音视频实战---音频重采样

1、使用swr_alloc()创建重采样实例

2、使用av_opt_set_int函数设置重采样输入输出参数

3、使用swr_init函数初始化重采样器

4、使用av_get_channel_layout_nb_channels函数计算输入源的通道数

5、给输入源分配内存空间–av_samples_alloc_array_and_samples

6、计算输出采样数量–av_rescale_rnd

7、分配输出缓存内存–av_samples_alloc_array_and_samples

8、计算输出缓冲区所需的延迟大小，以便进行合理的处理和同步。–swr_get_delay

9、计算加上延迟采样数后的输出采样数量—av_rescale_rnd

10、将音频进行重采样转换–swr_convert

11、获取重采样后的音频数据–av_samples_get_buffer_size

/** 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.*//*** @example resampling_audio.c* libswresample API use example.*/#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. 交错模式的*/
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)
{// 输入参数int64_t src_ch_layout = AV_CH_LAYOUT_STEREO;int src_rate = 48000;enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL;int src_nb_channels = 0;uint8_t **src_data = NULL;  // 二级指针int src_linesize;int src_nb_samples = 1024;// 输出参数int64_t dst_ch_layout = AV_CH_LAYOUT_STEREO;int dst_rate = 44100;enum AVSampleFormat dst_sample_fmt = AV_SAMPLE_FMT_S16;int dst_nb_channels = 0;uint8_t **dst_data = NULL;  //二级指针int dst_linesize;int dst_nb_samples;int max_dst_nb_samples;// 输出文件const char *dst_filename = NULL;    // 保存输出的pcm到本地，然后播放验证FILE *dst_file;int dst_bufsize;const char *fmt;// 重采样实例struct SwrContext *swr_ctx;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);}// 创建重采样器/* create resampler context */swr_ctx = swr_alloc();if (!swr_ctx) {fprintf(stderr, "Could not allocate resampler context\n");ret = AVERROR(ENOMEM);goto end;}// 设置重采样参数/* set options */// 输入参数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);// 输出参数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);// 初始化重采样/* 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 */// 计算出输入源的通道数量src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);// 给输入源分配内存空间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 */// 计算输出采样数量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);// 分配输出缓存内存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 */// 生成输入源fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);/* compute destination number of samples */int64_t delay = swr_get_delay(swr_ctx, src_rate);dst_nb_samples = av_rescale_rnd(delay + 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;}//        int fifo_size = swr_get_out_samples(swr_ctx,src_nb_samples);//        printf("fifo_size:%d\n", fifo_size);//        if(fifo_size < 1024)//            continue;/* convert to destination format *///        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);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);ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, NULL, 0);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("flush in:%d out:%d\n", 0, ret);fwrite(dst_data[0], 1, dst_bufsize, dst_file);if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)goto end;fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n""ffplay -f %s -channel_layout %lld -channels %d -ar %d %s\n",fmt, dst_ch_layout, 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;
}