summaryrefslogtreecommitdiffstats
path: root/libtoolame-dab/toolame.c
diff options
context:
space:
mode:
Diffstat (limited to 'libtoolame-dab/toolame.c')
-rw-r--r--libtoolame-dab/toolame.c1457
1 files changed, 1457 insertions, 0 deletions
diff --git a/libtoolame-dab/toolame.c b/libtoolame-dab/toolame.c
new file mode 100644
index 0000000..09eb69e
--- /dev/null
+++ b/libtoolame-dab/toolame.c
@@ -0,0 +1,1457 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#if defined(JACK_INPUT)
+# include <jack/jack.h>
+# include <jack/ringbuffer.h>
+#endif
+#include "common.h"
+#include "encoder.h"
+#include "musicin.h"
+#include "options.h"
+#include "audio_read.h"
+#include "bitstream.h"
+#include "mem.h"
+#include "crc.h"
+#include "psycho_n1.h"
+#include "psycho_0.h"
+#include "psycho_1.h"
+#include "psycho_2.h"
+#include "psycho_3.h"
+#include "psycho_4.h"
+#include "encode.h"
+#include "availbits.h"
+#include "subband.h"
+#include "encode_new.h"
+#include "toolame.h"
+#include "xpad.h"
+#include "utils.h"
+#include "vlc_input.h"
+#include "zmqoutput.h"
+
+#include <assert.h>
+
+music_in_t musicin;
+Bit_stream_struc bs;
+char *programName;
+char toolameversion[] = "0.2l-ODR";
+
+const int FPAD_LENGTH=2;
+
+void global_init (void)
+{
+ glopts.usepsy = TRUE;
+ glopts.usepadbit = TRUE;
+ glopts.quickmode = FALSE;
+ glopts.quickcount = 10;
+ glopts.downmix = FALSE;
+ glopts.byteswap = FALSE;
+ glopts.channelswap = FALSE;
+ glopts.vbr = FALSE;
+ glopts.vbrlevel = 0;
+ glopts.athlevel = 0;
+ glopts.verbosity = 2;
+ glopts.input_select = 0;
+}
+
+/************************************************************************
+ *
+ * main
+ *
+ * PURPOSE: MPEG II Encoder with
+ * psychoacoustic models 1 (MUSICAM) and 2 (AT&T)
+ *
+ * SEMANTICS: One overlapping frame of audio of up to 2 channels are
+ * processed at a time in the following order:
+ * (associated routines are in parentheses)
+ *
+ * 1. Filter sliding window of data to get 32 subband
+ * samples per channel.
+ * (window_subband,filter_subband)
+ *
+ * 2. If joint stereo mode, combine left and right channels
+ * for subbands above #jsbound#.
+ * (combine_LR)
+ *
+ * 3. Calculate scalefactors for the frame, and
+ * also calculate scalefactor select information.
+ * (*_scale_factor_calc)
+ *
+ * 4. Calculate psychoacoustic masking levels using selected
+ * psychoacoustic model.
+ * (psycho_i, psycho_ii)
+ *
+ * 5. Perform iterative bit allocation for subbands with low
+ * mask_to_noise ratios using masking levels from step 4.
+ * (*_main_bit_allocation)
+ *
+ * 6. If error protection flag is active, add redundancy for
+ * error protection.
+ * (*_CRC_calc)
+ *
+ * 7. Pack bit allocation, scalefactors, and scalefactor select
+ *headerrmation onto bitstream.
