qemu/audio/rate_template.h
Volker Rümelin 8933882da9 audio: make the resampling code greedy
Read the maximum possible number of audio frames instead of the
minimum necessary number of frames when the audio stream is
downsampled and the output buffer is limited. This makes the
function symmetrical to upsampling when the input buffer is
limited. The maximum possible number of frames is written here.

With this change it's easier to calculate the exact number of
audio frames the resample function will read or write. These two
functions will be introduced later.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-3-vr_qemu@t-online.de>
2023-03-06 10:30:23 +04:00

124 lines
3.5 KiB
C

/*
* QEMU Mixing engine
*
* Copyright (c) 2004-2005 Vassili Karpov (malc)
* Copyright (c) 1998 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
size_t *isamp, size_t *osamp)
{
struct rate *rate = opaque;
struct st_sample *istart, *iend;
struct st_sample *ostart, *oend;
struct st_sample ilast, icur, out;
#ifdef FLOAT_MIXENG
mixeng_real t;
#else
int64_t t;
#endif
istart = ibuf;
iend = ibuf + *isamp;
ostart = obuf;
oend = obuf + *osamp;
if (rate->opos_inc == (1ULL + UINT_MAX)) {
int i, n = *isamp > *osamp ? *osamp : *isamp;
for (i = 0; i < n; i++) {
OP (obuf[i].l, ibuf[i].l);
OP (obuf[i].r, ibuf[i].r);
}
*isamp = n;
*osamp = n;
return;
}
/* without input samples, there's nothing to do */
if (ibuf >= iend) {
*osamp = 0;
return;
}
ilast = rate->ilast;
while (true) {
/* read as many input samples so that ipos > opos */
while (rate->ipos <= (rate->opos >> 32)) {
ilast = *ibuf++;
rate->ipos++;
/* See if we finished the input buffer yet */
if (ibuf >= iend) {
goto the_end;
}
}
/* make sure that the next output sample can be written */
if (obuf >= oend) {
break;
}
icur = *ibuf;
/* wrap ipos and opos around long before they overflow */
if (rate->ipos >= 0x10001) {
rate->ipos = 1;
rate->opos &= 0xffffffff;
}
/* interpolate */
#ifdef FLOAT_MIXENG
#ifdef RECIPROCAL
t = (rate->opos & UINT_MAX) * (1.f / UINT_MAX);
#else
t = (rate->opos & UINT_MAX) / (mixeng_real) UINT_MAX;
#endif
out.l = (ilast.l * (1.0 - t)) + icur.l * t;
out.r = (ilast.r * (1.0 - t)) + icur.r * t;
#else
t = rate->opos & 0xffffffff;
out.l = (ilast.l * ((int64_t) UINT_MAX - t) + icur.l * t) >> 32;
out.r = (ilast.r * ((int64_t) UINT_MAX - t) + icur.r * t) >> 32;
#endif
/* output sample & increment position */
OP (obuf->l, out.l);
OP (obuf->r, out.r);
obuf += 1;
rate->opos += rate->opos_inc;
}
the_end:
*isamp = ibuf - istart;
*osamp = obuf - ostart;
rate->ilast = ilast;
}
#undef NAME
#undef OP