1 /*
2 * Copyright (c) 2012 Stefano Sabatini
3 *
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:
10 *
11 * The above copyright notice and this permission notice shall be included in
12 * all copies or substantial portions of the Software.
13 *
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
20 * THE SOFTWARE.
21 */
22
23 /**
24 * @example resampling_audio.c
25 * libswresample API use example.
26 */
27
28 #include <libavutil/opt.h>
29 #include <libavutil/channel_layout.h>
30 #include <libavutil/samplefmt.h>
31 #include <libswresample/swresample.h>
32
get_format_from_sample_fmt(const char ** fmt,enum AVSampleFormat sample_fmt)33 static int get_format_from_sample_fmt(const char **fmt,
34 enum AVSampleFormat sample_fmt)
35 {
36 int i;
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" },
45 };
46 *fmt = NULL;
47
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);
52 return 0;
53 }
54 }
55
56 fprintf(stderr,
57 "Sample format %s not supported as output format\n",
58 av_get_sample_fmt_name(sample_fmt));
59 return AVERROR(EINVAL);
60 }
61
62 /**
63 * Fill dst buffer with nb_samples, generated starting from t.
64 */
fill_samples(double * dst,int nb_samples,int nb_channels,int sample_rate,double * t)65 static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
66 {
67 int i, j;
68 double tincr = 1.0 / sample_rate, *dstp = dst;
69 const double c = 2 * M_PI * 440.0;
70
71 /* generate sin tone with 440Hz frequency and duplicated channels */
72 for (i = 0; i < nb_samples; i++) {
73 *dstp = sin(c * *t);
74 for (j = 1; j < nb_channels; j++)
75 dstp[j] = dstp[0];
76 dstp += nb_channels;
77 *t += tincr;
78 }
79 }
80
main(int argc,char ** argv)81 int main(int argc, char **argv)
82 {
83 AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_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;
91 FILE *dst_file;
92 int dst_bufsize;
93 const char *fmt;
94 struct SwrContext *swr_ctx;
95 char buf[64];
96 double t;
97 int ret;
98
99 if (argc != 2) {
100 fprintf(stderr, "Usage: %s output_file\n"
101 "API example program to show how to resample an audio stream with libswresample.\n"
102 "This program generates a series of audio frames, resamples them to a specified "
103 "output format and rate and saves them to an output file named output_file.\n",
104 argv[0]);
105 exit(1);
106 }
107 dst_filename = argv[1];
108
109 dst_file = fopen(dst_filename, "wb");
110 if (!dst_file) {
111 fprintf(stderr, "Could not open destination file %s\n", dst_filename);
112 exit(1);
113 }
114
115 /* create resampler context */
116 swr_ctx = swr_alloc();
117 if (!swr_ctx) {
118 fprintf(stderr, "Could not allocate resampler context\n");
119 ret = AVERROR(ENOMEM);
120 goto end;
121 }
122
123 /* set options */
124 av_opt_set_chlayout(swr_ctx, "in_chlayout", &src_ch_layout, 0);
125 av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
126 av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
127
128 av_opt_set_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 0);
129 av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
130 av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
131
132 /* initialize the resampling context */
133 if ((ret = swr_init(swr_ctx)) < 0) {
134 fprintf(stderr, "Failed to initialize the resampling context\n");
135 goto end;
136 }
137
138 /* allocate source and destination samples buffers */
139
140 src_nb_channels = src_ch_layout.nb_channels;
141 ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
142 src_nb_samples, src_sample_fmt, 0);
143 if (ret < 0) {
144 fprintf(stderr, "Could not allocate source samples\n");
145 goto end;
146 }
147
148 /* compute the number of converted samples: buffering is avoided
149 * ensuring that the output buffer will contain at least all the
150 * converted input samples */
151 max_dst_nb_samples = dst_nb_samples =
152 av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
153
154 /* buffer is going to be directly written to a rawaudio file, no alignment */
155 dst_nb_channels = dst_ch_layout.nb_channels;
156 ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
157 dst_nb_samples, dst_sample_fmt, 0);
158 if (ret < 0) {
159 fprintf(stderr, "Could not allocate destination samples\n");
160 goto end;
161 }
162
163 t = 0;
164 do {
165 /* generate synthetic audio */
166 fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
167
168 /* compute destination number of samples */
169 dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
170 src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
171 if (dst_nb_samples > max_dst_nb_samples) {
172 av_freep(&dst_data[0]);
173 ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
174 dst_nb_samples, dst_sample_fmt, 1);
175 if (ret < 0)
176 break;
177 max_dst_nb_samples = dst_nb_samples;
178 }
179
180 /* convert to destination format */
181 ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
182 if (ret < 0) {
183 fprintf(stderr, "Error while converting\n");
184 goto end;
185 }
186 dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
187 ret, dst_sample_fmt, 1);
188 if (dst_bufsize < 0) {
189 fprintf(stderr, "Could not get sample buffer size\n");
190 goto end;
191 }
192 printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
193 fwrite(dst_data[0], 1, dst_bufsize, dst_file);
194 } while (t < 10);
195
196 if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
197 goto end;
198 av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
199 fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
200 "ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
201 fmt, buf, dst_nb_channels, dst_rate, dst_filename);
202
203 end:
204 fclose(dst_file);
205
206 if (src_data)
207 av_freep(&src_data[0]);
208 av_freep(&src_data);
209
210 if (dst_data)
211 av_freep(&dst_data[0]);
212 av_freep(&dst_data);
213
214 swr_free(&swr_ctx);
215 return ret < 0;
216 }
217