swr: Add SOX resampler support

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 89 insertions, 0 deletions

diff --git a/libswresample/soxr_resample.c b/libswresample/soxr_resample.c
new file mode 100644
index 0000000000..0e02fcf56d
--- /dev/null
+++ b/libswresample/soxr_resample.c
@@ -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,
+};
+