mirror of
https://github.com/FFmpeg/FFmpeg.git
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swr: support float & int32 in the resampler
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
parent
605bcf6101
commit
7f1ae79d38
@ -26,39 +26,16 @@
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*/
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#include "libavutil/log.h"
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#include "libavutil/avassert.h"
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#include "swresample_internal.h"
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#ifndef CONFIG_RESAMPLE_HP
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#define FILTER_SHIFT 15
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#define FELEM int16_t
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#define FELEM2 int32_t
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#define FELEML int64_t
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#define FELEM_MAX INT16_MAX
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#define FELEM_MIN INT16_MIN
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#define WINDOW_TYPE 9
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#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
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#define FILTER_SHIFT 30
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#define FELEM int32_t
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#define FELEM2 int64_t
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#define FELEML int64_t
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#define FELEM_MAX INT32_MAX
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#define FELEM_MIN INT32_MIN
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#define WINDOW_TYPE 12
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#else
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#define FILTER_SHIFT 0
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#define FELEM double
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#define FELEM2 double
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#define FELEML double
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#define WINDOW_TYPE 24
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#endif
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typedef struct ResampleContext {
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const AVClass *av_class;
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FELEM *filter_bank;
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uint8_t *filter_bank;
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int filter_length;
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int ideal_dst_incr;
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int dst_incr;
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@ -70,6 +47,9 @@ typedef struct ResampleContext {
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int phase_mask;
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int linear;
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double factor;
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enum AVSampleFormat format;
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int felem_size;
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int filter_shift;
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} ResampleContext;
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/**
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@ -109,7 +89,7 @@ static double bessel(double x){
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* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
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* @return 0 on success, negative on error
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*/
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static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
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static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int phase_count, int scale, int type){
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int ph, i;
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double x, y, w;
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double *tab = av_malloc(tap_count * sizeof(*tab));
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@ -150,12 +130,19 @@ static int build_filter(FELEM *filter, double factor, int tap_count, int phase_c
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}
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/* normalize so that an uniform color remains the same */
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for(i=0;i<tap_count;i++) {
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#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
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filter[ph * tap_count + i] = tab[i] / norm;
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#else
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filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
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#endif
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switch(c->format){
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case AV_SAMPLE_FMT_S16:
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for(i=0;i<tap_count;i++)
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((int16_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);
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break;
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case AV_SAMPLE_FMT_S32:
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for(i=0;i<tap_count;i++)
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((int32_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX);
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break;
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case AV_SAMPLE_FMT_FLT:
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for(i=0;i<tap_count;i++)
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((float*)filter)[ph * tap_count + i] = tab[i] * scale / norm;
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break;
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}
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}
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#if 0
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@ -199,28 +186,48 @@ static int build_filter(FELEM *filter, double factor, int tap_count, int phase_c
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return 0;
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}
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ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
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ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat format){
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double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
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int phase_count= 1<<phase_shift;
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if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
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|| c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
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|| c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format) {
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c = av_mallocz(sizeof(*c));
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if (!c)
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return NULL;
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c->format= format;
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switch(c->format){
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case AV_SAMPLE_FMT_S16:
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c->felem_size = 2;
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c->filter_shift = 15;
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break;
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case AV_SAMPLE_FMT_S32:
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c->felem_size = 4;
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c->filter_shift = 30;
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break;
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case AV_SAMPLE_FMT_FLT:
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c->felem_size = 4;
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c->filter_shift = 0;
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break;
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default:
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av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
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return NULL;
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}
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c->phase_shift = phase_shift;
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c->phase_mask = phase_count - 1;
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c->linear = linear;
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c->factor = factor;
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c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
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c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
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c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*c->felem_size);
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if (!c->filter_bank)
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goto error;
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if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
