/* * audio resampling * Copyright (c) 2004-2012 Michael Niedermayer * 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 */ #include "libavutil/avassert.h" #include "libavutil/mem.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;iformat){ case AV_SAMPLE_FMT_S16P: for(i=0;ifilter_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< 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<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 + 1; int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr + 1; 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; ich_count; i++){ for(j=0; jch[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, };