blob: 8181c749b860b16781e35f8ac21832de81dc512c
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 = 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 | |
111 | static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src, |
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 | |
126 | struct Resampler const swri_soxr_resampler={ |
127 | create, destroy, process, flush, NULL /* set_compensation */, get_delay, |
128 | invert_initial_buffer, get_out_samples |
129 | }; |
130 | |
131 |