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
32
35 {
36 int i;
37 struct sample_fmt_entry {
39 } sample_fmt_entries[] = {
45 };
47
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",
60 }
61
62 /**
63 * Fill dst buffer with nb_samples, generated starting from t.
64 */
66 {
67 int i, j;
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);
75 dstp[j] = dstp[0];
77 *t += tincr;
78 }
79 }
80
81 int main(
int argc,
char **argv)
82 {
84 int src_rate = 48000, dst_rate = 44100;
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;
90 const char *dst_filename =
NULL;
91 FILE *dst_file;
92 int dst_bufsize;
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 */
116 if (!swr_ctx) {
117 fprintf(stderr, "Could not allocate resampler context\n");
120 }
121
122 /* set options */
126
130
131 /* initialize the resampling context */
132 if ((ret =
swr_init(swr_ctx)) < 0) {
133 fprintf(stderr, "Failed to initialize the resampling context\n");
135 }
136
137 /* allocate source and destination samples buffers */
138
141 src_nb_samples, src_sample_fmt, 0);
142 if (ret < 0) {
143 fprintf(stderr, "Could not allocate source samples\n");
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 =
152
153 /* buffer is going to be directly written to a rawaudio file, no alignment */
156 dst_nb_samples, dst_sample_fmt, 0);
157 if (ret < 0) {
158 fprintf(stderr, "Could not allocate destination samples\n");
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 */
170 if (dst_nb_samples > max_dst_nb_samples) {
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");
184 }
186 ret, dst_sample_fmt, 1);
187 if (dst_bufsize < 0) {
188 fprintf(stderr, "Could not get sample buffer size\n");
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
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
202 fclose(dst_file);
203
204 if (src_data)
207
208 if (dst_data)
211
213 return ret < 0;
214 }
static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat sample_fmt)
#define AV_CH_LAYOUT_SURROUND
#define AV_CH_LAYOUT_STEREO
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a data pointers array, samples buffer for nb_samples samples, and fill data pointers and lin...
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
static av_cold int end(AVCodecContext *avctx)
int main(int argc, char **argv)
libswresample public header
The libswresample context.
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
Fill dst buffer with nb_samples, generated starting from t.
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
int64_t swr_get_delay(struct SwrContext *s, int64_t base)
Gets the delay the next input sample will experience relative to the next output sample.
audio channel layout utility functions
int64_t av_rescale_rnd(int64_t a, int64_t b, int64_t c, enum AVRounding rnd)
Rescale a 64-bit integer with specified rounding.
#define FF_ARRAY_ELEMS(a)
int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a samples buffer for nb_samples samples, and fill data pointers and linesize accordingly...
AVSampleFormat
Audio sample formats.
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Get the required buffer size for the given audio parameters.
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)
int av_opt_set_sample_fmt(void *obj, const char *name, enum AVSampleFormat fmt, int search_flags)
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.