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Diffstat (limited to 'libswresample/resample.c')
| -rw-r--r-- | libswresample/resample.c | 617 |
1 files changed, 617 insertions, 0 deletions
diff --git a/libswresample/resample.c b/libswresample/resample.c new file mode 100644 index 0000000000..8f3428f512 --- /dev/null +++ b/libswresample/resample.c @@ -0,0 +1,617 @@ +/* + * 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; + 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 *= bessel(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 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) { + 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 void resample_free(ResampleContext **c){ + if(!*c) + return; + av_freep(&(*c)->filter_bank); + av_freep(c); +} + +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 && !c->compensation_distance); + + 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; + 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/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(need_emms) + emms_c(); + + 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) { + 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; + 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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