最简单的基于FFMPEG+SDL的音频播放器 2.2

/** * 最简单的基于FFmpeg的音频播放器 2 * Simplest FFmpeg Audio Player 2 * * 雷霄骅 Lei Xiaohua * leixiaohua1020@126.com * 中国传媒大学/数字电视技术 * Communication University of China / Digital TV Technology * http://blog.csdn.net/leixiaohua1020 * * 本程序实现了音频的解码和播放。 * 是最简单的FFmpeg音频解码方面的教程。 * 通过学习本例子可以了解FFmpeg的解码流程。 * * 该版本使用SDL 2.0替换了第一个版本中的SDL 1.0。 * 注意：SDL 2.0中音频解码的API并无变化。唯一变化的地方在于 * 其回调函数的中的Audio Buffer并没有完全初始化，需要手动初始化。 * 本例子中即SDL_memset(stream, 0, len); * * This software decode and play audio streams. * Suitable for beginner of FFmpeg. * * This version use SDL 2.0 instead of SDL 1.2 in version 1 * Note:The good news for audio is that, with one exception, * it's entirely backwards compatible with 1.2. * That one really important exception: The audio callback * does NOT start with a fully initialized buffer anymore. * You must fully write to the buffer in all cases. In this * example it is SDL_memset(stream, 0, len); * * Version 2.0 */ #include #include #include #define __STDC_CONSTANT_MACROS #ifdef _WIN32 //Windows extern "C" { #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libswresample/swresample.h" #include "SDL2/SDL.h" }; #else //Linux... #ifdef __cplusplus extern "C" { #endif #include #include #include #include #ifdef __cplusplus }; #endif #endif #define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio //Output PCM #define OUTPUT_PCM 1 //Use SDL #define USE_SDL 1 //Buffer: //|-----------|-------------| //chunk-------pos---len-----| 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 */ void fill_audio(void *udata,Uint8 *stream,int len){ //SDL 2.0 SDL_memset(stream, 0, len); if(audio_len==0) /* Only play if we have data left */ return; len=(len>audio_len?audio_len:len); /* Mix as much data as possible */ SDL_MixAudio(stream,audio_pos,len,SDL_MIX_MAXVOLUME); audio_pos += len; audio_len -= len; } //----------------- int main(int argc, char* argv[]) { AVFormatContext *pFormatCtx; int i, audioStream; AVCodecContext *pCodecCtx; AVCodec *pCodec; AVPacket *packet; uint8_t *out_buffer; AVFrame *pFrame; SDL_AudioSpec wanted_spec; int ret; uint32_t len = 0; int got_picture; int index = 0; int64_t in_channel_layout; struct SwrContext *au_convert_ctx; FILE *pFile=NULL; char url[]="WavinFlag.aac"; av_register_all(); avformat_network_init(); pFormatCtx = avformat_alloc_context(); //Open if(avformat_open_input(&pFormatCtx,url,NULL,NULL)!=0){ printf("Couldn't open input stream.\n"); return -1; } // Retrieve stream information if(avformat_find_stream_info(pFormatCtx,NULL)<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++) 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_open2(pCodecCtx, pCodec,NULL)<0){ printf("Could not open codec.\n"); return -1; } #if OUTPUT_PCM pFile=fopen("output.pcm", "wb"); #endif packet=(AVPacket *)av_malloc(sizeof(AVPacket)); av_init_packet(packet); //Out Audio Param uint64_t out_channel_layout=AV_CH_LAYOUT_STEREO; //nb_samples: AAC-1024 MP3-1152 int out_nb_samples=pCodecCtx->frame_size; AVSampleFormat out_sample_fmt=AV_SAMPLE_FMT_S16; int out_sample_rate=44100; int out_channels=av_get_channel_layout_nb_channels(out_channel_layout); //Out Buffer Size int out_buffer_size=av_samples_get_buffer_size(NULL,out_channels ,out_nb_samples,out_sample_fmt, 1); out_buffer=(uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE*2); pFrame=av_frame_alloc(); //SDL------------------ #if USE_SDL //Init if(SDL_Init(SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER)) { printf( "Could not initialize SDL - %s\n", SDL_GetError()); return -1; } //SDL_AudioSpec wanted_spec.freq = out_sample_rate; wanted_spec.format = AUDIO_S16SYS; wanted_spec.channels = out_channels; wanted_spec.silence = 0; wanted_spec.samples = out_nb_samples; wanted_spec.callback = fill_audio; wanted_spec.userdata = pCodecCtx; if (SDL_OpenAudio(&wanted_spec, NULL)<0){ printf("can't open audio.\n"); return -1; } #endif //FIX:Some Codec's Context Information is missing in_channel_layout=av_get_default_channel_layout(pCodecCtx->channels); //Swr au_convert_ctx = swr_alloc(); au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,out_channel_layout, out_sample_fmt, out_sample_rate, in_channel_layout,pCodecCtx->sample_fmt , pCodecCtx->sample_rate,0, NULL); swr_init(au_convert_ctx); while(av_read_frame(pFormatCtx, packet)>=0){ if(packet->stream_index==audioStream){ ret = avcodec_decode_audio4( pCodecCtx, pFrame,&got_picture, packet); if ( ret < 0 ) { printf("Error in decoding audio frame.\n"); return -1; } if ( got_picture > 0 ){ swr_convert(au_convert_ctx,&out_buffer, MAX_AUDIO_FRAME_SIZE,(const uint8_t **)pFrame->data , pFrame->nb_samples); #if 1 printf("index:%5d\t pts:%lld\t packet size:%d\n",index,packet->pts,packet->size); #endif #if OUTPUT_PCM //Write PCM fwrite(out_buffer, 1, out_buffer_size, pFile); #endif index++; } #if USE_SDL while(audio_len>0)//Wait until finish SDL_Delay(1); //Set audio buffer (PCM data) audio_chunk = (Uint8 *) out_buffer; //Audio buffer length audio_len =out_buffer_size; audio_pos = audio_chunk; //Play SDL_PauseAudio(0); #endif } av_free_packet(packet); } swr_free(&au_convert_ctx); #if USE_SDL SDL_CloseAudio();//Close SDL SDL_Quit(); #endif // Close file #if OUTPUT_PCM fclose(pFile); #endif av_free(out_buffer); // Close the codec avcodec_close(pCodecCtx); // Close the video file avformat_close_input(&pFormatCtx); return 0; }