/* * audio resampling with soxr * Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.net> * * 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 with soxr */ #include "libavutil/log.h" #include "swresample_internal.h" #include <soxr.h> static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){ soxr_error_t error; soxr_datatype_t type = format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S : format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I : format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S : format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I : format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S : format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I : format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S : format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1; soxr_io_spec_t io_spec = soxr_io_spec(type, type); soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby); q_spec.precision = linear? 0 : precision; #if !defined SOXR_VERSION /* Deprecated @ March 2013: */ q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc; #else q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end; #endif soxr_delete((soxr_t)c); c = (struct ResampleContext *) soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0); if (!c) av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error); return c; } static void destroy(struct ResampleContext * *c){ soxr_delete((soxr_t)*c); *c = NULL; } static int flush(struct SwrContext *s){ soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL); return 0; } static int process( struct ResampleContext * c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ size_t idone, odone; soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count)); if (!error) error = soxr_process((soxr_t)c, src->ch, (size_t)src_size, &idone, dst->ch, (size_t)dst_size, &odone); else idone = 0; *consumed = (int)idone; return error? -1 : odone; } static int64_t get_delay(struct SwrContext *s, int64_t base){ double delay_s = soxr_delay((soxr_t)s->resample) / s->out_sample_rate; return (int64_t)(delay_s * base + .5); } static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src, int in_count, int *out_idx, int *out_sz) { return 0; } struct Resampler const swri_soxr_resampler={ create, destroy, process, flush, NULL /* set_compensation */, get_delay, invert_initial_buffer, };