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if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, phase_count, 1<<c->filter_shift, WINDOW_TYPE))
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goto error;
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memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
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c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
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memcpy(c->filter_bank + (c->filter_length*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_length-1)*c->felem_size);
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memcpy(c->filter_bank + (c->filter_length*phase_count )*c->felem_size, c->filter_bank + (c->filter_length - 1)*c->felem_size, c->felem_size);
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}
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c->compensation_distance= 0;
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@ -268,100 +275,69 @@ int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensatio
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return 0;
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}
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int swri_resample(ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx){
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int dst_index, i;
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int index= c->index;
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int frac= c->frac;
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int dst_incr_frac= c->dst_incr % c->src_incr;
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int dst_incr= c->dst_incr / c->src_incr;
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int compensation_distance= c->compensation_distance;
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#define RENAME(N) N ## _int16
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#define FILTER_SHIFT 15
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#define DELEM int16_t
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#define FELEM int16_t
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#define FELEM2 int32_t
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#define FELEML int64_t
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#define FELEM_MAX INT16_MAX
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#define FELEM_MIN INT16_MIN
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#define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\
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d = (unsigned)(v + 32768) > 65535 ? (v>>31) ^ 32767 : v
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#include "resample_template.c"
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if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
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int64_t index2= ((int64_t)index)<<32;
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int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
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dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
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#undef RENAME
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#undef FELEM
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#undef FELEM2
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#undef DELEM
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#undef FELEML
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#undef OUT
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#undef FELEM_MIN
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#undef FELEM_MAX
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#undef FILTER_SHIFT
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for(dst_index=0; dst_index < dst_size; dst_index++){
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dst[dst_index] = src[index2>>32];
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index2 += incr;
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}
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index += dst_index * dst_incr;
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index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
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frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
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}else{
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for(dst_index=0; dst_index < dst_size; dst_index++){
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FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
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int sample_index= index >> c->phase_shift;
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FELEM2 val=0;
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if(sample_index + c->filter_length > src_size || -sample_index >= src_size){
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break;
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}else if(sample_index < 0){
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for(i=0; i<c->filter_length; i++)
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val += src[FFABS(sample_index + i)] * filter[i];
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}else if(c->linear){
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FELEM2 v2=0;
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for(i=0; i<c->filter_length; i++){
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val += src[sample_index + i] * (FELEM2)filter[i];
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v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
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}
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val+=(v2-val)*(FELEML)frac / c->src_incr;
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}else{
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for(i=0; i<c->filter_length; i++){
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val += src[sample_index + i] * (FELEM2)filter[i];
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}
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}
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#define RENAME(N) N ## _int32
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#define FILTER_SHIFT 30
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#define DELEM int32_t
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#define FELEM int32_t
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#define FELEM2 int64_t
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#define FELEML int64_t
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#define FELEM_MAX INT32_MAX
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#define FELEM_MIN INT32_MIN
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#define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\
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d = (uint64_t)(v + 0x80000000) > 0xFFFFFFFF ? (v>>63) ^ 0x7FFFFFFF : v
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#include "resample_template.c"
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#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
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dst[dst_index] = av_clip_int16(lrintf(val));
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#else
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val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
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dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
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#endif
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#undef RENAME
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#undef FELEM
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#undef FELEM2
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#undef DELEM
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#undef FELEML
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#undef OUT
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#undef FELEM_MIN
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#undef FELEM_MAX
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#undef FILTER_SHIFT
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frac += dst_incr_frac;
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index += dst_incr;
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if(frac >= c->src_incr){
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frac -= c->src_incr;
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index++;
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}
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if(dst_index + 1 == compensation_distance){
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compensation_distance= 0;
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dst_incr_frac= c->ideal_dst_incr % c->src_incr;
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dst_incr= c->ideal_dst_incr / c->src_incr;
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}
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}
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}
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*consumed= FFMAX(index, 0) >> c->phase_shift;
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if(index>=0) index &= c->phase_mask;
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#define RENAME(N) N ## _float
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#define FILTER_SHIFT 0
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#define DELEM float
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#define FELEM float
