313 lines
7.9 KiB
C
313 lines
7.9 KiB
C
/*
|
|
* Sample rate convertion for both audio and video
|
|
* Copyright (c) 2000 Gerard Lantau.
|
|
*
|
|
* This program is free software; you can redistribute it and/or modify
|
|
* it under the terms of the GNU General Public License as published by
|
|
* the Free Software Foundation; either version 2 of the License, or
|
|
* (at your option) any later version.
|
|
*
|
|
* This program is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
* GNU General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU General Public License
|
|
* along with this program; if not, write to the Free Software
|
|
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
|
|
*/
|
|
#include "avcodec.h"
|
|
|
|
typedef struct {
|
|
/* fractional resampling */
|
|
UINT32 incr; /* fractional increment */
|
|
UINT32 frac;
|
|
int last_sample;
|
|
/* integer down sample */
|
|
int iratio; /* integer divison ratio */
|
|
int icount, isum;
|
|
int inv;
|
|
} ReSampleChannelContext;
|
|
|
|
struct ReSampleContext {
|
|
ReSampleChannelContext channel_ctx[2];
|
|
float ratio;
|
|
/* channel convert */
|
|
int input_channels, output_channels, filter_channels;
|
|
};
|
|
|
|
|
|
#define FRAC_BITS 16
|
|
#define FRAC (1 << FRAC_BITS)
|
|
|
|
static void init_mono_resample(ReSampleChannelContext *s, float ratio)
|
|
{
|
|
ratio = 1.0 / ratio;
|
|
s->iratio = (int)floor(ratio);
|
|
if (s->iratio == 0)
|
|
s->iratio = 1;
|
|
s->incr = (int)((ratio / s->iratio) * FRAC);
|
|
s->frac = FRAC;
|
|
s->last_sample = 0;
|
|
s->icount = s->iratio;
|
|
s->isum = 0;
|
|
s->inv = (FRAC / s->iratio);
|
|
}
|
|
|
|
/* fractional audio resampling */
|
|
static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
|
|
{
|
|
unsigned int frac, incr;
|
|
int l0, l1;
|
|
short *q, *p, *pend;
|
|
|
|
l0 = s->last_sample;
|
|
incr = s->incr;
|
|
frac = s->frac;
|
|
|
|
p = input;
|
|
pend = input + nb_samples;
|
|
q = output;
|
|
|
|
l1 = *p++;
|
|
for(;;) {
|
|
/* interpolate */
|
|
*q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
|
|
frac = frac + s->incr;
|
|
while (frac >= FRAC) {
|
|
if (p >= pend)
|
|
goto the_end;
|
|
frac -= FRAC;
|
|
l0 = l1;
|
|
l1 = *p++;
|
|
}
|
|
}
|
|
the_end:
|
|
s->last_sample = l1;
|
|
s->frac = frac;
|
|
return q - output;
|
|
}
|
|
|
|
static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
|
|
{
|
|
short *q, *p, *pend;
|
|
int c, sum;
|
|
|
|
p = input;
|
|
pend = input + nb_samples;
|
|
q = output;
|
|
|
|
c = s->icount;
|
|
sum = s->isum;
|
|
|
|
for(;;) {
|
|
sum += *p++;
|
|
if (--c == 0) {
|
|
*q++ = (sum * s->inv) >> FRAC_BITS;
|
|
c = s->iratio;
|
|
sum = 0;
|
|
}
|
|
if (p >= pend)
|
|
break;
|
|
}
|
|
s->isum = sum;
|
|
s->icount = c;
|
|
return q - output;
|
|
}
|
|
|
|
/* n1: number of samples */
|
|
static void stereo_to_mono(short *output, short *input, int n1)
|
|
{
|
|
short *p, *q;
|
|
int n = n1;
|
|
|
|
p = input;
|
|
q = output;
|
|
while (n >= 4) {
|
|
q[0] = (p[0] + p[1]) >> 1;
|
|
q[1] = (p[2] + p[3]) >> 1;
|
|
q[2] = (p[4] + p[5]) >> 1;
|
|
q[3] = (p[6] + p[7]) >> 1;
|
|
q += 4;
|
|
p += 8;
|
|
n -= 4;
|
|
}
|
|
while (n > 0) {
|
|
q[0] = (p[0] + p[1]) >> 1;
|
|
q++;
|
|
p += 2;
|
|
n--;
|
|
}
|
|
}
|
|
|
|
/* n1: number of samples */
|
|
static void mono_to_stereo(short *output, short *input, int n1)
|
|
{
|
|
short *p, *q;
|
|
int n = n1;
|
|
int v;
|
|
|
|
p = input;
|
|
q = output;
|
|
while (n >= 4) {
|
|
v = p[0]; q[0] = v; q[1] = v;
|
|
v = p[1]; q[2] = v; q[3] = v;
|
|
v = p[2]; q[4] = v; q[5] = v;
|
|
v = p[3]; q[6] = v; q[7] = v;
|
|
q += 8;
|
|
p += 4;
|
|
n -= 4;
|
|
}
|
|
while (n > 0) {
|
|
v = p[0]; q[0] = v; q[1] = v;
|
|
q += 2;
|
|
p += 1;
|
|
n--;
|
|
}
|
|
}
|
|
|
|
/* XXX: should use more abstract 'N' channels system */
|
|
static void stereo_split(short *output1, short *output2, short *input, int n)
|
|
{
|
|
int i;
