comparison resample2.c @ 2967:ef2149182f1c libavcodec

COSMETICS: Remove all trailing whitespace.

2966:564788471dd4

 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 *
 */
 
/**
 * @file resample2.c
 * audio resampling
 * @author Michael Niedermayer <[e-mail]>
 */

 */
double bessel(double x){
    double v=1;
    double t=1;
    int i;
    
    for(i=1; i<50; i++){
        t *= i;
        v += pow(x*x/4, i)/(t*t);
    }
    return v;

 */
AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
    AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
    int phase_count= 1<<phase_shift;
    
    c->phase_shift= phase_shift;
    c->phase_mask= phase_count-1;
    c->linear= linear;

    c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);

 * @param sample_delta number of output samples which should be output less
 *
 * example: av_resample_compensate(c, 10, 500)
 * here instead of 510 samples only 500 samples would be output
 *
 * note, due to rounding the actual compensation might be slightly different, 
 * especially if the compensation_distance is large and the in_rate used during init is small
 */
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
//    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
    c->compensation_distance= compensation_distance;

  if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
        int64_t index2= ((int64_t)index)<<32;
        int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
        dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
        
        for(dst_index=0; dst_index < dst_size; dst_index++){
            dst[dst_index] = src[index2>>32];
            index2 += incr;
        }
        frac += dst_index * dst_incr_frac;

  }else{
    for(dst_index=0; dst_index < dst_size; dst_index++){
        FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
        int sample_index= index >> c->phase_shift;
        FELEM2 val=0;
                
        if(sample_index < 0){
            for(i=0; i<c->filter_length; i++)
                val += src[ABS(sample_index + i) % src_size] * filter[i];
        }else if(sample_index + c->filter_length > src_size){
            break;

        c->frac= frac;
        c->index= index;
        c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
        c->compensation_distance= compensation_distance;
    }
#if 0    
    if(update_ctx && !c->compensation_distance){
#undef rand
        av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
    }
#endif
    
    return dst_index;
}

2967:ef2149182f1c

 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 *
 */

/**
 * @file resample2.c
 * audio resampling
 * @author Michael Niedermayer <[e-mail]>
 */

 */
double bessel(double x){
    double v=1;
    double t=1;
    int i;

    for(i=1; i<50; i++){
        t *= i;
        v += pow(x*x/4, i)/(t*t);
    }
    return v;

 */
AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
    AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
    int phase_count= 1<<phase_shift;

    c->phase_shift= phase_shift;
    c->phase_mask= phase_count-1;
    c->linear= linear;

    c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);

 * @param sample_delta number of output samples which should be output less
 *
 * example: av_resample_compensate(c, 10, 500)
 * here instead of 510 samples only 500 samples would be output
 *
 * note, due to rounding the actual compensation might be slightly different,
 * especially if the compensation_distance is large and the in_rate used during init is small
 */
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
//    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
    c->compensation_distance= compensation_distance;

  if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
        int64_t index2= ((int64_t)index)<<32;
        int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
        dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);

        for(dst_index=0; dst_index < dst_size; dst_index++){
            dst[dst_index] = src[index2>>32];
            index2 += incr;
        }
        frac += dst_index * dst_incr_frac;

  }else{
    for(dst_index=0; dst_index < dst_size; dst_index++){
        FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
        int sample_index= index >> c->phase_shift;
        FELEM2 val=0;

        if(sample_index < 0){
            for(i=0; i<c->filter_length; i++)
                val += src[ABS(sample_index + i) % src_size] * filter[i];
        }else if(sample_index + c->filter_length > src_size){
            break;

        c->frac= frac;
        c->index= index;
        c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
        c->compensation_distance= compensation_distance;
    }
#if 0
    if(update_ctx && !c->compensation_distance){
#undef rand
        av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
    }
#endif

    return dst_index;
}