encode.c

ISO mp3 sources (distribution 10) Layer 1/2/3, C Source, 512 k Sources of the Mpeg 1,2 layer 1,2

/**********************************************************************
 * ISO MPEG Audio Subgroup Software Simulation Group (1996)
 * ISO 13818-3 MPEG-2 Audio Multichannel Encoder
 *
 * $Id: encode.c 1.9 1996/02/12 07:13:35 rowlands Exp $
 *
 * $Log: encode.c $
 * Revision 1.9  1996/02/12 07:13:35  rowlands
 * Release following Munich meeting
 *
 * Revision 1.6.2.1  1995/11/06  04:19:12  rowlands
 * Received from Uwe Felderhoff (IRT)
 *
 * Revision 1.8  1995/08/14  07:53:28  tenkate
 * ML-LSF added Warner ten Kate 7/8/95 (Philips)
 * variables "alloc" and "sblimit" changed into "alloc_ml" and
 * "sblimit_ml" where appropriate.
 * sample loops adapted to 12 or 6 depending on full or half fs.
 * II_a_bit_allocation separate summing for ML.
 *
 * Revision 1.7  1995/07/31  07:48:55  tenkate
 * addition of phantom coding, in void matricing(), 25/07/95 WtK
 *
 * Revision 1.4.3.1  1995/06/16  08:25:11  rowlands
 * Input from Sang Wook Kim (Samsung AIT)
 *
 * Revision 1.4.2.1  1995/06/16  03:46:42  rowlands
 * Input from Susanne Ritscher (IRT)
 *
 * Revision 1.4.1.6  1995/06/16  02:41:51  rowlands
 * Added dematrix procedure 2, corrected dematrixing values.
 *
 * Added support for dematrix procedure 2, without predistortion.
 * To implement predistortion will require a delay in the encoder
 * to be able to calculate the predistorted filtered surround signal
 * subband samples.
 *
 * Modified procedure matricing() to reuse the compatible stereo
 * channel signals computed by matricing_fft(). This was needed to
 * avoid calculating the low-pass-filtered surround signal in the
 * subband domain. The main function of procedure matricing() now
 * is to weight the multichannels. These channels are unweighted
 * prior to this for the benefit of the psycho model.
 *
 * Corrected weighting values for dematrixing procedures.
 * Deleted matrixing on multilingual channels.
 * In the quantization procedures, made quantizing to zero bits
 * a special case to skip some operations on undefined values.
 *
 **********************************************************************/

