/* * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> * * This file is part of Libav. * * Libav 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. * * Libav 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 Libav; 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 "audio_data.h" struct ResampleContext { AVAudioResampleContext *avr; AudioData *buffer; uint8_t *filter_bank; int filter_length; int ideal_dst_incr; int dst_incr; int index; int frac; int src_incr; int compensation_distance; int phase_shift; int phase_mask; int linear; enum AVResampleFilterType filter_type; int kaiser_beta; double factor; void (*set_filter)(void *filter, double *tab, int phase, int tap_count); void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0, int dst_index, const void *src0, int src_size, int index, int frac); }; /* 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) { int ph, i; double x, y, w, factor; 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); /* if upsampling, only need to interpolate, no filter */ factor = FFMIN(c->factor, 1.0); 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->factor = factor; 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 = resample_one_dbl; c->set_filter = set_filter_dbl; break; case AV_SAMPLE_FMT_FLTP: c->resample_one = resample_one_flt; c->set_filter = set_filter_flt; break; case AV_SAMPLE_FMT_S32P: c->resample_one = resample_one_s32; c->set_filter = set_filter_s32; break; case AV_SAMPLE_FMT_S16P: c->resample_one = resample_one_s16; c->set_filter = set_filter_s16; break; } 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) < 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->index = -phase_count * ((c->filter_length - 1) / 2); c->frac = 0; /* allocate internal buffer */ c->buffer = ff_audio_data_alloc(avr->resample_channels, 0, avr->internal_sample_fmt, "resample buffer"); if (!c->buffer) goto error; 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; AudioData *fifo_buf = NULL; int ret = 0; if (compensation_distance < 0) return AVERROR(EINVAL); if (!compensation_distance && sample_delta) return AVERROR(EINVAL); /* if resampling was not enabled previously, re-initialize the AVAudioResampleContext and force resampling */ if (!avr->resample_needed) { int fifo_samples; double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 }; /* buffer any remaining samples in the output FIFO before closing */ fifo_samples = av_audio_fifo_size(avr->out_fifo); if (fifo_samples > 0) { fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples, avr->out_sample_fmt, NULL); if (!fifo_buf) return AVERROR(EINVAL); ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf, fifo_samples); if (ret < 0) goto reinit_fail; } /* save the channel mixing matrix */ ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS); if (ret < 0) goto reinit_fail; /* close the AVAudioResampleContext */ avresample_close(avr); avr->force_resampling = 1; /* restore the channel mixing matrix */ ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS); if (ret < 0) goto reinit_fail; /* re-open the AVAudioResampleContext */ ret = avresample_open(avr); if (ret < 0) goto reinit_fail; /* restore buffered samples to the output FIFO */ if (fifo_samples > 0) { ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0, fifo_samples); if (ret < 0) goto reinit_fail; ff_audio_data_free(&fifo_buf); } } 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; reinit_fail: ff_audio_data_free(&fifo_buf); return ret; } static int resample(ResampleContext *c, void *dst, const void *src, int *consumed, int src_size, int dst_size, int update_ctx) { int dst_index; 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 (compensation_distance == 0 && c->filter_length == 1 && c->phase_shift == 0) { int64_t index2 = ((int64_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_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0); 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 || -sample_index >= src_size) break; if (dst) c->resample_one(c, 0, dst, dst_index, src, src_size, 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 = FFMAX(index, 0) >> c->phase_shift; if (update_ctx) { if (index >= 0) 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 *consumed) { int ch, in_samples, in_leftover, out_samples = 0; int ret = AVERROR(EINVAL); 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) { /* no remaining samples to flush */ return 0; } else { /* TODO: pad buffer to flush completely */ } /* 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); 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); } 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); av_dlog(c->avr, "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) { if (!avr->resample_needed || !avr->resample) return 0; return avr->resample->buffer->nb_samples; }