Loading...
Searching...
No Matches
resample_audio.c

Generate a synthetic audio signal, and Use libswresample API to perform audio resampling.

Generate a synthetic audio signal, and Use libswresample API to perform audio resampling. The output is written to a raw audio file to be played with ffplay.

/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file audio resampling API usage example
* @example resample_audio.c
*
* Generate a synthetic audio signal, and Use libswresample API to perform audio
* resampling. The output is written to a raw audio file to be played with
* ffplay.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
char buf[64];
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_chlayout(swr_ctx, "in_chlayout", &src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = src_ch_layout.nb_channels;
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = dst_ch_layout.nb_channels;
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
fmt, buf, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}
main
int main(int argc, char *argv[])
Definition avio_list_dir.c:119
channel_layout.h
Public libavutil channel layout APIs header.
get_format_from_sample_fmt
static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat sample_fmt)
Definition decode_audio.c:43
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition channel_layout.h:395
AV_CHANNEL_LAYOUT_SURROUND
#define AV_CHANNEL_LAYOUT_SURROUND
Definition channel_layout.h:398
av_channel_layout_describe
int av_channel_layout_describe(const AVChannelLayout *channel_layout, char *buf, size_t buf_size)
Get a human-readable string describing the channel layout properties.
AVERROR
#define AVERROR(e)
Definition error.h:45
av_rescale_rnd
int64_t av_rescale_rnd(int64_t a, int64_t b, int64_t c, enum AVRounding rnd) av_const
Rescale a 64-bit integer with specified rounding.
AV_ROUND_UP
@ AV_ROUND_UP
Round toward +infinity.
Definition mathematics.h:134
av_freep
void av_freep(void *ptr)
Free a memory block which has been allocated with a function of av_malloc() or av_realloc() family,...
av_samples_get_buffer_size
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.
av_get_sample_fmt_name
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.
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition samplefmt.h:55
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition samplefmt.h:60
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition samplefmt.h:59
AV_SAMPLE_FMT_U8
@ AV_SAMPLE_FMT_U8
unsigned 8 bits
Definition samplefmt.h:57
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition samplefmt.h:61
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition samplefmt.h:58
av_samples_alloc
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.
av_samples_alloc_array_and_samples
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...
swr_alloc
struct SwrContext * swr_alloc(void)
Allocate SwrContext.
SwrContext
struct SwrContext SwrContext
The libswresample context.
Definition swresample.h:189
swr_get_delay
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.
swr_free
void swr_free(struct SwrContext **s)
Free the given SwrContext and set the pointer to NULL.
swr_convert
int swr_convert(struct SwrContext *s, uint8_t *const *out, int out_count, const uint8_t *const *in, int in_count)
Convert audio.
swr_init
int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
av_opt_set_int
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
av_opt_set_chlayout
int av_opt_set_chlayout(void *obj, const char *name, const AVChannelLayout *layout, int search_flags)
av_opt_set_sample_fmt
int av_opt_set_sample_fmt(void *obj, const char *name, enum AVSampleFormat fmt, int search_flags)
AV_NE
#define AV_NE(be, le)
Definition macros.h:33
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition macros.h:53
M_PI
#define M_PI
Definition mathematics.h:67
opt.h
AVOptions.
fill_samples
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.
Definition resample_audio.c:69
samplefmt.h
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition channel_layout.h:319
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition channel_layout.h:329
swresample.h
libswresample public header