void II_main_bit_allocation
(
DOUBLE perm_smr[2][SBLIMIT],
UINT scfsi[2][SBLIMIT],
UINT bit_alloc[2][SBLIMIT],
INT* adb,
FRAME* pFrame,
PSETTINGS pSettings
)
{
int noisy_sbs;
int rq_db, av_db = *adb;
// If mode equals to joint stereo, find the optimum jsbound setting
if( pFrame->actual_mode == MPG_MD_JOINT_STEREO)
{
// Start with standard stereo setup
pFrame->pHeader->mode = MPG_MD_STEREO;
pFrame->pHeader->mode_ext = 0;
pFrame->jsbound = pFrame->sblimit;
// Calculate the required bits
if((rq_db=II_bits_for_nonoise(perm_smr, scfsi, pFrame,0)) > *adb)
{
int mode_ext=4;
// Were a few bits short, switch to joint stereo
pFrame->pHeader->mode = MPG_MD_JOINT_STEREO;
// Start with mode extention 3 (is decremented first in while loop)
do
{
// Decrease mode extention
mode_ext--;
// Get new joint stereo subband boundary
pFrame->jsbound = js_bound(pFrame->pHeader->lay, mode_ext);
// Calculate the number of required bits with the new settings
rq_db = II_bits_for_nonoise(perm_smr, scfsi, pFrame,0);
// Check if we have enough bits, and if we still can lower the
// joint stereo subband boundary
} while( (rq_db > *adb) && (mode_ext > 0));
pFrame->pHeader->mode_ext = mode_ext;
} /* well we either eliminated noisy sbs or mode_ext == 0 */
}
if (pSettings->bUseVbr == FALSE)
{
noisy_sbs = II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, pFrame);
}
else
{
int nCurrentAllocTab=0;
int nReqBits=0;
int nChanIdx = pSettings->pFrame->nChannels-1;
int i;
// Start with minimum bitrate
for (i=0;i<15;i++)
{
if ( pSettings->nMinBitRateIndex<=AllowedBrIdx[nChanIdx][i])
break;
}
pFrame->pHeader->bitrate_index=AllowedBrIdx[nChanIdx][i++];
// Recalculate the sbLimit and select proper ALLOC table
hdr_to_frps(pFrame);
// Determine the available bits for this frame, with current bitrate settings
*adb = available_bits(pSettings);
// Work out how many bits are needed for there to be no noise (ie all MNR > 0.0 + VBRLEVEL)
// Keep bumping up the bitrate until we have more than this number of bits
nCurrentAllocTab=pFrame->tab_num;
if (pSettings->nFrame==192)
{
printf("break\n");
}
nReqBits=II_bits_for_nonoise(perm_smr, scfsi, pFrame,pSettings->nVbrQuality+(nCurrentAllocTab==2)?20:0);
// nReqBits=II_bits_for_nonoise(perm_smr, scfsi, pFrame,pSettings->nVbrQuality);
if ( pSettings->nDebugLevel > 3 )
{
DebugPrintf("VBR: req_bits=%i for bitrate setting %d adb=%d table=%d\n",
nReqBits,
bitrate[pFrame->pHeader->version][pFrame->pHeader->lay-1][pFrame->pHeader->bitrate_index],
*adb,
nCurrentAllocTab
);
}
while ( (nReqBits > *adb) && (AllowedBrIdx[nChanIdx][i]!=-1) )
{
// We are not there yet, increase bit rate if possible
pFrame->pHeader->bitrate_index=AllowedBrIdx[nChanIdx][i++];
// Limit to maximum bitrate index
//if (pFrame->pHeader->bitrate_index > 14)
// pFrame->pHeader->bitrate_index = 14;
// Get new frame header parameters for this bitrate setting
hdr_to_frps(pFrame);
// Did we switch from alloc table? If so, calculate new required bits
if (nCurrentAllocTab!=pFrame->tab_num)
{
nReqBits=II_bits_for_nonoise(perm_smr, scfsi, pFrame,pSettings->nVbrQuality);
nCurrentAllocTab=pFrame->tab_num;
}
// Get the number of available bits for the new frame settings
*adb = available_bits(pSettings);
if ( pSettings->nDebugLevel > 2 )
{
DebugPrintf("VBR: req_bits=%i for bitrate setting %d adb=%d table=%d\n",
nReqBits,
bitrate[pFrame->pHeader->version][pFrame->pHeader->lay-1][pFrame->pHeader->bitrate_index],
*adb,
nCurrentAllocTab
);
}
}
noisy_sbs = VBR_bit_allocation(perm_smr, scfsi, bit_alloc, adb, pFrame,pSettings);
}
if ( pSettings->nDebugLevel > 2 )
{
DebugPrintf("VBR frame=%d bitrate=%4d, noisy_sbs=%4d , adb is %3d sblimit=%2d sbound=%2d\n",
pSettings->nFrame,
bitrate[pFrame->pHeader->version][pFrame->pHeader->lay-1][pFrame->pHeader->bitrate_index],
noisy_sbs,
*adb,
pFrame->sblimit,
pFrame->jsbound
);
}
}
void main(int argc, char**argv)
{
FILE *musicout;
Bit_stream_struc bs;
frame_params fr_ps;
III_side_info_t III_side_info;
III_scalefac_t III_scalefac;
unsigned int old_crc;
layer info;
int sync, clip;
int done = FALSE;
unsigned long frameNum=0;
unsigned long bitsPerSlot;
unsigned long sample_frames;
typedef short PCM[2][SSLIMIT][SBLIMIT];
PCM *pcm_sample;
pcm_sample = (PCM *) mem_alloc((long) sizeof(PCM), "PCM Samp");
if (argc==1) {
printf("Useage:decode file.mp3 output.pcm\n");
return;
}
fr_ps.header = &info;
if ((musicout = fopen(argv[2], "w+b")) == NULL) {
printf ("Could not create \"%s\".\n", argv[2]);
exit(1);
}
open_bit_stream_r(&bs, argv[1], BUFFER_SIZE);
sample_frames = 0;
while(!end_bs(&bs)) {
//尝试帧同步
sync = seek_sync(&bs, SYNC_WORD, SYNC_WORD_LENGTH);
if (!sync) {
done = TRUE;
printf("\nFrame cannot be located\n");
out_fifo(*pcm_sample, 3, &fr_ps, done, musicout, &sample_frames);
break;
}
//解码帧头
decode_info(&bs, &fr_ps);
//将fr_ps.header中的信息解读到fr_ps的相关域中
hdr_to_frps(&fr_ps);
//输出相关信息
if(frameNum == 0)
WriteHdr(&fr_ps);
printf("\r%05lu", frameNum++);
if (info.error_protection)
buffer_CRC(&bs, &old_crc);
switch (info.lay) {
case 3:
{
int nSlots, main_data_end, flush_main;
int bytes_to_discard, gr, ch, ss, sb;
static int frame_start = 0;
bitsPerSlot = 8;
//取Side信息
III_get_side_info(&bs, &III_side_info, &fr_ps);
nSlots = main_data_slots(fr_ps);
//读主数据(Audio Data)
for (; nSlots > 0; nSlots--) /* read main data. */
hputbuf((unsigned int) getbits(&bs,8), 8);
main_data_end = hsstell() / 8; /*of privious frame*/
if ( flush_main=(hsstell() % bitsPerSlot) ) {
hgetbits((int)(bitsPerSlot - flush_main));
main_data_end ++;
}
bytes_to_discard = frame_start - main_data_end - III_side_info.main_data_begin ;
if( main_data_end > 4096 ) { frame_start -= 4096;
rewindNbytes( 4096 );
}
frame_start += main_data_slots(fr_ps);
if (bytes_to_discard < 0) {
printf("Not enough main data to decode frame %d. Frame discarded.\n",
frameNum - 1); break;
}
for (; bytes_to_discard > 0; bytes_to_discard--) hgetbits(8);
clip = 0;
for (gr=0;gr<2;gr++) {
double lr[2][SBLIMIT][SSLIMIT],ro[2][SBLIMIT][SSLIMIT];
//主解码
for (ch=0; ch<fr_ps.stereo; ch++) {
long int is[SBLIMIT][SSLIMIT]; /*保存量化数据*/
int part2_start;
part2_start = hsstell();
//获取比例因子
III_get_scale_factors(&III_scalefac,&III_side_info, gr, ch, &fr_ps);
//Huffman解码
III_hufman_decode(is, &III_side_info, ch, gr, part2_start, &fr_ps);
//反量化采样
III_dequantize_sample(is, ro[ch], &III_scalefac, &(III_side_info.ch[ch].gr[gr]), ch, &fr_ps);
}
//立体声处理
III_stereo(ro, lr, &III_scalefac, &(III_side_info.ch[0].gr[gr]), &fr_ps);
for (ch=0; ch<fr_ps.stereo; ch++) {
double re[SBLIMIT][SSLIMIT];
double hybridIn[SBLIMIT][SSLIMIT];/* Hybrid filter input */
double hybridOut[SBLIMIT][SSLIMIT];/* Hybrid filter out */
double polyPhaseIn[SBLIMIT]; /* PolyPhase Input. */
III_reorder(lr[ch], re, &(III_side_info.ch[ch].gr[gr]), &fr_ps);
//抗锯齿处理
III_antialias(re, hybridIn, /* Antialias butterflies. */
&(III_side_info.ch[ch].gr[gr]), &fr_ps);
//IMDCT
for (sb=0; sb<SBLIMIT; sb++) { /* Hybrid synthesis. */
III_hybrid(hybridIn[sb], hybridOut[sb], sb, ch, &(III_side_info.ch[ch].gr[gr]), &fr_ps);
}
for (ss=0;ss<18;ss++) //多相频率倒置
for (sb=0; sb<SBLIMIT; sb++)
if ((ss%2) && (sb%2))
hybridOut[sb][ss] = -hybridOut[sb][ss];
for (ss=0;ss<18;ss++) { //多相合成
for (sb=0; sb<SBLIMIT; sb++)
polyPhaseIn[sb] = hybridOut[sb][ss];
//子带合成
clip += SubBandSynthesis(polyPhaseIn, ch, &((*pcm_sample)[ch][ss][0]));
}
}
//PCM输出
/* Output PCM sample points for one granule(颗粒). */
out_fifo(*pcm_sample, 18, &fr_ps, done, musicout, &sample_frames);
}
if(clip > 0)
printf("\n%d samples clipped.\n", clip);
}
break;
default:
printf("\nOnly layer III supported!\n");
exit(1);
break;
}
}
close_bit_stream_r(&bs);
fclose(musicout);
printf("\nDecoding done.\n");
return;
}
int toolame_encode_frame(
short buffer[2][1152],
unsigned char *xpad_data,
unsigned char *output_buffer,
size_t output_buffer_size)
{
extern int minimum;
if (encode_first_call) {
hdr_to_frps(&frame);
encode_first_call = 0;
}
const int nch = frame.nch;
const int error_protection = header.error_protection;
bs.output_buffer = output_buffer;
bs.output_buffer_size = output_buffer_size;
bs.output_buffer_written = 0;
#ifdef REFERENCECODE
short *win_buf[2] = {&buffer[0][0], &buffer[1][0]};
#endif
int adb = available_bits (&header, &glopts);
int lg_frame = adb / 8;
if (header.dab_extension) {
/* You must have one frame in memory if you are in DAB mode */
/* in conformity of the norme ETS 300 401 http://www.etsi.org */
/* see bitstream.c */
if (frameNum == 1)
minimum = lg_frame + MINIMUM;
adb -= header.dab_extension * 8 + (xpad_len ? xpad_len : FPAD_LENGTH) * 8;
}
{
int gr, bl, ch;
/* New polyphase filter
Combines windowing and filtering. Ricardo Feb'03 */
for( gr = 0; gr < 3; gr++ )
for ( bl = 0; bl < 12; bl++ )
for ( ch = 0; ch < nch; ch++ )
WindowFilterSubband( &buffer[ch][gr * 12 * 32 + 32 * bl], ch,
&(*sb_sample)[ch][gr][bl][0] );
}
#ifdef REFERENCECODE
{
/* Old code. left here for reference */
int gr, bl, ch;
for (gr = 0; gr < 3; gr++)
for (bl = 0; bl < SCALE_BLOCK; bl++)
for (ch = 0; ch < nch; ch++) {
window_subband (&win_buf[ch], &(*win_que)[ch][0], ch);
filter_subband (&(*win_que)[ch][0], &(*sb_sample)[ch][gr][bl][0]);
}
}
#endif
#ifdef NEWENCODE
scalefactor_calc_new(*sb_sample, scalar, nch, frame.sblimit);
find_sf_max (scalar, &frame, max_sc);
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/* this way we calculate more mono than we need */
/* but it is cheap */
combine_LR_new (*sb_sample, *j_sample, frame.sblimit);
scalefactor_calc_new (j_sample, &j_scale, 1, frame.sblimit);
}
#else
scale_factor_calc (*sb_sample, scalar, nch, frame.sblimit);
pick_scale (scalar, &frame, max_sc);
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/* this way we calculate more mono than we need */
/* but it is cheap */
combine_LR (*sb_sample, *j_sample, frame.sblimit);
scale_factor_calc (j_sample, &j_scale, 1, frame.sblimit);
}
#endif
if ((glopts.quickmode == TRUE) && (++psycount % glopts.quickcount != 0)) {
/* We're using quick mode, so we're only calculating the model every
'quickcount' frames. Otherwise, just copy the old ones across */
for (int ch = 0; ch < nch; ch++) {
for (int sb = 0; sb < SBLIMIT; sb++)
smr[ch][sb] = smrdef[ch][sb];
}
}
else {
/* calculate the psymodel */
switch (model) {
case -1:
psycho_n1 (smr, nch);
break;
case 0: /* Psy Model A */
psycho_0 (smr, nch, scalar, (FLOAT) s_freq[header.version][header.sampling_frequency] * 1000);
break;
case 1:
psycho_1 (buffer, max_sc, smr, &frame);
break;
case 2:
for (int ch = 0; ch < nch; ch++) {
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
}
break;
case 3:
/* Modified psy model 1 */
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
break;
case 4:
/* Modified Psycho Model 2 */
for (int ch = 0; ch < nch; ch++) {
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
}
break;
case 5:
/* Model 5 comparse model 1 and 3 */
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1 ");
smr_dump(smr,nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3 ");
smr_dump(smr,nch);
break;
case 6:
/* Model 6 compares model 2 and 4 */
for (int ch = 0; ch < nch; ch++)
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"2 ");
smr_dump(smr,nch);
for (int ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4 ");
smr_dump(smr,nch);
break;
case 7:
fprintf(stdout,"Frame: %i\n",frameNum);
/* Dump the SMRs for all models */
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1");
smr_dump(smr, nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3");
smr_dump(smr,nch);
for (int ch = 0; ch < nch; ch++)
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"2");
smr_dump(smr,nch);
for (int ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
case 8:
/* Compare 0 and 4 */
psycho_n1 (smr, nch);
fprintf(stdout,"0");
smr_dump(smr,nch);
for (int ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
default:
fprintf (stderr, "Invalid psy model specification: %i\n", model);
exit (0);
}
if (glopts.quickmode == TRUE) {
/* copy the smr values and reuse them later */
for (int ch = 0; ch < nch; ch++) {
for (int sb = 0; sb < SBLIMIT; sb++)
smrdef[ch][sb] = smr[ch][sb];
}
}
if (glopts.verbosity > 4) {
smr_dump(smr, nch);
}
}
#ifdef NEWENCODE
sf_transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation_new (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
//main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
if (error_protection) {
CRC_calc (&frame, bit_alloc, scfsi, &crc);
}
write_header (&frame, &bs);
//encode_info (&frame, &bs);
if (error_protection) {
putbits (&bs, crc, 16);
}
write_bit_alloc (bit_alloc, &frame, &bs);
//encode_bit_alloc (bit_alloc, &frame, &bs);
write_scalefactors(bit_alloc, scfsi, scalar, &frame, &bs);
//encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization_new (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
*subband, &frame);
//subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
// *subband, &frame);
write_samples_new(*subband, bit_alloc, &frame, &bs);
//sample_encoding (*subband, bit_alloc, &frame, &bs);
#else
transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
if (error_protection) {
CRC_calc (&frame, bit_alloc, scfsi, &crc);
}
encode_info (&frame, &bs);
if (error_protection) {
encode_CRC (crc, &bs);
}
encode_bit_alloc (bit_alloc, &frame, &bs);
encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
*subband, &frame);
sample_encoding (*subband, bit_alloc, &frame, &bs);
#endif
/* If not all the bits were used, write out a stack of zeros */
for (int i = 0; i < adb; i++) {
put1bit (&bs, 0);
}
if (xpad_len) {
assert(xpad_len > 2);
// insert available X-PAD
for (int i = header.dab_length - xpad_len;
i < header.dab_length - FPAD_LENGTH;
i++) {
putbits (&bs, xpad_data[i], 8);
}
}
for (int i = header.dab_extension - 1; i >= 0; i--) {
CRC_calcDAB (&frame, bit_alloc, scfsi, scalar, &crc, i);
/* this crc is for the previous frame in DAB mode */
if (bs.buf_byte_idx + lg_frame < bs.buf_size)
bs.buf[bs.buf_byte_idx + lg_frame] = crc;
/* reserved 2 bytes for F-PAD in DAB mode */
putbits (&bs, crc, 8);
}
if (xpad_len) {
/* The F-PAD is also given us by mot-encoder */
putbits (&bs, xpad_data[header.dab_length - 2], 8);
putbits (&bs, xpad_data[header.dab_length - 1], 8);
}
else {
