mirror of https://github.com/mstorsjo/fdk-aac.git
Merge "Prevent overflow in usac filters" am: 4053380094
am: eaad782db3
am: 18d4dcf255
Change-Id: Id1736dd6af5b656c158896f6d9fcb117231351a4
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commit
3f50499c9c
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@ -579,11 +579,11 @@ void Syn_filt(const FIXP_LPC a[], /* (i) : a[m] prediction coefficients Q12 */
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L_tmp = (FIXP_DBL)0;
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for (j = 0; j < M_LP_FILTER_ORDER; j++) {
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L_tmp -= fMultDiv2(a[j], y[i - (j + 1)]);
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L_tmp -= fMultDiv2(a[j], y[i - (j + 1)]) >> (LP_FILTER_SCALE - 1);
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}
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L_tmp = scaleValue(L_tmp, a_exp + 1);
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y[i] = L_tmp + x[i];
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L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE);
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y[i] = fAddSaturate(L_tmp, x[i]);
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}
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return;
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@ -631,10 +631,10 @@ void E_UTIL_residu(const FIXP_LPC *a, const INT a_exp, FIXP_DBL *x, FIXP_DBL *y,
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s = (FIXP_DBL)0;
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for (j = 0; j < M_LP_FILTER_ORDER; j++) {
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s += fMultDiv2(a[j], x[i - j - 1]);
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s += fMultDiv2(a[j], x[i - j - 1]) >> (LP_FILTER_SCALE - 1);
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}
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s = scaleValue(s, a_exp + 1);
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s = scaleValue(s, a_exp + LP_FILTER_SCALE);
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y[i] = fAddSaturate(s, x[i]);
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}
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@ -115,6 +115,7 @@ amm-info@iis.fraunhofer.de
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/* definitions which are independent of coreCoderFrameLength */
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#define M_LP_FILTER_ORDER 16 /* LP filter order */
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#define LP_FILTER_SCALE 4 /* LP filter scale */
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#define PIT_MIN_12k8 34 /* Minimum pitch lag with resolution 1/4 */
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#define PIT_MAX_12k8 231 /* Maximum pitch lag for fs=12.8kHz */
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@ -191,13 +191,11 @@ static void Syn_filt_zero(const FIXP_LPC a[], const INT a_exp, INT length,
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L_tmp = (FIXP_DBL)0;
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for (j = 0; j < fMin(i, M_LP_FILTER_ORDER); j++) {
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L_tmp -= fMultDiv2(a[j], x[i - (j + 1)]);
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L_tmp -= fMultDiv2(a[j], x[i - (j + 1)]) >> (LP_FILTER_SCALE - 1);
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}
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L_tmp = scaleValue(L_tmp, a_exp + 1);
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x[i] = scaleValueSaturate((x[i] >> 1) + (L_tmp >> 1),
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1); /* Avoid overflow issues and saturate. */
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L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE);
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x[i] = fAddSaturate(x[i], L_tmp);
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}
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}
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