--- /dev/null
+/*
+ (c) Copyright 1998-2000 - Tord Jansson
+ ======================================
+
+ This file is part of the BladeEnc MP3 Encoder, based on
+ ISO's reference code for MPEG Layer 3 compression, and might
+ contain smaller or larger sections that are directly taken
+ from ISO's reference code.
+
+ All changes to the ISO reference code herein are either
+ copyrighted by Tord Jansson (tord.jansson@swipnet.se)
+ or sublicensed to Tord Jansson by a third party.
+
+ BladeEnc is free software; you can redistribute this file
+ and/or modify it under the terms of the GNU Lesser General Public
+ License as published by the Free Software Foundation; either
+ version 2.1 of the License, or (at your option) any later version.
+
+
+
+ ------------ CHANGES ------------
+
+ 2000-11-22 Andre Piotrowski
+
+ - big fix: module - original dist10 error - l3_sb_sample has to be of type 'L3SBS' not 'SBS'.
+ - bug fix: codecInit() - don't reset fInit-huffman_read_flag - read_huffcodetab() is a memory eater!
+
+ 2000-12-11 ap
+
+ - reformatted
+ - speed up: use single buffer
+*/
+
+#include <stdlib.h>
+#include <assert.h>
+
+#include "common.h"
+#include "l3psy.h"
+#include "mdct.h"
+#include "reservoir.h"
+#include "formatbitstream2.h"
+#include "l3bitstream.h"
+#include "loop.h"
+#include "encoder.h"
+#include "codec.h"
+
+
+
+
+
+extern int fInit_fft;
+
+
+
+
+
+/************************************************************************/
+
+#define SAMPLES_PER_FRAME 1152
+
+static L3SBS l3_sb_sample;
+
+static layer info;
+
+
+
+#if ORG_BUFFERS
+static short buffer[2][1152];
+/* static float snr32[32]; */
+static short sam[2][2048];
+#else
+static FLOAT buffer[2][2048];
+static int buffer_idx;
+#endif
+
+
+
+static int whole_SpF;
+
+static double frac_SpF, slot_lag;
+
+static int stereo, error_protection;
+
+static III_side_info_t l3_side;
+static CodecInitOut sOut;
+
+static frame_params fr_ps;
+
+
+
+char *pEncodedOutput;
+int outputBit;
+
+volatile double avg_slots_per_frame;
+
+
+
+
+
+/*____ codecInit() ____________________________________________________________*/
+
+CodecInitOut *codecInit (CodecInitIn *psIn)
+{
+ int j;
+
+ /* Read psIn */
+
+ switch (psIn->frequency)
+ {
+ case 48000: info.sampling_frequency = 1; break;
+ case 44100: info.sampling_frequency = 0; break;
+ case 32000: info.sampling_frequency = 2; break;
+ default : return FALSE;
+ }
+
+ switch (psIn->mode)
+ {
+ case 0: info.mode = MPG_MD_STEREO ; info.mode_ext = 0; break;
+ case 2: info.mode = MPG_MD_DUAL_CHANNEL; info.mode_ext = 0; break;
+ case 3: info.mode = MPG_MD_MONO ; info.mode_ext = 0; break;
+ default: return FALSE;
+ }
+
+ j = 0;
+ while (j < 15 && bitratex[1][j] != psIn->bitrate)
