#include "avfilter.h"
#include "internal.h"
+typedef struct ThreadData {
+ AVFrame *in, *out;
+} ThreadData;
+
+typedef struct Pair {
+ int a, b;
+} Pair;
+
+typedef struct BiquadContext {
+ double a0, a1, a2;
+ double b0, b1, b2;
+ double i1, i2;
+ double o1, o2;
+} BiquadContext;
+
+typedef struct IIRChannel {
+ int nb_ab[2];
+ double *ab[2];
+ double g;
+ double *cache[2];
+ BiquadContext *biquads;
+ int clippings;
+} IIRChannel;
+
typedef struct AudioIIRContext {
const AVClass *class;
char *a_str, *b_str, *g_str;
double dry_gain, wet_gain;
int format;
+ int process;
+ int precision;
- int *nb_a, *nb_b;
- double **a, **b;
- double *g;
- double **input, **output;
- int clippings;
+ IIRChannel *iir;
int channels;
+ enum AVSampleFormat sample_format;
- void (*iir_frame)(AVFilterContext *ctx, AVFrame *in, AVFrame *out);
+ int (*iir_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs);
} AudioIIRContext;
static int query_formats(AVFilterContext *ctx)
{
+ AudioIIRContext *s = ctx->priv;
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
- static const enum AVSampleFormat sample_fmts[] = {
+ enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP,
- AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_S32P,
- AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE
};
int ret;
if (ret < 0)
return ret;
+ sample_fmts[0] = s->sample_format;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
-#define IIR_FRAME(name, type, min, max, need_clipping) \
-static void iir_frame_## name(AVFilterContext *ctx, AVFrame *in, AVFrame *out) \
+#define IIR_CH(name, type, min, max, need_clipping) \
+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; \
- int ch, n; \
+ 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]; \
+ 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]; \
+ int *clippings = &s->iir[ch].clippings; \
+ type *dst = (type *)out->extended_data[ch]; \
+ int n; \
\
- for (ch = 0; ch < out->channels; ch++) { \
- const type *src = (const type *)in->extended_data[ch]; \
- double *ic = (double *)s->input[ch]; \
- double *oc = (double *)s->output[ch]; \
- const int nb_a = s->nb_a[ch]; \
- const int nb_b = s->nb_b[ch]; \
- const double *a = s->a[ch]; \
- const double *b = s->b[ch]; \
- type *dst = (type *)out->extended_data[ch]; \
+ for (n = 0; n < in->nb_samples; n++) { \
+ double sample = 0.; \
+ int x; \
\
- for (n = 0; n < in->nb_samples; n++) { \
- double sample = 0.; \
- int x; \
+ memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
+ memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
+ ic[0] = src[n] * ig; \
+ for (x = 0; x < nb_b; x++) \
+ sample += b[x] * ic[x]; \
\
- memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
- memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
- ic[0] = src[n] * ig; \
- for (x = 0; x < nb_b; x++) \
- sample += b[x] * ic[x]; \
+ for (x = 1; x < nb_a; x++) \
+ sample -= a[x] * oc[x]; \
\
- for (x = 1; x < nb_a; x++) \
- sample -= a[x] * oc[x]; \
+ oc[0] = sample; \
+ sample *= og; \
+ if (need_clipping && sample < min) { \
+ (*clippings)++; \
+ dst[n] = min; \
+ } else if (need_clipping && sample > max) { \
+ (*clippings)++; \
+ dst[n] = max; \
+ } else { \
+ dst[n] = sample; \
+ } \
+ } \
\
- oc[0] = sample; \
- sample *= og; \
- if (need_clipping && sample < min) { \
- s->clippings++; \
+ return 0; \
+}
+
+IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
