/********************************************************************
 *                                                                  *
 * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
 * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
 * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
 * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
 *                                                                  *
 * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002             *
 * by the XIPHOPHORUS Company http://www.xiph.org/                  *
 *                                                                  *
 ********************************************************************

 function: simple example encoder
 last mod: $Id: encoder_example.c,v 1.50 2002/07/16 09:26:07 xiphmont Exp $

 ********************************************************************
  modified by AOYUMI
   //-   delete
   //+   add
   //+~, //~+   add range
   // /[asterisk]-~, [asterisk]/   delete range
   
   support only 16bit PCM .WAV file
   
   2004/04/26 
    In order to enable it to compile by Linux, a patch is applied to 
   a this code.  This patch is offer from Frederik Himpe. 
 *********************************************************************/

/* takes a stereo 16bit 44.1kHz WAV file from stdin and encodes it into
   a Vorbis bitstream */

/* Note that this is POSIX, not ANSI, code */

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <time.h>
#include <sys/types.h> //+
#include <sys/stat.h> //+
#include <math.h>
#include <vorbis/vorbisenc.h>

#ifdef _WIN32 /* We need the following two to set stdin/stdout to binary */
#include <io.h>
#include <fcntl.h>
#endif

#if defined(__MACOS__) && defined(__MWERKS__)
#include <console.h>      /* CodeWarrior's Mac "command-line" support */
#endif

#define READ 1024
#define DEF_Q 0  /* Default Quality Setting */  //+

signed char readbuffer[READ*4+44]; /* out of the data segment, not the stack */

//- int main(){
  int main(int argc, char *argv[]){ //+
  ogg_stream_state os; /* take physical pages, weld into a logical
			  stream of packets */
  ogg_page         og; /* one Ogg bitstream page.  Vorbis packets are inside */
  ogg_packet       op; /* one raw packet of data for decode */
  
  vorbis_info      vi; /* struct that stores all the static vorbis bitstream
			  settings */
  vorbis_comment   vc; /* struct that stores all the user comments */

  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
  vorbis_block     vb; /* local working space for packet->PCM decode */

  int eos=0,ret;
  int i, founddata;

//+~
  float qnum = DEF_Q;
  int qx = 1;
  if(argc < 2){
	puts("usage: encoder_example -q4 filename.wav");
	exit(1);
  }
  if( !strncmp(argv[1], "-q", 2) ){
	if( strlen(argv[1]) > 2 ){
		qnum = atof(argv[1]+2);
	}
	if( !((qnum >= -2) && (qnum <= 10)) ) qnum = DEF_Q;
	qx++;
  }
  FILE *FP;
  if( !(FP = fopen(argv[qx], "rb")) ){
	puts("source file not found");
	exit(1);
  }
//~+

#if defined(macintosh) && defined(__MWERKS__)
  int argc = 0;
  char **argv = NULL;
  argc = ccommand(&argv); /* get a "command line" from the Mac user */
                          /* this also lets the user set stdin and stdout */
#endif

  /* we cheat on the WAV header; we just bypass 44 bytes and never
     verify that it matches 16bit/stereo/44.1kHz.  This is just an
     example, after all. */

#ifdef _WIN32 /* We need to set stdin/stdout to binary mode. Damn windows. */
  /* if we were reading/writing a file, it would also need to in
     binary mode, eg, fopen("file.wav","wb"); */
  /* Beware the evil ifdef. We avoid these where we can, but this one we 
     cannot. Don't add any more, you'll probably go to hell if you do. */
  _setmode( _fileno( stdin ), _O_BINARY );
  _setmode( _fileno( stdout ), _O_BINARY );
#endif


