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Removed in-filter int to float conversion. af_ladspa now demands floats as 

that's what LADSPA filters use internally too. conversion from int, if needed,
is done by af_format as it's supposed to.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@16196 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
ivo 2005-08-10 23:27:39 +00:00
parent 452c2ba5da
commit 812ee3753e
1 changed files with 9 additions and 13 deletions
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22
libaf/af_ladspa.c
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@ -24,6 +24,7 @@
* Changelog
*
* 2005-06-21 Replaced erroneous use of mp_msg by af_msg
* 2005-05-30 Removed int16 to float conversion; leave that to af_format
* 2004-12-23 Added to CVS
* 2004-12-22 Cleaned up cosmetics
* Made conversion loops in play() more cache-friendly
@ -535,12 +536,12 @@ static int control(struct af_instance_s *af, int cmd, void *arg) {
if (!arg) return AF_ERROR;
/* for now, only accept 16 bit signed int */
/* accept FLOAT, let af_format do conversion */
af->data->rate = ((af_data_t*)arg)->rate;
af->data->nch = ((af_data_t*)arg)->nch;
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
/* arg->len is not set here yet, so init of buffers and connecting the
* filter, has to be done in play() :-/
@ -768,12 +769,11 @@ static void uninit(struct af_instance_s *af) {
static af_data_t* play(struct af_instance_s *af, af_data_t *data) {
af_ladspa_t *setup = af->setup;
const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
int16_t *audio = (int16_t*)data->audio;
int nsamples = data->len/2; /* /2 because it's int16_t */
float *audio = (float*)data->audio;
int nsamples = data->len/4; /* /4 because it's 32-bit float */
int nch = data->nch;
int rate = data->rate;
int i, p;
float v;
if (setup->status !=AF_OK)
return data;
@ -911,7 +911,7 @@ static af_data_t* play(struct af_instance_s *af, af_data_t *data) {
for (p=0; p<setup->bufsize; p++) {
for (i=0; i<nch; i++) {
setup->inbufs[i][p] = ( (float) audio[p*nch + i] ) / 32768.0f;
setup->inbufs[i][p] = audio[p*nch + i];
}
}
@ -923,15 +923,11 @@ static af_data_t* play(struct af_instance_s *af, af_data_t *data) {
i++;
}
/* Extract outbufs, hard clipping in case the filter exceeded [-1.0,1.0] */
/* Extract outbufs */
for (p=0; p<setup->bufsize; p++) {
for (i=0; i<nch; i++) {
v = setup->outbufs[i][p];
v *= 32768.0f;
v = (v > 32767.0f ? 32767.0f : v);
v = (v < -32768.0f ? -32768.0f : v);
audio[p*nch + i] = (int16_t) v;
audio[p*nch + i] = setup->outbufs[i][p];
}
}