X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavfilter%2Faf_aiir.c;h=a95f66d6bcf8742fd7b18227a81e652837cf6055;hb=a04ad248a05e7b613abe09b3bb067f555108d794;hp=20dea98cbbd758e6536286d9552b0e5b73c770f9;hpb=b2f32d60eeaf883bb7d9e1b8cc2fb9a983d08f72;p=ffmpeg diff --git a/libavfilter/af_aiir.c b/libavfilter/af_aiir.c index 20dea98cbbd..a95f66d6bcf 100644 --- a/libavfilter/af_aiir.c +++ b/libavfilter/af_aiir.c @@ -38,10 +38,9 @@ typedef struct Pair { } Pair; typedef struct BiquadContext { - double a0, a1, a2; - double b0, b1, b2; - double i1, i2; - double o1, o2; + double a[3]; + double b[3]; + double w1, w2; } BiquadContext; typedef struct IIRChannel { @@ -49,6 +48,7 @@ typedef struct IIRChannel { double *ab[2]; double g; double *cache[2]; + double fir; BiquadContext *biquads; int clippings; } IIRChannel; @@ -57,6 +57,8 @@ typedef struct AudioIIRContext { const AVClass *class; char *a_str, *b_str, *g_str; double dry_gain, wet_gain; + double mix; + int normalize; int format; int process; int precision; @@ -93,7 +95,7 @@ static int query_formats(AVFilterContext *ctx) AVFilterLink *videolink = ctx->outputs[1]; formats = ff_make_format_list(pix_fmts); - if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0) + if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0) return ret; } @@ -124,15 +126,17 @@ static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) AudioIIRContext *s = ctx->priv; \ const double ig = s->dry_gain; \ const double og = s->wet_gain; \ + const double mix = s->mix; \ ThreadData *td = arg; \ AVFrame *in = td->in, *out = td->out; \ const type *src = (const type *)in->extended_data[ch]; \ - double *ic = (double *)s->iir[ch].cache[0]; \ - double *oc = (double *)s->iir[ch].cache[1]; \ + double *oc = (double *)s->iir[ch].cache[0]; \ + double *ic = (double *)s->iir[ch].cache[1]; \ const int nb_a = s->iir[ch].nb_ab[0]; \ const int nb_b = s->iir[ch].nb_ab[1]; \ const double *a = s->iir[ch].ab[0]; \ const double *b = s->iir[ch].ab[1]; \ + const double g = s->iir[ch].g; \ int *clippings = &s->iir[ch].clippings; \ type *dst = (type *)out->extended_data[ch]; \ int n; \ @@ -151,7 +155,8 @@ static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) sample -= a[x] * oc[x]; \ \ oc[0] = sample; \ - sample *= og; \ + sample *= og * g; \ + sample = sample * mix + ic[0] * (1. - mix); \ if (need_clipping && sample < min) { \ (*clippings)++; \ dst[n] = min; \ @@ -171,42 +176,43 @@ IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) IIR_CH(fltp, float, -1., 1., 0) IIR_CH(dblp, double, -1., 1., 0) -#define SERIAL_IIR_CH(name, type, min, max, need_clipping) \ -static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \ +#define SERIAL_IIR_CH(name, type, min, max, need_clipping) \ +static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, \ + int ch, int nb_jobs) \ { \ AudioIIRContext *s = ctx->priv; \ const double ig = s->dry_gain; \ const double og = s->wet_gain; \ + const double mix = s->mix; \ + const double imix = 1. - mix; \ ThreadData *td = arg; \ AVFrame *in = td->in, *out = td->out; \ const type *src = (const type *)in->extended_data[ch]; \ type *dst = (type *)out->extended_data[ch]; \ IIRChannel *iir = &s->iir[ch]; \ + const double g = iir->g; \ int *clippings = &iir->clippings; \ int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \ int n, i; \ \ - for (i = 0; i < nb_biquads; i++) { \ - const double a1 = -iir->biquads[i].a1; \ - const double a2 = -iir->biquads[i].a2; \ - const double b0 = iir->biquads[i].b0; \ - const double b1 = iir->biquads[i].b1; \ - const double b2 = iir->biquads[i].b2; \ - double i1 = iir->biquads[i].i1; \ - double i2 = iir->biquads[i].i2; \ - double o1 = iir->biquads[i].o1; \ - double o2 = iir->biquads[i].o2; \ + for (i = nb_biquads - 1; i >= 0; i--) { \ + const double a1 = -iir->biquads[i].a[1]; \ + const double a2 = -iir->biquads[i].a[2]; \ + const double b0 = iir->biquads[i].b[0]; \ + const double b1 = iir->biquads[i].b[1]; \ + const double b2 = iir->biquads[i].b[2]; \ + double w1 = iir->biquads[i].w1; \ + double w2 = iir->biquads[i].w2; \ \ for (n = 0; n < in->nb_samples; n++) { \ - double sample = ig * (i ? dst[n] : src[n]); \ - double o0 = sample * b0 + i1 * b1 + i2 * b2 + o1 * a1 + o2 * a2; \ + double i0 = ig * (i ? dst[n] : src[n]); \ + double o0 = i0 * b0 + w1; \ \ - i2 = i1; \ - i1 = src[n]; \ - o2 = o1; \ - o1 = o0; \ - o0 *= og; \ + w1 = b1 * i0 + w2 + a1 * o0; \ + w2 = b2 * i0 + a2 * o0; \ + o0 *= og * g; \ \ + o0 = o0 * mix + imix * i0; \ if (need_clipping && o0 < min) { \ (*clippings)++; \ dst[n] = min; \ @@ -217,10 +223,8 @@ static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, int