libswresample库功能主要包括高度优化的音频重采样、rematrixing和样本格式转换操作。
以下是测试代码(test_ffmpeg_libswresample.cpp),对音频了解较少,测试代码是参考examples中的:
#include "funset.hpp"
#include <iostream>
#ifdef __cplusplus
extern "C" {
#endif
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
#ifdef __cplusplus
}
#endif
namespace {
int get_format_from_sample_fmt(const char** fmt, enum AVSampleFormat sample_fmt)
{
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 = nullptr;
for (int 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.
void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
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 (int i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (int j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
} // namespace
int test_ffmpeg_libswresample_resample()
{
// reference: doc/examples/resample_audio.c
fprintf(stdout, "swresample version: %d\n", swresample_version());
fprintf(stdout, "swresample configuration: %s\n", swresample_configuration());
fprintf(stdout, "swresample license: %s\n", swresample_license());
//create resampler context
struct SwrContext* swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "fail to swr_alloc\n");
return -1;
}
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
// set options
av_opt_set_int(swr_ctx, "in_channel_layout", 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_int(swr_ctx, "out_channel_layout", 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 (swr_init(swr_ctx) < 0) {
fprintf(stderr, "fail to swr_init\n");
return -1;
}
uint8_t **src_data = nullptr, **dst_data = nullptr;
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;
// allocate source and destination samples buffers
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
int 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, "fail to av_samples_alloc_array_and_samples\n");
return -1;
}
// 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 = av_get_channel_layout_nb_channels(dst_ch_layout);
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, "fail to av_samples_alloc_array_and_samples\n");
return -1;
}
#ifdef _MSC_VER
const char* file_name = "E:/GitCode/OpenCV_Test/test_images/xxx";
#else
const char* file_name = "test_images/xxx";
#endif
FILE* dst_file = fopen(file_name, "wb");
if (!dst_file) {
fprintf(stderr, "fail to open file: %s\n", file_name);
return -1;
}
double 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, "fail to swr_convert\n");
return -1;
}
int dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "fail to av_samples_get_buffer_size\n");
return -1;
}
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);
const char* fmt = nullptr;
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0) {
fprintf(stderr, "fail to get_format_from_sample_fmt");
return -1;
}
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 0;
}
执行结果如下所示: