typedef struct BiquadContext {
double a[3];
double b[3];
- double i1, i2;
- double o1, o2;
+ double w1, w2;
} BiquadContext;
typedef struct IIRChannel {
double *ab[2];
double g;
double *cache[2];
+ double fir;
BiquadContext *biquads;
int clippings;
} IIRChannel;
char *a_str, *b_str, *g_str;
double dry_gain, wet_gain;
double mix;
+ int normalize;
int format;
int process;
int precision;
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;
}
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]; \
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]; \
int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
int n, i; \
\
- for (i = 0; i < nb_biquads; i++) { \
+ 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 i1 = iir->biquads[i].i1; \
- double i2 = iir->biquads[i].i2; \
- double o1 = iir->biquads[i].o1; \
- double o2 = iir->biquads[i].o2; \
+ 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; \
+ w1 = b1 * i0 + w2 + a1 * o0; \
+ w2 = b2 * i0 + a2 * o0; \
o0 *= og * g; \
\
- o0 = o0 * mix + (1. - mix) * sample; \
+ o0 = o0 * mix + imix * i0; \
if (need_clipping && o0 < min) { \
(*clippings)++; \
dst[n] = min; \
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; \
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;
}
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);
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);
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);
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)
{
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]);
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)
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;
}
iir->nb_ab[0]++;
+ normalize_coeffs(ctx, ch);
+
fail:
av_free(topc);
av_free(botc);
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++) {
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].a[0] = 1.0;
+ 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]) * (current_biquad ? 1.0 : iir->g);
- iir->biquads[current_biquad].b[1] = (b[2] / a[4]) * (current_biquad ? 1.0 : iir->g);
- iir->biquads[current_biquad].b[2] = (b[0] / a[4]) * (current_biquad ? 1.0 : iir->g);
+ 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].a[0],
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;
}
}
+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;
}
}
+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 *r, double *i)
+ int nb_b, int nb_a, double *magnitude, double *phase)
{
double realz, realp;
double imagz, imagp;
div = realp * realp + imagp * imagp;
real = (realz * realp + imagz * imagp) / div;
imag = (imagz * realp - imagp * realz) / div;
- } else {
- real = 1;
- imag = 0;
- for (int x = 0; x < nb_a; x++) {
- double ore, oim, re, im;
-
- re = cos(w) - a[2 * x];
- im = sin(w) - a[2 * x + 1];
- ore = real;
- oim = imag;
+ *magnitude = hypot(real, imag);
+ *phase = atan2(imag, real);
+ } else {
+ double p = 1., z = 1.;
+ double acc = 0.;
- real = ore * re - oim * im;
- imag = ore * im + oim * re;
+ 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++) {
- double ore, oim, re, im;
-
- re = cos(w) - b[2 * x];
- im = sin(w) - b[2 * x + 1];
-
- ore = real;
- oim = imag;
- div = re * re + im * im;
-
- real = (ore * re + oim * im) / div;
- imag = (oim * re - ore * im) / div;
+ 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]);
}
- }
- *r = real;
- *i = imag;
+ *magnitude = z / p;
+ *phase = acc;
+ }
}
-static void draw_response(AVFilterContext *ctx, AVFrame *out)
+static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate)
{
AudioIIRContext *s = ctx->priv;
- float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
- float min_delay = FLT_MAX, max_delay = FLT_MIN;
+ 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;
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));
delay = av_malloc_array(s->w, sizeof(*delay));
- if (!mag || !phase || !delay)
+ if (!mag || !phase || !delay || !temp)
goto end;
ch = av_clip(s->ir_channel, 0, s->channels - 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 real, imag;
+ double m, p;
- get_response(ch, s->format, w, b, a, nb_b, nb_a, &real, &imag);
+ get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p);
- mag[i] = s->iir[ch].g * hypot(real, imag);
- phase[i] = atan2(imag, real);
- min = fminf(min, mag[i]);
- max = fmaxf(max, mag[i]);
+ mag[i] = s->iir[ch].g * m;
+ phase[i] = p;
+ min = fmin(min, mag[i]);
+ max = fmax(max, mag[i]);
}
+ temp[0] = 0.;
for (i = 0; i < s->w - 1; i++) {
- float dw = M_PI / (s->w - 1);
+ 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));
+ }
- delay[i] = -(phase[i + 1] - phase[i]) / dw;
- min_delay = fminf(min_delay, delay[i]);
- max_delay = fmaxf(max_delay, delay[i]);
+ 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]);
}
- delay[i] = delay[i - 1];
+ for (i = 0; i < s->w - 1; i++) {
+ double div = s->w / (double)sample_rate;
+
+ 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 ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
- int yphase = (0.5 * (1. + phase[i] / M_PI)) * (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);
snprintf(text, sizeof(text), "%.2f", min);
drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
- drawtext(out, 2, 22, "Max Delay:", 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, 22, text, 0xDDDDDDDD);
+ drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD);
- drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
+ drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min_delay);
- drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
+ drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD);
}
end:
av_free(delay);
+ av_free(temp);
av_free(phase);
av_free(mag);
}
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);
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");
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++) {
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;
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);
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;
}
}
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;
}
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);
}
#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 },
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),
projects / ffmpeg / blobdiff