_processing/aecm/aecm_core_c.c | 42 ++++++++----------- 2 files changed, 21 insertions(+), 28 deletions(-) diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c index de8ca2775..c95c1f2af 100644 --- a/webrtc/modules/audio_processing/aecm/aecm_core.c +++ b/webrtc/modules/audio_processing/aecm/aecm_core.c @@ -794,7 +794,7 @@ void WebRtcAecm_CalcEnergies(AecmCore* aecm, tmp16 = 2560 - aecm->farEnergyMin; if (tmp16 > 0) { - tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, FAR_ENERGY_VAD_REGION, 9); + tmp16 = (int16_t)((tmp16 * FAR_ENERGY_VAD_REGION) >> 9); } else { tmp16 = 0; @@ -1092,8 +1092,9 @@ void WebRtcAecm_UpdateChannel(AecmCore* aecm, aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld); } else { - aecm->mseThreshold += WEBRTC_SPL_MUL_16_16_RSFT(mseAdapt - - WEBRTC_SPL_MUL_16_16_RSFT(aecm->mseThreshold, 5, 3), 205, 8); + int scaled_threshold = aecm->mseThreshold * 5 / 8; + aecm->mseThreshold += + ((mseAdapt - scaled_threshold) * 205) >> 8; } } diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/webrtc/modules/audio_processing/aecm/aecm_core_c.c index c69193819..3a06bb669 100644 --- a/webrtc/modules/audio_processing/aecm/aecm_core_c.c +++ b/webrtc/modules/audio_processing/aecm/aecm_core_c.c @@ -73,14 +73,11 @@ static void WindowAndFFT(AecmCore* aecm, for (i = 0; i < PART_LEN; i++) { // Window time domain signal and insert into real part of // transformation array |fft| - fft[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( - (time_signal[i] << time_signal_scaling), - WebRtcAecm_kSqrtHanning[i], - 14); - fft[PART_LEN + i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( - (time_signal[i + PART_LEN] << time_signal_scaling), - WebRtcAecm_kSqrtHanning[PART_LEN - i], - 14); + int16_t scaled_time_signal = time_signal[i] << time_signal_scaling; + fft[i] = (int16_t)((scaled_time_signal * WebRtcAecm_kSqrtHanning[i]) >> 14); + scaled_time_signal = time_signal[i + PART_LEN] << time_signal_scaling; + fft[PART_LEN + i] = (int16_t)(( + scaled_time_signal * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14); } // Do forward FFT, then take only the first PART_LEN complex samples, @@ -124,9 +121,8 @@ static void InverseFFTAndWindow(AecmCore* aecm, tmp32no1 + aecm->outBuf[i], WEBRTC_SPL_WORD16_MIN); - tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(ifft_out[PART_LEN + i], - WebRtcAecm_kSqrtHanning[PART_LEN - i], - 14); + tmp32no1 = (ifft_out[PART_LEN + i] * + WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14; tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain); aecm->outBuf[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, @@ -250,8 +246,8 @@ static int TimeToFrequencyDomain(AecmCore* aecm, alpha = kAlpha3; beta = kBeta3; } - tmp16no1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(max_value, alpha, 15); - tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(min_value, beta, 15); + tmp16no1 = (int16_t)((max_value * alpha) >> 15); + tmp16no2 = (int16_t)((min_value * beta) >> 15); freq_signal_abs[i] = (uint16_t)tmp16no1 + (uint16_t)tmp16no2; #else #ifdef WEBRTC_ARCH_ARM_V7 @@ -561,7 +557,7 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm, // speech distortion in double-talk. for (i = 0; i < PART_LEN1; i++) { - hnl[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], hnl[i], 14); + hnl[i] = (int16_t)((hnl[i] * hnl[i]) >> 14); } for (i = kMinPrefBand; i <= kMaxPrefBand; i++) @@ -609,7 +605,7 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm, hnl[i] = ONE_Q14; } else { - hnl[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], nlpGain, 14); + hnl[i] = (int16_t)((hnl[i] * nlpGain) >> 14); } // multiply with Wiener coefficients @@ -744,9 +740,7 @@ static void ComfortNoise(AecmCore* aecm, noiseRShift16[i] = (int16_t)tmp32; tmp16 = ONE_Q14 - lambda[i]; - noiseRShift16[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, - noiseRShift16[i], - 14); + noiseRShift16[i] = (int16_t)((tmp16 * noiseRShift16[i]) >> 14); } // Generate a uniform random array on [0 2^15-1]. @@ -758,15 +752,13 @@ static void ComfortNoise(AecmCore* aecm, for (i = 1; i < PART_LEN1; i++) { // Get a random index for the cos and sin tables over [0 359]. - tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(359, randW16[i - 1], 15); + tmp16 = (int16_t)((359 * randW16[i - 1]) >> 15); // Tables are in Q13. - uReal[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(noiseRShift16[i], - WebRtcAecm_kCosTable[tmp16], - 13); - uImag[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(-noiseRShift16[i], - WebRtcAecm_kSinTable[tmp16], - 13); + uReal[i] = (int16_t)((noiseRShift16[i] * WebRtcAecm_kCosTable[tmp16]) >> + 13); + uImag[i] = (int16_t)((-noiseRShift16[i] * WebRtcAecm_kSinTable[tmp16]) >> + 13); } uImag[PART_LEN] = 0;