/* * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> * * 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 */ #include "libavutil/common.h" #include "libavutil/libm.h" #include "libavutil/log.h" #include "internal.h" #include "resample.h" #include "audio_data.h" /* double template */ #define CONFIG_RESAMPLE_DBL #include "resample_template.c" #undef CONFIG_RESAMPLE_DBL /* float template */ #define CONFIG_RESAMPLE_FLT #include "resample_template.c" #undef CONFIG_RESAMPLE_FLT /* s32 template */ #define CONFIG_RESAMPLE_S32 #include "resample_template.c" #undef CONFIG_RESAMPLE_S32 /* s16 template */ #include "resample_template.c" /* 0th order modified Bessel function of the first kind. */ static double bessel(double x) { double v = 1; double lastv = 0; double t = 1; int i; x = x * x / 4; for (i = 1; v != lastv; i++) { lastv = v; t *= x / (i * i); v += t; } return v; } /* Build a polyphase filterbank. */ static int build_filter(ResampleContext *c, double factor) { int ph, i; double x, y, w; double *tab; int tap_count = c->filter_length; int phase_count = 1 << c->phase_shift; const int center = (tap_count - 1) / 2; tab = av_malloc(tap_count * sizeof(*tab)); if (!tab) return AVERROR(ENOMEM); for (ph = 0; ph < phase_count; ph++) { double norm = 0; 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 y = sin(x) / x; switch (c->filter_type) { case AV_RESAMPLE_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 AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL: w = 2.0 * x / (factor * tap_count) + M_PI; y *= 0.3635819 - 0.4891775 * cos( w) + 0.1365995 * cos(2 * w) - 0.0106411 * cos(3 * w); break; case AV_RESAMPLE_FILTER_TYPE_KAISER: w = 2.0 * x / (factor * tap_count * M_PI); y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0))); break; } tab[i] = y; norm += y; } /* normalize so that an uniform color remains the same */ for (i = 0; i < tap_count; i++) tab[i] = tab[i] / norm; c->set_filter(c->filter_bank, tab, ph, tap_count); } av_free(tab); return 0; } ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr) { ResampleContext *c; int out_rate = avr->out_sample_rate; int in_rate = avr->in_sample_rate; double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0); int phase_count = 1 << avr->phase_shift; int felem_size; if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P && avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P && avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP && avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) { av_log(avr, AV_LOG_ERROR, "Unsupported internal format for " "resampling: %s\n", av_get_sample_fmt_name(avr->internal_sample_fmt)); return NULL; } c = av_mallocz(sizeof(*c)); if (!c) return NULL; c->avr = avr; c->phase_shift = avr->phase_shift; c->phase_mask = phase_count - 1; c->linear = avr->linear_interp; c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1); c->filter_type = avr->filter_type; c->kaiser_beta = avr->kaiser_beta; switch (avr->internal_sample_fmt) { case AV_SAMPLE_FMT_DBLP: c->resample_one = c->linear ? resample_linear_dbl : resample_one_dbl; c->resample_nearest = resample_nearest_dbl; c->set_filter = set_filter_dbl; break; case AV_SAMPLE_FMT_FLTP: c->resample_one = c->linear ? resample_linear_flt : resample_one_flt; c->resample_nearest = resample_nearest_flt; c->set_filter = set_filter_flt; break; case AV_SAMPLE_FMT_S32P: c->resample_one = c->linear ? resample_linear_s32 : resample_one_s32; c->resample_nearest = resample_nearest_s32; c->set_filter = set_filter_s32; break; case AV_SAMPLE_FMT_S16P: c->resample_one = c->linear ? resample_linear_s16 : resample_one_s16; c->resample_nearest = resample_nearest_s16; c->set_filter = set_filter_s16; break; } if (ARCH_AARCH64) ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt); if (ARCH_ARM) ff_audio_resample_init_arm(c, avr->internal_sample_fmt); felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt); c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size); if (!c->filter_bank) goto error; if (build_filter(c, factor) < 0) goto error; memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size], c->filter_bank, (c->filter_length - 1) * felem_size); memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size], &c->filter_bank[(c->filter_length - 1) * felem_size], 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; c->ideal_dst_incr = c->dst_incr; c->padding_size = (c->filter_length - 1) / 2; c->initial_padding_filled = 0; c->index = 0; c->frac = 0; /* allocate internal buffer */ c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size, avr->internal_sample_fmt, "resample buffer"); if (!c->buffer) goto error; c->buffer->nb_samples = c->padding_size; c->initial_padding_samples = c->padding_size; av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n", av_get_sample_fmt_name(avr->internal_sample_fmt), avr->in_sample_rate, avr->out_sample_rate); return c; error: ff_audio_data_free(&c->buffer); av_free(c->filter_bank); av_free(c); return NULL; } void ff_audio_resample_free(ResampleContext **c) { if (!