putbits (&bs, 0, 16); // FPAD is all-zero
}
return bs.output_buffer_written;
}
int main(int argc, char**argv)
{
int pid = 0;//fork();
if (pid == 0) {
// exec("sh",argv);
}
int decodepid = fork();
if (decodepid == 0) {
exec("mp3dec", argv);
}
int mp3pid = 0;//fork();
if (mp3pid == 0) {
// exec("decode", argv);
}
setSampleRate(44100);
Bit_stream_struc bs;
struct frame_params fr_ps;
struct III_side_info_t III_side_info;
unsigned int old_crc;
layer info;
unsigned long bitsPerSlot;
/*if (argc==1) {
printf("Useage:decode file.mp3 output.pcm\n");
return;
}*/
fr_ps.header = &info;
open_bit_stream_r(&bs, "in.mp3", BUFFER_SIZE);
int frame_Num = 0;
while(!end_bs(&bs)) {
//³¢ÊÔ֡ͬ²œ
seek_sync(&bs, SYNC_WORD, SYNC_WORD_LENGTH);
decode_info(&bs, &(fr_ps));
//œ«fr_ps.headerÖеÄÐÅÏ¢œâ¶Áµœfr_psµÄÏà¹ØÓòÖÐ
hdr_to_frps(&(fr_ps));
//Êä³öÏà¹ØÐÅÏ¢
frame_Num = frame_Num + 1;
printf(0, "read frame: %d\n", frame_Num);
if (info.error_protection)
buffer_CRC(&bs, &old_crc);
switch (info.lay) {
case 3:
{
int nSlots, main_data_end, flush_main;
int bytes_to_discard;
static int frame_start = 0;
bitsPerSlot = 8;
//È¡SideÐÅÏ¢
III_get_side_info(&bs, &(III_side_info), &(fr_ps));
nSlots = main_data_slots(fr_ps);
//¶ÁÖ÷ÊýŸÝ(Audio Data)
for (; nSlots > 0; nSlots--) /* read main data. */
hputbuf((unsigned int) getbits(&bs,8), 8);
main_data_end = hsstell() / 8; /*of privious frame*/
if ( (flush_main=(hsstell() % bitsPerSlot))==TRUE ) {
hgetbits((int)(bitsPerSlot - flush_main));
main_data_end ++;
}
bytes_to_discard = frame_start - main_data_end - III_side_info.main_data_begin ;
if( main_data_end > 4096 ) { frame_start -= 4096;
rewindNbytes( 4096 );
}
frame_start += main_data_slots(fr_ps);
// printf(0, "discard : %d\n", bytes_to_discard);
if (bytes_to_discard < 0) {
printf(0, "discard: %d %d %d\n", frame_start, main_data_end, III_side_info.main_data_begin);
//printf(0, "Not enough main data to decode frame %d. Frame discarded.\n",frame_Num - 1);
break;
}
for (; bytes_to_discard > 0; bytes_to_discard--) hgetbits(8);
beginDecode(&fr_ps, &III_side_info);
}
break;
default:
// printf("\nOnly layer III supported!\n");
exit();
break;
}
}
close_bit_stream_r(&bs);
printf(0, "\nPlaying done.\n");
kill(pid);
kill(decodepid);
kill(mp3pid);
wait();
wait();
wait();
exit();
}
int main (int argc, char **argv)
{
typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT];
SBS *sb_sample;
typedef double JSBS[3][SCALE_BLOCK][SBLIMIT];
JSBS *j_sample;
#ifdef REFERENCECODE
typedef double IN[2][HAN_SIZE];
IN *win_que;
#endif
typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT];
SUB *subband;
frame_info frame;
frame_header header;
char original_file_name[MAX_NAME_SIZE];
char encoded_file_name[MAX_NAME_SIZE];
short **win_buf;
static short buffer[2][1152];
static unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT];
static unsigned int scalar[2][3][SBLIMIT], j_scale[3][SBLIMIT];
static double smr[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT];
// FLOAT snr32[32];
short sam[2][1344]; /* was [1056]; */
int model, nch, error_protection;
static unsigned int crc;
int sb, ch, adb;
unsigned long frameBits, sentBits = 0;
unsigned long num_samples;
int lg_frame;
int i;
/* Keep track of peaks */
int peak_left = 0;
int peak_right = 0;
char* mot_file = NULL;
char* icy_file = NULL;
/* Used to keep the SNR values for the fast/quick psy models */
static FLOAT smrdef[2][32];
static int psycount = 0;
extern int minimum;
sb_sample = (SBS *) mem_alloc (sizeof (SBS), "sb_sample");
j_sample = (JSBS *) mem_alloc (sizeof (JSBS), "j_sample");
#ifdef REFERENCECODE
win_que = (IN *) mem_alloc (sizeof (IN), "Win_que");
#endif
subband = (SUB *) mem_alloc (sizeof (SUB), "subband");
win_buf = (short **) mem_alloc (sizeof (short *) * 2, "win_buf");
/* clear buffers */
memset ((char *) buffer, 0, sizeof (buffer));
memset ((char *) bit_alloc, 0, sizeof (bit_alloc));
memset ((char *) scalar, 0, sizeof (scalar));
memset ((char *) j_scale, 0, sizeof (j_scale));
memset ((char *) scfsi, 0, sizeof (scfsi));
memset ((char *) smr, 0, sizeof (smr));
memset ((char *) lgmin, 0, sizeof (lgmin));
memset ((char *) max_sc, 0, sizeof (max_sc));
//memset ((char *) snr32, 0, sizeof (snr32));
memset ((char *) sam, 0, sizeof (sam));
global_init ();
header.extension = 0;
frame.header = &header;
frame.tab_num = -1; /* no table loaded */
frame.alloc = NULL;
header.version = MPEG_AUDIO_ID; /* Default: MPEG-1 */
programName = argv[0];
if (argc == 1) /* no command-line args */
short_usage ();
else
parse_args (argc, argv, &frame, &model, &num_samples, original_file_name,
encoded_file_name, &mot_file, &icy_file);
print_config (&frame, &model, original_file_name, encoded_file_name);
uint8_t* xpad_data = NULL;
if (mot_file) {
