FFMPEG工程浩大,可以參考的書籍又不是很多,因此很多剛學習FFMPEG的人常常感覺到無從下手。
在此我把自己做項目過程中實現的一個非常簡單的音頻播放器大約200行代碼)源代碼傳上來,以作備忘,同時方便新手學習FFMPEG。
該播放器雖然簡單,但是幾乎包含了使用FFMPEG播放一個音頻所有必備的API,並且使用SDL輸出解碼出來的音頻。
並且支持流媒體等多種音頻輸入。
程序使用了新的FFMPEG類庫,和早期版本的FFMPEG類庫的API函數略有不同。
平台使用VC2010
注意:
1.程序輸出的解碼後PCM音頻數據可以使用Audition打開播放
2.m4a,aac文件可以直接播放。mp3文件需要調整SDL音頻幀大小為4608默認是4096),否則播放會不流暢
3.也可以播放視頻中的音頻
貼上程序代碼:
// //FFMPEG+SDL音頻解碼程序 //雷霄骅 //中國傳媒大學/數字電視技術 //[email protected] // // #include <stdlib.h> #include <string.h> extern "C" { #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" //SDL #include "sdl/SDL.h" #include "sdl/SDL_thread.h" }; #include "decoder.h" //#include "wave.h" //#define _WAVE_ //全局變量--------------------- static Uint8 *audio_chunk; static Uint32 audio_len; static Uint8 *audio_pos; //----------------- /* The audio function callback takes the following parameters: stream: A pointer to the audio buffer to be filled len: The length (in bytes) of the audio buffer (這是固定的4096?) 回調函數 注意:mp3為什麼播放不順暢? len=4096;audio_len=4608;兩個相差512!為了這512,還得再調用一次回調函數。。。 m4a,aac就不存在此問題(都是4096)! */ void fill_audio(void *udata,Uint8 *stream,int len){ /* Only play if we have data left */ if(audio_len==0) return; /* Mix as much data as possible */ len=(len>audio_len?audio_len:len); SDL_MixAudio(stream,audio_pos,len,SDL_MIX_MAXVOLUME); audio_pos += len; audio_len -= len; } //----------------- int decode_audio(char* no_use) { AVFormatContext *pFormatCtx; int i, audioStream; AVCodecContext *pCodecCtx; AVCodec *pCodec; char url[300]={0}; strcpy(url,no_use); //Register all available file formats and codecs av_register_all(); //支持網絡流輸入 avformat_network_init(); //初始化 pFormatCtx = avformat_alloc_context(); //有參數avdic //if(avformat_open_input(&pFormatCtx,url,NULL,&avdic)!=0){ if(avformat_open_input(&pFormatCtx,url,NULL,NULL)!=0){ printf("Couldn't open file.\n"); return -1; } // Retrieve stream information if(av_find_stream_info(pFormatCtx)<0) { printf("Couldn't find stream information.\n"); return -1; } // Dump valid information onto standard error av_dump_format(pFormatCtx, 0, url, false); // Find the first audio stream audioStream=-1; for(i=0; i < pFormatCtx->nb_streams; i++) //原為codec_type==CODEC_TYPE_AUDIO if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO) { audioStream=i; break; } if(audioStream==-1) { printf("Didn't find a audio stream.\n"); return -1; } // Get a pointer to the codec context for the audio stream pCodecCtx=pFormatCtx->streams[audioStream]->codec; // Find the decoder for the audio stream pCodec=avcodec_find_decoder(pCodecCtx->codec_id); if(pCodec==NULL) { printf("Codec not found.\n"); return -1; } // Open codec if(avcodec_open(pCodecCtx, pCodec)<0) { printf("Could not open codec.\n"); return -1; } /********* For output file ******************/ FILE *pFile; #ifdef _WAVE_ pFile=fopen("output.wav", "wb"); fseek(pFile, 44, SEEK_SET); //預留文件頭的位置 #else pFile=fopen("output.pcm", "wb"); #endif // Open the time stamp file FILE *pTSFile; pTSFile=fopen("audio_time_stamp.txt", "wb"); if(pTSFile==NULL) { printf("Could not open output file.\n"); return -1; } fprintf(pTSFile, "Time Base: %d/%d\n", pCodecCtx->time_base.num, pCodecCtx->time_base.den); /*** Write audio into file ******/ //把結構體改為指針 AVPacket *packet=(AVPacket *)malloc(sizeof(AVPacket)); av_init_packet(packet); //音頻和視頻解碼更加統一! //新加 AVFrame *pFrame; pFrame=avcodec_alloc_frame(); //---------SDL-------------------------------------- //初始化 if(SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) { printf( "Could not initialize SDL - %s\n", SDL_GetError()); exit(1); } //結構體,包含PCM數據的相關信息 SDL_AudioSpec wanted_spec; wanted_spec.freq = pCodecCtx->sample_rate; wanted_spec.format = AUDIO_S16SYS; wanted_spec.channels = pCodecCtx->channels; wanted_spec.silence = 0; wanted_spec.samples = 1024; //播放AAC,M4a,緩沖區的大小 //wanted_spec.samples = 1152; //播放MP3,WMA時候用 wanted_spec.callback = fill_audio; wanted_spec.userdata = pCodecCtx; if (SDL_OpenAudio(&wanted_spec, NULL)<0)//步驟2)打開音頻設備 { printf("can't open audio.