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use a single converter for both channels (if input data is stereo), for improved efficency; renamed MemoryAudioInputStream -> LinearMemoryStream and use some template voodoo to make the baseclass of Input8bitSignedStream etc. variable; added (commented out) draf of WrappedMemoryStream

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svn-id: r9209
This commit is contained in:
Max Horn 2003-07-28 01:13:31 +00:00
parent 12d872f754
commit e87bc6d89e
5 changed files with 174 additions and 88 deletions
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54
sound/resample.cpp
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@ -274,6 +274,16 @@ int st_resample_flow(eff_t effp, AudioInputStream &input, st_sample_t *obuf, st_
const long obufSize = *osamp;
#endif
TODO: adjust for the changes made to AudioInputStream; add support for stereo
initially, could just average the left/right channel -> bad for quality of course,
but easiest to implement and would get this going again.
Next step is to duplicate the X/Y buffers... a lot of computations don't care about
how many channels there are anyway, they could just be ran twice, e.g. SrcEX and SrcUD.
But better for efficiency would be to rewrite those to deal with 2 channels, too.
Because esp in SrcEX/SrcUD, only very few computations depend on the input data,
and dealing with both channels in parallel should only be a little slower than dealing
with them alone
// Constrain amount we actually process
//fprintf(stderr,"Xp %d, Xread %d\n",r->Xp, r->Xread);
@ -313,7 +323,6 @@ int st_resample_flow(eff_t effp, AudioInputStream &input, st_sample_t *obuf, st_
// Finally compute the effective number of bytes to process
Nproc = last - r->Xoff - r->Xp;
printf("FOO(3) Nproc %ld\n", Nproc);
if (Nproc <= 0) {
/* fill in starting here next time */
@ -356,7 +365,7 @@ printf("FOO(3) Nproc %ld\n", Nproc);
/* Copy back portion of input signal that must be re-used */
k = r->Xp - r->Xoff;
fprintf(stderr,"k %d, last %d\n",k,last);
//fprintf(stderr,"k %d, last %d\n",k,last);
for (i = 0; i < last - k; i++)
r->X[i] = r->X[i + k];
@ -370,7 +379,8 @@ printf("osamp = %ld, Nout = %ld\n", obufSize, Nout);
int sample = (int)(r->Y[i] * vol / 256);
clampedAdd(*obuf++, sample);
#if 1 // FIXME: Hack to generate stereo output
clampedAdd(*obuf++, sample);
// clampedAdd(*obuf++, sample);
*obuf++;
#endif
}
@ -514,12 +524,12 @@ static long SrcUD(resample_t r, long Nx) {
Factor = r->Factor;
time = r->Time;
dt = 1.0 / Factor; /* Output sampling period */
fprintf(stderr,"Factor %f, dt %f, ",Factor,dt);
fprintf(stderr,"Time %f, ",r->Time);
//fprintf(stderr,"Factor %f, dt %f, ",Factor,dt);
//fprintf(stderr,"Time %f, ",r->Time);
/* (Xh * dhb)>>La is max index into Imp[] */
/*fprintf(stderr,"ct=%d\n",ct);*/
fprintf(stderr,"ct=%.2f %d\n",(double)r->Nwing*Na/r->dhb, r->Xh);
fprintf(stderr,"ct=%ld, T=%.6f, dhb=%6f, dt=%.6f\n", r->Xh, time-floor(time),(double)r->dhb/Na,dt);
//fprintf(stderr,"ct=%.2f %d\n",(double)r->Nwing*Na/r->dhb, r->Xh);
//fprintf(stderr,"ct=%ld, T=%.6f, dhb=%6f, dt=%.6f\n", r->Xh, time-floor(time),(double)r->dhb/Na,dt);
Ystart = Y = r->Y + r->Yposition;
n = (int)ceil((double)Nx / dt);
while (n--) {
@ -722,3 +732,33 @@ static void LpFilter(double *c, long N, double frq, double Beta, long Num) {
}
}
}
#pragma mark -
ResampleRateConverter::ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality) {
// FIXME: quality is for now a nasty hack.
// Valid values are 0,1,2,3 (everything else is treated like 0 for now)
const char *arg = 0;
switch (quality) {
case 1: arg = "-qs"; break;
case 2: arg = "-q"; break;
case 3: arg = "-ql"; break;
}
st_resample_getopts(&effp, arg ? 1 : 0, &arg);
st_resample_start(&effp, inrate, outrate);
}
ResampleRateConverter::~ResampleRateConverter() {
st_resample_stop(&effp);
}
int ResampleRateConverter::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
return st_resample_flow(&effp, input, obuf, osamp, vol);
}
int ResampleRateConverter::drain(st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
return st_resample_drain(&effp, obuf, osamp, vol);
}