diff --git a/doc/filters.texi b/doc/filters.texi index 69c59d74a6..f6954c947c 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -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 diff --git a/libavfilter/af_aiir.c b/libavfilter/af_aiir.c index 62ad7ebfa1..87c3dadac8 100644 --- a/libavfilter/af_aiir.c +++ b/libavfilter/af_aiir.c @@ -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" },