RepoMirrors
/
ffmpeg
mirror of https://git.ffmpeg.org/ffmpeg.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity
ffmpeg/libavcodec/resample.c

352 lines
9.0 KiB
C
Raw Blame History

/*
* Sample rate convertion for both audio and video
* Copyright (c) 2000 Fabrice Bellard.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/**
* @file resample.c
* Sample rate convertion for both audio and video.
*/
#include "avcodec.h"
typedef struct {
/* fractional resampling */
uint32_t incr; /* fractional increment */
uint32_t 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)floorf(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) {
frac -= FRAC;
if (p >= pend)
goto the_end;
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 void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
{
int i;
short l,r;
for(i=0;i<n;i++) {
l=*input1++;
r=*input2++;
*output++ = l; /* left */
*output++ = (l/2)+(r/2); /* center */
*output++ = r; /* right */
*output++ = 0; /* left surround */
*output++ = 0; /* right surroud */
*output++ = 0; /* low freq */
}
}
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 ( input_channels > 2)
{
av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
return NULL;
}
s = av_mallocz(sizeof(ReSampleContext));
if (!s)
{
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
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;
/*
* ac3 output is the only case where filter_channels could be greater than 2.
* input channels can't be greater than 2, so resample the 2 channels and then
* expand to 6 channels after the resampling.
*/
if(s->filter_channels>2)
s->filter_channels = 2;
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);
} else if (s->output_channels == 6) {
ac3_5p1_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);
}
Reference in New Issue View Git Blame Copy Permalink