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ffmpeg/libswresample/resample.c

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/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
* bessel function: Copyright (c) 2006 Xiaogang Zhang
*
* 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>
*/
#include "libavutil/avassert.h"
#include "libavutil/cpu.h"
#include "resample.h"
/**
* builds a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param filter_type filter type
* @param kaiser_beta kaiser window beta
* @return 0 on success, negative on error
*/
static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
int filter_type, double kaiser_beta){
int ph, i;
int ph_nb = phase_count % 2 ? phase_count : phase_count / 2 + 1;
double x, y, w, t, s;
double *tab = av_malloc_array(tap_count+1, sizeof(*tab));
double *sin_lut = av_malloc_array(ph_nb, sizeof(*sin_lut));
const int center= (tap_count-1)/2;
double norm = 0;
int ret = AVERROR(ENOMEM);
if (!tab || !sin_lut)
goto fail;
av_assert0(tap_count == 1 || tap_count % 2 == 0);
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
if (factor == 1.0) {
for (ph = 0; ph < ph_nb; ph++)
sin_lut[ph] = sin(M_PI * ph / phase_count) * (center & 1 ? 1 : -1);
}
for(ph = 0; ph < ph_nb; ph++) {
s = sin_lut[ph];
for(i=0;i<tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else if (factor == 1.0)
y = s / x;
else
y = sin(x) / x;
switch(filter_type){
case SWR_FILTER_TYPE_CUBIC:{
const float d= -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
break;}
case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
w = 2.0*x / (factor*tap_count);
t = -cos(w);
y *= 0.3635819 - 0.4891775 * t + 0.1365995 * (2*t*t-1) - 0.0106411 * (4*t*t*t - 3*t);
break;
case SWR_FILTER_TYPE_KAISER:
w = 2.0*x / (factor*tap_count*M_PI);
y *= av_bessel_i0(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
break;
default:
av_assert0(0);
}
tab[i] = y;
s = -s;
if (!ph)
norm += y;
}
/* normalize so that an uniform color remains the same */
switch(c->format){
case AV_SAMPLE_FMT_S16P:
for(i=0;i<tap_count;i++)
((int16_t*)filter)[ph * alloc + i] = av_clip_int16(lrintf(tab[i] * scale / norm));
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i];
break;
case AV_SAMPLE_FMT_S32P:
for(i=0;i<tap_count;i++)
((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i];
break;
case AV_SAMPLE_FMT_FLTP:
for(i=0;i<tap_count;i++)
((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i];
break;
case AV_SAMPLE_FMT_DBLP:
for(i=0;i<tap_count;i++)
((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
if (phase_count % 2) break;
for (i = 0; i < tap_count; i++)
((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i];
break;
}
}
#if 0
{
#define LEN 1024
int j,k;
double sine[LEN + tap_count];
double filtered[LEN];
double maxff=-2, minff=2, maxsf=-2, minsf=2;
for(i=0; i<LEN; i++){
double ss=0, sf=0, ff=0;
for(j=0; j<LEN+tap_count; j++)
sine[j]= cos(i*j*M_PI/LEN);
for(j=0; j<LEN; j++){
double sum=0;
ph=0;
for(k=0; k<tap_count; k++)
sum += filter[ph * tap_count + k] * sine[k+j];
filtered[j]= sum / (1<<FILTER_SHIFT);
ss+= sine[j + center] * sine[j + center];
ff+= filtered[j] * filtered[j];
sf+= sine[j + center] * filtered[j];
}
ss= sqrt(2*ss/LEN);
ff= sqrt(2*ff/LEN);
sf= 2*sf/LEN;
maxff= FFMAX(maxff, ff);
minff= FFMIN(minff, ff);
maxsf= FFMAX(maxsf, sf);
minsf= FFMIN(minsf, sf);
if(i%11==0){
av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
minff=minsf= 2;
maxff=maxsf= -2;
}
}
}
#endif
ret = 0;
fail:
av_free(tab);
av_free(sin_lut);
return ret;
}
static void resample_free(ResampleContext **cc){
ResampleContext *c = *cc;
if(!c)
return;
av_freep(&c->filter_bank);
av_freep(cc);
}
static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta,
double precision, int cheby, int exact_rational)
{
double cutoff = cutoff0? cutoff0 : 0.97;
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
int phase_count_compensation = phase_count;
int filter_length = FFMAX((int)ceil(filter_size/factor), 1);
if (filter_length > 1)
filter_length = FFALIGN(filter_length, 2);
if (exact_rational) {
int phase_count_exact, phase_count_exact_den;
av_reduce(&phase_count_exact, &phase_count_exact_den, out_rate, in_rate, INT_MAX);
if (phase_count_exact <= phase_count) {
phase_count_compensation = phase_count_exact * (phase_count / phase_count_exact);
phase_count = phase_count_exact;
}
}
if (!c || c->phase_count != phase_count || c->linear!=linear || c->factor != factor
|| c->filter_length != filter_length || c->format != format
|| c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
resample_free(&c);
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
c->format= format;
c->felem_size= av_get_bytes_per_sample(c->format);
