2 * Copyright (c) 2012 Stefano Sabatini
4 * Permission is hereby granted, free of charge, to any person obtaining a copy
5 * of this software and associated documentation files (the "Software"), to deal
6 * in the Software without restriction, including without limitation the rights
7 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
8 * copies of the Software, and to permit persons to whom the Software is
9 * furnished to do so, subject to the following conditions:
11 * The above copyright notice and this permission notice shall be included in
12 * all copies or substantial portions of the Software.
14 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
15 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
16 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
17 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
18 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
19 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
24 * @example resampling_audio.c
25 * libswresample API use example.
28 #include <libavutil/opt.h>
29 #include <libavutil/channel_layout.h>
30 #include <libavutil/samplefmt.h>
31 #include <libswresample/swresample.h>
33 static int get_format_from_sample_fmt(const char **fmt
,
34 enum AVSampleFormat sample_fmt
)
37 struct sample_fmt_entry
{
38 enum AVSampleFormat sample_fmt
; const char *fmt_be
, *fmt_le
;
39 } sample_fmt_entries
[] = {
40 { AV_SAMPLE_FMT_U8
, "u8", "u8" },
41 { AV_SAMPLE_FMT_S16
, "s16be", "s16le" },
42 { AV_SAMPLE_FMT_S32
, "s32be", "s32le" },
43 { AV_SAMPLE_FMT_FLT
, "f32be", "f32le" },
44 { AV_SAMPLE_FMT_DBL
, "f64be", "f64le" },
48 for (i
= 0; i
< FF_ARRAY_ELEMS(sample_fmt_entries
); i
++) {
49 struct sample_fmt_entry
*entry
= &sample_fmt_entries
[i
];
50 if (sample_fmt
== entry
->sample_fmt
) {
51 *fmt
= AV_NE(entry
->fmt_be
, entry
->fmt_le
);
57 "Sample format %s not supported as output format\n",
58 av_get_sample_fmt_name(sample_fmt
));
59 return AVERROR(EINVAL
);
63 * Fill dst buffer with nb_samples, generated starting from t.
65 static void fill_samples(double *dst
, int nb_samples
, int nb_channels
, int sample_rate
, double *t
)
68 double tincr
= 1.0 / sample_rate
, *dstp
= dst
;
69 const double c
= 2 * M_PI
* 440.0;
71 /* generate sin tone with 440Hz frequency and duplicated channels */
72 for (i
= 0; i
< nb_samples
; i
++) {
74 for (j
= 1; j
< nb_channels
; j
++)
81 int main(int argc
, char **argv
)
83 int64_t src_ch_layout
= AV_CH_LAYOUT_STEREO
, dst_ch_layout
= AV_CH_LAYOUT_SURROUND
;
84 int src_rate
= 48000, dst_rate
= 44100;
85 uint8_t **src_data
= NULL
, **dst_data
= NULL
;
86 int src_nb_channels
= 0, dst_nb_channels
= 0;
87 int src_linesize
, dst_linesize
;
88 int src_nb_samples
= 1024, dst_nb_samples
, max_dst_nb_samples
;
89 enum AVSampleFormat src_sample_fmt
= AV_SAMPLE_FMT_DBL
, dst_sample_fmt
= AV_SAMPLE_FMT_S16
;
90 const char *dst_filename
= NULL
;
94 struct SwrContext
*swr_ctx
;
99 fprintf(stderr
, "Usage: %s output_file\n"
100 "API example program to show how to resample an audio stream with libswresample.\n"
101 "This program generates a series of audio frames, resamples them to a specified "
102 "output format and rate and saves them to an output file named output_file.\n",
106 dst_filename
