libswresample庫功能主要包括高度優化的音頻重采樣、rematrixing和樣本格式轉換操作。

以下是測試代碼(test_ffmpeg_libswresample.cpp)，對音頻了解較少，測試代碼是參考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
}
#endifnamespace {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 channelsfor (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;}
}} // namespaceint test_ffmpeg_libswresample_resample()
{// reference: doc/examples/resample_audio.cfprintf(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 contextstruct 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 optionsav_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 contextif (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 bufferssrc_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 samplesmax_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 alignmentdst_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_VERconst char* file_name = "E:/GitCode/OpenCV_Test/test_images/xxx";
#elseconst char* file_name = "test_images/xxx";	
#endifFILE* 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 audiofill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);// compute destination number of samplesdst_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 formatret = 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;
}

執行結果如下所示：

GitHub：?https://github.com/fengbingchun/OpenCV_Test?

總結

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