switch(c->format){
case AV_SAMPLE_FMT_S16P:
c->filter_shift = 15;
break;
case AV_SAMPLE_FMT_S32P:
c->filter_shift = 30;
break;
case AV_SAMPLE_FMT_FLTP:
case AV_SAMPLE_FMT_DBLP:
c->filter_shift = 0;
break;
default:
av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
av_assert0(0);
}
if (filter_size/factor > INT32_MAX/256) {
av_log(NULL, AV_LOG_ERROR, "Filter length too large\n");
goto error;
}
c->phase_count = phase_count;
c->linear = linear;
c->factor = factor;
c->filter_length = filter_length;
c->filter_alloc = FFALIGN(c->filter_length, 8);
c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size);
c->filter_type = filter_type;
c->kaiser_beta = kaiser_beta;
c->phase_count_compensation = phase_count_compensation;
if (!c->filter_bank)
goto error;
if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta))
goto error;
memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
}
c->compensation_distance= 0;
if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
goto error;
while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
c->dst_incr *= 2;
c->src_incr *= 2;
}
c->ideal_dst_incr = c->dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
c->index= -phase_count*((c->filter_length-1)/2);
c->frac= 0;
swri_resample_dsp_init(c);
return c;
error:
av_freep(&c->filter_bank);
av_free(c);
return NULL;
}
static int rebuild_filter_bank_with_compensation(ResampleContext *c)
{
uint8_t *new_filter_bank;
int new_src_incr, new_dst_incr;
int phase_count = c->phase_count_compensation;
int ret;
if (phase_count == c->phase_count)
return 0;
av_assert0(!c->frac && !c->dst_incr_mod);
new_filter_bank = av_calloc(c->filter_alloc, (phase_count + 1) * c->felem_size);
if (!new_filter_bank)
return AVERROR(ENOMEM);
ret = build_filter(c, new_filter_bank, c->factor, c->filter_length, c->filter_alloc,
phase_count, 1 << c->filter_shift, c->filter_type, c->kaiser_beta);
if (ret < 0) {
av_freep(&new_filter_bank);
return ret;
}
memcpy(new_filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, new_filter_bank, (c->filter_alloc-1)*c->felem_size);
memcpy(new_filter_bank + (c->filter_alloc*phase_count )*c->felem_size, new_filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
if (!av_reduce(&new_src_incr, &new_dst_incr, c->src_incr,
c->dst_incr * (int64_t)(phase_count/c->phase_count), INT32_MAX/2))
{
av_freep(&new_filter_bank);
return AVERROR(EINVAL);
}
c->src_incr = new_src_incr;
c->dst_incr = new_dst_incr;
while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
c->dst_incr *= 2;
c->src_incr *= 2;
}
c->ideal_dst_incr = c->dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
c->index *= phase_count / c->phase_count;
c->phase_count = phase_count;
av_freep(&c->filter_bank);
c->filter_bank = new_filter_bank;
return 0;
}
static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
int ret;
if (compensation_distance && sample_delta) {
ret = rebuild_filter_bank_with_compensation(c);
if (ret < 0)
return ret;
}
c->compensation_distance= compensation_distance;
if (compensation_distance)
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
else
c->dst_incr = c->ideal_dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
return 0;
}
static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
int i;
int64_t max_src_size = (INT64_MAX/2 / c->phase_count) / c->src_incr;
if (c->compensation_distance)
dst_size = FFMIN(dst_size, c->compensation_distance);
src_size = FFMIN(src_size, max_src_size);
*consumed = 0;
if (c->filter_length == 1 && c->phase_count == 1) {
int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*c->index;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
int new_size = (src_size * (int64_t)c->src_incr - c->frac + c->dst_incr - 1) / c->dst_incr;
dst_size = FFMAX(FFMIN(dst_size, new_size), 0);
if (dst_size > 0) {
for (i = 0; i < dst->ch_count; i++) {
c->dsp.resample_one(dst->ch[i], src->ch[i], dst_size, index2, incr);
if (i+1 == dst->ch_count) {
c->index += dst_size * c->dst_incr_div;
c->index += (c->frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr;
av_assert2(c->index >= 0);
*consumed = c->index;
c->frac = (c->frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr;
c->index = 0;
}
}
}
} else {
int64_t end_index = (1LL + src_size - c->filter_length) * c->phase_count;
int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;
int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
int (*resample_func)(struct ResampleContext *c, void *dst,
const void *src, int n, int update_ctx);
dst_size = FFMAX(FFMIN(dst_size, delta_n), 0);
if (dst_size > 0) {
/* resample_linear and resample_common should have same behavior
* when frac and dst_incr_mod are zero */
resample_func = (c->linear && (c->frac || c->dst_incr_mod)) ?