= argv
[1];
108 dst_file
= fopen(dst_filename
, "wb");
110 fprintf(stderr
, "Could not open destination file %s\n", dst_filename
);
114 /* create resampler context */
115 swr_ctx
= swr_alloc();
117 fprintf(stderr
, "Could not allocate resampler context\n");
118 ret
= AVERROR(ENOMEM
);
123 av_opt_set_int(swr_ctx
, "in_channel_layout", src_ch_layout
, 0);
124 av_opt_set_int(swr_ctx
, "in_sample_rate", src_rate
, 0);
125 av_opt_set_sample_fmt(swr_ctx
, "in_sample_fmt", src_sample_fmt
, 0);
127 av_opt_set_int(swr_ctx
, "out_channel_layout", dst_ch_layout
, 0);
128 av_opt_set_int(swr_ctx
, "out_sample_rate", dst_rate
, 0);
129 av_opt_set_sample_fmt(swr_ctx
, "out_sample_fmt", dst_sample_fmt
, 0);
131 /* initialize the resampling context */
132 if ((ret
= swr_init(swr_ctx
)) < 0) {
133 fprintf(stderr
, "Failed to initialize the resampling context\n");
137 /* allocate source and destination samples buffers */
139 src_nb_channels
= av_get_channel_layout_nb_channels(src_ch_layout
);
140 ret
= av_samples_alloc_array_and_samples(&src_data
, &src_linesize
, src_nb_channels
,
141 src_nb_samples
, src_sample_fmt
, 0);
143 fprintf(stderr
, "Could not allocate source samples\n");
147 /* compute the number of converted samples: buffering is avoided
148 * ensuring that the output buffer will contain at least all the
149 * converted input samples */
150 max_dst_nb_samples
= dst_nb_samples
=
151 av_rescale_rnd(src_nb_samples
, dst_rate
, src_rate
, AV_ROUND_UP
);
153 /* buffer is going to be directly written to a rawaudio file, no alignment */
154 dst_nb_channels
= av_get_channel_layout_nb_channels(dst_ch_layout
);
155 ret
= av_samples_alloc_array_and_samples(&dst_data
, &dst_linesize
, dst_nb_channels
,
156 dst_nb_samples
, dst_sample_fmt
, 0);
158 fprintf(stderr
, "Could not allocate destination samples\n");
164 /* generate synthetic audio */
165 fill_samples((double *)src_data
[0], src_nb_samples
, src_nb_channels
, src_rate
, &t
);
167 /* compute destination number of samples */
168 dst_nb_samples
= av_rescale_rnd(swr_get_delay(swr_ctx
, src_rate
) +
169 src_nb_samples
, dst_rate
, src_rate
, AV_ROUND_UP
);
170 if (dst_nb_samples
> max_dst_nb_samples
) {
171 av_freep(&dst_data
[0]);
172 ret
= av_samples_alloc(dst_data
, &dst_linesize
, dst_nb_channels
,
173 dst_nb_samples
, dst_sample_fmt
, 1);
176 max_dst_nb_samples
= dst_nb_samples
;
179 /* convert to destination format */
180 ret
= swr_convert(swr_ctx
, dst_data
, dst_nb_samples
, (const uint8_t **)src_data
, src_nb_samples
);
182 fprintf(stderr
, "Error while converting\n");
185 dst_bufsize
= av_samples_get_buffer_size(&dst_linesize
, dst_nb_channels
,
186 ret
, dst_sample_fmt
, 1);
187 if (dst_bufsize
< 0) {
188 fprintf(stderr
, "Could not get sample buffer size\n");
191 printf("t:%f in:%d out:%d\n", t
, src_nb_samples
, ret
);
192 fwrite(dst_data
[0], 1, dst_bufsize
, dst_file
);
195 if ((ret
= get_format_from_sample_fmt(&fmt
, dst_sample_fmt
)) < 0)
197 fprintf(stderr
, "Resampling succeeded. Play the output file with the command:\n"
198 "ffplay -f %s -channel_layout %"PRId64
" -channels %d -ar %d %s\n",
199 fmt
, dst_ch_layout
, dst_nb_channels
, dst_rate
, dst_filename
);
205 av_freep(&src_data
[0]);
209 av_freep(&dst_data
[0]);