/** 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>staticintget_format_from_sample_fmt(constchar**fmt,enumAVSampleFormat sample_fmt){int i;structsample_fmt_entry{enumAVSampleFormat sample_fmt;constchar*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++){structsample_fmt_entry*entry =&sample_fmt_entries[i];if(sample_fmt == entry->sample_fmt){*fmt =AV_NE(entry->fmt_be, entry->fmt_le);return0;}}fprintf(stderr,"Sample format %s not supported as output format\n",av_get_sample_fmt_name(sample_fmt));returnAVERROR(EINVAL);}/*** Fill dst buffer with nb_samples, generated starting from t. 交错模式的*/staticvoidfill_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;constdouble 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;}}intmain(int argc,char**argv){// 输入参数int64_t src_ch_layout = AV_CH_LAYOUT_STEREO;int src_rate =48000;enumAVSampleFormat 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;enumAVSampleFormat 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constchar*dst_filename =NULL;// 保存输出的pcm到本地，然后播放验证FILE *dst_file;int dst_bufsize;constchar*fmt;// 重采样实例structSwrContext*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,(constuint8_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;}
