| /* |
| * 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 "resample.h" |
| |
| static inline double eval_poly(const double *coeff, int size, double x) { |
| double sum = coeff[size-1]; |
| int i; |
| for (i = size-2; i >= 0; --i) { |
| sum *= x; |
| sum += coeff[i]; |
| } |
| return sum; |
| } |
| |
| /** |
| * 0th order modified bessel function of the first kind. |
| * Algorithm taken from the Boost project, source: |
| * https://searchcode.com/codesearch/view/14918379/ |
| * Use, modification and distribution are subject to the |
| * Boost Software License, Version 1.0 (see notice below). |
| * Boost Software License - Version 1.0 - August 17th, 2003 |
| Permission is hereby granted, free of charge, to any person or organization |
| obtaining a copy of the software and accompanying documentation covered by |
| this license (the "Software") to use, reproduce, display, distribute, |
| execute, and transmit the Software, and to prepare derivative works of the |
| Software, and to permit third-parties to whom the Software is furnished to |
| do so, all subject to the following: |
| |
| The copyright notices in the Software and this entire statement, including |
| the above license grant, this restriction and the following disclaimer, |
| must be included in all copies of the Software, in whole or in part, and |
| all derivative works of the Software, unless such copies or derivative |
| works are solely in the form of machine-executable object code generated by |
| a source language processor. |
| |
| THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
| IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
| FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT |
| SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE |
| FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE, |
| ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER |
| DEALINGS IN THE SOFTWARE. |
| */ |
| |
| static double bessel(double x) { |
| // Modified Bessel function of the first kind of order zero |
| // minimax rational approximations on intervals, see |
| // Blair and Edwards, Chalk River Report AECL-4928, 1974 |
| static const double p1[] = { |
| -2.2335582639474375249e+15, |
| -5.5050369673018427753e+14, |
| -3.2940087627407749166e+13, |
| -8.4925101247114157499e+11, |
| -1.1912746104985237192e+10, |
| -1.0313066708737980747e+08, |
| -5.9545626019847898221e+05, |
| -2.4125195876041896775e+03, |
| -7.0935347449210549190e+00, |
| -1.5453977791786851041e-02, |
| -2.5172644670688975051e-05, |
| -3.0517226450451067446e-08, |
| -2.6843448573468483278e-11, |
| -1.5982226675653184646e-14, |
| -5.2487866627945699800e-18, |
| }; |
| static const double q1[] = { |
| -2.2335582639474375245e+15, |
| 7.8858692566751002988e+12, |
| -1.2207067397808979846e+10, |
| 1.0377081058062166144e+07, |
| -4.8527560179962773045e+03, |
| 1.0, |
| }; |
| static const double p2[] = { |
| -2.2210262233306573296e-04, |
| 1.3067392038106924055e-02, |
| -4.4700805721174453923e-01, |
| 5.5674518371240761397e+00, |
| -2.3517945679239481621e+01, |
| 3.1611322818701131207e+01, |
| -9.6090021968656180000e+00, |
| }; |
| static const double q2[] = { |
| -5.5194330231005480228e-04, |
| 3.2547697594819615062e-02, |
| -1.1151759188741312645e+00, |
| 1.3982595353892851542e+01, |
| -6.0228002066743340583e+01, |
| 8.5539563258012929600e+01, |
| -3.1446690275135491500e+01, |
| 1.0, |
| }; |
| double y, r, factor; |
| if (x == 0) |
| return 1.0; |
| x = fabs(x); |
| if (x <= 15) { |
| y = x * x; |
| return eval_poly(p1, FF_ARRAY_ELEMS(p1), y) / eval_poly(q1, FF_ARRAY_ELEMS(q1), y); |
| } |
| else { |
| y = 1 / x - 1.0 / 15; |
| r = eval_poly(p2, FF_ARRAY_ELEMS(p2), y) / eval_poly(q2, FF_ARRAY_ELEMS(q2), y); |
| factor = exp(x) / sqrt(x); |
| return factor * r; |
| } |
| } |
| |
| /** |
| * 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; |
| double x, y, w, t, s; |
| double *tab = av_malloc_array(tap_count+1, sizeof(*tab)); |
| double *sin_lut = av_malloc_array(phase_count / 2 + 1, sizeof(*sin_lut)); |
| const int center= (tap_count-1)/2; |
| |
| if (!tab || !sin_lut) |
| goto fail; |
| |
| /* if upsampling, only need to interpolate, no filter */ |
| if (factor > 1.0) |
| factor = 1.0; |
| |
| av_assert0(phase_count == 1 || phase_count % 2 == 0); |
| |
| if (factor == 1.0) { |
| for (ph = 0; ph <= phase_count / 2; ph++) |
| sin_lut[ph] = sin(M_PI * ph / phase_count); |
| } |
| for(ph = 0; ph <= phase_count / 2; ph++) { |
| double norm = 0; |
| 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 *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0))); |
| break; |
| default: |
| av_assert0(0); |
| } |
| |
| tab[i] = y; |
| s = -s; |
| if (i < tap_count) |
| 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 (tap_count % 2 == 0) { |
| for (i = 0; i < tap_count; i++) |
| ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i]; |
| } |
| else { |
| for (i = 1; i <= tap_count; i++) |
| ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-i] = |