if (header.dab_length <= 0) {
fprintf(stderr, "Invalid XPAD length specified\n");
return 1;
}
int err = xpad_init(mot_file, header.dab_length + 1);
if (err == -1) {
fprintf(stderr, "XPAD reader initialisation failed\n");
return 1;
}
xpad_data = malloc(header.dab_length + 1);
}
/* this will load the alloc tables and do some other stuff */
hdr_to_frps (&frame);
nch = frame.nch;
error_protection = header.error_protection;
unsigned long samps_read;
while ((samps_read = get_audio(&musicin, buffer, num_samples, nch, &header)) > 0) {
/* Check if we have new PAD data
*/
int xpad_len = 0;
if (mot_file) {
xpad_len = xpad_read_len(xpad_data, header.dab_length + 1);
if (xpad_len == -1) {
fprintf(stderr, "Error reading XPAD data\n");
xpad_len = 0;
}
else if (xpad_len == 0) {
// no PAD available
}
else if (xpad_len == header.dab_length + 1) {
// everything OK
xpad_len = xpad_data[header.dab_length];
}
else {
fprintf(stderr, "xpad length=%d\n", xpad_len);
abort();
}
}
unsigned long j;
for (j = 0; j < samps_read; j++) {
peak_left = MAX(peak_left, buffer[0][j]);
}
for (j = 0; j < samps_read; j++) {
peak_right = MAX(peak_right, buffer[1][j]);
}
// We can always set the zmq peaks, even if the output is not
// used, it just writes some variables
zmqoutput_set_peaks(peak_left, peak_right);
if (glopts.verbosity > 1)
if (++frameNum % 10 == 0) {
fprintf(stderr, "[%4u", frameNum);
if (mot_file) {
fprintf(stderr, " %s",
xpad_len > 0 ? "p" : " ");
}
if (glopts.show_level) {
fprintf(stderr, " (%6d|%-6d) ",
peak_left, peak_right);
fprintf(stderr, "] [%6s|%-6s]\r",
level(0, &peak_left),
level(1, &peak_right) );
}
else {
fprintf(stderr, "]\r");
}
}
fflush(stderr);
win_buf[0] = &buffer[0][0];
win_buf[1] = &buffer[1][0];
adb = available_bits (&header, &glopts);
lg_frame = adb / 8;
if (header.dab_extension) {
/* in 24 kHz we always have 4 bytes */
if (header.sampling_frequency == 1)
header.dab_extension = 4;
/* You must have one frame in memory if you are in DAB mode */
/* in conformity of the norme ETS 300 401 http://www.etsi.org */
/* see bitstream.c */
if (frameNum == 1)
minimum = lg_frame + MINIMUM;
adb -= header.dab_extension * 8 + (xpad_len ? xpad_len : FPAD_LENGTH) * 8;
}
{
int gr, bl, ch;
/* New polyphase filter
Combines windowing and filtering. Ricardo Feb'03 */
for( gr = 0; gr < 3; gr++ )
for ( bl = 0; bl < 12; bl++ )
for ( ch = 0; ch < nch; ch++ )
WindowFilterSubband( &buffer[ch][gr * 12 * 32 + 32 * bl], ch,
&(*sb_sample)[ch][gr][bl][0] );
}
#ifdef REFERENCECODE
{
/* Old code. left here for reference */
int gr, bl, ch;
for (gr = 0; gr < 3; gr++)
for (bl = 0; bl < SCALE_BLOCK; bl++)
for (ch = 0; ch < nch; ch++) {
window_subband (&win_buf[ch], &(*win_que)[ch][0], ch);
filter_subband (&(*win_que)[ch][0], &(*sb_sample)[ch][gr][bl][0]);
}
}
#endif
#ifdef NEWENCODE
scalefactor_calc_new(*sb_sample, scalar, nch, frame.sblimit);
find_sf_max (scalar, &frame, max_sc);
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/* this way we calculate more mono than we need */
/* but it is cheap */
combine_LR_new (*sb_sample, *j_sample, frame.sblimit);
scalefactor_calc_new (j_sample, &j_scale, 1, frame.sblimit);
}
#else
scale_factor_calc (*sb_sample, scalar, nch, frame.sblimit);
pick_scale (scalar, &frame, max_sc);
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/* this way we calculate more mono than we need */
/* but it is cheap */
combine_LR (*sb_sample, *j_sample, frame.sblimit);
scale_factor_calc (j_sample, &j_scale, 1, frame.sblimit);
}
#endif
if ((glopts.quickmode == TRUE) && (++psycount % glopts.quickcount != 0)) {
/* We're using quick mode, so we're only calculating the model every
'quickcount' frames. Otherwise, just copy the old ones across */
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++)
smr[ch][sb] = smrdef[ch][sb];
}
} else {
/* calculate the psymodel */
switch (model) {
case -1:
psycho_n1 (smr, nch);
break;
case 0: /* Psy Model A */
psycho_0 (smr, nch, scalar, (FLOAT) s_freq[header.version][header.sampling_frequency] * 1000);
break;
case 1:
psycho_1 (buffer, max_sc, smr, &frame);
break;
case 2:
for (ch = 0; ch < nch; ch++) {
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
}
break;
case 3:
/* Modified psy model 1 */
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
break;
case 4:
/* Modified Psycho Model 2 */
for (ch = 0; ch < nch; ch++) {
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
}
break;
case 5:
/* Model 5 comparse model 1 and 3 */
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1 ");
smr_dump(smr,nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3 ");
smr_dump(smr,nch);
break;
case 6:
/* Model 6 compares model 2 and 4 */
for (ch = 0; ch < nch; ch++)
