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Refactor common_audio/signal_processing: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16

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The 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)

Also includes
- style changes
- replaced pointer operations with direct element access

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

Review URL: https://webrtc-codereview.appspot.com/48949004

Cr-Commit-Position: refs/heads/master@{#9075}
This commit is contained in:
Bjorn Volcker
2015-04-24 08:12:07 +02:00
parent e3827f27c3
commit affcfb2f16
10 changed files with 38 additions and 65 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
webrtc/common_audio/signal_processing/complex_fft.c
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@ -126,11 +126,9 @@ int WebRtcSpl_ComplexFFT(int16_t frfi[], int stages, int mode)
[wri]"r"(wri),
[cfftrnd]"r"(CFFTRND));
#else
tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
- WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CFFTRND;
ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CFFTRND;
#endif
tr32 >>= 15 - CFFTSFT;
@ -270,11 +268,9 @@ int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
);
#else
tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
- WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CIFFTRND;
ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+ WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CIFFTRND;
#endif
tr32 >>= 15 - CIFFTSFT;
ti32 >>= 15 - CIFFTSFT;

10
webrtc/common_audio/signal_processing/division_operations.c
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@ -106,8 +106,7 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
// result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
// tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
+ ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
tmpW32 = (den_hi * approx << 1) + ((den_low * approx >> 15) << 1);
// tmpW32 = den * approx
tmpW32 = (int32_t)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
@ -117,8 +116,7 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
tmp_low = (int16_t)((tmpW32 - ((int32_t)tmp_hi << 16)) >> 1);
// tmpW32 = 1/den in Q29
tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
>> 15)) << 1);
tmpW32 = (tmp_hi * approx + (tmp_low * approx >> 15)) << 1;
// 1/den in hi and low format
tmp_hi = (int16_t)(tmpW32 >> 16);
@ -130,8 +128,8 @@ int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
// num * (1/den) by 32 bit multiplication (result in Q28)
tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
>> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
tmpW32 = num_hi * tmp_hi + (num_hi * tmp_low >> 15) +
(num_low * tmp_hi >> 15);
// Put result in Q31 (convert from Q28)
tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);

8
webrtc/common_audio/signal_processing/filter_ar.c
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@ -51,13 +51,13 @@ int WebRtcSpl_FilterAR(const int16_t* a,
stop = (i < a_length) ? i + 1 : a_length;
for (j = 1; j < stop; j++)
{
o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
o -= *a_ptr * *filtered_ptr--;
oLOW -= *a_ptr++ * *filtered_low_ptr--;
}
for (j = i + 1; j < a_length; j++)
{
o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
o -= *a_ptr * *state_ptr--;
oLOW -= *a_ptr++ * *state_low_ptr--;
}
o += (oLOW >> 12);

8
webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
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@ -23,18 +23,14 @@ void WebRtcSpl_FilterMAFastQ12(const int16_t* in_ptr,
int16_t B_length,
int16_t length)
{
int32_t o;
int i, j;
for (i = 0; i < length; i++)
{
const int16_t* b_ptr = &B[0];
const int16_t* x_ptr = &in_ptr[i];
o = (int32_t)0;
int32_t o = 0;
for (j = 0; j < B_length; j++)
{
o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
o += B[j] * in_ptr[i - j];
}
// If output is higher than 32768, saturate it. Same with negative side

14
webrtc/common_audio/signal_processing/ilbc_specific_functions.c
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@ -68,16 +68,11 @@ void WebRtcSpl_AddAffineVectorToVector(int16_t *out, int16_t *in,
int16_t gain, int32_t add_constant,
int16_t right_shifts, int vector_length)
{
int16_t *inPtr;
int16_t *outPtr;
int i;
inPtr = in;
outPtr = out;
for (i = 0; i < vector_length; i++)
{
(*outPtr++) += (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
+ (int32_t)add_constant) >> right_shifts);
out[i] += (int16_t)((in[i] * gain + add_constant) >> right_shifts);
}
}
@ -85,15 +80,10 @@ void WebRtcSpl_AffineTransformVector(int16_t *out, int16_t *in,
int16_t gain, int32_t add_constant,
int16_t right_shifts, int vector_length)
{
int16_t *inPtr;
int16_t *outPtr;
int i;
inPtr = in;
outPtr = out;
for (i = 0; i < vector_length; i++)
{
(*outPtr++) = (int16_t)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
+ (int32_t)add_constant) >> right_shifts);
out[i] = (int16_t)((in[i] * gain + add_constant) >> right_shifts);
}
}