/**********************************************************************
 *   date   programmers         comment                               *
 * 3/01/91  Douglas Wong,       start of version 1.1 records          *
 *          Davis Pan                                                 *
 * 3/06/91  Douglas Wong        rename: setup.h to endef.h            *
 *                                      efilter to enfilter           *
 *                                      ewindow to enwindow           *
 *                              integrated "quantizer", "scalefactor",*
 *                              and "transmission" files              *
 *                              update routine "window_subband"       *
 * 3/31/91  Bill Aspromonte     replaced read_filter by               *
 *                              create_an_filter                      *
 * 5/10/91  W. Joseph Carter    Ported to Macintosh and Unix.         *
 *                              Incorporated Jean-Georges Fritsch's   *
 *                              "bitstream.c" package.                *
 *                              Incorporated Bill Aspromonte's        *
 *                              filterbank coefficient matrix         *
 *                              calculation routines and added        *
 *                              roundoff to coincide with specs.      *
 *                              Modified to strictly adhere to        *
 *                              encoded bitstream specs, including    *
 *                              "Berlin changes".                     *
 *                              Modified PCM sound file handling to   *
 *                              process all incoming samples and fill *
 *                              out last encoded frame with zeros     *
 *                              (silence) if needed.                  *
 *                              Located and fixed numerous software   *
 *                              bugs and table data errors.           *
 * 19jun91  dpwe (Aware)        moved "alloc_*" reader to common.c    *
 *                              Globals sblimit, alloc replaced by new*
 *                              struct 'frame_params' passed as arg.  *
 *                              Added JOINT STEREO coding, layers I,II*
 *                              Affects: *_bit_allocation,            *
 *                              subband_quantization, encode_bit_alloc*
 *                              sample_encoding                       *
 * 6/10/91  Earle Jennings      modified II_subband_quantization to   *
 *                              resolve type cast problem for MS_DOS  *
 * 6/11/91  Earle Jennings      modified to avoid overflow on MS_DOS  *
 *                              in routine filter_subband             *
 * 7/10/91  Earle Jennings      port to MsDos from MacIntosh version  *
 * 8/ 8/91  Jens Spille         Change for MS-C6.00                   *
 *10/ 1/91  S.I. Sudharsanan,   Ported to IBM AIX platform.           *
 *          Don H. Lee,                                               *
 *          Peter W. Farrett                                          *
 *10/ 3/91  Don H. Lee          implemented CRC-16 error protection   *
 *                              newly introduced function encode_CRC  *
 *11/ 8/91  Kathy Wang          Documentation of code                 *
 *                              All variablenames are referred to     *
 *                              with surrounding pound (#) signs      *
 * 2/11/92  W. Joseph Carter    Ported new code to Macintosh.  Most   *
 *                              important fixes involved changing     *
 *                              16-bit ints to long or unsigned in    *
 *                              bit alloc routines for quant of 65535 *
 *                              and passing proper function args.     *
 *                              Removed "Other Joint Stereo" option   *
 *                              and made bitrate be total channel     *
 *                              bitrate, irrespective of the mode.    *
 *                              Fixed many small bugs & reorganized.  *
 * 92-08-11 Soren H. Nielsen    Fixed bug: allocation of space in the *
 *                              bitstream for the CRC-word. Fixed     *
 *                              reading of window from file.          *
 * 92-11-06 Soren H. Nielsen	Fixed scalefactor calculation.        *
 **********************************************************************
 *                                                                    *
 *                                                                    *
 *  MPEG/audio Phase 2 coding/decoding multichannel                   *
 *                                                                    *
 *  7/27/93        Susanne Ritscher,  IRT Munich                      *
 *  8/10/93        changed matricing to 7 channels                    *
 *	           added void matricing_fft                           *
 *  8/12/93        added int required_bits,                           *
 *                       int max_alloc                                *
 *                 implemented the new mc_header (third working draft)*
 *  8/13/93        added channel-switching in required_bits and       *
 *                 II_subband_quantisation, II_encode_scale,          *
 *                 II_encode_bit_alloc, II_encode_sample,             *
 *                 encode_info                                        *
 *                 all channels normalized                            *
 *  9/20/93        channel-switching is only performed at a           *
 *                 certain limit of TC_ALLOC dB, which is included    *
 *                 in encoder.h                                       *
 *  1/04/94        get out some rubbish                               *
 *                                                                    *
 * 01/05/94        implemented the Committee Draft header             *
 *                                                                    *
 * 01/12/94        changed matricing procedure according to           *
 *                 Committee Draft                                    *
 *                                                                    *
 *  Version 1.0                                                       *
 *                                                                    *
 *  07/12/94       Susanne Ritscher,  IRT Munich                      *
 *                 Tel: [phone]                              *
 *                 Fax: [phone]                              *
 *                                                                    *
 *  Version 1.1                                                       *
 *                                                                    *
 *  02/23/95	   Susanne Ritscher,  IRT Munich                      *
 *                 corrected some bugs                                *
 *                 extension bitstream is working                     *
 *                                                                    *
 *  Version 2.0                                                       *
 *                                                                    *
 *  01/28/97       Frans de Bont, Philips Sound & Vision, Eindhoven   *
 *		    - dynamic crosstalk working for all configurations*
 *		    - prediction working for all configurations	      *
 *		    - extension bitstream fixed			      *
 *		    - fully compliant to DIS 13818-3.2                *
 *                                                                    *				      *		    
 **********************************************************************/
/**********************************************************************
 *                                                                    *
 *  06/06/95       Yeon Bae Thomas Kim, Samsung AIT                   *
 *                 ancillary data is working                          *
 *                                                                    *
 *  06/06/95       Sang Wook Kim, Samsung AIT                         *
 *                 corrected some bugs                                *
 *                                                                    *
 **********************************************************************/
#define VERY_FAST_FILTER  1	/* JMZ 08/03/1995 FILTER */

