Mercurial > mplayer.hg
changeset 16195:cb08f47d3a79
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.
author | ivo |
---|---|
date | Wed, 10 Aug 2005 23:27:39 +0000 |
parents | 167342641f0b |
children | d90f867b870d |
files | libaf/af_ladspa.c |
diffstat | 1 files changed, 9 insertions(+), 13 deletions(-) [+] |
line wrap: on
line diff
--- a/libaf/af_ladspa.c Wed Aug 10 22:48:32 2005 +0000 +++ b/libaf/af_ladspa.c Wed Aug 10 23:27:39 2005 +0000 @@ -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 @@ 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 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 @@ 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 @@ 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]; } }