+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) \
+{ \
+ AudioIIRContext *s = ctx->priv; \
+ const double ig = s->dry_gain; \
+ const double og = s->wet_gain; \
+ 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]; \
+ 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 (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; \
+ \
+ i2 = i1; \
+ i1 = src[n]; \
+ o2 = o1; \
+ o1 = o0; \
+ o0 *= og; \
+ \
+ if (need_clipping && o0 < min) { \
+ (*clippings)++; \
dst[n] = min; \
- } else if (need_clipping && sample > max) { \
- s->clippings++; \
+ } else if (need_clipping && o0 > max) { \
+ (*clippings)++; \
dst[n] = max; \
} else { \
- dst[n] = sample; \
+ dst[n] = o0; \
} \
} \
+ iir->biquads[i].i1 = i1; \
+ iir->biquads[i].i2 = i2; \
+ iir->biquads[i].o1 = o1; \
+ iir->biquads[i].o2 = o2; \
} \
+ \
+ return 0; \
}
-IIR_FRAME(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
-IIR_FRAME(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
-IIR_FRAME(fltp, float, -1., 1., 0)
-IIR_FRAME(dblp, double, -1., 1., 0)
+SERIAL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
+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)
static void count_coefficients(char *item_str, int *nb_items)
{
}
}
-static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
+static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items)
{
+ AudioIIRContext *s = ctx->priv;
char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
int i;
}
p = NULL;
- if (sscanf(arg, "%lf", &dst[i]) != 1) {
+ if (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);
return 0;
}
-static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
+static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format)
{
char *p, *arg, *old_str, *saveptr = NULL;
int i;
break;
p = NULL;
- if (sscanf(arg, "%lf %lfi", &dst[i*2], &dst[i*2+1]) != 2) {
+ if (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 int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache)
+static const char *format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd" };
+
+static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab)
{
AudioIIRContext *s = ctx->priv;
char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
if (!p)
return AVERROR(ENOMEM);
for (i = 0; i < channels; i++) {
+ IIRChannel *iir = &s->iir[i];
+
if (!(arg = av_strtok(p, "|", &saveptr)))
arg = prev_arg;
return AVERROR(EINVAL);
}
- count_coefficients(arg, &nb[i]);
+ count_coefficients(arg, &iir->nb_ab[ab]);
p = NULL;
- cache[i] = av_calloc(nb[i] + 1, sizeof(double));
- c[i] = av_calloc(nb[i] * (s->format + 1), sizeof(double));
- if (!c[i] || !cache[i]) {
+ iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double));
+ iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double));
+ if (!iir->ab[ab] || !iir->cache[ab]) {
av_freep(&old_str);
return AVERROR(ENOMEM);
}
if (s->format) {
- ret = read_zp_coefficients(ctx, arg, nb[i], c[i]);
+ ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]);
} else {
- ret = read_tf_coefficients(ctx, arg, nb[i], c[i]);
+ ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
}
if (ret < 0) {
av_freep(&old_str);
multiply(pz[2 * i], pz[2 * i + 1], nb, coeffs);
for (i = 0; i < nb + 1; i++) {
- if (fabs(coeffs[2 * i + 1]) > DBL_EPSILON) {
+ if (fabs(coeffs[2 * i + 1]) > FLT_EPSILON) {
av_log(ctx, AV_LOG_ERROR, "coeff: %lf of z^%d is not real; poles/zeros are not complex conjugates.\n",
coeffs[2 * i + 1], i);