#include "common.h"
#include "encoder.h"
 
/*=======================================================================\
|                                                                       |
| This segment contains all the core routines of the encoder,           |
| except for the psychoacoustic models.                                 |
|                                                                       |
| The user can select either one of the two psychoacoustic              |
| models. Model I is a simple tonal and noise masking threshold         |
| generator, and Model II is a more sophisticated cochlear masking      |
| threshold generator. Model I is recommended for lower complexity      |
| applications whereas Model II gives better subjective quality at low  |
| bit rates.                                                            |
|                                                                       |
| Layers I and II of mono, stereo, and joint stereo modes are supported.|
| Routines associated with a given layer are prefixed by "I_" for layer |
| 1 and "II_" for layer 2.                                              |
\=======================================================================*/
 
/************************************************************************/
/*
/* 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.
/*
/************************************************************************/

unsigned long read_samples(FILE *musicin, long int *sample_buffer, long unsigned int num_samples, long unsigned int frame_size, int
 *byte_per_sample, int *aiff)
{
unsigned long samples_read;
static unsigned long samples_to_read;
static char init = TRUE;
short pcm_sample_buffer[9216];        /*for correct reading of pcm-data*/
int i;

   if (init) {
        samples_to_read = num_samples;
        init = FALSE;
   }
   if (samples_to_read >= frame_size)
        samples_read = frame_size;
   else
		  samples_read = samples_to_read;

	if((*aiff==1) &&(*byte_per_sample !=2)){
	 
            if ((samples_read =
		  fread(sample_buffer, *byte_per_sample, (int)samples_read, musicin)) == 0)
		  if (verbosity >= 2) printf("Hit end of audio data\n");
	   
	}
	else{
	 if ((samples_read =
		  fread(pcm_sample_buffer, sizeof(short), (int)samples_read, musicin)) == 0)
		  if (verbosity >= 2) printf("Hit end of audio data\n");
            for(i = 0; i < 9216; ++i) sample_buffer[i] = pcm_sample_buffer[i];
        }

   samples_to_read -= samples_read;
   if (samples_read < frame_size && samples_read > 0) {
        if (verbosity >= 2) printf("Insufficient PCM input for one frame - fillout with zeros\n");
        for (; samples_read < frame_size; sample_buffer[samples_read++] = 0);
        samples_to_read = 0;
   }
   return(samples_read);
}

/************************************************************************/
/*
/* 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
/*
/*  SEMANTICS:
/* Calls read_samples() to read a frame of audio data from filepointer
/* #musicin# to #insampl[]#.  The data is shifted to make sure the data
/* is centered for the 1024pt window to be used by the psychoacoustic model,
/* and to compensate for the 256 sample delay from the filter bank. For
/* stereo, the channels are also demultiplexed into #buffer[0][]# and
/* #buffer[1][]#
/*
/* 21/03/1995 JMZ Multimode adaptations
/************************************************************************/
 
unsigned long
get_audio (
	FILE *musicin,
	double (*buffer)[1152],
	long unsigned int num_samples,
	int stereo,
	IFF_AIFF *aiff_ptr,
	int stereomc,
	int stereoaug,
	frame_params *fr_ps,
	int *aiff,
	int *byte_per_sample,
	double (*buffer_matr)[1152]
)
{
    int k, j, i;
    long insamp[9216];
    unsigned long samples_read;
    int lay;
    int lfe;
    layer *info = fr_ps->header;
    
    lay = info->lay;
    lfe = info->lfe;

