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loader using the SoX libst
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Cary Sandvig
parent
b7da1b768c
commit
2bba74fc8b
304
panda/src/audio/audio_load_st.cxx
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304
panda/src/audio/audio_load_st.cxx
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// Filename: audio_load_st.C
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// Created by: cary (03Oct00)
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//
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////////////////////////////////////////////////////////////////////
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#include <config.h>
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#include "audio_pool.h"
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#include "config_audio.h"
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#include <st.h>
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#include <patchlvl.h>
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#if (PATCHLEVEL == 16)
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#define FORMATS formats
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#define EFFECTS_TYPE struct effect
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#define STREAM_TYPE struct soundstream
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#define GETTYPE gettype
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#define COMPAT_EOF (-1)
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#define CHECKFORMAT checkformat
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#define SIZES sizes
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#define ENCODING styles
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#define COPYFORMAT copyformat
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#define GETEFFECT geteffect
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#define UPDATEEFFECT(a, b, c, d)
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#else /* PATCHLEVEL != 16 */
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#define FORMATS st_formats
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#define EFFECTS_TYPE struct st_effect
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#define STREAM_TYPE struct st_soundstream
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#define GETTYPE st_gettype
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#define COMPAT_EOF ST_EOF
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#define CHECKFORMAT st_checkformat
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#define SIZES st_sizes_str
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#define ENCODING st_encodings_str
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#define COPYFORMAT st_copyformat
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#define GETEFFECT st_geteffect
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#define UPDATEEFFECT st_updateeffect
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#endif /* PATCHLEVEL */
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Configure(audio_load_st);
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// the effects we will be using are to change the rate and number of channels
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static EFFECTS_TYPE efftab[5]; // left/mono channel effects
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static EFFECTS_TYPE efftabR[5]; // right channel effects
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static int neffects; // how many effects are in action
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static STREAM_TYPE informat; // holder for the input
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static STREAM_TYPE outformat; // holder for fake output;
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INLINE static void init_stream(void) {
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informat.info.rate = 0;
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informat.info.size = -1;
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informat.info.encoding = -1;
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informat.info.channels = -1;
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informat.comment = (char*)0L;
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informat.swap = 0;
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informat.filetype = (char*)0L;
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informat.fp = stdin;
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informat.filename = "input";
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outformat.info.rate = audio_mix_freq;
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outformat.info.size = -1;
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outformat.info.encoding = -1;
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outformat.info.channels = 2;
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outformat.comment = (char*)0L;
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outformat.swap = 0;
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outformat.filetype = (char*)0L;
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outformat.fp = stdout;
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outformat.filename = "output";
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}
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INLINE static void check_effects(void) {
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bool needchan = (informat.info.rate != audio_mix_freq);
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bool needrate = (informat.info.channels != 2);
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// efftab[0] is always the input stream and always exists
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neffects = 1;
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// if reducing the number of samples, it is faster to run all effects
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// after the resample effect
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if (needrate) {
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GETEFFECT(&efftab[neffects], "resample");
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// setup and give default opts
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(*efftab[neffects].h->getopts)(&efftab[neffects],(int)0,(char**)0L);
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// copy format info to effect table
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UPDATEEFFECT(&efftab[neffects], &informat, &outformat, 0);
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// rate can't handle multiple channels so be sure and account for that
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if (efftab[neffects].ininfo.channels > 1)
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memcpy(&efftabR[neffects], &efftab[neffects], sizeof(EFFECTS_TYPE));
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++neffects;
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}
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// if we ever have more then 2 channels in an input file, we will need to
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// deal with that somewhere here
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if (needchan) {
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GETEFFECT(&efftab[neffects], "avg");
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//setup and give default opts
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(*efftab[neffects].h->getopts)(&efftab[neffects],(int)0,(char**)0L);
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// copy format info to effect table
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UPDATEEFFECT(&efftab[neffects], &informat, &outformat, 0);
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++neffects;
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}
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}
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static byte* read_file(Filename filename) {
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int e, havedata;
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ostringstream out;
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init_stream();
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if ((informat.fp = fopen(filename.c_str(), READBINARY)) == NULL) {
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audio_cat->error() << "could not open '" << filename << "'" << endl;
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return (byte*)0L;
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}
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informat.filname = filename.c_str();
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informat.filetype = filename.get_extension();
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informat.comment = filename.c_str(); // for lack of anything better
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// now we start some more real work
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GETTYPE(&informat);
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// read and write starters can change their formats
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if ((*informat.h->startread)(&informat) == COMPAT_EOF) {
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audio_cat->error() << "failed to start read" << endl;
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return (byte*)0L;
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}
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CHECKFORMAT(&informat);
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if (audio_cat->is_debug())
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audio_cat->debug() << "Input file '" << informat.filename
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<< "': sample rate = " << informat.info.rate
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<< " size = " << SIZES[informat.info.size]
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<< " encoding = " << ENCODING[informat.info.encoding]
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<< " " << informat.info.channels
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<< (informat.info.channels > 1)?"channels":"channel"
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<< endl;
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if (audio_cat->is_debug())
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audio_cat->debug() << "Input file comment: '" << informat.comment << "'"
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<< endl;
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COPYFORMAT(&informat, &outformat);
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check_effects();
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// start all effects
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for (e=1; e<neffects; ++e) {
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(*efftab[e].h->start)(&efftab[e]);
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if (efftabR[e].name)
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(*efftabR[e].h->start)(&efftabR[e]);
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}
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// reserve output buffers for all effects
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for (e=0; e<neffects; e++) {
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efftab[e].obuf = new LONG[BUFSIZ*sizeof(LONG)];
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if (efftabR[e].name)
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efftabR[e].obuf = new LONG[BUFSIZ*sizeof(LONG)];
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}
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// get the main while loop ready, have some data for it to start on
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efftab[0].olen = (*informat.h->read)(&informat, efftab[0].obuf,
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(LONG)BUFSIZ);
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efftab[0].odone = 0;
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// run input data thru effects and get more until olen == 0
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while (efftab[0].olen > 0) {
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// mark chain as empty
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for (e=1; e<neffects; ++e)
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efftab[e].odone = efftab[e].olen = 0;
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do {
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ULONG w;
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// run entire chain backwards: pull, don't push
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// this is because buffering system isn't a nice queueing system
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for (e=neffects-1; e>0; --e)
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if (flow_effect(e))
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break;
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// add to output data
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if (efftab[neffects-1].olen>efftab[neffects-1].odone) {
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for (LONG i=0; i<efftab[neffects-1].olen; ++i) {
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LONG foo = efftab[neffects-1].obuf[i];
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signed short bar = (foo >> 16);
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unsigned char *b = &bar;
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out << *b << *(++b);
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}
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efftab[neffects-1].odone = efftab[neffects-1].olen;
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}
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// if there is still stuff in the pipeline, setup to flow effects again
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havedata = 0;
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for (e=0; e<neffects-1; ++e)
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if (efftab[e].odone < efftab[e].olen) {
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havedata = 1;
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break;
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}
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} while (havedata);
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// read another chunk
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efftab[0].olen = (*informat.h->read)(&informat, efftab[0].obuf,
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(LONG)BUFSIZ);
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efftab[0].odone = 0;
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}
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// drain the effects out first to last. push the residue thru subsequent
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// effects. suck.
