Android中间层分析2.【media】音视频基础知识-7.ffmpeg 简单音频播放器

哎，喜欢偷懒，这边直接抄袭下雷神的代码。雷神是个值得敬佩的程序员。

vs代码下载链接：
https://pan.baidu.com/s/1c2dIuYk 密码：ld4b

/* 
 *最简单的基于FFmpeg的音频播放器
 *Simplest FFmpeg Audio Player
 *
 *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。
 *
 */
#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"
};
#include "decoder.h"

//#define _WAVE_

//全局变量---------------------
	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 (这是固定的4096？)
	回调函数
	注意：mp3为什么播放不顺畅？
	len=4096;audio_len=4608;两个相差512！为了这512，还得再调用一次回调函数。。。
	m4a,aac就不存在此问题(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);

	av_register_all();
	//支持网络流输入ff
	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;
#ifdef _WAVE_
	pFile=fopen("output.wav", "wb");
	fseek(pFile, 44, SEEK_SET); //预留文件头的位置
#else
	pFile=fopen("output1.pcm", "wb");
#endif

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

	AVFrame	*pFrame;
	pFrame=avcodec_alloc_frame();
	//输出音频数据大小，一定小于输出内存。
	int out_linesize;
	//输出内存大小
	int out_buffer_size=av_samples_get_buffer_size(&out_linesize, pCodecCtx->channels,pCodecCtx->frame_size,pCodecCtx->sample_fmt, 1);
	uint8_t *out_buffer=new uint8_t[out_buffer_size];
	//---------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);
	}

	//结构体，包含PCM数据的相关信息
	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，缓冲区的大小
	wanted_spec.samples = 1152; //播放MP3，WMA时候用
	wanted_spec.callback = fill_audio; 
	wanted_spec.userdata = pCodecCtx; 

	if (SDL_OpenAudio(&wanted_spec, NULL)<0)//步骤（2）打开音频设备 
	{ 
		printf("can't open audio.\n"); 
		return 0; 
	} 
	//-----------------------------------------------------
	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,AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, 44100,
		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");
                exit(0);
            }
			if ( got_picture > 0 )
			{
#if 1
				swr_convert(au_convert_ctx,&out_buffer, out_linesize,(const uint8_t **)pFrame->data , pFrame->nb_samples);

				printf("index:%5d\t pts %5d\n", index,packet->pts);
#endif
				//直接写入
#if 1
				fwrite(out_buffer, 1, out_linesize, pFile);
#endif
				index++;
			}
#if 1
			//---------------------------------------
			//printf("begin....\n"); 
			//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_linesize;
			audio_len = 4096;
			//播放mp3的时候改为audio_len = 4096
			//则会比较流畅，但是声音会变调！MP3一帧长度4608
			//使用一次回调函数（4096字节缓冲）播放不完，所以还要使用一次回调函数，导致播放缓慢。。。
			//设置初始播放位置
			audio_pos = audio_chunk;
			//回放音频数据 
			SDL_PauseAudio(0);
			//printf("don't close, audio playing...\n"); 
			while(audio_len>0)//等待直到音频数据播放完毕! 
				SDL_Delay(1); 
			//---------------------------------------
#endif
		}
		av_free_packet(packet);
	}

#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();//关闭音频设备 
	// Close file
	fclose(pFile);
	// Close the codec
	avcodec_close(pCodecCtx);
	// Close the video file
	av_close_input_file(pFormatCtx);

	return 0;
}