    if (lay == 1)
    {
	if (stereo == 2)
	{ /* layer 1, stereo */
	    samples_read = read_samples (musicin, insamp, num_samples,
				         (unsigned long) 768, byte_per_sample, aiff);
	    for (j = 0; j < 448; j++) 
	    {
		if (j<64) 
		{
		    buffer[0][j] = buffer[0][j+384];
		    buffer[1][j] = buffer[1][j+384];
		    buffer[2][j] = 0;
		    buffer[3][j] = 0;
		    buffer[4][j] = 0;
		}
		else 
		{
		    buffer[0][j] = insamp[2*j-128];
		    buffer[1][j] = insamp[2*j-127];
		    buffer[2][j] = 0;
		    buffer[3][j] = 0;
		    buffer[4][j] = 0;
		}
	    }
	}
	else 
	{ /* layer 1, mono */
	    samples_read = read_samples (musicin, insamp, num_samples,
				         (unsigned long) 384, byte_per_sample, aiff);
	    for (j = 0; j < 448; j++)
	    {
		if (j < 64) 
		{
		    buffer[0][j] = buffer[0][j+384];
		    buffer[1][j] = 0;
		    buffer[2][j] = 0;
		    buffer[3][j] = 0;
		    buffer[4][j] = 0;
		}
		else 
		{
		    buffer[0][j] = insamp[j-64];
		    buffer[1][j] = 0;
		    buffer[2][j] = 0;
		    buffer[3][j] = 0;
		    buffer[4][j] = 0;
		}
	    }
	}
    }
    else 
    {
	if (*aiff == 1)
	{
	    k = aiff_ptr->numChannels;
	    samples_read = read_samples (musicin, insamp, num_samples,
					     (unsigned long) (k * 1152), byte_per_sample, aiff);

	    for (i = 0; i < k; i++)
		for (j = 0; j < 1152; j++)
		    buffer[i][j] = insamp[k*j+i];
	}
	else 
	{  /* layerII, stereo */
	    if (stereo == 2)
	    {
		samples_read = read_samples (musicin, insamp, num_samples,
			(unsigned long) ((2+lfe)*1152), byte_per_sample, aiff);
		for (j = 0; j < 1152; j++)
		{ /* fixed bug 28.6.93 S.R. */
		    buffer[0][j] = insamp[(2+lfe)*j];
		    buffer[1][j] = insamp[(2+lfe)*j+1];
		    buffer[2][j] = 0;
		    buffer[3+lfe][j] = 0;
		    buffer[4+lfe][j] = 0;
		    if (lfe)
		        buffer[3][j] = insamp[(2+lfe)*j+2]; /* ########### */
		}
	    }
	    else
	    { /* layer 2 (or 3), mono */
		samples_read = read_samples (musicin, insamp, num_samples,
			(unsigned long) 1152, byte_per_sample, aiff);
		for (j = 0; j < 1152; j++)
		{
		    buffer[0][j] = insamp[j];
		    buffer[1][j] = 0;
		    buffer[2][j] = 0;
		    buffer[3][j] = 0;
		    buffer[4][j] = 0;
		}        
	    }
	}
    }

    /*
     * If LFE is not enabled, "buffer" contains:
     *	buffer[0]	L
     *	buffer[1]	R
     *	buffer[2]	C
     *	buffer[3]	Ls
     *	buffer[4]	Rs
#ifdef Augmentation_7ch
     *  optional in 7.1 channel augmentation mode
     *	buffer[5]	Lc
     *	buffer[6]	Rc
#endif
     *
     * If LFE is enabled, "buffer" contains:
     *	buffer[0]	L
     *	buffer[1]	R
     *	buffer[2]	C
     *	buffer[3]	LFE
     *	buffer[4]	Ls
     *	buffer[5]	Rs
#ifdef Augmentation_7ch
     *  optional in 7.1 channel augmentation mode
     *	buffer[6]	Lc
     *	buffer[7]	Rc
#endif
     */

#ifdef Augmentation_7ch
    if (stereoaug == 2)
    	matricing_aug_fft (buffer, buffer_matr, fr_ps);
    else
#endif
	matricing_fft (buffer, buffer_matr, fr_ps);

    /*
     * After matrixing, "buffer_matr" contains:
     *	buffer_matr[0]	Lo
     *	buffer_matr[1]	Ro
     *	buffer_matr[2]	C
     *	buffer_matr[3]	Ls
     *	buffer_matr[4]	Rs
     *	buffer_matr[5]	L
     *	buffer_matr[6]	R
#ifdef Augmentation_7ch
     *  optional in 7.1 channel augmentation mode
     *	buffer_matr[7]	L7
     *	buffer_matr[8]	R7
     *	buffer_matr[9]	C7
     *	buffer_matr[10]	Lc
     *	buffer_matr[11]	Rc