return AVERROR(EINVAL);
static int convert_zp2tf(AVFilterContext *ctx, int channels)
{
AudioIIRContext *s = ctx->priv;
- int ch, i, j, ret;
+ int ch, i, j, ret = 0;
for (ch = 0; ch < channels; ch++) {
+ IIRChannel *iir = &s->iir[ch];
double *topc, *botc;
- topc = av_calloc((s->nb_b[ch] + 1) * 2, sizeof(*topc));
- botc = av_calloc((s->nb_a[ch] + 1) * 2, sizeof(*botc));
- if (!topc || !botc)
- return AVERROR(ENOMEM);
+ topc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*topc));
+ botc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*botc));
+ if (!topc || !botc) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
- ret = expand(ctx, s->a[ch], s->nb_a[ch], botc);
+ ret = expand(ctx, iir->ab[0], iir->nb_ab[0], botc);
if (ret < 0) {
- av_free(topc);
- av_free(botc);
- return ret;
+ goto fail;
}
- ret = expand(ctx, s->b[ch], s->nb_b[ch], topc);
+ ret = expand(ctx, iir->ab[1], iir->nb_ab[1], topc);
if (ret < 0) {
- av_free(topc);
- av_free(botc);
- return ret;
+ goto fail;
}
- for (j = 0, i = s->nb_b[ch]; i >= 0; j++, i--) {
- s->b[ch][j] = topc[2 * i];
+ for (j = 0, i = iir->nb_ab[1]; i >= 0; j++, i--) {
+ iir->ab[1][j] = topc[2 * i];
}
- s->nb_b[ch]++;
+ iir->nb_ab[1]++;
- for (j = 0, i = s->nb_a[ch]; i >= 0; j++, i--) {
- s->a[ch][j] = botc[2 * i];
+ for (j = 0, i = iir->nb_ab[0]; i >= 0; j++, i--) {
+ iir->ab[0][j] = botc[2 * i];
}
- s->nb_a[ch]++;
+ iir->nb_ab[0]++;
+fail:
av_free(topc);
av_free(botc);
+ if (ret < 0)
+ break;
+ }
+
+ return ret;
+}
+
+static int decompose_zp2biquads(AVFilterContext *ctx, int channels)
+{
+ AudioIIRContext *s = ctx->priv;
+ int ch, ret;
+
+ for (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 current_biquad = 0;
+
+ iir->biquads = av_calloc(nb_biquads, sizeof(BiquadContext));
+ if (!iir->biquads)
+ return AVERROR(ENOMEM);
+
+ while (nb_biquads--) {
+ Pair outmost_pole = { -1, -1 };
+ Pair nearest_zero = { -1, -1 };
+ double zeros[4] = { 0 };
+ double poles[4] = { 0 };
+ double b[6] = { 0 };
+ double a[6] = { 0 };
+ double min_distance = DBL_MAX;
+ double max_mag = 0;
+ int i;
+
+ for (i = 0; i < iir->nb_ab[0]; i++) {
+ double mag;
+
+ if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
+ continue;
+ mag = hypot(iir->ab[0][2 * i], iir->ab[0][2 * i + 1]);
+
+ if (mag > max_mag) {
+ max_mag = mag;
+ outmost_pole.a = i;
+ }
+ }
+
+ for (i = 0; i < iir->nb_ab[1]; i++) {
+ if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
+ continue;
+
+ if (iir->ab[0][2 * i ] == iir->ab[0][2 * outmost_pole.a ] &&
+ iir->ab[0][2 * i + 1] == -iir->ab[0][2 * outmost_pole.a + 1]) {
+ outmost_pole.b = i;
+ break;
+ }
+ }
+
+ av_log(ctx, AV_LOG_VERBOSE, "outmost_pole is %d.%d\n", outmost_pole.a, outmost_pole.b);
+
+ if (outmost_pole.a < 0 || outmost_pole.b < 0)
+ return AVERROR(EINVAL);
+
+ for (i = 0; i < iir->nb_ab[1]; i++) {
+ double distance;
+
+ if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
+ continue;
+ distance = hypot(iir->ab[0][2 * outmost_pole.a ] - iir->ab[1][2 * i ],
+ iir->ab[0][2 * outmost_pole.a + 1] - iir->ab[1][2 * i + 1]);
+
+ if (distance < min_distance) {
+ min_distance = distance;
+ nearest_zero.a = i;
+ }
+ }
+
+ for (i = 0; i < iir->nb_ab[1]; i++) {
+ if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
+ continue;
+
+ if (iir->ab[1][2 * i ] == iir->ab[1][2 * nearest_zero.a ] &&
+ iir->ab[1][2 * i + 1] == -iir->ab[1][2 * nearest_zero.a + 1]) {
+ nearest_zero.b = i;
+ break;
+ }
+ }
+