+ * (*_encode_bit_alloc,*_encode_scale,transmission_pattern)
+ *
+ * 8. Quantize subbands and pack them into bitstream
+ * (*_subband_quantization, *_sample_encoding)
+ *
+ ************************************************************************/
+
+static frame_info frame;
+static frame_header header;
+static int frameNum;
+static int xpad_len;
+static int psycount;
+static int model;
+static unsigned int crc;
+
+typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT];
+typedef double JSBS[3][SCALE_BLOCK][SBLIMIT];
+typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT];
+
+static SBS *sb_sample;
+static JSBS *j_sample;
+static SUB *subband;
+
+static unsigned int scalar[2][3][SBLIMIT];
+static unsigned int j_scale[3][SBLIMIT];
+
+static double smr[2][SBLIMIT];
+static double max_sc[2][SBLIMIT];
+static short sam[2][1344];
+
+/* Used to keep the SNR values for the fast/quick psy models */
+static FLOAT smrdef[2][32];
+
+static unsigned int scfsi[2][SBLIMIT];
+static unsigned int bit_alloc[2][SBLIMIT];
+
+static uint8_t* xpad_data;
+
+int toolame_init(void)
+{
+ frameNum = 0;
+ psycount = 0;
+
+ header.extension = 0;
+ frame.header = &header;
+ frame.tab_num = -1; /* no table loaded */
+ frame.alloc = NULL;
+ header.version = MPEG_AUDIO_ID; /* Default: MPEG-1 */
+
+ sb_sample = (SBS *) mem_alloc (sizeof (SBS), "sb_sample");
+ j_sample = (JSBS *) mem_alloc (sizeof (JSBS), "j_sample");
+ subband = (SUB *) mem_alloc (sizeof (SUB), "subband");
+ memset ((char *) scalar, 0, sizeof (scalar));
+ memset ((char *) j_scale, 0, sizeof (j_scale));
+ memset ((char *) smr, 0, sizeof (smr));
+ memset ((char *) max_sc, 0, sizeof (max_sc));
+ memset ((char *) sam, 0, sizeof (sam));
+ memset ((char *) scfsi, 0, sizeof (scfsi));
+ memset ((char *) bit_alloc, 0, sizeof (bit_alloc));
+
+ xpad_data = NULL;
+
+ return 0;
+}
+
+int toolame_encode_frame(short buffer[2][1152])
+{
+ extern int minimum;
+ const int nch = frame.nch;
+ const int error_protection = header.error_protection;
+
+ short *win_buf[2] = {&buffer[0][0], &buffer[1][0]};
+
+ 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
+ }
+}
+
+int oldmain (int argc, char **argv)
+{
+ SBS *sb_sample;
+ 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) {
+//#define XPAD_DEBUG
+#ifdef XPAD_DEBUG
+ fprintf(stderr, "XPAD:");
+ for (i = 0; i < xpad_len; i++)
+ fprintf(stderr, " %02X", xpad_data[i]);
+ fprintf(stderr, "\n");
+#endif
+ // everything OK
+ xpad_len = xpad_data[header.dab_length];
+ assert(xpad_len > 2);
+ }
+ else {
+ fprintf(stderr, "xpad length=%d\n", xpad_len);
+ abort();
+ }
+ }
+
+ unsigned long j;
+ for (j = 0; j < 1152; j++) {
+ peak_left = MAX(peak_left, buffer[0][j]);
+ }
+ for (j = 0; j < 1152; 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) {
+ /* 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);
+}
+
+/************************************************************************
+ *
+ * print_config
+ *
+ * PURPOSE: Prints the encoding parameters used
+ *
+ ************************************************************************/
+
+void print_config (frame_info * frame, int *psy, char *inPath,
+ char *outPath)
+{
+ frame_header *header = frame->header;
+
+ if (glopts.verbosity == 0)
+ return;
+
+ fprintf (stderr, "--------------------------------------------\n");
+ if (glopts.input_select == INPUT_SELECT_JACK) {
+ fprintf (stderr, "Input JACK\n");
+ fprintf (stderr, " name %s\n", musicin.jack_name);
+ }
+ else if (glopts.input_select == INPUT_SELECT_WAV) {
+ fprintf (stderr, "Input File : '%s' %.1f kHz\n",
+ (strcmp (inPath, "-") ? inPath : "stdin"),
+ s_freq[header->version][header->sampling_frequency]);
+ }
+ else if (glopts.input_select == INPUT_SELECT_VLC) {
+ fprintf (stderr, "Input VLC\n");
+ fprintf (stderr, " URI %s\n", inPath);
+ }
+
+ fprintf (stderr, "Output File: '%s'\n",
+ (strcmp (outPath, "-") ? outPath : "stdout"));
+ fprintf (stderr, "%d kbps ", bitrate[header->version][header->bitrate_index]);
+ fprintf (stderr, "%s ", version_names[header->version]);
+ if (header->mode != MPG_MD_JOINT_STEREO)
+ fprintf (stderr, "Layer II %s Psycho model=%d (Mode_Extension=%d)\n",
+ mode_names[header->mode], *psy, header->mode_ext);
+ else
+ fprintf (stderr, "Layer II %s Psy model %d \n", mode_names[header->mode],
+ *psy);
+
+ fprintf (stderr, "[De-emph:%s\tCopyright:%s\tOriginal:%s\tCRC:%s]\n",
+ ((header->emphasis) ? "On" : "Off"),
+ ((header->copyright) ? "Yes" : "No"),