\n"); return 0; } //----------------------------------------------------- printf("比特率 %3d\n", pFormatCtx->bit_rate); printf("解碼器名稱 %s\n", pCodecCtx->codec->long_name); printf("time_base %d \n", pCodecCtx->time_base); printf("聲道數 %d \n", pCodecCtx->channels); printf("sample per second %d \n", pCodecCtx->sample_rate); //新版不再需要 // short decompressed_audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]; // int decompressed_audio_buf_size; uint32_t ret,len = 0; int got_picture; int index = 0; while(av_read_frame(pFormatCtx, packet)>=0) { if(packet->stream_index==audioStream) { //decompressed_audio_buf_size = (AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2; //原為avcodec_decode_audio2 //ret = avcodec_decode_audio4( pCodecCtx, decompressed_audio_buf, //&decompressed_audio_buf_size, packet.data, packet.size ); //改為 ret = avcodec_decode_audio4( pCodecCtx, pFrame, &got_picture, packet); if ( ret < 0 ) // if error len = -1 { printf("Error in decoding audio frame.\n"); exit(0); } if ( got_picture > 0 ) { #if 1 printf("index %3d\n", index); printf("pts %5d\n", packet->pts); printf("dts %5d\n", packet->dts); printf("packet_size %5d\n", packet->size); //printf("test %s\n", rtmp->m_inChunkSize); #endif //直接寫入 //注意:數據是data0】,長度是linesize0】 #if 1 fwrite(pFrame->data[0], 1, pFrame->linesize[0], pFile); //fwrite(pFrame, 1, got_picture, pFile); //len+=got_picture; index++; //fprintf(pTSFile, "%4d,%5d,%8d\n", index, decompressed_audio_buf_size, packet.pts); #endif } #if 1 //--------------------------------------- //printf("begin....\n"); //設置音頻數據緩沖,PCM數據 audio_chunk = (Uint8*) pFrame->data[0]; //設置音頻數據長度 audio_len = pFrame->linesize[0]; //audio_len = 4096; //播放mp3的時候改為audio_len = 4096 //則會比較流暢,但是聲音會變調!MP3一幀長度4608 //使用一次回調函數4096字節緩沖)播放不完,所以還要使用一次回調函數,導致播放緩慢。。。 //設置初始播放位置 audio_pos = audio_chunk; //回放音頻數據 SDL_PauseAudio(0); //printf("don't close, audio playing...\n"); while(audio_len>0)//等待直到音頻數據播放完畢! SDL_Delay(1); //--------------------------------------- #endif } // Free the packet that was allocated by av_read_frame //已改 av_free_packet(packet); } //printf("The length of PCM data is %d bytes.\n", len); #ifdef _WAVE_ fseek(pFile, 0, SEEK_SET); struct WAVE_HEADER wh; memcpy(wh.header.RiffID, "RIFF", 4); wh.header.RiffSize = 36 + len; memcpy(wh.header.RiffFormat, "WAVE", 4); memcpy(wh.format.FmtID, "fmt ", 4); wh.format.FmtSize = 16; wh.format.wavFormat.FormatTag = 1; wh.format.wavFormat.Channels = pCodecCtx->channels; wh.format.wavFormat.SamplesRate = pCodecCtx->sample_rate; wh.format.wavFormat.BitsPerSample = 16; calformat(wh.format.wavFormat); //Calculate AvgBytesRate and BlockAlign memcpy(wh.data.DataID, "data", 4); wh.data.DataSize = len; fwrite(&wh, 1, sizeof(wh), pFile); #endif SDL_CloseAudio();//關閉音頻設備 // Close file fclose(pFile); // Close the codec avcodec_close(pCodecCtx); // Close the video file av_close_input_file(pFormatCtx); return 0; }
程序會打印每一幀的信息
運行截圖:
完整工程文件路徑:
http://down.51cto.com/data/949383
本文出自 “leixiaohua1020視音頻技術” 博客,請務必保留此出處http://leixiaohua1020.blog.51cto.com/3974648/1298616