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#define FELEM2 float
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#define FELEML float
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#define OUT(d, v) d = v
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#include "resample_template.c"
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if(compensation_distance){
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compensation_distance -= dst_index;
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assert(compensation_distance > 0);
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}
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if(update_ctx){
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c->frac= frac;
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c->index= index;
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c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
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c->compensation_distance= compensation_distance;
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}
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#if 0
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if(update_ctx && !c->compensation_distance){
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#undef rand
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av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
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av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
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}
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#endif
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return dst_index;
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}
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int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
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int i, ret= -1;
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for(i=0; i<dst->ch_count; i++){
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ret= swri_resample(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_S16) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_S32) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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if(c->format == AV_SAMPLE_FMT_FLT) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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}
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return ret;
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libswresample/resample_template.c
Normal file
113
libswresample/resample_template.c
Normal file
@ -0,0 +1,113 @@
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/*
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* audio resampling
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* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* audio resampling
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* @author Michael Niedermayer <michaelni@gmx.at>
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*/
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int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int *consumed, int src_size, int dst_size, int update_ctx){
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int dst_index, i;
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int index= c->index;
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int frac= c->frac;
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int dst_incr_frac= c->dst_incr % c->src_incr;
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int dst_incr= c->dst_incr / c->src_incr;
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int compensation_distance= c->compensation_distance;
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av_assert1(c->filter_shift == FILTER_SHIFT);
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av_assert1(c->felem_size == sizeof(FELEM));
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if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
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int64_t index2= ((int64_t)index)<<32;
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int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
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dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
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for(dst_index=0; dst_index < dst_size; dst_index++){
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dst[dst_index] = src[index2>>32];
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index2 += incr;
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}
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index += dst_index * dst_incr;
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index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
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frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
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}else{
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for(dst_index=0; dst_index < dst_size; dst_index++){
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FELEM *filter= ((FELEM*)c->filter_bank) + c->filter_length*(index & c->phase_mask);
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int sample_index= index >> c->phase_shift;
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FELEM2 val=0;
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if(sample_index + c->filter_length > src_size || -sample_index >= src_size){
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break;
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}else if(sample_index < 0){
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for(i=0; i<c->filter_length; i++)
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val += src[FFABS(sample_index + i)] * filter[i];
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}else if(c->linear){
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FELEM2 v2=0;
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for(i=0; i<c->filter_length; i++){
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val += src[sample_index + i] * (FELEM2)filter[i];
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v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
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}
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val+=(v2-val)*(FELEML)frac / c->src_incr;
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}else{
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for(i=0; i<c->filter_length; i++){
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val += src[sample_index + i] * (FELEM2)filter[i];
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}
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}
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OUT(dst[dst_index], val);
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frac += dst_incr_frac;
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index += dst_incr;
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if(frac >= c->src_incr){
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frac -= c->src_incr;
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index++;
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}
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if(dst_index + 1 == compensation_distance){
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compensation_distance= 0;
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dst_incr_frac= c->ideal_dst_incr % c->src_incr;
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dst_incr= c->ideal_dst_incr / c->src_incr;
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}
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}
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}
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*consumed= FFMAX(index, 0) >> c->phase_shift;
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if(index>=0) index &= c->phase_mask;
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if(compensation_distance){
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compensation_distance -= dst_index;
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assert(compensation_distance > 0);
|
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}
|
||||
if(update_ctx){
|
||||
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;
|
||||
}
|
@ -190,7 +190,7 @@ int swr_init(struct SwrContext *s){
|
||||
|
||||
|
||||
if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
|
||||
s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
|
||||
s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8, s->int_sample_fmt);
|
||||
}else
|
||||
swri_resample_free(&s->resample);
|
||||
if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
|
||||
|
@ -78,11 +78,13 @@ struct SwrContext {
|
||||
/* TODO: callbacks for ASM optimizations */
|
||||
};
|
||||
|
||||
struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff);
|
||||
struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat);
|
||||
void swri_resample_free(struct ResampleContext **c);
|
||||
int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
|
||||
void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance);
|
||||
int swri_resample(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
|
||||
int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
|
||||
int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
|
||||
int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
|
||||
|
||||
int swri_rematrix_init(SwrContext *s);
|
||||
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
|
||||
|
Loading…
Reference in New Issue
Block a user