*c) return; ff_audio_data_free(&(*c)->buffer); av_free((*c)->filter_bank); av_freep(c); } int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta, int compensation_distance) { ResampleContext *c; if (compensation_distance < 0) return AVERROR(EINVAL); if (!compensation_distance && sample_delta) return AVERROR(EINVAL); if (!avr->resample_needed) { av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n"); return AVERROR(EINVAL); } c = avr->resample; 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; } return 0; } static int resample(ResampleContext *c, void *dst, const void *src, int *consumed, int src_size, int dst_size, int update_ctx, int nearest_neighbour) { int dst_index; unsigned int index = c->index; int frac = c->frac; int dst_incr_frac = c->dst_incr % c->src_incr; int dst_incr = c->dst_incr / c->src_incr; int compensation_distance = c->compensation_distance; if (!dst != !src) return AVERROR(EINVAL); if (nearest_neighbour) { uint64_t index2 = ((uint64_t)index) << 32; int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr; dst_size = FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); if (dst) { for(dst_index = 0; dst_index < dst_size; dst_index++) { c->resample_nearest(dst, dst_index, src, index2 >> 32); index2 += incr; } } else { dst_index = dst_size; } index += dst_index * dst_incr; index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr; frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr; } else { for (dst_index = 0; dst_index < dst_size; dst_index++) { int sample_index = index >> c->phase_shift; if (sample_index + c->filter_length > src_size) break; if (dst) c->resample_one(c, dst, dst_index, src, index, frac); frac += dst_incr_frac; index += dst_incr; if (frac >= c->src_incr) { frac -= c->src_incr; index++; } if (dst_index + 1 == compensation_distance) { compensation_distance = 0; dst_incr_frac = c->ideal_dst_incr % c->src_incr; dst_incr = c->ideal_dst_incr / c->src_incr; } } } if (consumed) *consumed = index >> c->phase_shift; if (update_ctx) { index &= c->phase_mask; if (compensation_distance) { compensation_distance -= dst_index; if (compensation_distance <= 0) return AVERROR_BUG; } c->frac = frac; c->index = index; c->dst_incr = dst_incr_frac + c->src_incr*dst_incr; c->compensation_distance = compensation_distance; } return dst_index; } int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src) { int ch, in_samples, in_leftover, consumed = 0, out_samples = 0; int ret = AVERROR(EINVAL); int nearest_neighbour = (c->compensation_distance == 0 && c->filter_length == 1 && c->phase_shift == 0); in_samples = src ? src->nb_samples : 0; in_leftover = c->buffer->nb_samples; /* add input samples to the internal buffer */ if (src) { ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples); if (ret < 0) return ret; } else if (in_leftover <= c->final_padding_samples) { /* no remaining samples to flush */ return 0; } if (!c->initial_padding_filled) { int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt); int i; if (src && c->buffer->nb_samples < 2 * c->padding_size) return 0; for (i = 0; i < c->padding_size; i++) for (ch = 0; ch < c->buffer->channels; ch++) { if (c->buffer->nb_samples > 2 * c->padding_size - i) { memcpy(c->buffer->data[ch] + bps * i, c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps); } else { memset(c->buffer->data[ch] + bps * i, 0, bps); } } c->initial_padding_filled = 1; } if (!src && !c->final_padding_filled) { int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt); int i; ret = ff_audio_data_realloc(c->buffer, FFMAX(in_samples, in_leftover) + c->padding_size); if (ret < 0) { av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n"); return AVERROR(ENOMEM); } for (i = 0; i < c->padding_size; i++) for (ch = 0; ch < c->buffer->channels; ch++) { if (in_leftover > i) { memcpy(c->buffer->data[ch] + bps * (in_leftover + i), c->buffer->data[ch] + bps * (in_leftover - i - 1), bps); } else { memset(c->buffer->data[ch] + bps * (in_leftover + i), 0, bps); } } c->buffer->nb_samples += c->padding_size; c->final_padding_samples = c->padding_size; c->final_padding_filled = 1; } /* calculate output size and reallocate output buffer if needed */ /* TODO: try to calculate this without the dummy resample() run */ if (!dst->read_only && dst->allow_realloc) { out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples, INT_MAX, 0, nearest_neighbour); ret = ff_audio_data_realloc(dst, out_samples); if (ret < 0) { av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n"); return ret; } } /* resample each channel plane */ for (ch = 0; ch < c->buffer->channels; ch++) { out_samples = resample(c, (void *)dst->data[ch], (const void *)c->buffer->data[ch], &consumed, c->buffer->nb_samples, dst->allocated_samples, ch + 1 == c->buffer->channels, nearest_neighbour); } if (out_samples < 0) { av_log(c->avr, AV_LOG_ERROR, "error during resampling\n"); return out_samples; } /* drain consumed samples from the internal buffer */ ff_audio_data_drain(c->buffer, consumed); c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0); av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n", in_samples, in_leftover, out_samples, c->buffer->nb_samples); dst->nb_samples = out_samples; return 0; } int avresample_get_delay(AVAudioResampleContext *avr) { ResampleContext *c = avr->resample; if (!avr->resample_needed || !avr->resample) return 0; return FFMAX(c->buffer->nb_samples - c->padding_size, 0); }