comparison resample.c @ 64:5aa6292a1660 libavcodec

win32 fixes

63:8e2d8dbf89a5

 *
 * 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 <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include "avcodec.h"

#define NDEBUG
#include <assert.h>

typedef struct {
    /* fractional resampling */
    UINT32 incr; /* fractional increment */
    UINT32 frac;

    }
}

static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    short buf1[nb_samples];
    short *buftmp;

    /* first downsample by an integer factor with averaging filter */
    if (s->iratio > 1) {
        buftmp = buf1;
        nb_samples = integer_downsample(s, buftmp, input, nb_samples);

    if (s->incr != FRAC) {
        nb_samples = fractional_resample(s, output, buftmp, nb_samples);
    } else {
        memcpy(output, buftmp, nb_samples * sizeof(short));
    }
    return nb_samples;
}

ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
                                      int output_rate, int input_rate)

/* 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][nb_samples];
    short bufout[2][(int)(nb_samples * s->ratio) + 16]; /* make some zoom to avoid round pb */
    short *buftmp2[2], *buftmp3[2];

    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;
    }

    if (s->input_channels == 2 &&
        s->output_channels == 1) {
        buftmp2[0] = bufin[0];
        buftmp3[0] = output;

        mono_to_stereo(output, buftmp3[0], nb_samples1);
    } else if (s->output_channels == 2) {
        stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
    }

    return nb_samples1;
}

void audio_resample_close(ReSampleContext *s)
{

64:5aa6292a1660

 *
 * 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"
#include <math.h>

typedef struct {
    /* fractional resampling */
    UINT32 incr; /* fractional increment */
    UINT32 frac;

    }
}

static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    short *buf1;
    short *buftmp;

    buf1= (short*) 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);

    if (s->incr != FRAC) {
        nb_samples = fractional_resample(s, output, buftmp, nb_samples);
    } else {
        memcpy(output, buftmp, nb_samples * sizeof(short));
    }
    free(buf1);
    return nb_samples;
}

ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
                                      int output_rate, int input_rate)

/* 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*) malloc( nb_samples * sizeof(short) );
    bufin[1]= (short*) malloc( nb_samples * sizeof(short) );
    
    /* 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) {
        buftmp2[0] = bufin[0];
        buftmp3[0] = output;

        mono_to_stereo(output, buftmp3[0], nb_samples1);
    } else if (s->output_channels == 2) {
        stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
    }

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

    free(bufout[0]);
    free(bufout[1]);
    return nb_samples1;
}

void audio_resample_close(ReSampleContext *s)
{