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Code Issues Releases Wiki Activity

af_volume: some more cosmetics

This commit is contained in:
wm4 2013-10-26 14:04:38 +02:00
parent 13fcb1925a
commit 6ac5474790
1 changed files with 15 additions and 27 deletions
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42
audio/filter/af_volume.c
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@ -28,26 +28,19 @@
#include "af.h"
// Data for specific instances of this filter
typedef struct af_volume_s
{
struct priv {
float level[AF_NCH]; // Gain level for each channel
int soft; // Enable/disable soft clipping
int fast; // Use fix-point volume control
float cfg_volume;
} af_volume_t;
};
// Initialization and runtime control
static int control(struct af_instance *af, int cmd, void *arg)
{
af_volume_t *s = af->priv;
struct priv *s = af->priv;
switch (cmd) {
case AF_CONTROL_REINIT:
// Sanity check
if (!arg)
return AF_ERROR;
mp_audio_copy_config(af->data, (struct mp_audio *)arg);
if (s->fast && (((struct mp_audio *)arg)->format != AF_FORMAT_FLOAT_NE))
@ -66,16 +59,15 @@ static int control(struct af_instance *af, int cmd, void *arg)
return AF_UNKNOWN;
}
// Filter data through filter
static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
{
struct mp_audio *c = data; // Current working data
af_volume_t *s = af->priv;
int nch = c->nch; // Number of channels
struct mp_audio *c = data;
struct priv *s = af->priv;
int nch = c->nch;
if (af->data->format == AF_FORMAT_S16_NE) {
int16_t *a = (int16_t *)c->audio; // Audio data
int len = c->len / 2; // Number of samples
int16_t *a = c->audio;
int len = c->len / 2;
for (int ch = 0; ch < nch; ch++) {
int vol = 256.0 * s->level[ch];
if (vol != 256) {
@ -86,17 +78,13 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
}
}
} else if (af->data->format == AF_FORMAT_FLOAT_NE) {
float *a = (float *)c->audio; // Audio data
int len = c->len / 4; // Number of samples
float *a = c->audio;
int len = c->len / 4;
for (int ch = 0; ch < nch; ch++) {
// Volume control (fader)
if (s->level[ch] != 1.0) {
for (int i = ch; i < len; i += nch) {
float x = a[i];
// Set volume
x *= s->level[ch];
/* Soft clipping, the sound of a dream, thanks to Jon Wattes
post to Musicdsp.org */
a[i] = s->soft ? af_softclip(x) : clamp(x, -1.0, 1.0);
}
}
@ -107,19 +95,19 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
static int af_open(struct af_instance *af)
{
struct af_volume_s *priv = af->priv;
struct priv *s = af->priv;
af->control = control;
af->play = play;
af->mul = 1;
af->data = talloc_zero(af, struct mp_audio);
float level;
af_from_dB(1, &priv->cfg_volume, &level, 20.0, -200.0, 60.0);
af_from_dB(1, &s->cfg_volume, &level, 20.0, -200.0, 60.0);
for (int i = 0; i < AF_NCH; i++)
priv->level[i] = level;
s->level[i] = level;
return AF_OK;
}
#define OPT_BASE_STRUCT struct af_volume_s
#define OPT_BASE_STRUCT struct priv
// Description of this filter
struct af_info af_info_volume = {
@ -127,7 +115,7 @@ struct af_info af_info_volume = {
.name = "volume",
.flags = AF_FLAGS_NOT_REENTRANT,
.open = af_open,
.priv_size = sizeof(struct af_volume_s),
.priv_size = sizeof(struct priv),
.options = (const struct m_option[]) {
OPT_FLOATRANGE("volumedb", cfg_volume, 0, -200, 60),
OPT_FLAG("softclip", soft, 0),