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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 | ||
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 | */ | |
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 | ||
81 | int main(int argc, char **argv) | |
82 | { | |
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; | |
91 | FILE *dst_file; | |
92 | int dst_bufsize; | |
93 | const char *fmt; | |
94 | struct SwrContext *swr_ctx; | |
95 | double t; | |
96 | int ret; | |
97 | ||
98 | if (argc != 2) { | |
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", | |
103 | argv[0]); | |
104 | exit(1); | |
105 | } | |
106 | dst_filename = argv[1]; | |
107 | ||
108 | dst_file = fopen(dst_filename, "wb"); | |
109 | if (!dst_file) { | |
110 | fprintf(stderr, "Could not open destination file %s\n", dst_filename); | |
111 | exit(1); | |
112 | } | |
113 | ||
114 | /* create resampler context */ | |
115 | swr_ctx = swr_alloc(); | |
116 | if (!swr_ctx) { | |
117 | fprintf(stderr, "Could not allocate resampler context\n"); | |
118 | ret = AVERROR(ENOMEM); | |
119 | goto end; | |
120 | } | |
121 | ||
122 | /* set options */ | |
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); | |
126 | ||
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); | |
130 | ||
131 | /* initialize the resampling context */ | |
132 | if ((ret = swr_init(swr_ctx)) < 0) { | |
133 | fprintf(stderr, "Failed to initialize the resampling context\n"); | |
134 | goto end; | |
135 | } | |
136 | ||
137 | /* allocate source and destination samples buffers */ | |
138 | ||
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); | |
142 | if (ret < 0) { | |
143 | fprintf(stderr, "Could not allocate source samples\n"); | |
144 | goto end; | |
145 | } | |
146 | ||
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); | |
152 | ||
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); | |
157 | if (ret < 0) { | |
158 | fprintf(stderr, "Could not allocate destination samples\n"); | |
159 | goto end; | |
160 | } | |
161 | ||
162 | t = 0; | |
163 | do { | |
164 | /* generate synthetic audio */ | |
165 | fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t); | |
166 | ||
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); | |
174 | if (ret < 0) | |
175 | break; | |
176 | max_dst_nb_samples = dst_nb_samples; | |
177 | } | |
178 | ||
179 | /* convert to destination format */ | |
180 | ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples); | |
181 | if (ret < 0) { | |
182 | fprintf(stderr, "Error while converting\n"); | |
183 | goto end; | |
184 | } | |
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"); | |
189 | goto end; | |
190 | } | |
191 | printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret); | |
192 | fwrite(dst_data[0], 1, dst_bufsize, dst_file); | |
193 | } while (t < 10); | |
194 | ||
195 | if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0) | |
196 | goto end; | |
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); | |
200 | ||
201 | end: | |
202 | fclose(dst_file); | |
203 | ||
204 | if (src_data) | |
205 | av_freep(&src_data[0]); | |
206 | av_freep(&src_data); | |
207 | ||
208 | if (dst_data) | |
209 | av_freep(&dst_data[0]); | |
210 | av_freep(&dst_data); | |
211 | ||
212 | swr_free(&swr_ctx); | |
213 | return ret < 0; | |
214 | } |