| av_clip_int16(lrintf(tab[i] * scale / (norm - tab[0] + tab[tap_count]))); |
| } |
| 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 (tap_count % 2 == 0) { |
| for (i = 0; i < tap_count; i++) |
| ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i]; |
| } |
| else { |
| for (i = 1; i <= tap_count; i++) |
| ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-i] = |
| av_clipl_int32(llrint(tab[i] * scale / (norm - tab[0] + tab[tap_count]))); |
| } |
| break; |
| case AV_SAMPLE_FMT_FLTP: |
| for(i=0;i<tap_count;i++) |
| ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm; |
| if (tap_count % 2 == 0) { |
| for (i = 0; i < tap_count; i++) |
| ((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i]; |
| } |
| else { |
| for (i = 1; i <= tap_count; i++) |
| ((float*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]); |
| } |
| break; |
| case AV_SAMPLE_FMT_DBLP: |
| for(i=0;i<tap_count;i++) |
| ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm; |
| if (tap_count % 2 == 0) { |
| for (i = 0; i < tap_count; i++) |
| ((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i]; |
| } |
| else { |
| for (i = 1; i <= tap_count; i++) |
| ((double*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]); |
| } |
| 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 |
| |
| fail: |
| av_free(tab); |
| av_free(sin_lut); |
| return 0; |
| } |
| |
| 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) |
| { |
| double cutoff = cutoff0? cutoff0 : 0.97; |
| 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->format != format |
| || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { |
| 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_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_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; |
| 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 void resample_free(ResampleContext **c){ |
| if(!*c) |
| return; |
| av_freep(&(*c)->filter_bank); |
| av_freep(c); |
| } |
| |
| static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){ |
| 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 swri_resample(ResampleContext *c, |
| uint8_t *dst, const uint8_t *src, int *consumed, |
| int src_size, int dst_size, int update_ctx) |
| { |
| if (c->filter_length == 1 && c->phase_shift == 0) { |
| int index= c->index; |
| int frac= c->frac; |
| int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index; |
| int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; |
| int new_size = (src_size * (int64_t)c->src_incr - frac + c->dst_incr - 1) / c->dst_incr; |
| |
| dst_size= FFMIN(dst_size, new_size); |
| c->dsp.resample_one(dst, src, dst_size, index2, incr); |
| |
| index += dst_size * c->dst_incr_div; |
| index += (frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr; |
| av_assert2(index >= 0); |
| *consumed= index; |
| if (update_ctx) { |
| c->frac = (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_shift; |
| 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; |
| |
| dst_size = FFMIN(dst_size, delta_n); |
| if (dst_size > 0) { |
| *consumed = c->dsp.resample(c, dst, src, dst_size, update_ctx); |
| } else { |
| *consumed = 0; |
| } |
| } |
| |
| return dst_size; |
| } |
| |
| static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ |
| int i, ret= -1; |
| int av_unused mm_flags = av_get_cpu_flags(); |
| int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 && |
| (mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2; |
| int64_t max_src_size = (INT64_MAX >> (c->phase_shift+1)) / c->src_incr; |
| |
| if (c->compensation_distance) |
| dst_size = FFMIN(dst_size, c->compensation_distance); |
| src_size = FFMIN(src_size, max_src_size); |
| |
| for(i=0; i<dst->ch_count; i++){ |
| ret= swri_resample(c, dst->ch[i], src->ch[i], |
| consumed, src_size, dst_size, i+1==dst->ch_count); |
| } |
| if(need_emms) |
| emms_c(); |
| |
| if (c->compensation_distance) { |
| c->compensation_distance -= ret; |
| 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 ret; |
| } |
| |
| 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 *= 1 << c->phase_shift; |
| 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_shift); |
| } |
| |
| 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 *= 1 << c->phase_shift; |
| num -= c->index; |
| num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) << c->phase_shift, 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) { |
| AudioData *a= &s->in_buffer; |
| int i, j, ret; |
| if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0) |
| return ret; |
| av_assert0(a->planar); |
| for(i=0; i<a->ch_count; i++){ |
| for(j=0; j<s->in_buffer_count; 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 += (s->in_buffer_count+1)/2; |
| 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 + (c->index >> c->phase_shift); |
| *out_sz = FFMAX(*out_sz + c->filter_length, |
| 1 + c->filter_length * 2) - *out_idx; |
| c->index &= c->phase_mask; |
| |
| 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, |
| }; |
