get_audio.c

MPEG-4编解码的实现(包括MPEG4视音频编解码)

#include "util.h"
#include "get_audio.h"
#ifdef HAVEGTK
#include "gtkanal.h"
#include <gtk/gtk.h>
#endif

#if (defined LIBSNDFILE || defined LAMESNDFILE)

#ifdef _WIN32
/* needed to set stdin to binary on windoze machines */
#include <io.h>
#endif



static FILE *musicin=NULL;  /* input file pointer */
static unsigned long num_samples;
static int samp_freq;
static int input_bitrate;
static int num_channels;
static int count_samples_carefully;

int read_samples_pcm(lame_global_flags *gfp,short sample_buffer[2304],int frame_size, int samples_to_read);
int read_samples_mp3(lame_global_flags *gfp,FILE *musicin,short int mpg123pcm[2][1152],int num_chan);





void lame_init_infile(lame_global_flags *gfp)
{
  /* open the input file */
  count_samples_carefully=0;
  OpenSndFile(gfp,gfp->inPath,gfp->in_samplerate,gfp->num_channels);  
  /* if GetSndSampleRate is non zero, use it to overwrite the default */
  if (GetSndSampleRate()) gfp->in_samplerate=GetSndSampleRate();
  if (GetSndChannels()) gfp->num_channels=GetSndChannels();
  gfp->num_samples = GetSndSamples();
}
void lame_close_infile(lame_global_flags *gfp)
{
  CloseSndFile(gfp);
}




/************************************************************************
*
* lame_readframe()
*
* PURPOSE:  reads a frame of audio data from a file to the buffer,
*   aligns the data for future processing, and separates the
*   left and right channels
*
*
************************************************************************/
int lame_readframe(lame_global_flags *gfp,short int Buffer[2][1152])
{
  int iread;

  /* note: if input is gfp->stereo and output is mono, get_audio() 
   * will return  .5*(L+R) in channel 0,  and nothing in channel 1. */
  iread = get_audio(gfp,Buffer,gfp->stereo);
  
  /* check to see if we overestimated/underestimated totalframes */
  if (iread==0)  gfp->totalframes = Min(gfp->totalframes,gfp->frameNum+2);
  if (gfp->frameNum > (gfp->totalframes-1)) gfp->totalframes = gfp->frameNum;
  return iread;
}





/************************************************************************
*
* get_audio()
*
* PURPOSE:  reads a frame of audio data from a file to the buffer,
*   aligns the data for future processing, and separates the
*   left and right channels
*
*
************************************************************************/
int get_audio(lame_global_flags *gfp,short buffer[2][1152],int stereo)
{

  int		j;
  short	insamp[2304];
  int samples_read;
  int framesize,samples_to_read;
  static unsigned long num_samples_read;
  unsigned long remaining;
  int num_channels = gfp->num_channels;

  if (gfp->frameNum==0) {
    num_samples_read=0;
    num_samples= GetSndSamples();
  }
  framesize = gfp->mode_gr*576;

  samples_to_read = framesize;
  if (count_samples_carefully) { 
    /* if this flag has been set, then we are carefull to read 
     * exactly num_samples and no more.  This is usefull for .wav and .aiff
     * files which have id3 or other tags at the end.  Note that if you
     * are using LIBSNDFILE, this is not necessary */
    remaining=num_samples-Min(num_samples,num_samples_read);
    if (remaining < (unsigned long)framesize) 
      samples_to_read = remaining;
  }


  if (gfp->input_format==sf_mp3) {
    /* decode an mp3 file for the input */
    samples_read=read_samples_mp3(gfp,musicin,buffer,num_channels);
  }else{
    samples_read = read_samples_pcm(gfp,insamp,num_channels*framesize,num_channels*samples_to_read);
    samples_read /=num_channels;

    for(j=0;j<framesize;j++) {
      buffer[0][j] = insamp[num_channels*j];
      if (num_channels==2) buffer[1][j] = insamp[2*j+1];
      else buffer[1][j]=0;
    }
  }

  /* dont count things in this case to avoid overflows */
  if (num_samples!=MAX_U_32_NUM) num_samples_read += samples_read;
  return(samples_read);

}
  






  
int GetSndBitrate(void)
{
	return input_bitrate;
}






int read_samples_mp3(lame_global_flags *gfp,FILE *musicin,short int mpg123pcm[2][1152],int stereo)
{
#if (defined  AMIGA_MPEGA || defined HAVEMPGLIB) 
  int j,out=0;
#ifdef HAVEGTK
  static int framesize=0;
  int ch;
#endif

  out=lame_decode_fromfile(musicin,mpg123pcm[0],mpg123pcm[1]);
  /* out = -1:  error, probably EOF */
  /* out = 0:   not possible with lame_decode_fromfile() */
  /* out = number of output samples */

  if (out==-1) {
    for ( j = 0; j < 1152; j++ ) {
      mpg123pcm[0][j] = 0;
      mpg123pcm[1][j] = 0;
    }
  }