+ j++;
+ info.bitrate_index = j;
+
+ info.version = 1; /* Default: MPEG-1 */
+ info.emphasis = psIn->emphasis;
+ info.extension = psIn->fPrivate;
+ info.copyright = psIn->fCopyright;
+ info.original = psIn->fOriginal;
+ info.error_protection = psIn->fCRC;
+
+
+/*_______ Static-fix _______________*/
+
+
+ fInit_mdct_sub = 0;
+ fInit_mdct = 0;
+
+ fInit_fft = 0;
+
+ fInit_iteration_loop = 0;
+/* DON'T DO THAT ALL THE TIME
+ fInit_huffman_read_flag = 0;
+*/
+
+ fixStatic_loop();
+
+ l3_side.main_data_begin = 0;
+ fixStatic_reservoir();
+
+
+/*___________________________________*/
+
+ psycho_anal_init (psIn->frequency);
+ initWindowFilterSubband ();
+ initFormatBitstream ();
+
+/* clear buffers */
+ memset ((char *) l3_sb_sample, 0, sizeof(l3_sb_sample));
+ memset((char *) buffer, 0, sizeof(buffer));
+/* memset((char *) snr32, 0, sizeof(snr32));*/
+#if ORG_BUFFERS
+ memset((char *) sam, 0, sizeof(sam));
+#endif
+
+
+
+ fr_ps.header = &info;
+ fr_ps.tab_num = -1; /* no table loaded */
+ fr_ps.alloc = NULL;
+ fr_ps.actual_mode = info.mode;
+ fr_ps.stereo = (info.mode == MPG_MD_MONO) ? 1 : 2;
+ fr_ps.sblimit = SBLIMIT;
+ fr_ps.jsbound = SBLIMIT;
+
+
+ stereo = fr_ps.stereo;
+ error_protection = info.error_protection;
+
+ avg_slots_per_frame =
+ ((double) SAMPLES_PER_FRAME / s_freq[1][info.sampling_frequency]) *
+ ((double) bitratex[1][info.bitrate_index] / 8.0);
+ whole_SpF = (int) avg_slots_per_frame;
+ frac_SpF = avg_slots_per_frame - (double) whole_SpF;
+ slot_lag = -frac_SpF;
+
+/* if (frac_SpF == 0)
+ info.padding = 0;
+*/
+ genNoisePowTab();
+
+/*________________________*/
+
+
+ if( stereo != 2 )
+ sOut.nSamples = SAMPLES_PER_FRAME;
+ else
+ sOut.nSamples = SAMPLES_PER_FRAME*2;
+
+ sOut.bufferSize = 2048;
+
+ return &sOut; /* How many samples we want in each chunk... */
+}
+
+
+
+
+
+/*____ codecEncodeChunk() _____________________________________________________*/
+
+unsigned int codecEncodeChunk
+(
+ int nSamples,
+ short *pSamples,
+ char *pDest
+)
+{
+ static double xr[2][2][576];
+ static double xr_dec[2][2][576];
+ static double pe[2][2];
+ static int l3_enc[2][2][576];
+ static III_psy_ratio ratio;
+ static III_scalefac_t scalefac;
+ int gr, ch;
+ int mean_bits, sideinfo_len;
+ int bitsPerFrame;
+ int j;
+
+/*
+ static int fFirst = TRUE;
+
+ if (fFirst)
+ {
+ memset ((char *) &xr , 0, sizeof(xr));
+ memset ((char *) &xr_dec , 0, sizeof(xr_dec));
+ memset ((char *) &pe , 0, sizeof(pe));
+ memset ((char *) &l3_enc , 0, sizeof(l3_enc));
+ memset ((char *) &ratio , 0, sizeof(ratio));
+ memset ((char *) &l3_side , 0, sizeof(l3_side));
+ memset ((char *) &scalefac, 0, sizeof(scalefac));
+
+ fFirst = FALSE;
+ }
+*/
+
+
+/* rebuffer audio */
+
+#if ORG_BUFFERS
