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avfilter/af_aiir: rename options, provide gains in separate option 

This way it can be also used for other format.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
tags/n4.0
Paul B Mahol 7 years ago
parent
6c65de3db0
commit
2d3df8e2e9
2 changed files with 77 additions and 38 deletions
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  1. +11
    -9
      doc/filters.texi
  2. +66
    -29
      libavfilter/af_aiir.c

+ 11
- 9
doc/filters.texi View File

@@ -1067,11 +1067,14 @@ Apply an arbitrary Infinite Impulse Response filter.
It accepts the following parameters:

@table @option
@item a
@item z
Set numerator/zeros coefficients.

@item p
Set denominator/poles coefficients.

@item b
Set numerator/zeros coefficients.
@item k
Set channels gains.

@item dry_gain
Set input gain.
@@ -1089,11 +1092,10 @@ order.

Coefficients in @code{zp} format are separated by spaces and order of coefficients
doesn't matter. Coefficients in @code{zp} format are complex numbers with @var{i}
imaginary unit, also first number in numerator, option @var{b}, is not complex but
real number and sets overall gain for channel.
imaginary unit.

Different coefficients can be provided for every channel, in such case
use '|' to separate coefficients. Last provided coefficients will be
Different coefficients and gains can be provided for every channel, in such case
use '|' to separate coefficients or gains. Last provided coefficients will be
used for all remaining channels.

@subsection Examples
@@ -1102,13 +1104,13 @@ used for all remaining channels.
@item
Apply 2 pole elliptic notch at arround 5000Hz for 48000 Hz sample rate:
@example
aiir=b=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:a=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf
aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf
@end example

@item
Same as above but in @code{zp} format:
@example
aiir=b=0.79575848078096756 0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:a=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp
aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp
@end example
@end itemize



+ 66
- 29
libavfilter/af_aiir.c View File

@@ -29,12 +29,13 @@

typedef struct AudioIIRContext {
const AVClass *class;
char *a_str, *b_str;
char *a_str, *b_str, *g_str;
double dry_gain, wet_gain;
int format;

int *nb_a, *nb_b;
double **a, **b;
double *g;
double **input, **output;
int clippings;
int channels;
@@ -140,6 +141,38 @@ static void count_coefficients(char *item_str, int *nb_items)
}
}

static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
{
char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
int i;

p = old_str = av_strdup(item_str);
if (!p)
return AVERROR(ENOMEM);
for (i = 0; i < nb_items; i++) {
if (!(arg = av_strtok(p, "|", &saveptr)))
arg = prev_arg;

if (!arg) {
av_freep(&old_str);
return AVERROR(EINVAL);
}

p = NULL;
if (sscanf(arg, "%lf", &dst[i]) != 1) {
av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg);
av_freep(&old_str);
return AVERROR(EINVAL);
}

prev_arg = arg;
}

av_freep(&old_str);

return 0;
}

static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
{
char *p, *arg, *old_str, *saveptr = NULL;
@@ -155,6 +188,7 @@ static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
p = NULL;
if (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);
}
}
@@ -164,7 +198,7 @@ static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
return 0;
}

static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, int is_zeros)
static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
{
char *p, *arg, *old_str, *saveptr = NULL;
int i;
@@ -177,16 +211,10 @@ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
break;

p = NULL;
if (i == 0 && is_zeros) {
if (sscanf(arg, "%lf", &dst[i]) != 1) {
av_log(ctx, AV_LOG_ERROR, "Invalid gain supplied: %s\n", arg);
return AVERROR(EINVAL);
}
} else {
if (sscanf(arg, "%lf %lfi", &dst[i*2], &dst[i*2+1]) != 2) {
av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
return AVERROR(EINVAL);
}
if (sscanf(arg, "%lf %lfi", &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);
}
}

@@ -195,7 +223,7 @@ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
return 0;
}

static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache, int is_zeros)
static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache)
{
AudioIIRContext *s = ctx->priv;
char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
@@ -208,26 +236,30 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str,
if (!(arg = av_strtok(p, "|", &saveptr)))
arg = prev_arg;

if (!arg)
if (!arg) {
av_freep(&old_str);
return AVERROR(EINVAL);
}

count_coefficients(arg, &nb[i]);

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])
if (!c[i] || !cache[i]) {
av_freep(&old_str);
return AVERROR(ENOMEM);
}

if (s->format) {
ret = read_zp_coefficients(ctx, arg, nb[i], c[i], is_zeros);
if (is_zeros)
nb[i]--;
ret = read_zp_coefficients(ctx, arg, nb[i], c[i]);
} else {
ret = read_tf_coefficients(ctx, arg, nb[i], c[i]);
}
if (ret < 0)
if (ret < 0) {
av_freep(&old_str);
return ret;
}
prev_arg = arg;
}

@@ -288,7 +320,7 @@ static int convert_zp2tf(AVFilterContext *ctx, int channels)
int ch, i, j, ret;

for (ch = 0; ch < channels; ch++) {
double *topc, *botc, gain;
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));
@@ -302,16 +334,15 @@ static int convert_zp2tf(AVFilterContext *ctx, int channels)
return ret;
}

ret = expand(ctx, &s->b[ch][2], s->nb_b[ch], topc);
ret = expand(ctx, s->b[ch], s->nb_b[ch], topc);
if (ret < 0) {
av_free(topc);
av_free(botc);
return ret;
}

gain = s->b[ch][0];
for (j = 0, i = s->nb_b[ch]; i >= 0; j++, i--) {
s->b[ch][j] = topc[2 * i] * gain;
s->b[ch][j] = topc[2 * i];
}
s->nb_b[ch]++;

@@ -337,6 +368,7 @@ static int config_output(AVFilterLink *outlink)
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));
@@ -344,11 +376,15 @@ static int config_output(AVFilterLink *outlink)
if (!s->a || !s->b || !s->nb_a || !s->nb_b || !s->input || !s->output)
return AVERROR(ENOMEM);

ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output, 0);
ret = read_gains(ctx, s->g_str, inlink->channels, s->g);
if (ret < 0)
return ret;

ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output);
if (ret < 0)
return ret;

ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input, 1);
ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input);
if (ret < 0)
return ret;

@@ -364,7 +400,7 @@ static int config_output(AVFilterLink *outlink)
}

for (i = 0; i < s->nb_b[ch]; i++) {
s->b[ch][i] /= s->a[ch][0];
s->b[ch][i] *= s->g[ch] / s->a[ch][0];
}
}

@@ -412,7 +448,7 @@ static av_cold int init(AVFilterContext *ctx)
{
AudioIIRContext *s = ctx->priv;

if (!s->a_str || !s->b_str) {
if (!s->a_str || !s->b_str || !s->g_str) {
av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n");
return AVERROR(EINVAL);
}
@@ -470,8 +506,9 @@ static const AVFilterPad outputs[] = {
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption aiir_options[] = {
{ "a", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
{ "b", "set B/numerator/zeros coefficients", OFFSET(b_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 1"}, 0, 0, AF },
{ "p", "set A/denominator/poles coefficients", OFFSET(a_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=0}, 0, 1, AF, "format" },


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