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最简单的基于FFmpeg的音频播放器系列文章列表:
《最简单的基于FFMPEG+SDL的音频播放器》
《最简单的基于FFMPEG+SDL的音频播放器 ver2 (采用SDL2.0)》
《最简单的基于FFMPEG+SDL的音频播放器:拆分-解码器和播放器》
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本文补充记录《最简单的基于FFMPEG+SDL的音频播放器》中的两个例子:FFmpeg音频解码器和SDL音频采样数据播放器。这两个部分是从音频播放器中拆分出来的两个例子。FFmpeg音频解码器实现了视频数据到PCM采样数据的解码,而SDL音频采样数据播放器实现了PCM数据到音频设备的播放。简而言之,原先的FFmpeg+SDL音频播放器实现了:
音频数据->PCM->音频设备
FFmpeg音频解码器实现了:
音频数据->PCM
SDL音频采样数据播放器实现了:
/** * 最简单的基于FFmpeg的音频解码器 * Simplest FFmpeg Audio Decoder * * 雷霄骅 Lei Xiaohua * [email protected] * 中国传媒大学/数字电视技术 * Communication University of China / Digital TV Technology * http://blog.csdn.net/leixiaohua1020 * * 本程序可以将音频码流(mp3,AAC等)解码为PCM采样数据。 * 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。 * * This software decode audio streams (AAC,MP3 ...) to PCM data. * Suitable for beginner of FFmpeg. * */ #include <stdio.h> #include <stdlib.h> #include <string.h> #define __STDC_CONSTANT_MACROS #ifdef _WIN32 //Windows extern "C" { #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libswresample/swresample.h" }; #else //Linux... #ifdef __cplusplus extern "C" { #endif #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <libswresample/swresample.h> #ifdef __cplusplus }; #endif #endif #define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio int main(int argc, char* argv[]) { AVFormatContext *pFormatCtx; int i, audioStream; AVCodecContext *pCodecCtx; AVCodec *pCodec; AVPacket *packet; uint8_t *out_buffer; AVFrame *pFrame; int ret; uint32_t len = 0; int got_picture; int index = 0; int64_t in_channel_layout; struct SwrContext *au_convert_ctx; FILE *pFile=fopen("output.pcm", "wb"); char url[]="skycity1.mp3"; 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; } 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(); //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); printf("index:%5d\t pts:%lld\t packet size:%d\n",index,packet->pts,packet->size); //Write PCM fwrite(out_buffer, 1, out_buffer_size, pFile); index++; } } av_free_packet(packet); } swr_free(&au_convert_ctx); fclose(pFile); av_free(out_buffer); // Close the codec avcodec_close(pCodecCtx); // Close the video file avformat_close_input(&pFormatCtx); return 0; }
/** * 最简单的SDL2播放音频的例子(SDL2播放PCM) * Simplest Audio Play SDL2 (SDL2 play PCM) * * 雷霄骅 Lei Xiaohua * [email protected] * 中国传媒大学/数字电视技术 * Communication University of China / Digital TV Technology * http://blog.csdn.net/leixiaohua1020 * * 本程序使用SDL2播放PCM音频采样数据。SDL实际上是对底层绘图 * API(Direct3D,OpenGL)的封装,使用起来明显简单于直接调用底层 * API。 * * 函数调用步骤如下: * * [初始化] * SDL_Init(): 初始化SDL。 * SDL_OpenAudio(): 根据参数(存储于SDL_AudioSpec)打开音频设备。 * * [循环播放数据] * SDL_PauseAudio(): 播放音频数据。 * SDL_Delay(): 延时等待播放完成。 * * This software plays PCM raw audio data using SDL2. * SDL is a wrapper of low-level API (DirectSound). * Use SDL is much easier than directly call these low-level API. * * The process is shown as follows: * * [Init] * SDL_Init(): Init SDL. * SDL_OpenAudio(): Opens the audio device with the desired * parameters (In SDL_AudioSpec). * * [Loop to play data] * SDL_PauseAudio(): Play Audio. * SDL_Delay(): Wait for completetion of playback. */ #include <stdio.h> #include <tchar.h> extern "C" { #include "sdl/SDL.h" }; //Buffer: //|-----------|-------------| //chunk-------pos---len-----| static Uint8 *audio_chunk; static Uint32 audio_len; static Uint8 *audio_pos; /* Audio Callback * 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[]) { //Init if(SDL_Init(SDL_INIT_AUDIO | SDL_INIT_TIMER)) { printf( "Could not initialize SDL - %s\n", SDL_GetError()); return -1; } //SDL_AudioSpec SDL_AudioSpec wanted_spec; wanted_spec.freq = 44100; wanted_spec.format = AUDIO_S16SYS; wanted_spec.channels = 2; wanted_spec.silence = 0; wanted_spec.samples = 1024; wanted_spec.callback = fill_audio; if (SDL_OpenAudio(&wanted_spec, NULL)<0){ printf("can't open audio.\n"); return -1; } FILE *fp=fopen("NocturneNo2inEflat_44.1k_s16le.pcm","rb+"); if(fp==NULL){ printf("cannot open this file\n"); return -1; } //For YUV420P int pcm_buffer_size=4096; char *pcm_buffer=(char *)malloc(pcm_buffer_size); int data_count=0; while(1){ if (fread(pcm_buffer, 1, pcm_buffer_size, fp) != pcm_buffer_size){ // Loop fseek(fp, 0, SEEK_SET); fread(pcm_buffer, 1, pcm_buffer_size, fp); data_count=0; } printf("Now Playing %10d Bytes data.\n",data_count); data_count+=pcm_buffer_size; //Set audio buffer (PCM data) audio_chunk = (Uint8 *) pcm_buffer; //Audio buffer length audio_len =pcm_buffer_size; audio_pos = audio_chunk; //Play SDL_PauseAudio(0); while(audio_len>0)//Wait until finish SDL_Delay(1); } free(pcm_buffer); SDL_Quit(); return 0; }
Github:https://github.com/leixiaohua1020/simplest_ffmpeg_audio_player
开源中国:http://git.oschina.net/leixiaohua1020/simplest_ffmpeg_audio_player
CSDN下载地址:http://download.csdn.net/detail/leixiaohua1020/8924329
simplest_ffmpeg_audio_player:基于FFmpeg+SDL的音频解码器
simplest_ffmpeg_audio_decoder:音频解码器。使用了libavcodec和libavformat。
simplest_audio_play_sdl2:使用SDL2播放PCM采样数据的例子。