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ffmpeg.wasm-core
Commit 7f1ae79d authored by Michael Niedermayer's avatar Michael Niedermayer
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swr: support float & int32 in the resampler 

Signed-off-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parent 605bcf61
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Showing with 220 additions and 129 deletions
libswresample/resample.c
View file @ 7f1ae79d
......@@ -26,39 +26,16 @@
*/
#include "libavutil/log.h"
#include "libavutil/avassert.h"
#include "swresample_internal.h"
#ifndef CONFIG_RESAMPLE_HP
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define WINDOW_TYPE 9
#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define WINDOW_TYPE 12
#else
#define FILTER_SHIFT 0
#define FELEM double
#define FELEM2 double
#define FELEML double
#define WINDOW_TYPE 24
#endif
typedef struct ResampleContext {
const AVClass *av_class;
FELEM *filter_bank;
uint8_t *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
......@@ -70,6 +47,9 @@ typedef struct ResampleContext {
int phase_mask;
int linear;
double factor;
enum AVSampleFormat format;
int felem_size;
int filter_shift;
} ResampleContext;
/**
......@@ -109,7 +89,7 @@ static double bessel(double x){
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
* @return 0 on success, negative on error
*/
static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i;
double x, y, w;
double *tab = av_malloc(tap_count * sizeof(*tab));
......@@ -150,12 +130,19 @@ static int build_filter(FELEM *filter, double factor, int tap_count, int phase_c
}
/* normalize so that an uniform color remains the same */
for(i=0;i<tap_count;i++) {
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
filter[ph * tap_count + i] = tab[i] / norm;
#else
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
#endif
switch(c->format){
case AV_SAMPLE_FMT_S16:
for(i=0;i<tap_count;i++)
((int16_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);
break;
case AV_SAMPLE_FMT_S32:
for(i=0;i<tap_count;i++)
((int32_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX);
break;
case AV_SAMPLE_FMT_FLT:
for(i=0;i<tap_count;i++)
((float*)filter)[ph * tap_count + i] = tab[i] * scale / norm;
break;
}
}
#if 0
......@@ -199,28 +186,48 @@ static int build_filter(FELEM *filter, double factor, int tap_count, int phase_c
return 0;
}
ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
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){
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
|| c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
|| c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format) {
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
c->format= format;
switch(c->format){
case AV_SAMPLE_FMT_S16:
c->felem_size = 2;
c->filter_shift = 15;
break;
case AV_SAMPLE_FMT_S32:
c->felem_size = 4;
c->filter_shift = 30;
break;
case AV_SAMPLE_FMT_FLT:
c->felem_size = 4;
c->filter_shift = 0;
break;
default:
av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
return NULL;
}
c->phase_shift = phase_shift;
c->phase_mask = phase_count - 1;
c->linear = linear;
c->factor = factor;
c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*c->felem_size);
if (!c->filter_bank)
goto error;
if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, phase_count, 1<<c->filter_shift, WINDOW_TYPE))
goto error;
memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
memcpy(c->filter_bank + (c->filter_length*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_length-1)*c->felem_size);
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);
}
c->compensation_distance= 0;
......@@ -268,100 +275,69 @@ int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensatio
return 0;
}
int swri_resample(ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int compensation_distance= c->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;
}
index += dst_index * dst_incr;
index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
}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 + c->filter_length > src_size || -sample_index >= src_size){
break;
}else if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[FFABS(sample_index + i)] * filter[i];
}else if(c->linear){
FELEM2 v2=0;
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
}
val+=(v2-val)*(FELEML)frac / c->src_incr;
}else{
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
}
}
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
#endif
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
if(dst_index + 1 == compensation_distance){
compensation_distance= 0;
dst_incr_frac= c->ideal_dst_incr % c->src_incr;
dst_incr= c->ideal_dst_incr / c->src_incr;
}
}
}
*consumed= FFMAX(index, 0) >> c->phase_shift;
if(index>=0) index &= c->phase_mask;
if(compensation_distance){
compensation_distance -= dst_index;
assert(compensation_distance > 0);
}
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
#define RENAME(N) N ## _int16
#define FILTER_SHIFT 15
#define DELEM int16_t
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\
d = (unsigned)(v + 32768) > 65535 ? (v>>31) ^ 32767 : v
#include "resample_template.c"
#undef RENAME
#undef FELEM
#undef FELEM2
#undef DELEM
#undef FELEML
#undef OUT
#undef FELEM_MIN
#undef FELEM_MAX
#undef FILTER_SHIFT
#define RENAME(N) N ## _int32
#define FILTER_SHIFT 30
#define DELEM int32_t
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\
d = (uint64_t)(v + 0x80000000) > 0xFFFFFFFF ? (v>>63) ^ 0x7FFFFFFF : v
#include "resample_template.c"
#undef RENAME
#undef FELEM
#undef FELEM2
#undef DELEM
#undef FELEML
#undef OUT
#undef FELEM_MIN
#undef FELEM_MAX
#undef FILTER_SHIFT
#define RENAME(N) N ## _float
#define FILTER_SHIFT 0
#define DELEM float
#define FELEM float
#define FELEM2 float
#define FELEML float
#define OUT(d, v) d = v
#include "resample_template.c"
return dst_index;
}
int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
int i, ret= -1;
for(i=0; i<dst->ch_count; i++){
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);
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);
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);
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);
}
return ret;
......
libswresample/resample_template.c 0 → 100644
View file @ 7f1ae79d
/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int compensation_distance= c->compensation_distance;
av_assert1(c->filter_shift == FILTER_SHIFT);
av_assert1(c->felem_size == sizeof(FELEM));
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;
}
index += dst_index * dst_incr;
index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
}else{
for(dst_index=0; dst_index < dst_size; dst_index++){
FELEM *filter= ((FELEM*)c->filter_bank) + c->filter_length*(index & c->phase_mask);
int sample_index= index >> c->phase_shift;
FELEM2 val=0;
if(sample_index + c->filter_length > src_size || -sample_index >= src_size){
break;
}else if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[FFABS(sample_index + i)] * filter[i];
}else if(c->linear){
FELEM2 v2=0;
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
}
val+=(v2-val)*(FELEML)frac / c->src_incr;
}else{
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
}
}
OUT(dst[dst_index], val);
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
if(dst_index + 1 == compensation_distance){
compensation_distance= 0;
dst_incr_frac= c->ideal_dst_incr % c->src_incr;
dst_incr= c->ideal_dst_incr / c->src_incr;
}
}
}
*consumed= FFMAX(index, 0) >> c->phase_shift;
if(index>=0) index &= c->phase_mask;
if(compensation_distance){
compensation_distance -= dst_index;
assert(compensation_distance > 0);
}
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;
}
libswresample/swresample.c
View file @ 7f1ae79d
......@@ -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){
......
libswresample/swresample_internal.h
View file @ 7f1ae79d
......@@ -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);
......
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