  /* we cheat on the WAV header; we just bypass the header and never
     verify that it matches 16bit/stereo/44.1kHz.  This is just an
     example, after all. */

  unsigned long footer = 0; //+
  unsigned long sa_rate = 44100; //+
  unsigned long channel; //+
  struct stat sbuf; //+
  stat(argv[qx], &sbuf); //+

  readbuffer[0] = '\0';
//-  for (i=0, founddata=0; i<30 && ! feof(stdin) && ! ferror(stdin); i++)
  for (i=0, founddata=0; i<30 && ! feof(FP) && ! ferror(FP); i++) //+
  {
//-    fread(readbuffer,1,2,stdin);
    fread(readbuffer,1,2,FP); //+
//+~
    if ( ! memcmp(readbuffer, "fm", 2) ){
       fread(readbuffer,1,8,FP); //+
       fread(&channel,1,2,FP); //+
       fread(&sa_rate,1,4,FP); //+
    }
//~+
//-    if ( ! strncmp((char*)readbuffer, "da", 2) )
    if ( ! memcmp(readbuffer, "da", 2) ) //+
    {
      founddata = 1;
//-      fread(readbuffer,1,6,stdin);
      fread(readbuffer,1,6,FP); //+
      memcpy(&footer, readbuffer+2, 4); //+
      footer += ftell(FP); //+
      break;
    }
  }
  footer = sbuf.st_size - footer; //+
  printf("\n%dHz\n", sa_rate); //+

  /********** Encode setup ************/

  vorbis_info_init(&vi);

  /* choose an encoding mode.  A few possibilities commented out, one
     actually used: */

  /*********************************************************************
   Encoding using a VBR quality mode.  The usable range is -.1
   (lowest quality, smallest file) to 1. (highest quality, largest file).
   Example quality mode .4: 44kHz stereo coupled, roughly 128kbps VBR 
  
   ret = vorbis_encode_init_vbr(&vi,2,44100,.4);

   ---------------------------------------------------------------------

   Encoding using an average bitrate mode (ABR).
   example: 44kHz stereo coupled, average 128kbps VBR 
  
   ret = vorbis_encode_init(&vi,2,44100,-1,128000,-1);

   ---------------------------------------------------------------------

   Encode using a qulity mode, but select that quality mode by asking for
   an approximate bitrate.  This is not ABR, it is true VBR, but selected
   using the bitrate interface, and then turning bitrate management off:

   ret = ( vorbis_encode_setup_managed(&vi,2,44100,-1,128000,-1) ||
           vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE_AVG,NULL) ||
           vorbis_encode_setup_init(&vi));

   *********************************************************************/

//-  ret=vorbis_encode_init_vbr(&vi,2,44100,.4);
  ret=vorbis_encode_init_vbr(&vi,channel,sa_rate, qnum/10); //+
  /* do not continue if setup failed; this can happen if we ask for a
     mode that libVorbis does not support (eg, too low a bitrate, etc,
     will return 'OV_EIMPL') */

//-  if(ret)exit(1);
//+~
  if(ret){
	puts("encode init error");
	exit(1);
  }
  char wfile[1024];
  strcpy(wfile, argv[qx]);
  strcat(wfile, ".ogg");
  FILE *FP2;
  if( !(FP2 = fopen(wfile, "wb")) ){
	puts("file open error");
	exit(1);
  }
  
  printf("Quality %f\n Encoding...", qnum);
//~+
  /* add a comment */
  vorbis_comment_init(&vc);
  vorbis_comment_add_tag(&vc,"ENCODER","encoder_example.c");

  /* set up the analysis state and auxiliary encoding storage */
  vorbis_analysis_init(&vd,&vi);
  vorbis_block_init(&vd,&vb);
  
  /* set up our packet->stream encoder */
  /* pick a random serial number; that way we can more likely build
     chained streams just by concatenation */
  srand(time(NULL));
  ogg_stream_init(&os,rand());

  /* Vorbis streams begin with three headers; the initial header (with
     most of the codec setup parameters) which is mandated by the Ogg
     bitstream spec.  The second header holds any comment fields.  The
     third header holds the bitstream codebook.  We merely need to
     make the headers, then pass them to libvorbis one at a time;
     libvorbis handles the additional Ogg bitstream constraints */

