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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, double kaiser_beta, double precision, int cheby, int exact_rational){
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  s->delayed_samples_fixup = soxr_delay((soxr_t)s->resample);
71 
72  soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
73 
74  {
75  float f;
76  size_t idone, odone;
77  soxr_process((soxr_t)s->resample, &f, 0, &idone, &f, 0, &odone);
78  s->delayed_samples_fixup -= soxr_delay((soxr_t)s->resample);
79  }
80 
81  return 0;
82 }
83 
84 static int process(
85  struct ResampleContext * c, AudioData *dst, int dst_size,
86  AudioData *src, int src_size, int *consumed){
87  size_t idone, odone;
88  soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));
89  if (!error)
90  error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
91  &idone, dst->ch, (size_t)dst_size, &odone);
92  else
93  idone = 0;
94 
95  *consumed = (int)idone;
96  return error? -1 : odone;
97 }
98 
99 static int64_t get_delay(struct SwrContext *s, int64_t base){
100  double delayed_samples = soxr_delay((soxr_t)s->resample);
101  double delay_s;
102 
103  if (s->flushed)
104  delayed_samples += s->delayed_samples_fixup;
105 
106  delay_s = delayed_samples / s->out_sample_rate;
107 
108  return (int64_t)(delay_s * base + .5);
109 }
110 
112  int in_count, int *out_idx, int *out_sz){
113  return 0;
114 }
115 
116 static int64_t get_out_samples(struct SwrContext *s, int in_samples){
117  double out_samples = (double)s->out_sample_rate / s->in_sample_rate * in_samples;
118  double delayed_samples = soxr_delay((soxr_t)s->resample);
119 
120  if (s->flushed)
121  delayed_samples += s->delayed_samples_fixup;
122 
123  return (int64_t)(out_samples + delayed_samples + 1 + .5);
124 }
125 
127  create, destroy, process, flush, NULL /* set_compensation */, get_delay,
129 };
130 
float, planar
Definition: samplefmt.h:69
#define NULL
Definition: coverity.c:32
static int64_t get_out_samples(struct SwrContext *s, int in_samples)
const char * s
Definition: avisynth_c.h:768
int out_sample_rate
output sample rate
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
static int linear(InterplayACMContext *s, unsigned ind, unsigned col)
Definition: interplayacm.c:116
int ch_count
number of channels
SwrFilterType
Resampling Filter Types.
Definition: swresample.h:169
double delayed_samples_fixup
soxr 0.1.1: needed to fixup delayed_samples after flush has been called.
double, planar
Definition: samplefmt.h:70
static int64_t get_delay(struct SwrContext *s, int64_t base)
Definition: soxr_resample.c:99
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:84
enum AVResampleFilterType filter_type
Definition: resample.h:42
signed 32 bits
Definition: samplefmt.h:62
#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:94
#define FFMIN(a, b)
Definition: common.h:96
signed 32 bits, planar
Definition: samplefmt.h:68
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, double kaiser_beta, double precision, int cheby, int exact_rational)
Definition: soxr_resample.c:32
int in_sample_rate
input sample rate
#define src
Definition: vp9dsp.c:530
static int flush(struct SwrContext *s)
Definition: soxr_resample.c:69
static void destroy(struct ResampleContext **c)
Definition: soxr_resample.c:64
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
GLint GLenum type
Definition: opengl_enc.c:105
static const char * format
Definition: movenc.c:47
int flushed
1 if data is to be flushed and no further input is expected
signed 16 bits
Definition: samplefmt.h:61
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)
signed 16 bits, planar
Definition: samplefmt.h:67
struct Resampler const swri_soxr_resampler
uint8_t * ch[SWR_CH_MAX]
samples buffer per channel