+ av_log(ctx, AV_LOG_VERBOSE, "nearest_zero is %d.%d\n", nearest_zero.a, nearest_zero.b);
+
+ if (nearest_zero.a < 0 || nearest_zero.b < 0)
+ return AVERROR(EINVAL);
+
+ poles[0] = iir->ab[0][2 * outmost_pole.a ];
+ poles[1] = iir->ab[0][2 * outmost_pole.a + 1];
+
+ zeros[0] = iir->ab[1][2 * nearest_zero.a ];
+ zeros[1] = iir->ab[1][2 * nearest_zero.a + 1];
+
+ if (nearest_zero.a == nearest_zero.b && outmost_pole.a == outmost_pole.b) {
+ zeros[2] = 0;
+ zeros[3] = 0;
+
+ poles[2] = 0;
+ poles[3] = 0;
+ } else {
+ poles[2] = iir->ab[0][2 * outmost_pole.b ];
+ poles[3] = iir->ab[0][2 * outmost_pole.b + 1];
+
+ zeros[2] = iir->ab[1][2 * nearest_zero.b ];
+ zeros[3] = iir->ab[1][2 * nearest_zero.b + 1];
+ }
+
+ ret = expand(ctx, zeros, 2, b);
+ if (ret < 0)
+ return ret;
+
+ ret = expand(ctx, poles, 2, a);
+ if (ret < 0)
+ return ret;
+
+ iir->ab[0][2 * outmost_pole.a] = iir->ab[0][2 * outmost_pole.a + 1] = NAN;
+ iir->ab[0][2 * outmost_pole.b] = iir->ab[0][2 * outmost_pole.b + 1] = NAN;
+ 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);
+
+ av_log(ctx, AV_LOG_VERBOSE, "a=%lf %lf %lf:b=%lf %lf %lf\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);
+
+ current_biquad++;
+ }
}
return 0;
}
+static void convert_pr2zp(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 r = iir->ab[0][2*n];
+ double angle = iir->ab[0][2*n+1];
+
+ iir->ab[0][2*n] = r * cos(angle);
+ iir->ab[0][2*n+1] = r * sin(angle);
+ }
+
+ for (n = 0; n < iir->nb_ab[1]; n++) {
+ double r = iir->ab[1][2*n];
+ double angle = iir->ab[1][2*n+1];
+
+ iir->ab[1][2*n] = r * cos(angle);
+ iir->ab[1][2*n+1] = r * sin(angle);
+ }
+ }
+}
+
+static void convert_pd2zp(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 r = iir->ab[0][2*n];
+ double angle = M_PI*iir->ab[0][2*n+1]/180.;
+
+ iir->ab[0][2*n] = r * cos(angle);
+ iir->ab[0][2*n+1] = r * sin(angle);
+ }
+
+ for (n = 0; n < iir->nb_ab[1]; n++) {
+ double r = iir->ab[1][2*n];
+ double angle = M_PI*iir->ab[1][2*n+1]/180.;
+
+ iir->ab[1][2*n] = r * cos(angle);
+ iir->ab[1][2*n+1] = r * sin(angle);
+ }
+ }
+}
+
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
int ch, ret, i;
s->channels = inlink->channels;
- s->a = av_calloc(inlink->channels, sizeof(*s->a));
- s->b = av_calloc(inlink->channels, sizeof(*s->b));
- s->g = av_calloc(inlink->channels, sizeof(*s->g));
- s->nb_a = av_calloc(inlink->channels, sizeof(*s->nb_a));
- s->nb_b = av_calloc(inlink->channels, sizeof(*s->nb_b));
- s->input = av_calloc(inlink->channels, sizeof(*s->input));
- s->output = av_calloc(inlink->channels, sizeof(*s->output));
- if (!s->a || !s->b || !s->nb_a || !s->nb_b || !s->input || !s->output)
+ s->iir = av_calloc(s->channels, sizeof(*s->iir));
+ if (!s->iir)
return AVERROR(ENOMEM);
- ret = read_gains(ctx, s->g_str, inlink->channels, s->g);
+ ret = read_gains(ctx, s->g_str, inlink->channels);
if (ret < 0)
return ret;
- ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output);
+ ret = read_channels(ctx, inlink->channels, s->a_str, 0);
if (ret < 0)
return ret;
- ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input);
+ ret = read_channels(ctx, inlink->channels, s->b_str, 1);
if (ret < 0)
return ret;
- if (s->format) {
+ if (s->format == 2) {
+ convert_pr2zp(ctx, inlink->channels);
+ } else if (s->format == 3) {
+ convert_pd2zp(ctx, inlink->channels);
+ }
+
+ if (s->format == 0)
+ av_log(ctx, AV_LOG_WARNING, "tf 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");
