mirror of
https://github.com/xenia-project/FFmpeg.git
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313 lines
8.0 KiB
C
313 lines
8.0 KiB
C
/*
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* Sample rate convertion for both audio and video
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* Copyright (c) 2000 Fabrice Bellard.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "avcodec.h"
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typedef struct {
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/* fractional resampling */
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UINT32 incr; /* fractional increment */
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UINT32 frac;
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int last_sample;
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/* integer down sample */
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int iratio; /* integer divison ratio */
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int icount, isum;
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int inv;
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} ReSampleChannelContext;
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struct ReSampleContext {
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ReSampleChannelContext channel_ctx[2];
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float ratio;
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/* channel convert */
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int input_channels, output_channels, filter_channels;
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};
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#define FRAC_BITS 16
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#define FRAC (1 << FRAC_BITS)
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static void init_mono_resample(ReSampleChannelContext *s, float ratio)
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{
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ratio = 1.0 / ratio;
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s->iratio = (int)floor(ratio);
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if (s->iratio == 0)
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s->iratio = 1;
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s->incr = (int)((ratio / s->iratio) * FRAC);
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s->frac = FRAC;
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s->last_sample = 0;
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s->icount = s->iratio;
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s->isum = 0;
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s->inv = (FRAC / s->iratio);
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}
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/* fractional audio resampling */
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static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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{
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unsigned int frac, incr;
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int l0, l1;
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short *q, *p, *pend;
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l0 = s->last_sample;
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incr = s->incr;
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frac = s->frac;
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p = input;
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pend = input + nb_samples;
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q = output;
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l1 = *p++;
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for(;;) {
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/* interpolate */
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*q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
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frac = frac + s->incr;
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while (frac >= FRAC) {
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if (p >= pend)
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goto the_end;
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frac -= FRAC;
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l0 = l1;
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l1 = *p++;
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}
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}
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the_end:
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s->last_sample = l1;
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s->frac = frac;
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return q - output;
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}
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static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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{
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short *q, *p, *pend;
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int c, sum;
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p = input;
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pend = input + nb_samples;
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q = output;
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c = s->icount;
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sum = s->isum;
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for(;;) {
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sum += *p++;
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if (--c == 0) {
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*q++ = (sum * s->inv) >> FRAC_BITS;
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c = s->iratio;
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sum = 0;
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}
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if (p >= pend)
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break;
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}
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s->isum = sum;
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s->icount = c;
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return q - output;
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}
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/* n1: number of samples */
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static void stereo_to_mono(short *output, short *input, int n1)
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{
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short *p, *q;
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int n = n1;
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p = input;
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q = output;
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while (n >= 4) {
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q[0] = (p[0] + p[1]) >> 1;
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q[1] = (p[2] + p[3]) >> 1;
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q[2] = (p[4] + p[5]) >> 1;
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q[3] = (p[6] + p[7]) >> 1;
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q += 4;
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p += 8;
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n -= 4;
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}
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while (n > 0) {
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q[0] = (p[0] + p[1]) >> 1;
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q++;
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p += 2;
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n--;
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}
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}
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/* n1: number of samples */
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static void mono_to_stereo(short *output, short *input, int n1)
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{
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short *p, *q;
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int n = n1;
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int v;
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p = input;
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q = output;
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while (n >= 4) {
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v = p[0]; q[0] = v; q[1] = v;
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v = p[1]; q[2] = v; q[3] = v;
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v = p[2]; q[4] = v; q[5] = v;
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v = p[3]; q[6] = v; q[7] = v;
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q += 8;
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p += 4;
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n -= 4;
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}
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while (n > 0) {
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v = p[0]; q[0] = v; q[1] = v;
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q += 2;
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p += 1;
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n--;
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}
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}
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/* XXX: should use more abstract 'N' channels system */
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static void stereo_split(short *output1, short *output2, short *input, int n)
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{
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int i;
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for(i=0;i<n;i++) {
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*output1++ = *input++;
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*output2++ = *input++;
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}
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}
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static void stereo_mux(short *output, short *input1, short *input2, int n)
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{
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int i;
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for(i=0;i<n;i++) {
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*output++ = *input1++;
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*output++ = *input2++;
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}
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}
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static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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{
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short *buf1;
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short *buftmp;
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buf1= (short*)av_malloc( nb_samples * sizeof(short) );
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/* first downsample by an integer factor with averaging filter */
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if (s->iratio > 1) {
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buftmp = buf1;
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nb_samples = integer_downsample(s, buftmp, input, nb_samples);
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} else {
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buftmp = input;
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}
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/* then do a fractional resampling with linear interpolation */
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if (s->incr != FRAC) {
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nb_samples = fractional_resample(s, output, buftmp, nb_samples);
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} else {
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memcpy(output, buftmp, nb_samples * sizeof(short));
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}
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av_free(buf1);
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return nb_samples;
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}
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ReSampleContext *audio_resample_init(int output_channels, int input_channels,
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int output_rate, int input_rate)
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{
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ReSampleContext *s;
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int i;
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if (output_channels > 2 || input_channels > 2)
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return NULL;
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s = av_mallocz(sizeof(ReSampleContext));
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if (!s)
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return NULL;
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s->ratio = (float)output_rate / (float)input_rate;
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s->input_channels = input_channels;
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s->output_channels = output_channels;
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s->filter_channels = s->input_channels;
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if (s->output_channels < s->filter_channels)
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s->filter_channels = s->output_channels;
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for(i=0;i<s->filter_channels;i++) {
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init_mono_resample(&s->channel_ctx[i], s->ratio);
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}
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return s;
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}
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/* resample audio. 'nb_samples' is the number of input samples */
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/* XXX: optimize it ! */
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/* XXX: do it with polyphase filters, since the quality here is
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HORRIBLE. Return the number of samples available in output */
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int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
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{
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int i, nb_samples1;
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short *bufin[2];
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short *bufout[2];
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short *buftmp2[2], *buftmp3[2];
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int lenout;
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if (s->input_channels == s->output_channels && s->ratio == 1.0) {
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/* nothing to do */
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memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
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return nb_samples;
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}
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/* XXX: move those malloc to resample init code */
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bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
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bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
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/* make some zoom to avoid round pb */
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lenout= (int)(nb_samples * s->ratio) + 16;
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bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
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bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
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if (s->input_channels == 2 &&
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s->output_channels == 1) {
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buftmp2[0] = bufin[0];
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buftmp3[0] = output;
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stereo_to_mono(buftmp2[0], input, nb_samples);
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} else if (s->output_channels == 2 && s->input_channels == 1) {
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buftmp2[0] = input;
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buftmp3[0] = bufout[0];
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} else if (s->output_channels == 2) {
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buftmp2[0] = bufin[0];
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buftmp2[1] = bufin[1];
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buftmp3[0] = bufout[0];
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buftmp3[1] = bufout[1];
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stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
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} else {
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buftmp2[0] = input;
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buftmp3[0] = output;
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}
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/* resample each channel */
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nb_samples1 = 0; /* avoid warning */
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for(i=0;i<s->filter_channels;i++) {
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nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
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}
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if (s->output_channels == 2 && s->input_channels == 1) {
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mono_to_stereo(output, buftmp3[0], nb_samples1);
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} else if (s->output_channels == 2) {
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stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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}
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av_free(bufin[0]);
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av_free(bufin[1]);
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av_free(bufout[0]);
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av_free(bufout[1]);
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return nb_samples1;
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}
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void audio_resample_close(ReSampleContext *s)
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{
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av_free(s);
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}
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