| 0,0 → 1,93 |
|---|
| /* |
| * 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)); |
| error = soxr_process((soxr_t)c, src->ch, (size_t)src_size, |
| &idone, dst->ch, (size_t)dst_size, &odone); |
| *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); |
| } |
| |
| struct Resampler const soxr_resampler={ |
| create, destroy, process, flush, NULL /* set_compensation */, get_delay, |
| }; |
| |