Commit 8dd3a53d authored by Paul B Mahol's avatar Paul B Mahol

avfilter/af_crystalizer: add support for more sample formats

Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent d535e0c1
......@@ -29,6 +29,8 @@ typedef struct CrystalizerContext {
float mult;
int clip;
AVFrame *prev;
void (*filter)(void **dst, void **prv, const void **src,
int nb_samples, int channels, float mult, int clip);
} CrystalizerContext;
#define OFFSET(x) offsetof(CrystalizerContext, x)
......@@ -46,10 +48,18 @@ static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
(ret = ff_set_common_formats(ctx , formats )) < 0)
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
layouts = ff_all_channel_counts();
......@@ -64,16 +74,123 @@ static int query_formats(AVFilterContext *ctx)
return ff_set_common_samplerates(ctx, formats);
}
static void filter_flt(void **d, void **p, const void **s,
int nb_samples, int channels,
float mult, int clip)
{
const float *src = s[0];
float *dst = d[0];
float *prv = p[0];
int n, c;
for (n = 0; n < nb_samples; n++) {
for (c = 0; c < channels; c++) {
float current = src[c];
dst[c] = current + (current - prv[c]) * mult;
prv[c] = current;
if (clip) {
dst[c] = av_clipf(dst[c], -1, 1);
}
}
dst += c;
src += c;
}
}
static void filter_dbl(void **d, void **p, const void **s,
int nb_samples, int channels,
float mult, int clip)
{
const double *src = s[0];
double *dst = d[0];
double *prv = p[0];
int n, c;
for (n = 0; n < nb_samples; n++) {
for (c = 0; c < channels; c++) {
double current = src[c];
dst[c] = current + (current - prv[c]) * mult;
prv[c] = current;
if (clip) {
dst[c] = av_clipd(dst[c], -1, 1);
}
}
dst += c;
src += c;
}
}
static void filter_fltp(void **d, void **p, const void **s,
int nb_samples, int channels,
float mult, int clip)
{
int n, c;
for (c = 0; c < channels; c++) {
const float *src = s[c];
float *dst = d[c];
float *prv = p[c];
for (n = 0; n < nb_samples; n++) {
float current = src[n];
dst[n] = current + (current - prv[0]) * mult;
prv[0] = current;
if (clip) {
dst[n] = av_clipf(dst[n], -1, 1);
}
}
}
}
static void filter_dblp(void **d, void **p, const void **s,
int nb_samples, int channels,
float mult, int clip)
{
int n, c;
for (c = 0; c < channels; c++) {
const double *src = s[c];
double *dst = d[c];
double *prv = p[c];
for (n = 0; n < nb_samples; n++) {
double current = src[n];
dst[n] = current + (current - prv[0]) * mult;
prv[0] = current;
if (clip) {
dst[n] = av_clipd(dst[n], -1, 1);
}
}
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
CrystalizerContext *s = ctx->priv;
switch (inlink->format) {
case AV_SAMPLE_FMT_FLT: s->filter = filter_flt; break;
case AV_SAMPLE_FMT_DBL: s->filter = filter_dbl; break;
case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
CrystalizerContext *s = ctx->priv;
const float *src = (const float *)in->data[0];
const float mult = s->mult;
AVFrame *out;
float *dst, *prv;
int n, c;
if (!s->prev) {
s->prev = ff_get_audio_buffer(inlink, 1);
......@@ -94,22 +211,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
av_frame_copy_props(out, in);
}
dst = (float *)out->data[0];
prv = (float *)s->prev->data[0];
for (n = 0; n < in->nb_samples; n++) {
for (c = 0; c < in->channels; c++) {
float current = src[c];
dst[c] = current + (current - prv[c]) * mult;
prv[c] = current;
if (s->clip) {
dst[c] = av_clipf(dst[c], -1, 1);
}
}
dst += c;
src += c;
}
s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data,
in->nb_samples, in->channels, s->mult, s->clip);
if (out != in)
av_frame_free(&in);
......@@ -129,6 +232,7 @@ static const AVFilterPad inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
{ NULL }
};
......
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