#ifdef HAVEGTK
  if (gfp->gtkflag) {
    framesize=1152;
    if (out==576) framesize=576;
    
    /* add a delay of framesize-DECDELAY, which will make the total delay
     * exactly one frame, so we can sync MP3 output with WAV input */
    
    for ( ch = 0; ch < stereo; ch++ ) {
      for ( j = 0; j < framesize-DECDELAY; j++ )
	pinfo->pcmdata2[ch][j] = pinfo->pcmdata2[ch][j+framesize];
      for ( j = 0; j < framesize; j++ ) 
	pinfo->pcmdata2[ch][j+framesize-DECDELAY] = mpg123pcm[ch][j];
    }
  
  pinfo->frameNum123 = gfp->frameNum-1;
  pinfo->frameNum = gfp->frameNum;
  }
#endif
  if (out==-1) return 0;
  else return out;
#else
  fprintf(stderr,"Error: libmp3lame was not compiled with I/O support \n");
  exit(1);
#endif
}
#endif  /* LAMESNDFILE or LIBSNDFILE */




#ifdef LIBSNDFILE 
/*
** Copyright (C) 1999 Albert Faber
**  
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.	 See the GNU
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */


#include <stdio.h>

/* External references */

static SNDFILE*	gs_pSndFileIn=NULL;
static SF_INFO	gs_wfInfo;


unsigned long GetSndSamples(void)
{
       return gs_wfInfo.samples;
}
int GetSndSampleRate(void)
{
	return gs_wfInfo.samplerate;
}

int GetSndChannels(void)
{
	return gs_wfInfo.channels;
}

void CloseSndFile(lame_global_flags *gfp)
{
  if (gfp->input_format==sf_mp3) {
#ifndef AMIGA_MPEGA
    if (fclose(musicin) != 0){
      fprintf(stderr, "Could not close audio input file\n");
      exit(2);
    }
#endif
  }else{
	if (gs_pSndFileIn)
	{
		if (sf_close(gs_pSndFileIn) !=0)
		{
			fprintf(stderr, "Could not close sound file \n");
			exit(2);
		}
	}
  }
}



FILE * OpenSndFile(lame_global_flags *gfp,const char* lpszFileName, int default_samp,
int default_channels)
{
  input_bitrate=0;
  if (gfp->input_format==sf_mp3) {
#ifdef AMIGA_MPEGA
    if (-1==lame_decode_initfile(lpszFileName,&num_channels,&samp_freq,&input_bitrate,&num_samples)) {
      fprintf(stderr,"Error reading headers in mp3 input file %s.\n", lpszFileName);
      exit(1);
    }
#endif
#ifdef HAVEMPGLIB
    if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
      fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
      exit(1);
    }
    if (-1==lame_decode_initfile(musicin,&num_channels,&samp_freq,&input_bitrate,&num_samples)) {
      fprintf(stderr,"Error reading headers in mp3 input file %s.\n", lpszFileName);
      exit(1);
    }
#endif
    gs_wfInfo.samples=num_samples;
    gs_wfInfo.channels=num_channels;
    gs_wfInfo.samplerate=samp_freq;

  } else {

    /* Try to open the sound file */
    /* set some defaults incase input is raw PCM */
    gs_wfInfo.seekable=(gfp->input_format!=sf_raw);  /* if user specified -r, set to not seekable */
    gs_wfInfo.samplerate=default_samp;
    gs_wfInfo.pcmbitwidth=16;
    gs_wfInfo.channels=default_channels;
    if (DetermineByteOrder()==order_littleEndian) {
      if (gfp->swapbytes) gs_wfInfo.format=SF_FORMAT_RAW_BE;
      else gs_wfInfo.format=SF_FORMAT_RAW_LE;
    } else {
      if (gfp->swapbytes) gs_wfInfo.format=SF_FORMAT_RAW_LE;
      else gs_wfInfo.format=SF_FORMAT_RAW_BE;
    }

    gs_pSndFileIn=sf_open_read(lpszFileName,&gs_wfInfo);

        /* Check result */
	if (gs_pSndFileIn==NULL)
	{
	        sf_perror(gs_pSndFileIn);
		fprintf(stderr, "Could not open sound file \"%s\".\n", lpszFileName);
		exit(1);
	}

    if ((gs_wfInfo.format==SF_FORMAT_RAW_LE) || 
	(gs_wfInfo.format==SF_FORMAT_RAW_BE)) 
      gfp->input_format=sf_raw;

