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audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>

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The implementation of WEBRTC_SPL_RSHIFT_W16 is simply >>. This CL removes the macro usage in audio_processing and signal_processing.

Affected components:
* aecm
* agc
* nsx

Indirectly affecting (through signal_processing changes)
* codecs/cng
* codecs/isac/fix
* codecs/isac/main

BUG=3348,3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7432 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2014-10-13 13:01:13 +00:00
parent 396a5e0001
commit d4fe824862
9 changed files with 24 additions and 29 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/common_audio/signal_processing/refl_coef_to_lpc.c
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@ -27,7 +27,7 @@ void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a)
kptr = k;
*a = 4096; // i.e., (Word16_MAX >> 3)+1.
*any = *a;
a[1] = WEBRTC_SPL_RSHIFT_W16((*k), 3);
a[1] = *k >> 3;
for (m = 1; m < use_order; m++)
{
@ -38,7 +38,7 @@ void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a)
anyptr = any;
anyptr++;
any[m + 1] = WEBRTC_SPL_RSHIFT_W16((*kptr), 3);
any[m + 1] = *kptr >> 3;
for (i = 0; i < m; i++)
{
*anyptr = (*aptr)

12
webrtc/common_audio/signal_processing/spl_sqrt.c
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@ -17,6 +17,8 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include <assert.h>
int32_t WebRtcSpl_SqrtLocal(int32_t in);
int32_t WebRtcSpl_SqrtLocal(int32_t in)
@ -154,15 +156,15 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
x_norm = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH
nshift = WEBRTC_SPL_RSHIFT_W16(sh, 1); // nshift = sh>>1
nshift = -nshift; // Negate the power for later de-normalization
nshift = (sh / 2);
assert(nshift >= 0);
A = (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)x_norm, 16);
A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
if ((-2 * nshift) == sh)
{ // Even shift value case
if (2 * nshift == sh) {
// Even shift value case
t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); // t16 = AH
@ -178,7 +180,7 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
}
A = A & ((int32_t)0x0000ffff);
A = (int32_t)WEBRTC_SPL_SHIFT_W32(A, nshift); // De-normalize the result
A >>= nshift; // De-normalize the result.
return A;
}

4
webrtc/modules/audio_processing/aecm/aecm_core.c
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@ -691,10 +691,10 @@ int16_t WebRtcAecm_AsymFilt(const int16_t filtOld, const int16_t inVal,
retVal = filtOld;
if (filtOld > inVal)
{
retVal -= WEBRTC_SPL_RSHIFT_W16(filtOld - inVal, stepSizeNeg);
retVal -= (filtOld - inVal) >> stepSizeNeg;
} else
{
retVal += WEBRTC_SPL_RSHIFT_W16(inVal - filtOld, stepSizePos);
retVal += (inVal - filtOld) >> stepSizePos;
}
return retVal;

5
webrtc/modules/audio_processing/aecm/aecm_core_c.c
View File

@ -483,10 +483,7 @@ int WebRtcAecm_ProcessBlock(AecmCore_t * aecm,
if (zeros32 > tmp16no1)
{
echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
(uint16_t)WEBRTC_SPL_RSHIFT_W16(
supGain,
tmp16no1)
);
supGain >> tmp16no1);
} else
{
// Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)

2
webrtc/modules/audio_processing/aecm/aecm_core_mips.c
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@ -988,7 +988,7 @@ int WebRtcAecm_ProcessBlock(AecmCore_t* aecm,
if (zeros32 > tmp16no1) {
echoEst32Gained = WEBRTC_SPL_UMUL_32_16(
(uint32_t)aecm->echoFilt[i],
(uint16_t)WEBRTC_SPL_RSHIFT_W16(supGain, tmp16no1));
supGain >> tmp16no1);
} else {
// Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
echoEst32Gained = WEBRTC_SPL_UMUL_32_16(