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for (e=1; e<neffects; ++e)
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while (1) {
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if (drain_effect(e) == 0)
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break; // get out of while loop
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if (efftab[neffects-1].olen > 0) {
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for (LONG i=0; i<efftab[neffects-1].olen; ++i) {
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LONG foo = efftab[neffects-1].obuf[i];
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signed short bar = (foo >> 16);
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unsigned char *b = &bar;
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out << *b << *(++b);
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}
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}
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if (efftab[e].olen != BUFSIZ)
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break;
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}
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// stop all effects. In so doing, some may generate more data
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for (e=1; e<neffects; ++e) {
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(*efftab[e].h->stop)(&efftab[e]);
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if (efftabR[e].name)
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(*efftabR[e].h->stop)(&efftabR[e]);
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}
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// stop reading the file
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if ((*informat.h->stopread)(&informat) == COMPAT_EOF) {
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audio_cat->error() << "error stoping input file" << endl;
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}
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fclose(informat.fp);
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// generate output
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string s = out.str();
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int slen = s.length();
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byte* ret = new byte[slen];
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memcpy(ret, s.data(), slen);
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return ret;
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}
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#ifdef USE_MIKMOD
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void AudioDestroySt(AudioTraits::SampleClass* sample) {
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delete sample;
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}
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void AudioLoadSt(AudioTraits::SampleClass** sample,
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AudioTraits::PlayerClass** player,
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AudioTraits::DeleteSampleFunc** destroy, Filename) {
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audio_cat->warning() << "MikMod doesn't support reading raw data yet"
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<< endl;
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*sample = (AudioTraits::SampleClass*)0L;
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*player = (AudioTraits::PlayerClass*)0L;
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*destroy = AudioDestroySt;
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}
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#else /* no MikMod */
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ifdef PENV_WIN32
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void AudioDestroySt(AudioTraits::SampleClass* sample) {
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delete sample;
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}
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void AudioLoadSt(AudioTraits::SampleClass** sample,
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AudioTraits::PlayerClass** player,
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AudioTraits::DeleteSampleFunc** destroy, Filename) {
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audio_cat->warning() << "win32 doesn't support reading raw data yet"
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<< endl;
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*sample = (AudioTraits::SampleClass*)0L;
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*player = (AudioTraits::PlayerClass*)0L;
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*destroy = AudioDestroySt;
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}
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#else /* no win32 */
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#ifdev PENV_LINUX
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void AudioDestroySt(AudioTraits::SampleClass* sample) {
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delete sample;
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}
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void AudioLoadSt(AudioTraits::SampleClass** sample,
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AudioTraits::PlayerClass** player,
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AudioTraits::DeleteSampleFunc** destroy, Filename) {
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audio_cat->warning() << "linux doesn't support reading raw data yet"
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<< endl;
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*sample = (AudioTraits::SampleClass*)0L;
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*player = (AudioTraits::PlayerClass*)0L;
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*destroy = AudioDestroySt;
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}
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#else /* no linux */
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// Null driver
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#include "audio_null_traits.h"
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void AudioDestroySt(AudioTraits::SampleClass* sample) {
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delete sample;
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}
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void AudioLoadSt(AudioTraits::SampleClass** sample,
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AudioTraits::PlayerClass** player,
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AudioTraits::DeleteSampleFunc** destroy, Filename) {
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*sample = new NullSample();
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*player = new NullPlayer();
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*destroy = AudioDestroySt;
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}
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#endif /* linux */
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#endif /* win32 */
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#endif /* MikMod */
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ConfigureFn(audio_load_st) {
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for (int i=0; FORMATS[i].names != (char**)0L; ++i)
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for (int j=0; FORMATS[i].names[j] != (char*)0L; ++j) {
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if (audio_cat->is_debug())
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audio_cat->debug() << "adding reader for '." << FORMATS[i].names[j]
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<< "'" << endl;
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AudioPool::register_sample_loader(FORMATS[i].names[j], AudioLoadSt);
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}
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}
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