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soxr_resample.c
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1 /*
2  * audio resampling with soxr
3  * Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.net>
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * audio resampling with soxr
25  */
26 
27 #include "libavutil/log.h"
28 #include "swresample_internal.h"
29 
30 #include <soxr.h>
31 
32 static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
33  double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){
34  soxr_error_t error;
35 
36  soxr_datatype_t type =
37  format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S :
38  format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I :
39  format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S :
40  format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I :
41  format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S :
42  format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I :
43  format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S :
44  format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1;
45 
46  soxr_io_spec_t io_spec = soxr_io_spec(type, type);
47 
48  soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby);
49  q_spec.precision = linear? 0 : precision;
50 #if !defined SOXR_VERSION /* Deprecated @ March 2013: */
51  q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc;
52 #else
53  q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end;
54 #endif
55 
56  soxr_delete((soxr_t)c);
57  c = (struct ResampleContext *)
58  soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0);
59  if (!c)
60  av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error);
61  return c;
62 }
63 
64 static void destroy(struct ResampleContext * *c){
65  soxr_delete((soxr_t)*c);
66  *c = NULL;
67 }
68 
69 static int flush(struct SwrContext *s){
70  soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
71  return 0;
72 }
73 
74 static int process(
75  struct ResampleContext * c, AudioData *dst, int dst_size,
76  AudioData *src, int src_size, int *consumed){
77  size_t idone, odone;
78  soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));
79  if (!error)
80  error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
81  &idone, dst->ch, (size_t)dst_size, &odone);
82  else
83  idone = 0;
84 
85  *consumed = (int)idone;
86  return error? -1 : odone;
87 }
88 
89 static int64_t get_delay(struct SwrContext *s, int64_t base){
90  double delay_s = soxr_delay((soxr_t)s->resample) / s->out_sample_rate;
91  return (int64_t)(delay_s * base + .5);
92 }
93 
94 static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src,
95  int in_count, int *out_idx, int *out_sz)
96 {
97  return 0;
98 }
99 
101  create, destroy, process, flush, NULL /* set_compensation */, get_delay,
103 };
104 
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)
Definition: soxr_resample.c:32
float, planar
Definition: samplefmt.h:70
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:631
int out_sample_rate
output sample rate
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
int ch_count
number of channels
SwrFilterType
Resampling Filter Types.
Definition: swresample.h:170
double, planar
Definition: samplefmt.h:71
static int64_t get_delay(struct SwrContext *s, int64_t base)
Definition: soxr_resample.c:89
struct ResampleContext * resample
resampling context
static int process(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed)
Definition: soxr_resample.c:74
enum AVSampleFormat format
Definition: resample.h:49
enum AVResampleFilterType filter_type
Definition: resample.h:42
signed 32 bits
Definition: samplefmt.h:63
#define av_log(a,...)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
The libswresample context.
#define FFMAX(a, b)
Definition: common.h:64
#define FFMIN(a, b)
Definition: common.h:66
signed 32 bits, planar
Definition: samplefmt.h:69
static int flush(struct SwrContext *s)
Definition: soxr_resample.c:69
static void destroy(struct ResampleContext **c)
Definition: soxr_resample.c:64
AVS_Value src
Definition: avisynth_c.h:482
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
GLint GLenum type
Definition: opengl_enc.c:105
signed 16 bits
Definition: samplefmt.h:62
static double c[64]
static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src, int in_count, int *out_idx, int *out_sz)
Definition: soxr_resample.c:94
signed 16 bits, planar
Definition: samplefmt.h:68
struct Resampler const swri_soxr_resampler
uint8_t * ch[SWR_CH_MAX]
samples buffer per channel