+ rebuffer_audio (buffer, pSamples, nSamples, stereo);
+#else
+ rebuffer_audio (pSamples, buffer, &buffer_idx, nSamples, stereo);
+#endif
+
+
+/* psychoacoustic model */
+
+ for (gr = 0; gr < 2; gr++)
+ for (ch = 0; ch < stereo; ch++)
+ psycho_anal
+ (
+#if ORG_BUFFERS
+ &buffer[ch][gr*576],
+ &sam[ch][0],
+#else
+ buffer[ch],
+ (buffer_idx+gr*576) & 2047,
+#endif
+ ch,
+ 3,
+ /*snr32,*/
+ &ratio.l[gr][ch][0],
+ &ratio.s[gr][ch][0],
+ &pe[gr][ch],
+ &l3_side.gr[gr].ch[ch].tt
+ );
+
+
+/* polyphase filtering */
+
+ for (gr = 0; gr < 2; gr++)
+ {
+ int gr_plus_1 = gr_idx[gr+1];
+
+ for (ch = 0; ch < stereo; ch++)
+ {
+ for (j = 0; j < 18; j++)
+ {
+ windowFilterSubband
+ (
+#if ORG_BUFFERS
+ &buffer[ch][gr*18*32+32*j],
+ ch,
+#else
+ buffer[ch],
+ (buffer_idx+768-480+gr*18*32+32*j) & 2047,
+#endif
+ l3_sb_sample[ch][gr_plus_1][j]
+ );
+ }
+ }
+ }
+
+
+/* apply mdct to the polyphase outputs */
+
+ mdct_sub (&l3_sb_sample, xr, stereo, &l3_side, 2);
+
+
+ pEncodedOutput = pDest;
+ outputBit = 8;
+ pEncodedOutput[0] = 0;
+
+
+ if (frac_SpF != 0)
+ {
+ if (slot_lag > (frac_SpF-1.0))
+ {
+ slot_lag -= frac_SpF;
+ info.padding = 0;
+ }
+ else
+ {
+ info.padding = 1;
+ slot_lag += (1-frac_SpF);
+ }
+ }
+
+ bitsPerFrame = 8 * whole_SpF + (info.padding * 8);
+
+
+/* determine the mean bitrate for main data */
+
+ sideinfo_len = 32;
+
+ if (stereo == 1)
+ sideinfo_len += 136;
+ else
+ sideinfo_len += 256;
+
+ if (info.error_protection)
+ sideinfo_len += 16;
+
+ mean_bits = (bitsPerFrame - sideinfo_len) / 2;
+
+
+/* bit and noise allocation */
+
+ iteration_loop
+ (
+ pe,
+ xr,
+ &ratio,
+ &l3_side,
+ l3_enc,
+ mean_bits,
+ stereo,
+ xr_dec,
+ &scalefac,
+ &fr_ps,
+ 0,
+ bitsPerFrame
+ );
+
+
+/* write the frame to the bitstream */
+
+ III_format_bitstream
+ (
+ bitsPerFrame,
+ &fr_ps,
+ l3_enc,
+ &l3_side,
+ &scalefac,
+ xr,
+ NULL,
+ 0
+ );
+
+
+ return pEncodedOutput - pDest;
+}
+
+
+
+
+
+/*____ codecExit() ____________________________________________________________*/
+
+unsigned int codecExit (char *pDest)
+{
+ pEncodedOutput = pDest;
+ outputBit = 8;
+ pEncodedOutput[0] = 0;
+
+ psycho_anal_exit ();
+ exitFormatBitstream ();
+ III_FlushBitstream ();
+
+ return pEncodedOutput - pDest;
+}
+
+
+
+
+
+/*____ codecFlush() _________________________________________________________*/
+
+unsigned int codecFlush (char *pDest)
+{
+ pEncodedOutput = pDest;
+ outputBit = 8;
+ pEncodedOutput[0] = 0;
+
+ flushFrame ();
+
+ whole_SpF = (int) avg_slots_per_frame;
+ frac_SpF = avg_slots_per_frame - (double) whole_SpF;
+ slot_lag = -frac_SpF;
+
+ l3_side.main_data_begin = 0;
+ fixStatic_reservoir ();
+
+ return pEncodedOutput - pDest;
+}
+
+
+
git.asbjorn.biz / swftools.git / blobdiff