ch, int nb dst[n] = o0; \ } \ } \ - iir->biquads[i].i1 = i1; \ - iir->biquads[i].i2 = i2; \ - iir->biquads[i].o1 = o1; \ - iir->biquads[i].o2 = o2; \ + iir->biquads[i].w1 = w1; \ + iir->biquads[i].w2 = w2; \ } \ \ return 0; \ @@ -231,6 +235,127 @@ SERIAL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) SERIAL_IIR_CH(fltp, float, -1., 1., 0) SERIAL_IIR_CH(dblp, double, -1., 1., 0) +#define PARALLEL_IIR_CH(name, type, min, max, need_clipping) \ +static int iir_ch_parallel_## name(AVFilterContext *ctx, void *arg, \ + int ch, int nb_jobs) \ +{ \ + AudioIIRContext *s = ctx->priv; \ + const double ig = s->dry_gain; \ + const double og = s->wet_gain; \ + const double mix = s->mix; \ + const double imix = 1. - mix; \ + ThreadData *td = arg; \ + AVFrame *in = td->in, *out = td->out; \ + const type *src = (const type *)in->extended_data[ch]; \ + type *dst = (type *)out->extended_data[ch]; \ + IIRChannel *iir = &s->iir[ch]; \ + const double g = iir->g; \ + const double fir = iir->fir; \ + int *clippings = &iir->clippings; \ + int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \ + int n, i; \ + \ + for (i = 0; i < nb_biquads; i++) { \ + const double a1 = -iir->biquads[i].a[1]; \ + const double a2 = -iir->biquads[i].a[2]; \ + const double b1 = iir->biquads[i].b[1]; \ + const double b2 = iir->biquads[i].b[2]; \ + double w1 = iir->biquads[i].w1; \ + double w2 = iir->biquads[i].w2; \ + \ + for (n = 0; n < in->nb_samples; n++) { \ + double i0 = ig * src[n]; \ + double o0 = w1; \ + \ + w1 = b1 * i0 + w2 + a1 * o0; \ + w2 = b2 * i0 + a2 * o0; \ + o0 *= og * g; \ + o0 += dst[n]; \ + \ + if (need_clipping && o0 < min) { \ + (*clippings)++; \ + dst[n] = min; \ + } else if (need_clipping && o0 > max) { \ + (*clippings)++; \ + dst[n] = max; \ + } else { \ + dst[n] = o0; \ + } \ + } \ + iir->biquads[i].w1 = w1; \ + iir->biquads[i].w2 = w2; \ + } \ + \ + for (n = 0; n < in->nb_samples; n++) { \ + dst[n] += fir * src[n]; \ + dst[n] = dst[n] * mix + imix * src[n]; \ + } \ + \ + return 0; \ +} + +PARALLEL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) +PARALLEL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) +PARALLEL_IIR_CH(fltp, float, -1., 1., 0) +PARALLEL_IIR_CH(dblp, double, -1., 1., 0) + +#define LATTICE_IIR_CH(name, type, min, max, need_clipping) \ +static int iir_ch_lattice_## name(AVFilterContext *ctx, void *arg, \ + int ch, int nb_jobs) \ +{ \ + AudioIIRContext *s = ctx->priv; \ + const double ig = s->dry_gain; \ + const double og = s->wet_gain; \ + const double mix = s->mix; \ + ThreadData *td = arg; \ + AVFrame *in = td->in, *out = td->out; \ + const type *src = (const type *)in->extended_data[ch]; \ + double n0, n1, p0, *x = (double *)s->iir[ch].cache[0]; \ + const int nb_stages = s->iir[ch].nb_ab[1]; \ + const double *v = s->iir[ch].ab[0]; \ + const double *k = s->iir[ch].ab[1]; \ + const double g = s->iir[ch].g; \ + int *clippings = &s->iir[ch].clippings; \ + type *dst = (type *)out->extended_data[ch]; \ + int n; \ + \ + for (n = 0; n < in->nb_samples; n++) { \ + const double in = src[n] * ig; \ + double out = 0.; \ + \ + n1 = in; \ + for (int i = nb_stages - 1; i >= 0; i--) { \ + n0 = n1 - k[i] * x[i]; \ + p0 = n0 * k[i] + x[i]; \ + out += p0 * v[i+1]; \ + x[i] = p0; \ + n1 = n0; \ + } \ + \ + out += n1 * v[0]; \ + memmove(&x[1], &x[0], nb_stages * sizeof(*x)); \ + x[0] = n1; \ + out *= og * g; \ + out = out * mix + in * (1. - mix); \ + if (need_clipping && out < min) { \ + (*clippings)++; \ + dst[n] = min; \ + } else if (need_clipping && out > max) { \ + (*clippings)++; \ + dst[n] = max; \ + } else { \ + dst[n] = out; \ + } \ + } \ + \ + return 0; \ +} + +LATTICE_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) +LATTICE_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) +LATTICE_IIR_CH(fltp, float, -1., 1., 0) +LATTICE_IIR_CH(dblp, double, -1., 1., 0) + static void count_coefficients(char *item_str, int *nb_items) { char *p; @@ -264,7 +389,7 @@ static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items) } p = NULL; - if (sscanf(arg, "%lf", &s->iir[i].g) != 1) { + if (av_sscanf(arg, "%lf", &s->iir[i].g) != 1) { av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg); av_freep(&old_str); return AVERROR(EINVAL); @@ -291,7 +416,7 @@ static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite break; p = NULL; - if (sscanf(arg, "%lf", &dst[i]) != 1) { + if (av_sscanf(arg, "%lf", &dst[i]) != 1) { av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg); av_freep(&old_str); return AVERROR(EINVAL); @@ -316,7 +441,7 @@ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite break; p = NULL; - if (sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) { + if (av_sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) { av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg); av_freep(&old_str); return AVERROR(EINVAL); @@ -328,7 +453,7 @@ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite return 0; } -static const char *format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd" }; +static const char *const format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd", "%lf %lfi" }; static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab) { @@ -360,7 +485,7 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, return AVERROR(ENOMEM); } - if (s->format) { + if (s->format > 0) { ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]); } else { ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]); @@ -377,50 +502,65 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, return 0; } -static void multiply(double wre, double wim, int npz, double *coeffs) +static void cmul(double re, double im, double re2, double im2, double *RE, double *IM) { - double nwre = -wre, nwim = -wim; - double cre, cim; - int i; + *RE = re * re2 - im * im2; + *IM = re * im2 + re2 * im; +} + +static int expand(AVFilterContext *ctx, double *pz, int n, double *coefs) +{ + coefs[2 * n] = 1.0; + + for (int i = 1; i <= n; i++) { + for (int j = n - i; j < n; j++) { + double re, im; - for (i = npz; i >= 1; i--) { - cre = coeffs[2 * i + 0]; - cim = coeffs[2 * i + 1]; + cmul(coefs[2 * (j + 1)], coefs[2 * (j + 1) + 1], + pz[2 * (i - 1)], pz[2 * (i - 1) + 1], &re, &im); - coeffs[2 * i + 0] = (nwre * cre - nwim * cim) + coeffs[2 * (i - 1) + 0]; - coeffs[2 * i + 1] = (nwre * cim + nwim * cre) + coeffs[2 * (i - 1) + 1]; + coefs[2 * j] -= re; + coefs[2 * j + 1] -= im; + } } - cre = coeffs[0]; - cim = coeffs[1]; - coeffs[0] = nwre * cre - nwim * cim; - coeffs[1] = nwre * cim + nwim * cre; + for (int i = 0; i < n + 1; i++) { + if (fabs(coefs[2 * i + 1]) > FLT_EPSILON) { + av_log(ctx, AV_LOG_ERROR, "coefs: %f of z^%d is not real; poles/zeros are not complex conjugates.\n", + coefs[2 * i + 1], i); + return AVERROR(EINVAL); + } + } + + return 0; } -static int expand(AVFilterContext *ctx, double *pz, int nb, double *coeffs) +static void normalize_coeffs(AVFilterContext *ctx, int ch) { - int i; + AudioIIRContext *s = ctx->priv; + IIRChannel *iir = &s->iir[ch]; + double sum_den = 0.; - coeffs[0] = 1.0; - coeffs[1] = 0.0; + if (!s->normalize) + return; - for (i = 0; i < nb; i++) { - coeffs[2 * (i + 1) ] = 0.0; - coeffs[2 * (i + 1) + 1] = 0.0; + for (int i = 0; i < iir->nb_ab[1]; i++) { + sum_den += iir->ab[1][i]; } - for (i = 0; i < nb; i++) - multiply(pz[2 * i], pz[2 * i + 1], nb, coeffs); + if (sum_den > 1e-6) { + double factor, sum_num = 0.; - for (i = 0; i < nb + 1; i++) { - if (fabs(coeffs[2 * i + 1]) > FLT_EPSILON) { - av_log(ctx, AV_LOG_ERROR, "coeff: %f of z^%d is not real; poles/zeros are not complex conjugates.\n", - coeffs[2 * i + 1], i); - return AVERROR(EINVAL); + for (int i = 0; i < iir->nb_ab[0]; i++) { + sum_num += iir->ab[0][i]; } - } - return 0; + factor = sum_num / sum_den; + + for (int i = 0; i < iir->nb_ab[1]; i++) { + iir->ab[1][i] *= factor; + } + } } static int convert_zp2tf(AVFilterContext *ctx, int channels) @@ -432,8 +572,8 @@ static int convert_zp2tf(AVFilterContext *ctx, int channels) IIRChannel *iir = &s->iir[ch]; double *topc, *botc; - topc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*topc)); - botc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*botc)); + topc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*topc)); + botc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*botc)); if (!topc || !botc) { ret = AVERROR(ENOMEM); goto fail; @@ -459,6 +599,8 @@ static int convert_zp2tf(AVFilterContext *ctx, int channels) } iir->nb_ab[0]++; + normalize_coeffs(ctx, ch); + fail: av_free(topc); av_free(botc); @@ -492,6 +634,7 @@ static int decompose_zp2biquads(AVFilterContext *ctx, int channels) double a[6] = { 0 }; double min_distance = DBL_MAX; double max_mag = 0; + double factor; int i; for (i = 0; i < iir->nb_ab[0]; i++) { @@ -507,7 +650,7 @@ static int decompose_zp2biquads(AVFilterContext *ctx, int channels) } } - for (i = 0; i < iir->nb_ab[1]; i++) { + for (i = 0; i < iir->nb_ab[0]; i++) { if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1])) continue; @@ -586,20 +729,42 @@ static int decompose_zp2biquads(AVFilterContext *ctx, int channels) iir->ab[1][2 * nearest_zero.a] = iir->ab[1][2 * nearest_zero.a + 1] = NAN; iir->ab[1][2 * nearest_zero.b] = iir->ab[1][2 * nearest_zero.b + 1] = NAN; - iir->biquads[current_biquad].a0 = 1.0; - iir->biquads[current_biquad].a1 = a[2] / a[4]; - iir->biquads[current_biquad].a2 = a[0] / a[4]; - iir->biquads[current_biquad].b0 = b[4] / a[4] * (current_biquad ? 