#ifdef _DEBUG_SND_FILE
	printf("\n\nSF_INFO structure\n");
	printf("samplerate        :%d\n",gs_wfInfo.samplerate);
	printf("samples           :%d\n",gs_wfInfo.samples);
	printf("channels          :%d\n",gs_wfInfo.channels);
	printf("pcmbitwidth       :%d\n",gs_wfInfo.pcmbitwidth);
	printf("format            :");

	/* new formats from [e-mail]  1/2000 */
        if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_WAV)
	  printf("Microsoft WAV format (big endian). ");
	if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_AIFF)
	  printf("Apple/SGI AIFF format (little endian). ");
	if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_AU)
	  printf("Sun/NeXT AU format (big endian). ");
	if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_AULE)
	  printf("DEC AU format (little endian). ");
	if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_RAW)
	  printf("RAW PCM data. ");
	if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_PAF)
	  printf("Ensoniq PARIS file format. ");
	if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_SVX)
	  printf("Amiga IFF / SVX8 / SV16 format. ");
	if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_NIST)
	  printf("Sphere NIST format. ");

	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM)
	  printf("PCM data in 8, 16, 24 or 32 bits.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_FLOAT)
	  printf("32 bit Intel x86 floats.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_ULAW)
	  printf("U-Law encoded.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_ALAW)
	  printf("A-Law encoded.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_IMA_ADPCM)
	  printf("IMA ADPCM.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_MS_ADPCM)
	  printf("Microsoft ADPCM.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_BE)
	  printf("Big endian PCM data.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_LE)
	  printf("Little endian PCM data.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_S8)
	  printf("Signed 8 bit PCM.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_U8)
	  printf("Unsigned 8 bit PCM.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_SVX_FIB)
	  printf("SVX Fibonacci Delta encoding.");
	if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_SVX_EXP)
	  printf("SVX Exponential Delta encoding.");




	printf("\n");
	printf("pcmbitwidth       :%d\n",gs_wfInfo.pcmbitwidth);
	printf("sections          :%d\n",gs_wfInfo.sections);
	printf("seekable          :\n",gs_wfInfo.seekable);
#endif
  }

  if (gs_wfInfo.samples==MAX_U_32_NUM) {
    struct stat sb;
    /* try to figure out num_samples */
    if (0==stat(lpszFileName,&sb)) {
      /* try file size, assume 2 bytes per sample */
      if (gfp->input_format == sf_mp3) {
	FLOAT totalseconds = (sb.st_size*8.0/(1000.0*GetSndBitrate()));
	gs_wfInfo.samples= totalseconds*GetSndSampleRate();
      }else{
	gs_wfInfo.samples = sb.st_size/(2*GetSndChannels());
      }
    }
  }
  return musicin;    
}


/************************************************************************
*
* read_samples()
*
* PURPOSE:  reads the PCM samples from a file to the buffer
*
*  SEMANTICS:
* Reads #samples_read# number of shorts from #musicin# filepointer
* into #sample_buffer[]#.  Returns the number of samples read.
*
************************************************************************/

int read_samples_pcm(lame_global_flags *gfp,short sample_buffer[2304],int frame_size,int samples_to_read)
{
    int 		samples_read;
    int			rcode;

    samples_read=sf_read_short(gs_pSndFileIn,sample_buffer,samples_to_read);
    
    rcode = samples_read;
    if (samples_read < frame_size)
      {
	/*fprintf(stderr,"Insufficient PCM input for one frame - fillout with zeros\n"); 
	*/
	if (samples_read<0) samples_read=0;
	for (; samples_read < frame_size; sample_buffer[samples_read++] = 0);
      }

	if (8==gs_wfInfo.pcmbitwidth)
	  for (; samples_read >= 0; sample_buffer[samples_read] = sample_buffer[samples_read--] * 256);

    return(rcode);
}


#endif /* ifdef LIBSNDFILE */
#ifdef LAMESNDFILE 

/************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 *
 * OLD ISO/LAME routines follow.  Used if you dont have LIBSNDFILE
 * or for stdin/stdout support
 *
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************/

/* Replacement for forward fseek(,,SEEK_CUR), because fseek() fails on pipes */
int fskip(FILE *sf,long num_bytes,int dummy)
{
  char data[1024];
  int nskip = 0;
  while (num_bytes > 0) {
    nskip = (num_bytes>1024) ? 1024 : num_bytes;
    num_bytes -= fread(data,(size_t)1,(size_t)nskip,sf);
  }
  /* return 0 if last read was successful */