[FFmpeg-cvslog] avfilter/af_aiir: rename options, provide gains in separate option
Paul B Mahol
git at videolan.org
Sun Jan 7 22:23:44 EET 2018
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun Jan 7 21:16:25 2018 +0100| [2d3df8e2e9e60829c6cd392e334bf0302b8b59bb] | committer: Paul B Mahol
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 at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2d3df8e2e9e60829c6cd392e334bf0302b8b59bb
---
doc/filters.texi | 20 ++++++-----
libavfilter/af_aiir.c | 95 +++++++++++++++++++++++++++++++++++----------------
2 files changed, 77 insertions(+), 38 deletions(-)
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
- at item a
+ at item z
+Set numerator/zeros coefficients.
+
+ at item p
Set denominator/poles coefficients.
- at item b
-Set numerator/zeros coefficients.
+ at 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" },
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