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Diffstat (limited to 'libswresample/soxr_resample.c')
| -rw-r--r-- | libswresample/soxr_resample.c | 93 |
1 files changed, 93 insertions, 0 deletions
diff --git a/libswresample/soxr_resample.c b/libswresample/soxr_resample.c new file mode 100644 index 0000000000..4c000db0ca --- /dev/null +++ b/libswresample/soxr_resample.c @@ -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, +}; + |