|
|
@@ -0,0 +1,89 @@
|
|
|
+/*
|
|
|
+ * audio resampling with soxr
|
|
|
+ * Copyright (c) 2012 Rob Sykes <aquegg@yahoo.co.uk>
|
|
|
+ *
|
|
|
+ * 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.bits = linear? 0 : precision;
|
|
|
+ q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc;
|
|
|
+
|
|
|
+ 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,
|
|
|
+};
|
|
|
+
|
Rob Sykes