32 #if FF_API_AVCODEC_RESAMPLE
34 #ifndef CONFIG_RESAMPLE_HP
35 #define FILTER_SHIFT 15
40 #define FELEM_MAX INT16_MAX
41 #define FELEM_MIN INT16_MIN
43 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
44 #define FILTER_SHIFT 30
47 #define FELEM2 int64_t
48 #define FELEML int64_t
49 #define FELEM_MAX INT32_MAX
50 #define FELEM_MIN INT32_MIN
51 #define WINDOW_TYPE 12
53 #define FILTER_SHIFT 0
58 #define WINDOW_TYPE 24
87 for(i=1; v != lastv; i++){
106 const int center= (tap_count-1)/2;
115 for(ph=0;ph<phase_count;ph++) {
117 for(i=0;i<tap_count;i++) {
118 x =
M_PI * ((double)(i - center) - (double)ph / phase_count) *
factor;
124 x = fabs(((
double)(i - center) - (
double)ph / phase_count) * factor);
125 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
126 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
129 w = 2.0*x / (factor*tap_count) +
M_PI;
130 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
133 w = 2.0*x / (factor*tap_count*
M_PI);
143 for(i=0;i<tap_count;i++) {
144 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
145 filter[ph * tap_count + i] = tab[i] / norm;
155 double sine[
LEN + tap_count];
156 double filtered[
LEN];
157 double maxff=-2, minff=2, maxsf=-2, minsf=2;
158 for(i=0; i<
LEN; i++){
159 double ss=0, sf=0, ff=0;
160 for(j=0; j<LEN+tap_count; j++)
161 sine[j]= cos(i*j*
M_PI/LEN);
162 for(j=0; j<
LEN; j++){
165 for(k=0; k<tap_count; k++)
166 sum += filter[ph * tap_count + k] * sine[k+j];
168 ss+= sine[j + center] * sine[j + center];
169 ff+= filtered[j] * filtered[j];
170 sf+= sine[j + center] * filtered[j];
175 maxff=
FFMAX(maxff, ff);
176 minff=
FFMIN(minff, ff);
177 maxsf=
FFMAX(maxsf, sf);
178 minsf=
FFMIN(minsf, sf);
180 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);
194 double factor=
FFMIN(out_rate * cutoff / in_rate, 1.0);
195 int phase_count= 1<<phase_shift;
246 int64_t index2= ((int64_t)index)<<32;
250 for(dst_index=0; dst_index < dst_size; dst_index++){
251 dst[dst_index] = src[index2>>32];
254 index += dst_index * dst_incr;
255 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->
src_incr;
256 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->
src_incr;
258 for(dst_index=0; dst_index < dst_size; dst_index++){
263 if(sample_index < 0){
265 val += src[
FFABS(sample_index + i) % src_size] * filter[i];
271 val += src[sample_index + i] * (
FELEM2)filter[i];
277 val += src[sample_index + i] * (
FELEM2)filter[i];
281 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
282 dst[dst_index] = av_clip_int16(
lrintf(val));
285 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
288 frac += dst_incr_frac;
295 if(dst_index + 1 == compensation_distance){
296 compensation_distance= 0;
305 if(compensation_distance){
306 compensation_distance -= dst_index;
const char const char void * val
static int linear(InterplayACMContext *s, unsigned ind, unsigned col)
int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx)
Resample an array of samples using a previously configured context.
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
static void filter(int16_t *output, ptrdiff_t out_stride, int16_t *low, ptrdiff_t low_stride, int16_t *high, ptrdiff_t high_stride, int len, uint8_t clip)
void av_resample_close(AVResampleContext *c)
static double bessel(double x)
0th order modified bessel function of the first kind.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
simple assert() macros that are a bit more flexible than ISO C assert().
int compensation_distance
int av_reduce(int *dst_num, int *dst_den, int64_t num, int64_t den, int64_t max)
Reduce a fraction.
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance)
Compensate samplerate/timestamp drift.
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type)
Build a polyphase filterbank.
Libavcodec external API header.
Describe the class of an AVClass context structure.
static const int factor[16]
AVResampleContext * av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff)
Initialize an audio resampler.
common internal and external API header
static const struct twinvq_data tab
static void * av_mallocz_array(size_t nmemb, size_t size)
#define av_malloc_array(a, b)
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...