+ 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;
}
for (ch = 0; ch < inlink->channels; ch++) {
- for (i = 1; i < s->nb_a[ch]; i++) {
- s->a[ch][i] /= s->a[ch][0];
+ IIRChannel *iir = &s->iir[ch];
+
+ for (i = 1; i < iir->nb_ab[0]; i++) {
+ iir->ab[0][i] /= iir->ab[0][0];
}
- for (i = 0; i < s->nb_b[ch]; i++) {
- s->b[ch][i] *= s->g[ch] / s->a[ch][0];
+ for (i = 0; i < iir->nb_ab[1]; i++) {
+ iir->ab[1][i] *= iir->g / iir->ab[0][0];
}
}
switch (inlink->format) {
- case AV_SAMPLE_FMT_DBLP: s->iir_frame = iir_frame_dblp; break;
- case AV_SAMPLE_FMT_FLTP: s->iir_frame = iir_frame_fltp; break;
- case AV_SAMPLE_FMT_S32P: s->iir_frame = iir_frame_s32p; break;
- case AV_SAMPLE_FMT_S16P: s->iir_frame = iir_frame_s16p; break;
+ 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;
}
return 0;
AVFilterContext *ctx = inlink->dst;
AudioIIRContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
+ ThreadData td;
AVFrame *out;
+ int ch;
if (av_frame_is_writable(in)) {
out = in;
av_frame_copy_props(out, in);
}
- s->iir_frame(ctx, in, out);
+ td.in = in;
+ td.out = out;
+ ctx->internal->execute(ctx, s->iir_channel, &td, NULL, outlink->channels);
- if (s->clippings > 0)
- av_log(ctx, AV_LOG_WARNING, "clipping %d times. Please reduce gain.\n", s->clippings);
- s->clippings = 0;
+ for (ch = 0; ch < outlink->channels; ch++) {
+ if (s->iir[ch].clippings > 0)
+ av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n",
+ ch, s->iir[ch].clippings);
+ s->iir[ch].clippings = 0;
+ }
if (in != out)
av_frame_free(&in);
return AVERROR(EINVAL);
}
+ switch (s->precision) {
+ case 0: s->sample_format = AV_SAMPLE_FMT_DBLP; break;
+ case 1: s->sample_format = AV_SAMPLE_FMT_FLTP; break;
+ case 2: s->sample_format = AV_SAMPLE_FMT_S32P; break;
+ case 3: s->sample_format = AV_SAMPLE_FMT_S16P; break;
+ default: return AVERROR_BUG;
+ }
+
return 0;
}
AudioIIRContext *s = ctx->priv;
int ch;
- if (s->a) {
+ if (s->iir) {
for (ch = 0; ch < s->channels; ch++) {
- av_freep(&s->a[ch]);
- av_freep(&s->output[ch]);
+ IIRChannel *iir = &s->iir[ch];
+ av_freep(&iir->ab[0]);
+ av_freep(&iir->ab[1]);
+ av_freep(&iir->cache[0]);
+ av_freep(&iir->cache[1]);
+ av_freep(&iir->biquads);
}
}
- av_freep(&s->a);
-
- if (s->b) {
- for (ch = 0; ch < s->channels; ch++) {
- av_freep(&s->b[ch]);
- av_freep(&s->input[ch]);
- }
- }
- av_freep(&s->b);
-
- av_freep(&s->input);
- av_freep(&s->output);
-
- av_freep(&s->nb_a);
- av_freep(&s->nb_b);
+ av_freep(&s->iir);
}
static const AVFilterPad inputs[] = {
#define AF AV_OPT_FLAG_AUDIO_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 1"}, 0, 0, AF },
- { "p", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
+ { "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 },
{ "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=0}, 0, 1, AF, "format" },
+ { "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" },
{ "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" },
+ { "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" },
+ { "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" },
{ NULL },
};
.name = "aiir",
.description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."),
.priv_size = sizeof(AudioIIRContext),
+ .priv_class = &aiir_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = inputs,
.outputs = outputs,
- .priv_class = &aiir_class,
+ .flags = AVFILTER_FLAG_SLICE_THREADS,
};
projects / ffmpeg / blobdiff