mirror of https://git.ffmpeg.org/ffmpeg.git
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>
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@ -1067,11 +1067,14 @@ Apply an arbitrary Infinite Impulse Response filter.
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It accepts the following parameters:
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@table @option
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@item a
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@item z
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Set numerator/zeros coefficients.
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@item p
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Set denominator/poles coefficients.
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@item b
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Set numerator/zeros coefficients.
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@item k
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Set channels gains.
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@item dry_gain
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Set input gain.
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@ -1089,11 +1092,10 @@ order.
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Coefficients in @code{zp} format are separated by spaces and order of coefficients
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doesn't matter. Coefficients in @code{zp} format are complex numbers with @var{i}
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imaginary unit, also first number in numerator, option @var{b}, is not complex but
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real number and sets overall gain for channel.
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imaginary unit.
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Different coefficients can be provided for every channel, in such case
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use '|' to separate coefficients. Last provided coefficients will be
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Different coefficients and gains can be provided for every channel, in such case
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use '|' to separate coefficients or gains. Last provided coefficients will be
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used for all remaining channels.
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@subsection Examples
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@ -1102,13 +1104,13 @@ used for all remaining channels.
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@item
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Apply 2 pole elliptic notch at arround 5000Hz for 48000 Hz sample rate:
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@example
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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
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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
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@end example
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@item
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Same as above but in @code{zp} format:
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@example
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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
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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
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@end example
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@end itemize
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@ -29,12 +29,13 @@
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typedef struct AudioIIRContext {
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const AVClass *class;
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char *a_str, *b_str;
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char *a_str, *b_str, *g_str;
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double dry_gain, wet_gain;
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int format;
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int *nb_a, *nb_b;
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double **a, **b;
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double *g;
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double **input, **output;
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int clippings;
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int channels;
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@ -140,6 +141,38 @@ static void count_coefficients(char *item_str, int *nb_items)
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}
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}
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static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
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{
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char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
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int i;
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p = old_str = av_strdup(item_str);
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if (!p)
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return AVERROR(ENOMEM);
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for (i = 0; i < nb_items; i++) {
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if (!(arg = av_strtok(p, "|", &saveptr)))
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arg = prev_arg;
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if (!arg) {
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av_freep(&old_str);
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return AVERROR(EINVAL);
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}
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p = NULL;
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if (sscanf(arg, "%lf", &dst[i]) != 1) {
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av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg);
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av_freep(&old_str);
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return AVERROR(EINVAL);
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}
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prev_arg = arg;
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}
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av_freep(&old_str);
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return 0;
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}
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static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
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{
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char *p, *arg, *old_str, *saveptr = NULL;
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@ -155,6 +188,7 @@ static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
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p = NULL;
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if (sscanf(arg, "%lf", &dst[i]) != 1) {
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av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
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av_freep(&old_str);
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return AVERROR(EINVAL);
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}
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}
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@ -164,7 +198,7 @@ static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
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return 0;
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}
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static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, int is_zeros)
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static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
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{
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char *p, *arg, *old_str, *saveptr = NULL;
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int i;
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@ -177,25 +211,19 @@ static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_ite
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break;
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p = NULL;
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if (i == 0 && is_zeros) {
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if (sscanf(arg, "%lf", &dst[i]) != 1) {
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av_log(ctx, AV_LOG_ERROR, "Invalid gain supplied: %s\n", arg);
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return AVERROR(EINVAL);
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}
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} else {
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if (sscanf(arg, "%lf %lfi", &dst[i*2], &dst[i*2+1]) != 2) {
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av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
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av_freep(&old_str);
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return AVERROR(EINVAL);
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}
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}
