source: contrib/API/lib/resample.c@ 541

/*
 * Sample rate convertion for both audio and video.
 *
 * This file is part of uniaud.dll.
 *
 * Copyright (c) 2010 Mensys BV
 * Copyright (c) 2007 Vlad Stelmahovsky aka Vladest
 * 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 3 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 and the GNU General Public License along with this library.
 * If not, see <http://www.gnu.org/licenses/>.
 */
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "resample.h"

/* 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 */
        }
}

struct ReSampleContext *audio_resample_init(int output_channels, int input_channels,
                                                                          int output_rate, int input_rate)
{
        struct ReSampleContext *s = NULL;

        if ( input_channels > 2) {
                printf("Resampling with input channels greater than 2 unsupported.");
                return NULL;
        }

        s = malloc(sizeof(struct ReSampleContext));
        if (!s) {
                printf("Can't allocate memory for resample context.");
                return NULL;
        }

        memset(s,0, sizeof(struct ReSampleContext));

        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;

        s->resample_context= (AVResampleContext *)av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0);

        return s;
}

/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
int audio_resample(struct 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 && 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 */
        for(i=0; i<s->filter_channels; i++){
                bufin[i]= (short*) malloc( (nb_samples + s->temp_len) * sizeof(short) );
                memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
                buftmp2[i] = bufin[i] + s->temp_len;
        }

        /* make some zoom to avoid round pb */
        lenout= (int)(nb_samples * s->ratio) + 16;
        bufout[0]= (short*) malloc( lenout * sizeof(short) );
        bufout[1]= (short*) malloc( lenout * sizeof(short) );

        if (s->input_channels == 2 &&
                s->output_channels == 1) {
                buftmp3[0] = output;
                stereo_to_mono(buftmp2[0], input, nb_samples);
        } else if (s->output_channels >= 2 && s->input_channels == 1) {
                buftmp3[0] = bufout[0];
                memcpy(buftmp2[0], input, nb_samples*sizeof(short));
        } else if (s->output_channels >= 2) {
                buftmp3[0] = bufout[0];
                buftmp3[1] = bufout[1];
                stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
        } else {
                buftmp3[0] = output;
                memcpy(buftmp2[0], input, nb_samples*sizeof(short));
        }

        nb_samples += s->temp_len;

        /* resample each channel */
        nb_samples1 = 0; /* avoid warning */
        for(i=0;i<s->filter_channels;i++) {
                int consumed;
                int is_last= i+1 == s->filter_channels;

                nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
                s->temp_len= nb_samples - consumed;
                s->temp[i]= realloc(s->temp[i], s->temp_len*sizeof(short));
                memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
        }

        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);
        }

        for(i=0; i<s->filter_channels; i++)
                free(bufin[i]);

        free(bufout[0]);
        free(bufout[1]);

        return nb_samples1;
}

void audio_resample_close(struct ReSampleContext *s)
{
        av_resample_close(s->resample_context);
        free(s->temp[0]);
        free(s->temp[1]);
        free(s);
}