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"2 ");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4 ");
smr_dump(smr,nch);
break;
case 7:
fprintf(stdout,"Frame: %i\n",frameNum);
/* Dump the SMRs for all models */
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1");
smr_dump(smr, nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"2");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
case 8:
/* Compare 0 and 4 */
psycho_n1 (smr, nch);
fprintf(stdout,"0");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
default:
fprintf (stderr, "Invalid psy model specification: %i\n", model);
exit (0);
}
if (glopts.quickmode == TRUE)
/* copy the smr values and reuse them later */
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++)
smrdef[ch][sb] = smr[ch][sb];
}
if (glopts.verbosity > 4)
smr_dump(smr, nch);
}
#ifdef NEWENCODE
sf_transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation_new (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
//main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
if (error_protection)
CRC_calc (&frame, bit_alloc, scfsi, &crc);
write_header (&frame, &bs);
//encode_info (&frame, &bs);
if (error_protection)
putbits (&bs, crc, 16);
write_bit_alloc (bit_alloc, &frame, &bs);
//encode_bit_alloc (bit_alloc, &frame, &bs);
write_scalefactors(bit_alloc, scfsi, scalar, &frame, &bs);
//encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization_new (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
*subband, &frame);
//subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
// *subband, &frame);
write_samples_new(*subband, bit_alloc, &frame, &bs);
//sample_encoding (*subband, bit_alloc, &frame, &bs);
#else
transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
if (error_protection)
CRC_calc (&frame, bit_alloc, scfsi, &crc);
encode_info (&frame, &bs);
if (error_protection)
encode_CRC (crc, &bs);
encode_bit_alloc (bit_alloc, &frame, &bs);
encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
*subband, &frame);
sample_encoding (*subband, bit_alloc, &frame, &bs);
#endif
/* If not all the bits were used, write out a stack of zeros */
for (i = 0; i < adb; i++)
put1bit (&bs, 0);
if (xpad_len) {
assert(xpad_len > 2);
// insert available X-PAD
for (i = header.dab_length - xpad_len; i < header.dab_length - FPAD_LENGTH; i++)
putbits (&bs, xpad_data[i], 8);
}
for (i = header.dab_extension - 1; i >= 0; i--) {
CRC_calcDAB (&frame, bit_alloc, scfsi, scalar, &crc, i);
/* this crc is for the previous frame in DAB mode */
if (bs.buf_byte_idx + lg_frame < bs.buf_size)
bs.buf[bs.buf_byte_idx + lg_frame] = crc;
/* reserved 2 bytes for F-PAD in DAB mode */
putbits (&bs, crc, 8);
}
if (xpad_len) {
/* The F-PAD is also given us by mot-encoder */
putbits (&bs, xpad_data[header.dab_length - 2], 8);
putbits (&bs, xpad_data[header.dab_length - 1], 8);
}
else {
putbits (&bs, 0, 16); // FPAD is all-zero
}
#if defined(VLC_INPUT)
if (glopts.input_select == INPUT_SELECT_VLC) {
vlc_in_write_icy();
}
#endif
frameBits = sstell (&bs) - sentBits;
if (frameBits % 8) { /* a program failure */
fprintf (stderr, "Sent %ld bits = %ld slots plus %ld\n", frameBits,
frameBits / 8, frameBits % 8);
fprintf (stderr, "If you are reading this, the program is broken\n");
fprintf (stderr, "Please report a bug.\n");
exit(1);
}
sentBits += frameBits;
// Reset peak measurement
peak_left = 0;
peak_right = 0;
}
fprintf(stdout, "Main loop has quit with samps_read = %zu\n", samps_read);
close_bit_stream_w (&bs);
if ((glopts.verbosity > 1) && (glopts.vbr == TRUE)) {
int i;
#ifdef NEWENCODE
extern int vbrstats_new[15];
#else
extern int vbrstats[15];
#endif
fprintf (stdout, "VBR stats:\n");
for (i = 1; i < 15; i++)
fprintf (stdout, "%4i ", bitrate[header.version][i]);
fprintf (stdout, "\n");
for (i = 1; i < 15; i++)
#ifdef NEWENCODE
fprintf (stdout,"%4i ",vbrstats_new[i]);
#else
fprintf (stdout, "%4i ", vbrstats[i]);
#endif
fprintf (stdout, "\n");
}
fprintf (stderr,
"Avg slots/frame = %.3f; b/smp = %.2f; bitrate = %.3f kbps\n",
(FLOAT) sentBits / (frameNum * 8),
(FLOAT) sentBits / (frameNum * 1152),
(FLOAT) sentBits / (frameNum * 1152) *
s_freq[header.version][header.sampling_frequency]);
if (glopts.input_select == INPUT_SELECT_WAV) {
if ( fclose (musicin.wav_input) != 0) {
fprintf (stderr, "Could not close \"%s\".\n", original_file_name);
exit (2);
}
}
fprintf (stderr, "\nDone\n");
exit (0);
}
int main (int argc, char **argv)
{
typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT];
SBS *sb_sample;
typedef double JSBS[3][SCALE_BLOCK][SBLIMIT];
JSBS *j_sample;
typedef double IN[2][HAN_SIZE];
IN *win_que;
typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT];
SUB *subband;
frame_info frame;
frame_header header;
char original_file_name[MAX_NAME_SIZE];
