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audio: Make the simple resampler operate in-place. 

This allows us to avoid an extra copy, allocate less memory and reduce cache
pressure. On the downside: we have to do a lot of tapdancing to resample the
buffer in reverse when the output is growing.
This commit is contained in:
Ryan C. Gordon 2017-01-22 23:48:15 -05:00
parent 64056e81cd
commit 8855daac66
1 changed files with 48 additions and 21 deletions
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69
src/audio/SDL_audiocvt.c
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@ -245,27 +245,60 @@ SDL_ResampleAudioSimple(const int chans, const double rate_incr,
const int finalpos = (total * chans) - chans;
const int dest_samples = (int)(((double)total) * rate_incr);
const double src_incr = 1.0 / rate_incr;
float *dst = outbuf;
float *target = (dst + (dest_samples * chans));
double idx = 0.0;
float *dst;
double idx;
int i;
SDL_assert((dest_samples * framelen) <= outbuflen);
SDL_assert((inbuflen % framelen) == 0);
while (dst < target) {
const int pos = ((int)idx) * chans;
const float *src = &inbuf[pos];
SDL_assert(pos <= finalpos);
for (i = 0; i < chans; i++) {
const float val = *(src++);
*(dst++) = (val + last_sample[i]) * 0.5f;
last_sample[i] = val;
if (rate_incr > 1.0) {
float *target = (outbuf + chans);
const float *earlier_sample = &inbuf[finalpos];
float final_sample[8];
dst = outbuf + (dest_samples * chans);
idx = (double) total;
/* save this off so we can correctly maintain state between runs. */
SDL_memcpy(final_sample, &inbuf[finalpos], framelen);
while (dst > target) {
const int pos = ((int) idx) * chans;
const float *src = &inbuf[pos];
SDL_assert(pos >= 0.0);
for (i = chans - 1; i >= 0; i--) {
const float val = *(--src);
*(--dst) = (val + earlier_sample[i]) * 0.5f;
}
earlier_sample = src;
idx -= src_incr;
}
/* do last sample, interpolated against previous run's state. */
for (i = chans - 1; i >= 0; i--) {
const float val = inbuf[i];
*(--dst) = (val + last_sample[i]) * 0.5f;
}
SDL_memcpy(last_sample, final_sample, framelen);
dst = (outbuf + (dest_samples * chans)) - 1;
} else {
float *target = (outbuf + (dest_samples * chans));
dst = outbuf;
idx = 0.0;
while (dst < target) {
const int pos = ((int) idx) * chans;
const float *src = &inbuf[pos];
SDL_assert(pos <= finalpos);
for (i = 0; i < chans; i++) {
const float val = *(src++);
*(dst++) = (val + last_sample[i]) * 0.5f;
last_sample[i] = val;
}
idx += src_incr;
}
idx += src_incr;
}
return (int) ((dst - outbuf) * (int)sizeof(float));
return (int) ((dst - outbuf) * ((int) sizeof (float)));
}
@ -420,8 +453,8 @@ SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format
{
const float *src = (const float *) cvt->buf;
const int srclen = cvt->len_cvt;
float *dst = (float *) (cvt->buf + srclen);
const int dstlen = (cvt->len * cvt->len_mult) - srclen;
float *dst = (float *) cvt->buf;
const int dstlen = (cvt->len * cvt->len_mult);
float state[8];
SDL_assert(format == AUDIO_F32SYS);
@ -429,8 +462,6 @@ SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format
SDL_memcpy(state, src, chans*sizeof(*src));
cvt->len_cvt = SDL_ResampleAudioSimple(chans, cvt->rate_incr, state, src, srclen, dst, dstlen);
SDL_memcpy(cvt->buf, dst, cvt->len_cvt);
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, format);
}
@ -492,10 +523,6 @@ SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
}
/* the buffer is big enough to hold the destination now, but
we need it large enough to hold a separate scratch buffer. */
cvt->len_mult *= 2;
return 1; /* added a converter. */
}