c->dsp.resample_linear : c->dsp.resample_common;
for (i = 0; i < dst->ch_count; i++)
*consumed = resample_func(c, dst->ch[i], src->ch[i], dst_size, i+1 == dst->ch_count);
}
}
if (c->compensation_distance) {
c->compensation_distance -= dst_size;
if (!c->compensation_distance) {
c->dst_incr = c->ideal_dst_incr;
c->dst_incr_div = c->dst_incr / c->src_incr;
c->dst_incr_mod = c->dst_incr % c->src_incr;
}
}
return dst_size;
}
static int64_t get_delay(struct SwrContext *s, int64_t base){
ResampleContext *c = s->resample;
int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
num *= c->phase_count;
num -= c->index;
num *= c->src_incr;
num -= c->frac;
return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr * c->phase_count);
}
static int64_t get_out_samples(struct SwrContext *s, int in_samples) {
ResampleContext *c = s->resample;
// The + 2 are added to allow implementations to be slightly inaccurate, they should not be needed currently.
// They also make it easier to proof that changes and optimizations do not
// break the upper bound.
int64_t num = s->in_buffer_count + 2LL + in_samples;
num *= c->phase_count;
num -= c->index;
num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) * c->phase_count, AV_ROUND_UP) + 2;
if (c->compensation_distance) {
if (num > INT_MAX)
return AVERROR(EINVAL);
num = FFMAX(num, (num * c->ideal_dst_incr - 1) / c->dst_incr + 1);
}
return num;
}
static int resample_flush(struct SwrContext *s) {
ResampleContext *c = s->resample;
AudioData *a= &s->in_buffer;
int i, j, ret;
int reflection = (FFMIN(s->in_buffer_count, c->filter_length) + 1) / 2;
if((ret = swri_realloc_audio(a, s->in_buffer_index + s->in_buffer_count + reflection)) < 0)
return ret;
av_assert0(a->planar);
for(i=0; i<a->ch_count; i++){
for(j=0; j<reflection; j++){
memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
}
}
s->in_buffer_count += reflection;
return 0;
}
// in fact the whole handle multiple ridiculously small buffers might need more thinking...
static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src,
int in_count, int *out_idx, int *out_sz)
{
int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;
if (c->index >= 0)
return 0;
if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)
return res;
// copy
for (n = *out_sz; n < num; n++) {
for (ch = 0; ch < src->ch_count; ch++) {
memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);
}
}
// if not enough data is in, return and wait for more
if (num < c->filter_length + 1) {
*out_sz = num;
*out_idx = c->filter_length;
return INT_MAX;
}
// else invert
for (n = 1; n <= c->filter_length; n++) {
for (ch = 0; ch < src->ch_count; ch++) {
memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),
dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
c->felem_size);
}
}
res = num - *out_sz;
*out_idx = c->filter_length;
while (c->index < 0) {
--*out_idx;
c->index += c->phase_count;
}
*out_sz = FFMAX(*out_sz + c->filter_length,
1 + c->filter_length * 2) - *out_idx;
return FFMAX(res, 0);
}
struct Resampler const swri_resampler={
resample_init,
resample_free,
multiple_resample,
resample_flush,
set_compensation,
get_delay,
invert_initial_buffer,
get_out_samples,
};
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