6
webrtc/modules/audio_processing/agc/analog_agc.c
View File

@ -296,7 +296,7 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
ptr = stt->Rxx16w32_array[0];
}
for (i = 0; i < WEBRTC_SPL_RSHIFT_W16(M, 1); i++)
for (i = 0; i < M / 2; i++)
{
if (stt->fs == 16000)
{
@ -546,7 +546,7 @@ void WebRtcAgc_UpdateAgcThresholds(Agc_t *stt)
{
/* Lower the analog target level since we have reached its maximum */
zeros = WebRtcSpl_NormW32(stt->Rxx160_LPw32);
stt->targetIdxOffset = WEBRTC_SPL_RSHIFT_W16((3 * zeros) - stt->targetIdx - 2, 2);
stt->targetIdxOffset = (3 * zeros - stt->targetIdx - 2) / 4;
}
#endif
@ -696,7 +696,7 @@ void WebRtcAgc_SpeakerInactiveCtrl(Agc_t *stt)
if (stt->vadMic.stdLongTerm < 4500)
{
/* Scale between min and max threshold */
vadThresh += WEBRTC_SPL_RSHIFT_W16(4500 - stt->vadMic.stdLongTerm, 1);
vadThresh += (4500 - stt->vadMic.stdLongTerm) / 2;
}
/* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */

2
webrtc/modules/audio_processing/agc/digital_agc.c
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@ -507,7 +507,7 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
{
if (gate < 2500)
{
gain_adj = WEBRTC_SPL_RSHIFT_W16(2500 - gate, 5);
gain_adj = (2500 - gate) >> 5;
} else
{
gain_adj = 0;

12
webrtc/modules/audio_processing/ns/nsx_core.c
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@ -407,13 +407,11 @@ static void NoiseEstimationC(NsxInst_t* inst,
// +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
// CounterDiv=1/(inst->counter[s]+1) in Q15
tmp16 += 2;
tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
inst->noiseEstLogQuantile[offset + i] += tmp16no1;
inst->noiseEstLogQuantile[offset + i] += tmp16 / 4;
} else {
tmp16 += 1;
tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
// *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16 / 2, 3, 1);
inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
if (inst->noiseEstLogQuantile[offset + i] < logval) {
// This is the smallest fixed point representation we can
@ -611,7 +609,7 @@ void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t* inst,
// Piecewise linear approximation of 'b' in
// 2^(int_part+frac_part) = 2^int_part * (1 + b)
// 'b' is given in Q11 and below stored in frac_part.
if (WEBRTC_SPL_RSHIFT_W16(frac_part, 10)) {
if (frac_part >> 10) {
// Upper fractional part
tmp32no2 = WEBRTC_SPL_MUL_16_16(2048 - frac_part, 1244); // Q21
tmp32no2 = 2048 - WEBRTC_SPL_RSHIFT_W32(tmp32no2, 10);
@ -669,7 +667,7 @@ int32_t WebRtcNsx_InitCore(NsxInst_t* inst, uint32_t fs) {
inst->maxLrt = 0x0080000;
inst->minLrt = 104858;
}
inst->anaLen2 = WEBRTC_SPL_RSHIFT_W16(inst->anaLen, 1);
inst->anaLen2 = inst->anaLen / 2;
inst->magnLen = inst->anaLen2 + 1;
if (inst->real_fft != NULL) {
@ -1410,7 +1408,7 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
tmpU32no1 >>= zeros;
}
tmp_2_w32 -= (int32_t)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
matrix_determinant >>= zeros; // Q(-zeros)
tmp_2_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q11
tmp_2_w32 += (int32_t)net_norm << 11; // Q11
if (tmp_2_w32 < 0) {

6
webrtc/modules/audio_processing/ns/nsx_core_neon.c
View File

@ -313,13 +313,11 @@ void WebRtcNsx_NoiseEstimationNeon(NsxInst_t* inst,
// +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
// CounterDiv=1/(inst->counter[s]+1) in Q15
tmp16 += 2;
tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
inst->noiseEstLogQuantile[offset + i] += tmp16no1;
inst->noiseEstLogQuantile[offset + i] += tmp16 / 4;
} else {
tmp16 += 1;
tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
// *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16 / 2, 3, 1);
inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
if (inst->noiseEstLogQuantile[offset + i] < logval) {
// logval is the smallest fixed point representation we can have.