1.0 : iir->g); - iir->biquads[current_biquad].b1 = b[2] / a[4] * (current_biquad ? 1.0 : iir->g); - iir->biquads[current_biquad].b2 = b[0] / a[4] * (current_biquad ? 1.0 : iir->g); + iir->biquads[current_biquad].a[0] = 1.; + iir->biquads[current_biquad].a[1] = a[2] / a[4]; + iir->biquads[current_biquad].a[2] = a[0] / a[4]; + iir->biquads[current_biquad].b[0] = b[4] / a[4]; + iir->biquads[current_biquad].b[1] = b[2] / a[4]; + iir->biquads[current_biquad].b[2] = b[0] / a[4]; + + if (s->normalize && + fabs(iir->biquads[current_biquad].b[0] + + iir->biquads[current_biquad].b[1] + + iir->biquads[current_biquad].b[2]) > 1e-6) { + factor = (iir->biquads[current_biquad].a[0] + + iir->biquads[current_biquad].a[1] + + iir->biquads[current_biquad].a[2]) / + (iir->biquads[current_biquad].b[0] + + iir->biquads[current_biquad].b[1] + + iir->biquads[current_biquad].b[2]); + + av_log(ctx, AV_LOG_VERBOSE, "factor=%f\n", factor); + + iir->biquads[current_biquad].b[0] *= factor; + iir->biquads[current_biquad].b[1] *= factor; + iir->biquads[current_biquad].b[2] *= factor; + } + + iir->biquads[current_biquad].b[0] *= (current_biquad ? 1.0 : iir->g); + iir->biquads[current_biquad].b[1] *= (current_biquad ? 1.0 : iir->g); + iir->biquads[current_biquad].b[2] *= (current_biquad ? 1.0 : iir->g); av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n", - iir->biquads[current_biquad].a0, - iir->biquads[current_biquad].a1, - iir->biquads[current_biquad].a2, - iir->biquads[current_biquad].b0, - iir->biquads[current_biquad].b1, - iir->biquads[current_biquad].b2); + iir->biquads[current_biquad].a[0], + iir->biquads[current_biquad].a[1], + iir->biquads[current_biquad].a[2], + iir->biquads[current_biquad].b[0], + iir->biquads[current_biquad].b[1], + iir->biquads[current_biquad].b[2]); current_biquad++; } @@ -608,6 +773,128 @@ static int decompose_zp2biquads(AVFilterContext *ctx, int channels) return 0; } +static void biquad_process(double *x, double *y, int length, + double b0, double b1, double b2, + double a1, double a2) +{ + double w1 = 0., w2 = 0.; + + a1 = -a1; + a2 = -a2; + + for (int n = 0; n < length; n++) { + double out, in = x[n]; + + y[n] = out = in * b0 + w1; + w1 = b1 * in + w2 + a1 * out; + w2 = b2 * in + a2 * out; + } +} + +static void solve(double *matrix, double *vector, int n, double *y, double *x, double *lu) +{ + double sum = 0.; + + for (int i = 0; i < n; i++) { + for (int j = i; j < n; j++) { + sum = 0.; + for (int k = 0; k < i; k++) + sum += lu[i * n + k] * lu[k * n + j]; + lu[i * n + j] = matrix[j * n + i] - sum; + } + for (int j = i + 1; j < n; j++) { + sum = 0.; + for (int k = 0; k < i; k++) + sum += lu[j * n + k] * lu[k * n + i]; + lu[j * n + i] = (1. / lu[i * n + i]) * (matrix[i * n + j] - sum); + } + } + + for (int i = 0; i < n; i++) { + sum = 0.; + for (int k = 0; k < i; k++) + sum += lu[i * n + k] * y[k]; + y[i] = vector[i] - sum; + } + + for (int i = n - 1; i >= 0; i--) { + sum = 0.; + for (int k = i + 1; k < n; k++) + sum += lu[i * n + k] * x[k]; + x[i] = (1 / lu[i * n + i]) * (y[i] - sum); + } +} + +static int convert_serial2parallel(AVFilterContext *ctx, int channels) +{ + AudioIIRContext *s = ctx->priv; + int ret = 0; + + for (int ch = 0; ch < channels; ch++) { + IIRChannel *iir = &s->iir[ch]; + int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; + int length = nb_biquads * 2 + 1; + double *impulse = av_calloc(length, sizeof(*impulse)); + double *y = av_calloc(length, sizeof(*y)); + double *resp = av_calloc(length, sizeof(*resp)); + double *M = av_calloc((length - 1) * 2 * nb_biquads, sizeof(*M)); + double *W = av_calloc((length - 1) * 2 * nb_biquads, sizeof(*W)); + + if (!impulse || !y || !resp || !M) { + av_free(impulse); + av_free(y); + av_free(resp); + av_free(M); + av_free(W); + return AVERROR(ENOMEM); + } + + impulse[0] = 1.; + + for (int n = 0; n < nb_biquads; n++) { + BiquadContext *biquad = &iir->biquads[n]; + + biquad_process(n ? y : impulse, y, length, + biquad->b[0], biquad->b[1], biquad->b[2], + biquad->a[1], biquad->a[2]); + } + + for (int n = 0; n < nb_biquads; n++) { + BiquadContext *biquad = &iir->biquads[n]; + + biquad_process(impulse, resp, length - 1, + 1., 0., 0., biquad->a[1], biquad->a[2]); + + memcpy(M + n * 2 * (length - 1), resp, sizeof(*resp) * (length - 1)); + memcpy(M + n * 2 * (length - 1) + length, resp, sizeof(*resp) * (length - 2)); + memset(resp, 