  {
    ogg_packet header;
    ogg_packet header_comm;
    ogg_packet header_code;

    vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
    ogg_stream_packetin(&os,&header); /* automatically placed in its own
					 page */
    ogg_stream_packetin(&os,&header_comm);
    ogg_stream_packetin(&os,&header_code);

	/* This ensures the actual
	 * audio data will start on a new page, as per spec
	 */
	while(!eos){
		int result=ogg_stream_flush(&os,&og);
		if(result==0)break;
//-		fwrite(og.header,1,og.header_len,stdout);
		fwrite(og.header,1,og.header_len,FP2); //+
//-		fwrite(og.body,1,og.body_len,stdout);
		fwrite(og.body,1,og.body_len,FP2); //+
	}

  }

  while(!eos){
    long i;
//-   long bytes=fread(readbuffer,1,READ*4,stdin); /* stereo hardwired here */
    long bytes=fread(readbuffer,1,READ*(channel*2),FP); //+
    if(bytes < (READ*(channel*2))) bytes -= footer; //+
    if(bytes < 0) bytes = 0; //+
    if(bytes==0){
     /* end of file.  this can be done implicitly in the mainline,
        but it's easier to see here in non-clever fashion.
        Tell the library we're at end of stream so that it can handle
        the last frame and mark end of stream in the output properly */
       vorbis_analysis_wrote(&vd,0);

    }else{
      /* data to encode */

      /* expose the buffer to submit data */
      float **buffer=vorbis_analysis_buffer(&vd,READ);
      
      /* uninterleave samples */
/*-~
      for(i=0;i<bytes/4;i++){
	buffer[0][i]=((readbuffer[i*4+1]<<8)|
		      (0x00ff&(int)readbuffer[i*4]))/32768.f;
	buffer[1][i]=((readbuffer[i*4+3]<<8)|
		      (0x00ff&(int)readbuffer[i*4+2]))/32768.f;
      }
*/
//+~
      for(i=0;i<bytes/(channel*2);i++){
	buffer[0][i]=((readbuffer[i*(channel*2)+1]<<8)|
		      (0x00ff&(int)readbuffer[i*(channel*2)]))/32768.f;
	if(channel == 2)
	buffer[1][i]=((readbuffer[i*(channel*2)+3]<<8)|
		      (0x00ff&(int)readbuffer[i*(channel*2)+2]))/32768.f;
      }
//~+
      /* tell the library how much we actually submitted */
      vorbis_analysis_wrote(&vd,i);
    }

    /* vorbis does some data preanalysis, then divvies up blocks for
       more involved (potentially parallel) processing.  Get a single
       block for encoding now */
    while(vorbis_analysis_blockout(&vd,&vb)==1){

      /* analysis, assume we want to use bitrate management */
      vorbis_analysis(&vb,NULL);
      vorbis_bitrate_addblock(&vb);

      while(vorbis_bitrate_flushpacket(&vd,&op)){
	
	/* weld the packet into the bitstream */
	ogg_stream_packetin(&os,&op);
	
	/* write out pages (if any) */
	while(!eos){
	  int result=ogg_stream_pageout(&os,&og);
	  if(result==0)break;
//-	  fwrite(og.header,1,og.header_len,stdout);
	  fwrite(og.header,1,og.header_len,FP2); //+
//-	  fwrite(og.body,1,og.body_len,stdout);
	  fwrite(og.body,1,og.body_len,FP2); //+
	  
	  /* this could be set above, but for illustrative purposes, I do
	     it here (to show that vorbis does know where the stream ends) */
	  
	  if(ogg_page_eos(&og))eos=1;
	}
      }
    }
  }

  /* clean up and exit.  vorbis_info_clear() must be called last */
  
  ogg_stream_clear(&os);
  vorbis_block_clear(&vb);
  vorbis_dsp_clear(&vd);
  vorbis_comment_clear(&vc);
  vorbis_info_clear(&vi);
  
  /* ogg_page and ogg_packet structs always point to storage in
     libvorbis.  They're never freed or manipulated directly */
  fclose(FP); //+
  fclose(FP2); //+
  fprintf(stderr,"Done.\n");
  return(0);
}
