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];
    short *output_bak = NULL;
    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;
    }

    if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
        int istride[1] = { s->sample_size[0] };
        int ostride[1] = { 2 };
        const void *ibuf[1] = { input };
        void       *obuf[1];
        unsigned input_size = nb_samples*s->input_channels*2;

        if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
            av_free(s->buffer[0]);
            s->buffer_size[0] = input_size;
            s->buffer[0] = av_malloc(s->buffer_size[0]);
            if (!s->buffer[0]) {
                av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
                return 0;
            }
        }

        obuf[0] = s->buffer[0];

        if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
                             ibuf, istride, nb_samples*s->input_channels) < 0) {
            av_log(s, AV_LOG_ERROR, "Audio sample format conversion failed\n");
            return 0;
        }

        input  = s->buffer[0];
    }

    lenout= 4*nb_samples * s->ratio + 16;

    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
        output_bak = output;

        if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
            av_free(s->buffer[1]);
            s->buffer_size[1] = lenout;
            s->buffer[1] = av_malloc(s->buffer_size[1]);
            if (!s->buffer[1]) {
                av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
                return 0;
            }
        }

        output = s->buffer[1];
    }

    /* XXX: move those malloc to resample init code */
    for(i=0; i<s->filter_channels; i++){
        bufin[i]= av_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 */
    bufout[0]= av_malloc( lenout * sizeof(short) );
    bufout[1]= av_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]= av_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);
    }

    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
        int istride[1] = { 2 };
        int ostride[1] = { s->sample_size[1] };
        const void *ibuf[1] = { output };
        void       *obuf[1] = { output_bak };

        if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
                             ibuf, istride, nb_samples1*s->output_channels) < 0) {
            av_log(s, AV_LOG_ERROR, "Audio sample format convertion failed\n");
            return 0;
        }
    }

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

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