|
|
|
|
for(i=0;i<n;i++) {
|
|
*output1++ = *input++;
|
|
*output2++ = *input++;
|
|
}
|
|
}
|
|
|
|
static void stereo_mux(short *output, short *input1, short *input2, int n)
|
|
{
|
|
int i;
|
|
|
|
for(i=0;i<n;i++) {
|
|
*output++ = *input1++;
|
|
*output++ = *input2++;
|
|
}
|
|
}
|
|
|
|
static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
|
|
{
|
|
short *buf1;
|
|
short *buftmp;
|
|
|
|
buf1= (short*)av_malloc( nb_samples * sizeof(short) );
|
|
|
|
/* first downsample by an integer factor with averaging filter */
|
|
if (s->iratio > 1) {
|
|
buftmp = buf1;
|
|
nb_samples = integer_downsample(s, buftmp, input, nb_samples);
|
|
} else {
|
|
buftmp = input;
|
|
}
|
|
|
|
/* then do a fractional resampling with linear interpolation */
|
|
if (s->incr != FRAC) {
|
|
nb_samples = fractional_resample(s, output, buftmp, nb_samples);
|
|
} else {
|
|
memcpy(output, buftmp, nb_samples * sizeof(short));
|
|
}
|
|
av_free(buf1);
|
|
return nb_samples;
|
|
}
|
|
|
|
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
|
|
int output_rate, int input_rate)
|
|
{
|
|
ReSampleContext *s;
|
|
int i;
|
|
|
|
if (output_channels > 2 || input_channels > 2)
|
|
return NULL;
|
|
|
|
s = av_mallocz(sizeof(ReSampleContext));
|
|
if (!s)
|
|
return NULL;
|
|
|
|
s->ratio = (float)output_rate / (float)input_rate;
|
|
|
|
s->input_channels = input_channels;
|
|
s->output_channels = output_channels;
|
|
|
|
s->filter_channels = s->input_channels;
|
|
if (s->output_channels < s->filter_channels)
|
|
s->filter_channels = s->output_channels;
|
|
|
|
for(i=0;i<s->filter_channels;i++) {
|
|
init_mono_resample(&s->channel_ctx[i], s->ratio);
|
|
}
|
|
return s;
|
|
}
|
|
|
|
/* resample audio. 'nb_samples' is the number of input samples */
|
|
/* XXX: optimize it ! */
|
|
/* XXX: do it with polyphase filters, since the quality here is
|
|
HORRIBLE. Return the number of samples available in output */
|
|
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
|
|
{
|
|
int i, nb_samples1;
|
|
short *bufin[2];
|
|
short *bufout[2];
|
|
short *buftmp2[2], *buftmp3[2];
|
|
int lenout;
|
|
|
|
if (s->input_channels == s->output_channels && s->ratio == 1.0) {
|
|
/* nothing to do */
|
|
memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
|
|
return nb_samples;
|
|
}
|
|
|
|
/* XXX: move those malloc to resample init code */
|
|
bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
|
|
bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
|
|
|
|
/* make some zoom to avoid round pb */
|
|
lenout= (int)(nb_samples * s->ratio) + 16;
|
|
bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
|
|
bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
|
|
|
|
if (s->input_channels == 2 &&
|
|
s->output_channels == 1) {
|
|
buftmp2[0] = bufin[0];
|
|
buftmp3[0] = output;
|
|
stereo_to_mono(buftmp2[0], input, nb_samples);
|
|
} else if (s->output_channels == 2 && s->input_channels == 1) {
|
|
buftmp2[0] = input;
|
|
buftmp3[0] = bufout[0];
|
|
} else if (s->output_channels == 2) {
|
|
buftmp2[0] = bufin[0];
|
|
buftmp2[1] = bufin[1];
|
|
buftmp3[0] = bufout[0];
|
|
buftmp3[1] = bufout[1];
|
|
stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
|
|
} else {
|
|
buftmp2[0] = input;
|
|
buftmp3[0] = output;
|
|
}
|
|
|
|
/* resample each channel */
|
|
nb_samples1 = 0; /* avoid warning */
|
|
for(i=0;i<s->filter_channels;i++) {
|
|
nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
|
|
}
|
|
|
|
if (s->output_channels == 2 && s->input_channels == 1) {
|
|
mono_to_stereo(output, buftmp3[0], nb_samples1);
|
|
} else if (s->output_channels == 2) {
|
|
stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
|
|
}
|
|
|
|
av_free(bufin[0]);
|
|
av_free(bufin[1]);
|
|
|
|
av_free(bufout[0]);
|
|
av_free(bufout[1]);
|
|
return nb_samples1;
|
|
}
|
|
|
|
void audio_resample_close(ReSampleContext *s)
|
|
{
|
|
av_free(s);
|
|
}
|