www.pudn.com > MP3Dec2Wav_04.rar > MP3Dec2Wav.c

 
#include  
//#include "mpglibdll.h" 
 
// wfz_added headerfile----- 
//#include  
//#include  
//#include  
//#include  
#include  
#include "audio.h" 
#include "mpglib\mpg123.h" 
 
 
 
//HINSTANCE    hmpglibdll; 
//INITMP3      InitMP3; 
//EXITMP3      ExitMP3; 
//DECODEMP3    decodeMP3; 
 
#define NO_DLL_FOUND -1 
 
char buf[16384]; 
struct mpstr mp; 
 
extern long freqs[9]; // wfz_added 
 
int main (void) 
{ 
	int size; 
	char out[8192]; 
	int len,ret; 
	FILE *pFileRead; 
 	audio_file *aufile; 
 
	char mp3FileName[100]; 
	char wavFileName[100]; 
	int samplesperframe; 
	int bytespersample; 
 
	int samplerate   = 44100; 
	int channels     = 2; 
	int outputFormat = FAAD_FMT_16BIT;  //only support 16bit now, otherwise error 
	int fileType     =  OUTPUT_WAV; 
	int channelMask  = 0; 
	BOOL firstbuf = TRUE; 
	//------print log info----- 
	FILE *pLogFile = fopen("d:\\Mp3toWav.txt","wa"); 
	char log[150]; 
 
 
	switch(outputFormat) 
	{ 
	case FAAD_FMT_16BIT: 
		bytespersample = 2; 
		break; 
	case FAAD_FMT_24BIT: 
		bytespersample = 3; 
		break; 
	case FAAD_FMT_32BIT: 
		bytespersample = 4; 
		break; 
	default: 
		bytespersample = 2; 
	} 
	strcpy(mp3FileName, "phonexb.mp2"); 
	strcpy(wavFileName, "result.wav"); 
 
//------ 
	// Open file for input:  
	if( (pFileRead = fopen( mp3FileName, "rb" )) == NULL ) 
	{ 
		perror( "open jsb.mp3 failed on input file" ); 
		exit( 1 ); 
	} 
	// Open file for output:  
	aufile = open_audio_file(wavFileName, samplerate, channels, 
		outputFormat, fileType, channelMask); 
	 
	if(aufile->sndfile  == NULL) 
	{ 
		perror( "Open failed on output file" ); 
		exit( 1 ); 
	} 
//------- 
 
 
	InitMP3(&mp); 
 
	while( !feof(pFileRead) ) 
	{ 
		len = fread(buf, sizeof(char), 10000, pFileRead); 
		if(len <= 0) 
			break; 
		ret = decodeMP3(&mp,buf,len,out,8192,&size); 
		if(firstbuf) 
		{ 
			aufile->samplerate = freqs[mp.fr.sampling_frequency];  
			firstbuf = FALSE; 
		} 
		while(ret == MP3_OK) 
		{ 
			samplesperframe = size/bytespersample; 
       		write_audio_file(aufile, out, samplesperframe, 0); 
			if(samplesperframe != 2304) // 1152 * 2 channels = 2304 
			{ 
				sprintf(log, " samples of current frame is %d \n", samplesperframe); 
				fwrite(log, 1, strlen(log), pLogFile);  
			} 
			ret = decodeMP3(&mp,NULL,0,out,8192,&size); 
		} 
	} 
 
	//aufile->samplerate = mp.fr.sampling_frequency; //save the real samplerate 
	ExitMP3(&mp); 
	 
	fclose(pFileRead); 
	fclose(pLogFile); 
    close_audio_file(aufile); 
 
	return (0); 
}