0, length * sizeof(*resp)); + } + + solve(M, &y[1], length - 1, &impulse[1], resp, W); + + iir->fir = y[0]; + + for (int n = 0; n < nb_biquads; n++) { + BiquadContext *biquad = &iir->biquads[n]; + + biquad->b[0] = 0.; + biquad->b[1] = resp[n * 2 + 0]; + biquad->b[2] = resp[n * 2 + 1]; + } + + av_free(impulse); + av_free(y); + av_free(resp); + av_free(M); + av_free(W); + + if (ret < 0) + return ret; + } + + return 0; +} + static void convert_pr2zp(AVFilterContext *ctx, int channels) { AudioIIRContext *s = ctx->priv; @@ -635,6 +922,87 @@ static void convert_pr2zp(AVFilterContext *ctx, int channels) } } +static void convert_sp2zp(AVFilterContext *ctx, int channels) +{ + AudioIIRContext *s = ctx->priv; + int ch; + + for (ch = 0; ch < channels; ch++) { + IIRChannel *iir = &s->iir[ch]; + int n; + + for (n = 0; n < iir->nb_ab[0]; n++) { + double sr = iir->ab[0][2*n]; + double si = iir->ab[0][2*n+1]; + + iir->ab[0][2*n] = exp(sr) * cos(si); + iir->ab[0][2*n+1] = exp(sr) * sin(si); + } + + for (n = 0; n < iir->nb_ab[1]; n++) { + double sr = iir->ab[1][2*n]; + double si = iir->ab[1][2*n+1]; + + iir->ab[1][2*n] = exp(sr) * cos(si); + iir->ab[1][2*n+1] = exp(sr) * sin(si); + } + } +} + +static double fact(double i) +{ + if (i <= 0.) + return 1.; + return i * fact(i - 1.); +} + +static double coef_sf2zf(double *a, int N, int n) +{ + double z = 0.; + + for (int i = 0; i <= N; i++) { + double acc = 0.; + + for (int k = FFMAX(n - N + i, 0); k <= FFMIN(i, n); k++) { + acc += ((fact(i) * fact(N - i)) / + (fact(k) * fact(i - k) * fact(n - k) * fact(N - i - n + k))) * + ((k & 1) ? -1. : 1.); + } + + z += a[i] * pow(2., i) * acc; + } + + return z; +} + +static void convert_sf2tf(AVFilterContext *ctx, int channels) +{ + AudioIIRContext *s = ctx->priv; + int ch; + + for (ch = 0; ch < channels; ch++) { + IIRChannel *iir = &s->iir[ch]; + double *temp0 = av_calloc(iir->nb_ab[0], sizeof(*temp0)); + double *temp1 = av_calloc(iir->nb_ab[1], sizeof(*temp1)); + + if (!temp0 || !temp1) + goto next; + + memcpy(temp0, iir->ab[0], iir->nb_ab[0] * sizeof(*temp0)); + memcpy(temp1, iir->ab[1], iir->nb_ab[1] * sizeof(*temp1)); + + for (int n = 0; n < iir->nb_ab[0]; n++) + iir->ab[0][n] = coef_sf2zf(temp0, iir->nb_ab[0] - 1, n); + + for (int n = 0; n < iir->nb_ab[1]; n++) + iir->ab[1][n] = coef_sf2zf(temp1, iir->nb_ab[1] - 1, n); + +next: + av_free(temp0); + av_free(temp1); + } +} + static void convert_pd2zp(AVFilterContext *ctx, int channels) { AudioIIRContext *s = ctx->priv; @@ -662,6 +1030,25 @@ static void convert_pd2zp(AVFilterContext *ctx, int channels) } } +static void check_stability(AVFilterContext *ctx, int channels) +{ + AudioIIRContext *s = ctx->priv; + int ch; + + for (ch = 0; ch < channels; ch++) { + IIRChannel *iir = &s->iir[ch]; + + for (int n = 0; n < iir->nb_ab[0]; n++) { + double pr = hypot(iir->ab[0][2*n], iir->ab[0][2*n+1]); + + if (pr >= 1.) { + av_log(ctx, AV_LOG_WARNING, "pole %d at channel %d is unstable\n", n, ch); + break; + } + } + } +} + static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color) { const uint8_t *font; @@ -711,100 +1098,140 @@ static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t col } } -static void draw_response(AVFilterContext *ctx, AVFrame *out) +static double distance(double x0, double x1, double y0, double y1) +{ + return hypot(x0 - x1, y0 - y1); +} + +static void get_response(int channel, int format, double w, + const double *b, const double *a, + int nb_b, int nb_a, double *magnitude, double *phase) +{ + double realz, realp; + double imagz, imagp; + double real, imag; + double div; + + if (format == 0) { + realz = 0., realp = 0.; + imagz = 0., imagp = 0.; + for (int x = 0; x < nb_a; x++) { + realz += cos(-x * w) * a[x]; + imagz += sin(-x * w) * a[x]; + } + + for (int x = 0; x < nb_b; x++) { + realp += cos(-x * w) * b[x]; + imagp += sin(-x * w) * b[x]; + } + + div = realp * realp + imagp * imagp; + real = (realz * realp + imagz * imagp) / div; + imag = (imagz * realp - imagp * realz) / div; + + *magnitude = hypot(real, imag); + *phase = atan2(imag, real); + } else { + double p = 1., z = 1.; + double acc = 0.; + + for (int x = 0; x < nb_a; x++) { + z *= distance(cos(w), a[2 * x], sin(w), a[2 * x + 1]); + acc += atan2(sin(w) - a[2 * x + 1], cos(w) - a[2 * x]); + } + + for (int x = 0; x < nb_b; x++) { + p *= distance(cos(w), b[2 * x], sin(w), b[2 * x + 1]); + acc -= atan2(sin(w) - b[2 * x + 1], cos(w) - b[2 * x]); + } + + *magnitude = z / p; + *phase = acc; + } +} + +static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate) { AudioIIRContext *s = ctx->priv; - float *mag, *phase, min = FLT_MAX, max = FLT_MIN; - int prev_ymag = -1, prev_yphase = -1; + double *mag, *phase, *temp, *delay, min = DBL_MAX, max = -DBL_MAX; + double min_delay = DBL_MAX, max_delay = -DBL_MAX, min_phase, max_phase; + int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1; char text[32]; - int ch, i, x; + int ch, i; memset(out->data[0], 0, s->h * out->linesize[0]); phase = av_malloc_array(s->w, sizeof(*phase)); + temp = av_malloc_array(s->w, sizeof(*temp)); mag = av_malloc_array(s->w, sizeof(*mag)); - if (!mag || !phase) + delay = av_malloc_array(s->w, sizeof(*delay)); + if (!mag || !phase || !delay || !temp) goto end; ch = av_clip(s->ir_channel, 0, s->channels - 1); for (i = 0; i < s->w; i++) { const double *b = s->iir[ch].ab[0]; const double *a = s->iir[ch].ab[1]; + const int nb_b = s->iir[ch].nb_ab[0]; + const int nb_a = s->iir[ch].nb_ab[1]; double w = i * M_PI / (s->w - 1); - double realz, realp; - double imagz, imagp; - double real, imag, div; - - if (s->format == 0) { - realz = 0., realp = 0.; - imagz = 0., imagp = 0.; - for (x = 0; x < s->iir[ch].nb_ab[1]; x++) { - realz += cos(-x * w) * a[x]; - imagz += sin(-x * w) * a[x]; - } + double m, p; - for (x = 0; x < s->iir[ch].nb_ab[0]; x++) { - realp += cos(-x * w) * b[x]; - imagp += sin(-x * w) * b[x]; - } - - div = realp * realp + imagp * imagp; - real = (realz * realp + imagz * imagp) / div; - imag = (imagz * realp - imagp * realz) / div; - } else { - real = 1; - imag = 0; - for (x = 0; x < s->iir[ch].nb_ab[1]; x++) { - double ore, oim, re, im; - - re = cos(w) - a[2 * x]; - im = sin(w) - a[2 * x + 1]; + get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p); - ore = real; - oim = imag; - - real = ore * re - oim * im; - imag = ore * im + oim * re; - } - - for (x = 0; x < s->iir[ch].nb_ab[0]; x++) { - double ore, oim, re, im; + mag[i] = s->iir[ch].g * m; + phase[i] = p; + min = fmin(min, mag[i]); + max = fmax(max, mag[i]); + } - re = cos(w) - b[2 * x]; - im = sin(w) - b[2 * x + 1]; + temp[0] = 0.; + for (i = 0; i < s->w - 1; i++) { + double d = phase[i] - phase[i + 1]; + temp[i + 1] = ceil(fabs(d) / (2. * M_PI)) * 2. * M_PI * ((d > M_PI) - (d < -M_PI)); + } - ore = real; - oim = imag; - div = re * re + im * im; + min_phase = phase[0]; + max_phase = phase[0]; + for (i = 1; i < s->w; i++) { + temp[i] += temp[i - 1]; + phase[i] += temp[i]; + min_phase = fmin(min_phase, phase[i]); + max_phase = fmax(max_phase, phase[i]); + } - real = (ore * re + oim * im) / div; - imag = (oim * re - ore * im) / div; - } - } + for (i = 0; i < s->w - 1; i++) { + double div = s->w / (double)sample_rate; - mag[i] = s->iir[ch].g * hypot(real, imag); - phase[i] = atan2(imag, real); - min = fminf(min, mag[i]); - max = fmaxf(max, mag[i]); + delay[i + 1] = -(phase[i] - phase[i + 1]) / div; + min_delay = fmin(min_delay, delay[i + 1]); + max_delay = fmax(max_delay, delay[i + 1]); } + delay[0] = delay[1]; for (i = 0; i < s->w; i++) { int ymag = mag[i] / max * (s->h - 1); - int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1); + int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1); + int yphase = (phase[i] - min_phase) / (max_phase - min_phase) * (s->h - 1); ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1); yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1); + ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1); if (prev_ymag < 0) prev_ymag = ymag; if (prev_yphase < 0) prev_yphase = yphase; + if (prev_ydelay < 0) + prev_ydelay = ydelay; draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF); draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00); + draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF); prev_ymag = ymag; prev_yphase = yphase; + prev_ydelay = ydelay; } if (s->w > 400 && s->h > 100) { @@ -815,9 +1242,27 @@ static void draw_response(AVFilterContext *ctx, AVFrame *out) drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD); snprintf(text, sizeof(text), "%.2f", min); drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD); + + drawtext(out, 2, 22, "Max Phase:", 0xDDDDDDDD); + snprintf(text, sizeof(text), "%.2f", max_phase); + drawtext(out, 15 * 8 + 2, 22, text, 0xDDDDDDDD); + + drawtext(out, 2, 32, "Min Phase:", 0xDDDDDDDD); + snprintf(text, sizeof(text), "%.2f", min_phase); + drawtext(out, 15 * 8 + 2, 32, text, 0xDDDDDDDD); + + drawtext(out, 2, 42, "Max Delay:", 0xDDDDDDDD); + snprintf(text, sizeof(text), "%.2f", max_delay); + drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD); + + drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD); + snprintf(text, sizeof(text), "%.2f", min_delay); + drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD); } end: + av_free(delay); + av_free(temp); av_free(phase); av_free(mag); } @@ -846,10 +1291,18 @@ static int config_output(AVFilterLink *outlink) if (ret < 0) return ret; - if (s->format == 2) { + if (s->format == -1) { + convert_sf2tf(ctx, inlink->channels); + s->format = 0; + } else if (s->format == 2) { convert_pr2zp(ctx, inlink->channels); } else if (s->format == 3) { convert_pd2zp(ctx, inlink->channels); + } else if (s->format == 4) { + convert_sp2zp(ctx, inlink->channels); + } + if (s->format > 0) { + check_stability(ctx, inlink->channels); } av_frame_free(&s->video); @@ -858,11 +1311,11 @@ static int config_output(AVFilterLink *outlink) if (!s->video) return AVERROR(ENOMEM); - draw_response(ctx, s->video); + draw_response(ctx, s->video, inlink->sample_rate); } if (s->format == 0) - av_log(ctx, AV_LOG_WARNING, "tf coefficients format is not recommended for too high number of zeros/poles.\n"); + av_log(ctx, AV_LOG_WARNING, "transfer function coefficients format is not recommended for too high number of zeros/poles.\n"); if (s->format > 0 && s->process == 0) { av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n"); @@ -870,16 +1323,37 @@ static int config_output(AVFilterLink *outlink) ret = convert_zp2tf(ctx, inlink->channels); if (ret < 0) return ret; - } else if (s->format == 0 && s->process == 1) { - av_log(ctx, AV_LOG_ERROR, "Serial cascading is not implemented for transfer function.\n"); + } else if (s->format == -2 && s->process > 0) { + av_log(ctx, AV_LOG_ERROR, "Only direct processing is implemented for lattice-ladder function.\n"); + return AVERROR_PATCHWELCOME; + } else if (s->format <= 0 && s->process == 1) { + av_log(ctx, AV_LOG_ERROR, "Serial processing is not implemented for transfer function.\n"); + return AVERROR_PATCHWELCOME; + } else if (s->format <= 0 && s->process == 2) { + av_log(ctx, AV_LOG_ERROR, "Parallel processing is not implemented for transfer function.\n"); return AVERROR_PATCHWELCOME; } else if (s->format > 0 && s->process == 1) { - if (inlink->format == AV_SAMPLE_FMT_S16P) - av_log(ctx, AV_LOG_WARNING, "Serial cascading is not recommended for i16 precision.\n"); - ret = decompose_zp2biquads(ctx, inlink->channels); if (ret < 0) return ret; + } else if (s->format > 0 && s->process == 2) { + if (s->precision > 1) + av_log(ctx, AV_LOG_WARNING, "Parallel processing is not recommended for fixed-point precisions.\n"); + ret = decompose_zp2biquads(ctx, inlink->channels); + if (ret < 0) + return ret; + ret = convert_serial2parallel(ctx, inlink->channels); + if (ret < 0) + return ret; + } + + for (ch = 0; s->format == -2 && ch < inlink->channels; ch++) { + IIRChannel *iir = &s->iir[ch]; + + if (iir->nb_ab[0] != iir->nb_ab[1] + 1) { + av_log(ctx, AV_LOG_ERROR, "Number of ladder coefficients must be one more than number of reflection coefficients.\n"); + return AVERROR(EINVAL); + } } for (ch = 0; s->format == 0 && ch < inlink->channels; ch++) { @@ -889,16 +1363,28 @@ static int config_output(AVFilterLink *outlink) iir->ab[0][i] /= iir->ab[0][0]; } + iir->ab[0][0] = 1.0; for (i = 0; i < iir->nb_ab[1]; i++) { - iir->ab[1][i] *= iir->g / iir->ab[0][0]; + iir->ab[1][i] *= iir->g; } + + normalize_coeffs(ctx, ch); } switch (inlink->format) { - case AV_SAMPLE_FMT_DBLP: s->iir_channel = s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break; - case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break; - case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break; - case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 1 ? iir_ch_serial_s16p : iir_ch_s16p; break; + case AV_SAMPLE_FMT_DBLP: s->iir_channel = s->process == 2 ? iir_ch_parallel_dblp : s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break; + case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 2 ? iir_ch_parallel_fltp : s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break; + case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s32p : s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break; + case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s16p : s->process == 1 ? iir_ch_serial_s16p : iir_ch_s16p; break; + } + + if (s->format == -2) { + switch (inlink->format) { + case AV_SAMPLE_FMT_DBLP: s->iir_channel = iir_ch_lattice_dblp; break; + case AV_SAMPLE_FMT_FLTP: s->iir_channel = iir_ch_lattice_fltp; break; + case AV_SAMPLE_FMT_S32P: s->iir_channel = iir_ch_lattice_s32p; break; + case AV_SAMPLE_FMT_S16P: s->iir_channel = iir_ch_lattice_s16p; break; + } } return 0; @@ -913,7 +1399,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFrame *out; int ch, ret; - if (av_frame_is_writable(in)) { + if (av_frame_is_writable(in) && s->process != 2) { out = in; } else { out = ff_get_audio_buffer(outlink, in->nb_samples); @@ -944,8 +1430,13 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) int64_t new_pts = av_rescale_q(out->pts, ctx->inputs[0]->time_base, outlink->time_base); if (new_pts > old_pts) { + AVFrame *clone; + s->video->pts = new_pts; - ret = ff_filter_frame(outlink, av_frame_clone(s->video)); + clone = av_frame_clone(s->video); + if (!clone) + return AVERROR(ENOMEM); + ret = ff_filter_frame(outlink, clone); if (ret < 0) return ret; } @@ -988,29 +1479,22 @@ static av_cold int init(AVFilterContext *ctx) } pad = (AVFilterPad){ - .name = av_strdup("default"), + .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }; - if (!pad.name) - return AVERROR(ENOMEM); + ret = ff_insert_outpad(ctx, 0, &pad); + if (ret < 0) + return ret; if (s->response) { vpad = (AVFilterPad){ - .name = av_strdup("filter_response"), + .name = "filter_response", .type = AVMEDIA_TYPE_VIDEO, .config_props = config_video, }; - if (!vpad.name) - return AVERROR(ENOMEM); - } - - ret = ff_insert_outpad(ctx, 0, &pad); - if (ret < 0) - return ret; - if (s->response) { ret = ff_insert_outpad(ctx, 1, &vpad); if (ret < 0) return ret; @@ -1036,9 +1520,6 @@ static av_cold void uninit(AVFilterContext *ctx) } av_freep(&s->iir); - av_freep(&ctx->output_pads[0].name); - if (s->response) - av_freep(&ctx->output_pads[1].name); av_frame_free(&s->video); } @@ -1056,24 +1537,37 @@ static const AVFilterPad inputs[] = { #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption aiir_options[] = { - { "z", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, - { "p", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, + { "zeros", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, + { "z", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, + { "poles", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, + { "p", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, + { "gains", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF }, { "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF }, { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF }, { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF }, - { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 3, AF, "format" }, - { "tf", "transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "format" }, + { "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, "format" }, + { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, "format" }, + { "ll", "lattice-ladder function", 0, AV_OPT_TYPE_CONST, {.i64=-2}, 0, 0, AF, "format" }, + { "sf", "analog transfer function", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "format" }, + { "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "format" }, { "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "format" }, { "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "format" }, { "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "format" }, - { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "process" }, + { "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "format" }, + { "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "process" }, + { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "process" }, { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "process" }, - { "s", "serial cascading", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" }, + { "s", "serial", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" }, + { "p", "parallel", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "process" }, + { "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" }, { "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" }, { "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" }, { "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" }, { "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" }, { "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision" }, + { "normalize", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, + { "n", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, + { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF }, { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF }, { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF }, { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF }, @@ -1083,7 +1577,7 @@ static const AVOption aiir_options[] = { AVFILTER_DEFINE_CLASS(aiir); -AVFilter ff_af_aiir = { +const AVFilter ff_af_aiir = { .name = "aiir", .description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."), .priv_size = sizeof(AudioIIRContext),