char encoded_file_name[MAX_NAME_SIZE];
short **win_buf;
static short buffer[2][1152];
static unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT];
static unsigned int scalar[2][3][SBLIMIT], j_scale[3][SBLIMIT];
static double smr[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT];
// FLOAT snr32[32];
short sam[2][1344]; /* was [1056]; */
int model, nch, error_protection;
static unsigned int crc;
int sb, ch, adb;
unsigned long frameBits, sentBits = 0;
unsigned long num_samples;
int lg_frame;
int i;
/* Used to keep the SNR values for the fast/quick psy models */
static FLOAT smrdef[2][32];
static int psycount = 0;
extern int minimum;
sb_sample = (SBS *) mem_alloc (sizeof (SBS), "sb_sample");
j_sample = (JSBS *) mem_alloc (sizeof (JSBS), "j_sample");
win_que = (IN *) mem_alloc (sizeof (IN), "Win_que");
subband = (SUB *) mem_alloc (sizeof (SUB), "subband");
win_buf = (short **) mem_alloc (sizeof (short *) * 2, "win_buf");
/* clear buffers */
memset ((char *) buffer, 0, sizeof (buffer));
memset ((char *) bit_alloc, 0, sizeof (bit_alloc));
memset ((char *) scalar, 0, sizeof (scalar));
memset ((char *) j_scale, 0, sizeof (j_scale));
memset ((char *) scfsi, 0, sizeof (scfsi));
memset ((char *) smr, 0, sizeof (smr));
memset ((char *) lgmin, 0, sizeof (lgmin));
memset ((char *) max_sc, 0, sizeof (max_sc));
//memset ((char *) snr32, 0, sizeof (snr32));
memset ((char *) sam, 0, sizeof (sam));
global_init ();
header.extension = 0;
frame.header = &header;
frame.tab_num = -1; /* no table loaded */
frame.alloc = NULL;
header.version = MPEG_AUDIO_ID; /* Default: MPEG-1 */
programName = argv[0];
if (argc == 1) /* no command-line args */
short_usage ();
else
parse_args (argc, argv, &frame, &model, &num_samples, original_file_name,
encoded_file_name);
print_config (&frame, &model, original_file_name, encoded_file_name);
/* this will load the alloc tables and do some other stuff */
hdr_to_frps (&frame);
nch = frame.nch;
error_protection = header.error_protection;
while (get_audio (musicin, buffer, num_samples, nch, &header) > 0) {
if (glopts.verbosity > 1)
if (++frameNum % 10 == 0)
fprintf (stderr, "[%4u]\r", frameNum);
fflush (stderr);
win_buf[0] = &buffer[0][0];
win_buf[1] = &buffer[1][0];
adb = available_bits (&header, &glopts);
lg_frame = adb / 8;
if (header.dab_extension) {
/* in 24 kHz we always have 4 bytes */
if (header.sampling_frequency == 1)
header.dab_extension = 4;
/* You must have one frame in memory if you are in DAB mode */
/* in conformity of the norme ETS 300 401 http://www.etsi.org */
/* see bitstream.c */
if (frameNum == 1)
minimum = lg_frame + MINIMUM;
adb -= header.dab_extension * 8 + header.dab_length * 8 + 16;
}
{
int gr, bl, ch;
/* New polyphase filter
Combines windowing and filtering. Ricardo Feb'03 */
for( gr = 0; gr < 3; gr++ )
for ( bl = 0; bl < 12; bl++ )
for ( ch = 0; ch < nch; ch++ )
WindowFilterSubband( &buffer[ch][gr * 12 * 32 + 32 * bl], ch,
&(*sb_sample)[ch][gr][bl][0] );
}
#ifdef REFERENCECODE
{
/* Old code. left here for reference */
int gr, bl, ch;
for (gr = 0; gr < 3; gr++)
for (bl = 0; bl < SCALE_BLOCK; bl++)
for (ch = 0; ch < nch; ch++) {
window_subband (&win_buf[ch], &(*win_que)[ch][0], ch);
filter_subband (&(*win_que)[ch][0], &(*sb_sample)[ch][gr][bl][0]);
}
}
#endif
#ifdef NEWENCODE
scalefactor_calc_new(*sb_sample, scalar, nch, frame.sblimit);
find_sf_max (scalar, &frame, max_sc);
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/* this way we calculate more mono than we need */
/* but it is cheap */
combine_LR_new (*sb_sample, *j_sample, frame.sblimit);
scalefactor_calc_new (j_sample, &j_scale, 1, frame.sblimit);
}
#else
scale_factor_calc (*sb_sample, scalar, nch, frame.sblimit);
pick_scale (scalar, &frame, max_sc);
if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
/* this way we calculate more mono than we need */
/* but it is cheap */
combine_LR (*sb_sample, *j_sample, frame.sblimit);
scale_factor_calc (j_sample, &j_scale, 1, frame.sblimit);
}
#endif
if ((glopts.quickmode == TRUE) && (++psycount % glopts.quickcount != 0)) {
/* We're using quick mode, so we're only calculating the model every
'quickcount' frames. Otherwise, just copy the old ones across */
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++)
smr[ch][sb] = smrdef[ch][sb];
}
} else {
/* calculate the psymodel */
switch (model) {
case -1:
psycho_n1 (smr, nch);
break;
case 0: /* Psy Model A */
psycho_0 (smr, nch, scalar, (FLOAT) s_freq[header.version][header.sampling_frequency] * 1000);
break;
case 1:
psycho_1 (buffer, max_sc, smr, &frame);
break;
case 2:
for (ch = 0; ch < nch; ch++) {
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
}
break;
case 3:
/* Modified psy model 1 */
