基于FFmpeg， 实现 8k 16bit 立体声 转码 16k 16bit 单声道。

基于FFmpeg， 实现 8k 16bit 立体声 转码 16k 16bit 单声道。

命令 🐳​

	ffmpeg -i original.wav -ac 1 -ar 16000 new.wav

CODE 🚀​

1. 设置重采样数据

   1. 输入: 立体声, 双通道, 8k, 16bit
输出: 单声道, 单通道, 16k, 16bit

   2. // 准备重采样
        	struct SwrContext* swr = swr_alloc();
        	av_opt_set_int(swr, "in_channel_count", pCodecCtx->channels, 0);
        	av_opt_set_int(swr, "out_channel_count", DST_CHANNELS, 0);
        	av_opt_set_int(swr, "in_channel_layout", pCodecCtx->channel_layout, 0);
        	av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_MONO, 0);
        	av_opt_set_int(swr, "in_sample_rate", pCodecCtx->sample_rate, 0);
        	av_opt_set_int(swr, "out_sample_rate", DST_SAMPLE_RATE, 0);
        	av_opt_set_sample_fmt(swr, "in_sample_fmt", pCodecCtx->sample_fmt, 0);
        	av_opt_set_sample_fmt(swr, "out_sample_fmt", DST_SAMPLE_FMT, 0);
        	// 初始化
        	swr_init(swr);
        	if (!swr_is_initialized(swr)) {
        		fprintf(stderr, "Resampler has not been properly initialized\n");
        		return NULL;
        	}
      

2. 重采样

   1. 解码一帧音频

      1. // 解码一帧音频
         if (avcodec_decode_audio4(pCodecCtx, frame, &gotFrame, &packet) < 0) 
           	break;
           		
         if (!gotFrame) {
           	printf("avcodec_decode_audio4 error. \n");
           	continue;
          }
         

   2. 计算实际重采样后得到的数据大小, 然后重新分配缓存

      1. int dst_nb_samples = av_rescale_rnd(swr_get_delay(swr, DST_SAMPLE_RATE) +  frame->nb_samples,DST_SAMPLE_RATE, pCodecCtx->sample_rate, AV_ROUND_UP);
         

   3. 由于双声道转单声道，所以输入的通道个数等于输出通道个数的2倍。

      1. int outSize = dst_nb_samples * 2;
         int inSize = frame->nb_samples;
         

   4. 据音频格式分配相应大小的内存空间

      1. uint8_t** buffer;
         av_samples_alloc((uint8_t**)&buffer, NULL, DST_CHANNELS, outSize,  DST_SAMPLE_FMT, 0);
         

   5. 进行重采样

      1. // 重采样
          int frame_count = swr_convert(swr, (uint8_t**)&buffer, outSize,
         	(const uint8_t**)frame->data, inSize);
         
          if (frame_count < 0) {
         	av_freep(&buffer);
         	printf("swr_convert error\n");
         	break;
           }
         

   6. 获取给定音频参数所需的缓冲区大小。

      1. dst_bufsize = av_samples_get_buffer_size(&dst_linesize, DST_CHANNELS,
           			frame_count, DST_SAMPLE_FMT, 0);
         if (dst_bufsize < 0) {
           	av_freep(&buffer);
           	printf("av_samples_get_buffer_size error\n");
           	return -1;
          }
         

   7. 写入 wav 文件

      1. audioFileSize += dst_bufsize;
         fwrite(buffer, 1, dst_bufsize, p);
         fflush(p);
         av_freep(&buffer);
         

   8. 写入文件长度和数据长度 ，最后关闭文件

      1. fseek(p, 40, SEEK_SET);
         fwrite(&audioFileSize, 1, sizeof(int32_t), p);
         audioFileSize += 36;
         fseek(p, 4, SEEK_SET);
         fwrite(&audioFileSize, 1, sizeof(int32_t), p);
         fclose(p);
         

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DEMO: https://download.csdn.net/download/haiyangyunbao813/17984367

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参考博客 🅱️​

1. https://blog.csdn.net/weixin_41353840/article/details/112568105?utm_medium=distribute.pc_relevant_download.none-task-blog-baidujs-1.nonecase&depth_1-utm_source=distribute.pc_relevant_download.none-task-blog-baidujs-1.nonecase
2. https://blog.csdn.net/sun007700/article/details/115034806