FFmpeg
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:861
#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:198
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:83
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:149
AV_SAMPLE_FMT_U8
AVOptions.
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
#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
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
BYTE * dstp
Definition: avisynth_c.h:908
The libswresample context.
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:568
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:49
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:867
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.
Definition: mathematics.c:58
#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:173
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:137
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:119
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:714
signed 16 bits
Definition: samplefmt.h:61
printf("static const uint8_t my_array[100] = {\n")
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
int nb_channels
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
int av_opt_set_sample_fmt(void *obj, const char *name, enum AVSampleFormat fmt, int search_flags)
Definition: opt.c:686
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152