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
break;
case 4:
/* Modified Psycho Model 2 */
for (ch = 0; ch < nch; ch++) {
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
}
break;
case 5:
/* Model 5 comparse model 1 and 3 */
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1 ");
smr_dump(smr,nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3 ");
smr_dump(smr,nch);
break;
case 6:
/* Model 6 compares model 2 and 4 */
for (ch = 0; ch < nch; ch++)
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"2 ");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4 ");
smr_dump(smr,nch);
break;
case 7:
fprintf(stdout,"Frame: %i\n",frameNum);
/* Dump the SMRs for all models */
psycho_1 (buffer, max_sc, smr, &frame);
fprintf(stdout,"1");
smr_dump(smr, nch);
psycho_3 (buffer, max_sc, smr, &frame, &glopts);
fprintf(stdout,"3");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"2");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
case 8:
/* Compare 0 and 4 */
psycho_n1 (smr, nch);
fprintf(stdout,"0");
smr_dump(smr,nch);
for (ch = 0; ch < nch; ch++)
psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
(FLOAT) s_freq[header.version][header.sampling_frequency] *
1000, &glopts);
fprintf(stdout,"4");
smr_dump(smr,nch);
break;
default:
fprintf (stderr, "Invalid psy model specification: %i\n", model);
exit (0);
}
if (glopts.quickmode == TRUE)
/* copy the smr values and reuse them later */
for (ch = 0; ch < nch; ch++) {
for (sb = 0; sb < SBLIMIT; sb++)
smrdef[ch][sb] = smr[ch][sb];
}
if (glopts.verbosity > 4)
smr_dump(smr, nch);
}
#ifdef NEWENCODE
sf_transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation_new (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
//main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
if (error_protection)
CRC_calc (&frame, bit_alloc, scfsi, &crc);
write_header (&frame, &bs);
//encode_info (&frame, &bs);
if (error_protection)
putbits (&bs, crc, 16);
write_bit_alloc (bit_alloc, &frame, &bs);
//encode_bit_alloc (bit_alloc, &frame, &bs);
write_scalefactors(bit_alloc, scfsi, scalar, &frame, &bs);
//encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization_new (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
*subband, &frame);
//subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
// *subband, &frame);
write_samples_new(*subband, bit_alloc, &frame, &bs);
//sample_encoding (*subband, bit_alloc, &frame, &bs);
#else
transmission_pattern (scalar, scfsi, &frame);
main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
if (error_protection)
CRC_calc (&frame, bit_alloc, scfsi, &crc);
encode_info (&frame, &bs);
if (error_protection)
encode_CRC (crc, &bs);
encode_bit_alloc (bit_alloc, &frame, &bs);
encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
*subband, &frame);
sample_encoding (*subband, bit_alloc, &frame, &bs);
#endif
/* If not all the bits were used, write out a stack of zeros */
for (i = 0; i < adb; i++)
put1bit (&bs, 0);
if (header.dab_extension) {
/* Reserve some bytes for X-PAD in DAB mode */
putbits (&bs, 0, header.dab_length * 8);
for (i = header.dab_extension - 1; i >= 0; i--) {
CRC_calcDAB (&frame, bit_alloc, scfsi, scalar, &crc, i);
/* this crc is for the previous frame in DAB mode */
if (bs.buf_byte_idx + lg_frame < bs.buf_size)
bs.buf[bs.buf_byte_idx + lg_frame] = crc;
/* reserved 2 bytes for F-PAD in DAB mode */
putbits (&bs, crc, 8);
}
putbits (&bs, 0, 16);
}
frameBits = sstell (&bs) - sentBits;
if (frameBits % 8) { /* a program failure */
fprintf (stderr, "Sent %ld bits = %ld slots plus %ld\n", frameBits,
frameBits / 8, frameBits % 8);
fprintf (stderr, "If you are reading this, the program is broken\n");
fprintf (stderr, "email [mfc at NOTplanckenerg.com] without the NOT\n");
fprintf (stderr, "with the command line arguments and other info\n");
exit (0);
}
sentBits += frameBits;
}
close_bit_stream_w (&bs);
if ((glopts.verbosity > 1) && (glopts.vbr == TRUE)) {
int i;
#ifdef NEWENCODE
extern int vbrstats_new[15];
#else
extern int vbrstats[15];
#endif
fprintf (stdout, "VBR stats:\n");
for (i = 1; i < 15; i++)
fprintf (stdout, "%4i ", bitrate[header.version][i]);
fprintf (stdout, "\n");
for (i = 1; i < 15; i++)
#ifdef NEWENCODE
fprintf (stdout,"%4i ",vbrstats_new[i]);
#else
fprintf (stdout, "%4i ", vbrstats[i]);
#endif
fprintf (stdout, "\n");
}
fprintf (stderr,
"Avg slots/frame = %.3f; b/smp = %.2f; bitrate = %.3f kbps\n",
(FLOAT) sentBits / (frameNum * 8),
(FLOAT) sentBits / (frameNum * 1152),
(FLOAT) sentBits / (frameNum * 1152) *
s_freq[header.version][header.sampling_frequency]);
if (fclose (musicin) != 0) {
fprintf (stderr, "Could not close \"%s\".\n", original_file_name);
exit (2);
}
fprintf (stderr, "\nDone\n");
exit (0);
}
