comparison resample.c @ 396:fce0a2520551 libavcodec

removed useless header includes - use av memory functions

395:80518daaab05

 * 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)

        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)
{
    free(s);
}

396:fce0a2520551

 * 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;

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);

    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)

        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;

        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);
}