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resampling_audio.c
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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>
30 #include <libavutil/samplefmt.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 }
#define NULL
Definition: coverity.c:32
static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat sample_fmt)
const char * fmt
Definition: avisynth_c.h:632
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.
Definition: mathematics.c:58
#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...
Definition: samplefmt.c:196
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
uint8_t
Round toward +infinity.
Definition: mathematics.h:74
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:148
AV_SAMPLE_FMT_U8
AVOptions.
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
#define AV_NE(be, le)
Definition: common.h:50
signed 32 bits
Definition: samplefmt.h:62
int main(int argc, char **argv)
libswresample public header
BYTE * dstp
Definition: avisynth_c.h:676
#define AVERROR(e)
Definition: error.h:43
The libswresample context.
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:533
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.
Definition: samplefmt.c:47
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.
Definition: swresample.c:848
audio channel layout utility functions
#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...
Definition: samplefmt.c:171
sample_rate
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:140
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.
Definition: samplefmt.c:117
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)
Definition: swresample.c:695
signed 16 bits
Definition: samplefmt.h:61
static double c[64]
#define av_freep(p)
#define M_PI
Definition: mathematics.h:46
int nb_channels
int av_opt_set_sample_fmt(void *obj, const char *name, enum AVSampleFormat fmt, int search_flags)
Definition: opt.c:651
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:155