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}
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av_freep(&old_str);
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return 0;
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}
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static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache, int is_zeros)
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static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache)
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{
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AudioIIRContext *s = ctx->priv;
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char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
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@ -208,26 +236,30 @@ static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str,
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if (!(arg = av_strtok(p, "|", &saveptr)))
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arg = prev_arg;
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if (!arg)
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if (!arg) {
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av_freep(&old_str);
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return AVERROR(EINVAL);
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}
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count_coefficients(arg, &nb[i]);
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p = NULL;
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cache[i] = av_calloc(nb[i] + 1, sizeof(double));
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c[i] = av_calloc(nb[i] * (s->format + 1), sizeof(double));
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if (!c[i] || !cache[i])
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if (!c[i] || !cache[i]) {
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av_freep(&old_str);
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return AVERROR(ENOMEM);
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}
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if (s->format) {
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ret = read_zp_coefficients(ctx, arg, nb[i], c[i], is_zeros);
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if (is_zeros)
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nb[i]--;
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ret = read_zp_coefficients(ctx, arg, nb[i], c[i]);
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} else {
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ret = read_tf_coefficients(ctx, arg, nb[i], c[i]);
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}
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if (ret < 0)
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if (ret < 0) {
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av_freep(&old_str);
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return ret;
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}
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prev_arg = arg;
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}
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@ -288,7 +320,7 @@ static int convert_zp2tf(AVFilterContext *ctx, int channels)
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int ch, i, j, ret;
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for (ch = 0; ch < channels; ch++) {
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double *topc, *botc, gain;
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double *topc, *botc;
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topc = av_calloc((s->nb_b[ch] + 1) * 2, sizeof(*topc));
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botc = av_calloc((s->nb_a[ch] + 1) * 2, sizeof(*botc));
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@ -302,16 +334,15 @@ static int convert_zp2tf(AVFilterContext *ctx, int channels)
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return ret;
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}
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ret = expand(ctx, &s->b[ch][2], s->nb_b[ch], topc);
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ret = expand(ctx, s->b[ch], s->nb_b[ch], topc);
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if (ret < 0) {
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av_free(topc);
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av_free(botc);
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return ret;
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}
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gain = s->b[ch][0];
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for (j = 0, i = s->nb_b[ch]; i >= 0; j++, i--) {
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s->b[ch][j] = topc[2 * i] * gain;
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s->b[ch][j] = topc[2 * i];
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}
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s->nb_b[ch]++;
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@ -337,6 +368,7 @@ static int config_output(AVFilterLink *outlink)
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s->channels = inlink->channels;
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s->a = av_calloc(inlink->channels, sizeof(*s->a));
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s->b = av_calloc(inlink->channels, sizeof(*s->b));
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s->g = av_calloc(inlink->channels, sizeof(*s->g));
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s->nb_a = av_calloc(inlink->channels, sizeof(*s->nb_a));
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s->nb_b = av_calloc(inlink->channels, sizeof(*s->nb_b));
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s->input = av_calloc(inlink->channels, sizeof(*s->input));
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@ -344,11 +376,15 @@ static int config_output(AVFilterLink *outlink)
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if (!s->a || !s->b || !s->nb_a || !s->nb_b || !s->input || !s->output)
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return AVERROR(ENOMEM);
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ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output, 0);
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ret = read_gains(ctx, s->g_str, inlink->channels, s->g);
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if (ret < 0)
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return ret;
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ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input, 1);
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ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output);
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if (ret < 0)
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return ret;
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ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input);
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if (ret < 0)
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return ret;
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@ -364,7 +400,7 @@ static int config_output(AVFilterLink *outlink)
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}
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for (i = 0; i < s->nb_b[ch]; i++) {
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s->b[ch][i] /= s->a[ch][0];
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s->b[ch][i] *= s->g[ch] / s->a[ch][0];
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}
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}
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@ -412,7 +448,7 @@ static av_cold int init(AVFilterContext *ctx)
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{
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AudioIIRContext *s = ctx->priv;
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if (!s->a_str || !s->b_str) {
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if (!s->a_str || !s->b_str || !s->g_str) {
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av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n");
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return AVERROR(EINVAL);
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}
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@ -470,8 +506,9 @@ static const AVFilterPad outputs[] = {
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption aiir_options[] = {
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{ "a", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
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{ "b", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
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{ "z", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
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{ "p", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
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{ "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
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{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
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{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
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{ "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "format" },
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