最简单的基于FFMPEG+SDL的音频播放器 ver2 （采用SDL2.0）

标签：ffmpeg   音频   sdl2   播放   解码   

简介

之前做过一个简单的音频播放器：《最简单的基于FFMPEG+SDL的音频播放器》，采用的是SDL1.2。前两天刚把原先做的《最简单的基于FFMPEG+SDL的视频播放器》更新采用了SDL2.0，于是顺手也把音频播放器更新成为SDL2.0.

SourceForge项目主页：https://sourceforge.net/projects/simplestffmpegaudioplayer/

完整工程下载地址：http://download.csdn.net/detail/leixiaohua1020/7850021

需要注意的是，与播放视频有很大的不同，SDL2.0播放音频的函数相对于SDL1.2来说变化很小。基本上保持了不变。

除了使用SDL2.0之外，修改了如下地方：

*重建了工程，删掉了不必要的代码，把代码修改得更规范更易懂。

*可以通过宏控制是否使用SDL，以及是否输出PCM。

*支持MP3,AAC等多种格式

源代码

/**
 * 最简单的基于FFmpeg的音频播放器 2 (SDL 2.0)
 * Simplest FFmpeg Audio Player 2 (SDL 2.0)
 *
 * 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 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);
 *
 * 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。
 *
 * This software decode and play audio streams.
 *
 * Version 2.0
 */
#include "stdafx.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libswresample/swresample.h"
//SDL
#include "sdl/SDL.h"
#include "sdl/SDL_thread.h"
};

#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio


//Output PCM
#define OUTPUT_PCM 0
//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 
 * 回调函数
*/ 
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 _tmain(int argc, _TCHAR* argv[])
{
	AVFormatContext	*pFormatCtx;
	int				i, audioStream;
	AVCodecContext	*pCodecCtx;
	AVCodec			*pCodec;

	char url[]="WavinFlag.aac";

	av_register_all();
	//支持网络流输入
	avformat_network_init();
	//初始化
	pFormatCtx = avformat_alloc_context();
	//打开
	if(avformat_open_input(&pFormatCtx,url,NULL,NULL)!=0){
		printf("Couldn‘t open input stream.\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++)
		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;
	}

	FILE *pFile=NULL;
#if OUTPUT_PCM
	pFile=fopen("output.pcm", "wb");
#endif

	AVPacket *packet=(AVPacket *)malloc(sizeof(AVPacket));
	av_init_packet(packet);

	//Out Audio Param
	uint64_t out_channel_layout=AV_CH_LAYOUT_STEREO;
	int out_nb_samples=1024;
	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);
	//输出内存大小
	int out_buffer_size=av_samples_get_buffer_size(NULL,out_channels ,out_nb_samples,out_sample_fmt, 1);

	uint8_t *out_buffer=(uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE*2);

	AVFrame	*pFrame;
	pFrame=avcodec_alloc_frame();
//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
	SDL_AudioSpec wanted_spec;
	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
	printf("Bitrate:\t %3d\n", pFormatCtx->bit_rate);
	printf("Decoder Name:\t %s\n", pCodecCtx->codec->long_name);
	printf("Channels:\t %d\n", pCodecCtx->channels);
	printf("Sample per Second\t %d \n", pCodecCtx->sample_rate);

	uint32_t ret,len = 0;
	int got_picture;
	int index = 0;
	struct SwrContext *au_convert_ctx;
	au_convert_ctx = swr_alloc();
	au_convert_ctx=swr_alloc_set_opts(au_convert_ctx,out_channel_layout, out_sample_fmt, out_sample_rate,
		pCodecCtx->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:%10d\t packet size:%d\n",index,packet->pts,packet->size);
#endif
#if OUTPUT_PCM
				//Write PCM
				fwrite(out_buffer, 1, out_linesize, pFile);
#endif
				//FIX:FLAC,MP3,AAC Different number of samples
				if(wanted_spec.samples!=pFrame->nb_samples){
					SDL_CloseAudio();
					out_nb_samples=pFrame->nb_samples;
					out_buffer_size=av_samples_get_buffer_size(NULL,out_channels ,out_nb_samples,out_sample_fmt, 1);
					wanted_spec.samples=out_nb_samples;
					SDL_OpenAudio(&wanted_spec, NULL);
				}
				index++;
			}
//SDL------------------
#if USE_SDL
			//设置音频数据缓冲,PCM数据
			audio_chunk = (Uint8 *) out_buffer; 
			//设置音频数据长度
			audio_len =out_buffer_size;

			audio_pos = audio_chunk;
			//回放音频数据 
			SDL_PauseAudio(0);
			while(audio_len>0)//等待直到音频数据播放完毕! 
				SDL_Delay(1); 
#endif
		}
		av_free_packet(packet);
	}

	swr_free(&au_convert_ctx);

#if USE_SDL
	SDL_CloseAudio();//关闭音频设备 
	SDL_Quit();
#endif
	// Close file
	fclose(pFile);

	av_free(out_buffer);
	// Close the codec
	avcodec_close(pCodecCtx);
	// Close the video file
	av_close_input_file(pFormatCtx);

	return 0;
}

最简单的基于FFMPEG+SDL的音频播放器 ver2 （采用SDL2.0）

标签：ffmpeg   音频   sdl2   播放   解码   

原文地址：http://blog.csdn.net/leixiaohua1020/article/details/38979615