29
webrtc/common_audio/signal_processing/levinson_durbin.c
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@ -76,8 +76,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
// Alpha = R[0] * (1-K^2)
temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
<< 1); // temp1W32 = k^2 in Q31
temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) << 1; // = k^2 in Q31
temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
temp1W32 = (int32_t)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
@ -87,9 +86,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
// Calculate Alpha in Q31
temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
+ (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
+ (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
temp1W32 = (R_hi[0] * tmp_hi + (R_hi[0] * tmp_low >> 15) +
(R_low[0] * tmp_hi >> 15)) << 1;
// Normalize Alpha and put it in hi and low format
@ -113,10 +111,10 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
for (j = 1; j < i; j++)
{
// temp1W32 is in Q31
temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
+ (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
+ (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
// temp1W32 is in Q31
temp1W32 += (R_hi[j] * A_hi[i - j] << 1) +
(((R_hi[j] * A_low[i - j] >> 15) +
(R_low[j] * A_hi[i - j] >> 15)) << 1);
}
temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
@ -177,9 +175,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
+ WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[j],1);
// temp1W32 += K*A[i-j] in Q27
temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
+ (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
+ (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
temp1W32 += (K_hi * A_hi[i - j] + (K_hi * A_low[i - j] >> 15) +
(K_low * A_hi[i - j] >> 15)) << 1;
// Put Anew in hi and low format
A_upd_hi[j] = (int16_t)(temp1W32 >> 16);
@ -197,8 +194,7 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
// Alpha = Alpha * (1-K^2)
temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
+ WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) << 1; // K*K in Q31
temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
temp1W32 = (int32_t)0x7fffffffL - temp1W32; // 1 - K*K in Q31
@ -208,9 +204,8 @@ int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
// Calculate Alpha = Alpha * (1-K^2) in Q31
temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
+ (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
+ (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
temp1W32 = (Alpha_hi * tmp_hi + (Alpha_hi * tmp_low >> 15) +
(Alpha_low * tmp_hi >> 15)) << 1;
// Normalize Alpha and store it on hi and low format

5
webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
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@ -30,7 +30,7 @@ void WebRtcSpl_LpcToReflCoef(int16_t* a16, int use_order, int16_t* k16)
for (m = use_order - 1; m > 0; m--)
{
// (1 - k^2) in Q30
tmp_inv_denom32 = ((int32_t)1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]);
tmp_inv_denom32 = 1073741823 - k16[m] * k16[m];
// (1 - k^2) in Q15
tmp_inv_denom16 = (int16_t)(tmp_inv_denom32 >> 15);
@ -39,8 +39,7 @@ void WebRtcSpl_LpcToReflCoef(int16_t* a16, int use_order, int16_t* k16)
// tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
// [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
tmp32[k] = WEBRTC_SPL_LSHIFT_W32((int32_t)a16[k], 16)
- WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
tmp32[k] = (a16[k] << 16) - (k16[m] * a16[m - k + 1] << 1);
tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
}

10
webrtc/common_audio/signal_processing/spl_sqrt.c
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@ -49,16 +49,16 @@ int32_t WebRtcSpl_SqrtLocal(int32_t in)
A >>= 16;
A = A * A * 2; // A = (x/2)^4
t16 = (int16_t)(A >> 16);
B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
B += -20480 * t16 * 2; // B = B - 0.625*A
// After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
A = x_half * t16 * 2; // A = (x/2)^5
t16 = (int16_t)(A >> 16);
B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
B += 28672 * t16 * 2; // B = B + 0.875*A
// After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
t16 = (int16_t)(x2 >> 16);
A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
A = x_half * t16 * 2; // A = x/2^3
B = B + (A >> 1); // B = B + 0.5*A
// After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
@ -166,7 +166,7 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
t16 = (int16_t)(A >> 16); // t16 = AH
A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
A = k_sqrt_2 * t16 * 2; // A = 1/sqrt(2)*t16
A = A + ((int32_t)32768); // Round off
A = A & ((int32_t)0x7fff0000); // Round off

2
webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
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@ -27,7 +27,7 @@ void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, int vector_length,
for (m = 0; m < vector_length; m++)
{
tmp = xQ15[m];
sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
sq = tmp * tmp; // x^2 in Q30
sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
yQ15[m] = (int16_t)sq;

5
webrtc/common_audio/signal_processing/vector_scaling_operations.c
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@ -157,9 +157,8 @@ int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
for (i = 0; i < length; i++) {
out_vector[i] = (int16_t)((
WEBRTC_SPL_MUL_16_16(in_vector1[i], in_vector1_scale)
+ WEBRTC_SPL_MUL_16_16(in_vector2[i], in_vector2_scale)
+ round_value) >> right_shifts);
in_vector1[i] * in_vector1_scale + in_vector2[i] * in_vector2_scale +
round_value) >> right_shifts);
}
return 0;