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ffmpeg.wasm-core
Commit 47df8716 authored by Paul B Mahol's avatar Paul B Mahol
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avfilter/af_aphaser: use the name 's' for the pointer to the private context 

Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent e030d3c6
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Showing with 61 additions and 61 deletions
libavfilter/af_aphaser.c
View file @ 47df8716
......@@ -47,7 +47,7 @@ typedef struct AudioPhaserContext {
int delay_pos, modulation_pos;
void (*phaser)(struct AudioPhaserContext *p,
void (*phaser)(struct AudioPhaserContext *s,
uint8_t * const *src, uint8_t **dst,
int nb_samples, int channels);
} AudioPhaserContext;
......@@ -73,11 +73,11 @@ AVFILTER_DEFINE_CLASS(aphaser);
static av_cold int init(AVFilterContext *ctx)
{
AudioPhaserContext *p = ctx->priv;
AudioPhaserContext *s = ctx->priv;
if (p->in_gain > (1 - p->decay * p->decay))
if (s->in_gain > (1 - s->decay * s->decay))
av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
if (s->in_gain / (1 - s->decay) > 1 / s->out_gain)
av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
return 0;
......@@ -119,75 +119,75 @@ static int query_formats(AVFilterContext *ctx)
#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
#define PHASER_PLANAR(name, type) \
static void phaser_## name ##p(AudioPhaserContext *p, \
uint8_t * const *src, uint8_t **dst, \
static void phaser_## name ##p(AudioPhaserContext *s, \
uint8_t * const *ssrc, uint8_t **ddst, \
int nb_samples, int channels) \
{ \
int i, c, delay_pos, modulation_pos; \
\
av_assert0(channels > 0); \
for (c = 0; c < channels; c++) { \
type *s = (type *)src[c]; \
type *d = (type *)dst[c]; \
double *buffer = p->delay_buffer + \
c * p->delay_buffer_length; \
type *src = (type *)ssrc[c]; \
type *dst = (type *)ddst[c]; \
double *buffer = s->delay_buffer + \
c * s->delay_buffer_length; \
\
delay_pos = p->delay_pos; \
modulation_pos = p->modulation_pos; \
delay_pos = s->delay_pos; \
modulation_pos = s->modulation_pos; \
\
for (i = 0; i < nb_samples; i++, s++, d++) { \
double v = *s * p->in_gain + buffer[ \
MOD(delay_pos + p->modulation_buffer[ \
for (i = 0; i < nb_samples; i++, src++, dst++) { \
double v = *src * s->in_gain + buffer[ \
MOD(delay_pos + s->modulation_buffer[ \
modulation_pos], \
p->delay_buffer_length)] * p->decay; \
s->delay_buffer_length)] * s->decay; \
\
modulation_pos = MOD(modulation_pos + 1, \
p->modulation_buffer_length); \
delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
s->modulation_buffer_length); \
delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
buffer[delay_pos] = v; \
\
*d = v * p->out_gain; \
*dst = v * s->out_gain; \
} \
} \
\
p->delay_pos = delay_pos; \
p->modulation_pos = modulation_pos; \
s->delay_pos = delay_pos; \
s->modulation_pos = modulation_pos; \
}
#define PHASER(name, type) \
static void phaser_## name (AudioPhaserContext *p, \
uint8_t * const *src, uint8_t **dst, \
static void phaser_## name (AudioPhaserContext *s, \
uint8_t * const *ssrc, uint8_t **ddst, \
int nb_samples, int channels) \
{ \
int i, c, delay_pos, modulation_pos; \
type *s = (type *)src[0]; \
type *d = (type *)dst[0]; \
double *buffer = p->delay_buffer; \
type *src = (type *)ssrc[0]; \
type *dst = (type *)ddst[0]; \
double *buffer = s->delay_buffer; \
\
delay_pos = p->delay_pos; \
modulation_pos = p->modulation_pos; \
delay_pos = s->delay_pos; \
modulation_pos = s->modulation_pos; \
\
for (i = 0; i < nb_samples; i++) { \
int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
p->delay_buffer_length) * channels; \
int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
s->delay_buffer_length) * channels; \
int npos; \
\
delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
npos = delay_pos * channels; \
for (c = 0; c < channels; c++, s++, d++) { \
double v = *s * p->in_gain + buffer[pos + c] * p->decay; \
for (c = 0; c < channels; c++, src++, dst++) { \
double v = *src * s->in_gain + buffer[pos + c] * s->decay; \
\
buffer[npos + c] = v; \
\
*d = v * p->out_gain; \
*dst = v * s->out_gain; \
} \
\
modulation_pos = MOD(modulation_pos + 1, \
p->modulation_buffer_length); \
s->modulation_buffer_length); \
} \
\
p->delay_pos = delay_pos; \
p->modulation_pos = modulation_pos; \
s->delay_pos = delay_pos; \
s->modulation_pos = modulation_pos; \
}
PHASER_PLANAR(dbl, double)
......@@ -202,36 +202,36 @@ PHASER(s32, int32_t)
static int config_output(AVFilterLink *outlink)
{
AudioPhaserContext *p = outlink->src->priv;
AudioPhaserContext *s = outlink->src->priv;
AVFilterLink *inlink = outlink->src->inputs[0];
p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
if (p->delay_buffer_length <= 0) {
s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
if (s->delay_buffer_length <= 0) {
av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
return AVERROR(EINVAL);
}
p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));
s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->channels);
s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer));
if (!p->modulation_buffer || !p->delay_buffer)
if (!s->modulation_buffer || !s->delay_buffer)
return AVERROR(ENOMEM);
ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
p->modulation_buffer, p->modulation_buffer_length,
1., p->delay_buffer_length, M_PI / 2.0);
ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32,
s->modulation_buffer, s->modulation_buffer_length,
1., s->delay_buffer_length, M_PI / 2.0);
p->delay_pos = p->modulation_pos = 0;
s->delay_pos = s->modulation_pos = 0;
switch (inlink->format) {
case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break;
case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break;
case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break;
case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break;
case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
default: av_assert0(0);
}
......@@ -240,7 +240,7 @@ static int config_output(AVFilterLink *outlink)
static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
{
AudioPhaserContext *p = inlink->dst->priv;
AudioPhaserContext *s = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
AVFrame *outbuf;
......@@ -253,7 +253,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
av_frame_copy_props(outbuf, inbuf);
}
p->phaser(p, inbuf->extended_data, outbuf->extended_data,
s->phaser(s, inbuf->extended_data, outbuf->extended_data,
outbuf->nb_samples, av_frame_get_channels(outbuf));
if (inbuf != outbuf)
......@@ -264,10 +264,10 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
static av_cold void uninit(AVFilterContext *ctx)
{
AudioPhaserContext *p = ctx->priv;
AudioPhaserContext *s = ctx->priv;
av_freep(&p->delay_buffer);
av_freep(&p->modulation_buffer);
av_freep(&s->delay_buffer);
av_freep(&s->modulation_buffer);
}
static const AVFilterPad aphaser_inputs[] = {
......
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