+ ((header->original) ? "Yes" : "No"),
+ ((header->error_protection) ? "On" : "Off"));
+
+ fprintf (stderr, "[Padding:%s\tByte-swap:%s\tChanswap:%s\tDAB:%s]\n",
+ ((glopts.usepadbit) ? "Normal" : "Off"),
+ ((glopts.byteswap) ? "On" : "Off"),
+ ((glopts.channelswap) ? "On" : "Off"),
+ ((glopts.dab) ? "On" : "Off"));
+
+ if (glopts.vbr == TRUE)
+ fprintf (stderr, "VBR Enabled. Using MNR boost of %f\n", glopts.vbrlevel);
+ fprintf(stderr,"ATH adjustment %f\n",glopts.athlevel);
+
+ fprintf (stderr, "--------------------------------------------\n");
+}
+
+
+/************************************************************************
+ *
+ * usage
+ *
+ * PURPOSE: Writes command line syntax to the file specified by #stderr#
+ *
+ ************************************************************************/
+
+void usage (void)
+{ /* print syntax & exit */
+ /* FIXME: maybe have an option to display better definitions of help codes, and
+ long equivalents of the flags */
+ fprintf (stdout, "\nToolame-DAB version %s\n (http://opendigitalradio.org)\n",
+ toolameversion);
+ fprintf (stdout, "MPEG Audio Layer II encoder for DAB\n\n");
+ fprintf (stdout, "usage: \n");
+ fprintf (stdout, "\t%s [options] (<infile>|-j <jackname>|-V <libvlc url>) <output>\n\n", programName);
+
+ fprintf (stdout, "Options:\n");
+ fprintf (stdout, "Input\n");
+ fprintf (stdout, "\t-s sfrq input smpl rate in kHz (dflt %4.1f)\n",
+ DFLT_SFQ);
+ fprintf (stdout, "\t-a downmix from stereo to mono\n");
+ fprintf (stdout, "\t-x force byte-swapping of input\n");
+ fprintf (stdout, "\t-g swap channels of input file\n");
+
+#if defined(JACK_INPUT)
+ fprintf (stdout, "\t-j use jack input\n");
+#else
+ fprintf (stdout, "\t-j DISABLED: JACK input not compiled in\n");
+#endif
+
+#if defined(VLC_INPUT)
+ fprintf (stdout, "\t-V use libvlc input\n");
+#else
+ fprintf (stdout, "\t-V DISABLED: libvlc input not compiled in\n");
+#endif
+
+ fprintf (stdout, "\t-W file when using libvlc input, write the ICY-Text to file\n");
+ fprintf (stdout, "\t-L enable audio level display\n");
+ fprintf (stdout, "Output\n");
+ fprintf (stdout, "\t-m mode channel mode : s/d/j/m (dflt %4c)\n",
+ DFLT_MOD);
+ fprintf (stdout, "\t-y psy psychoacoustic model 0/1/2/3 (dflt %4u)\n",
+ DFLT_PSY);
+ fprintf (stdout, "\t-b br total bitrate in kbps (dflt 192)\n");
+ fprintf (stdout, "\t-v lev vbr mode\n");
+ fprintf (stdout, "\t-l lev ATH level (dflt 0)\n");
+ fprintf (stdout, "Operation\n");
+ // fprintf (stdout, "\t-f fast mode (turns off psy model)\n");
+ // deprecate the -f switch. use "-y 0" instead.
+ fprintf (stdout,
+ "\t-q num quick mode. only calculate psy model every num frames\n");
+ fprintf (stdout, "Misc\n");
+ fprintf (stdout, "\t-d emp de-emphasis n/5/c (dflt %4c)\n",
+ DFLT_EMP);
+ fprintf (stdout, "\t-c mark as copyright\n");
+ fprintf (stdout, "\t-o mark as original\n");
+ fprintf (stdout, "\t-e add error protection\n");
+ fprintf (stdout, "\t-r force padding bit/frame off\n");
+ fprintf (stdout, "\t-p len "
+ "enable PAD, and read len bytes of X-PAD data per frame\n");
+ fprintf (stdout, "\t-P file "
+ "read X-PAD data from mot-encoder from the specified file\n");
+ fprintf (stdout, "\t-t talkativity 0=no messages (dflt 2)\n");
+ fprintf (stdout, "Files\n");
+ fprintf (stdout,
+ "\tinput input sound file. (WAV,AIFF,PCM or use '/dev/stdin')\n");
+ fprintf (stdout, "\toutput output bit stream of encoded audio\n");
+ fprintf (stdout, "\t prefix with tcp:// to use a ZMQ output\n");
+ fprintf (stdout, "\t Several ZMQ destinations can be given,\n");
+ fprintf (stdout, "\t separated by semicolons.\n");
+ fprintf (stdout,
+ "\n\tAllowable bitrates for 16, 22.05 and 24kHz sample input\n");
+ fprintf (stdout,
+ "\t8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160\n");
+ fprintf (stdout,
+ "\n\tAllowable bitrates for 32, 44.1 and 48kHz sample input\n");
+ fprintf (stdout,
+ "\t32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384\n");
+ exit (1);
+}
+
+/*********************************************
+ * void short_usage(void)
+ ********************************************/
+void short_usage (void)
+{
+ /* print a bit of info about the program */
+ fprintf (stderr, "Toolame-DAB version %s\n (http://opendigitalradio.org)\n",
+ toolameversion);
+ fprintf (stderr, "MPEG Audio Layer II encoder for DAB\n\n");
+#if defined(JACK_INPUT) && defined(VLC_INPUT)
+ fprintf (stderr, "USAGE: %s [options] (<infile>|-j <jackname>|-V <libvlc url>) [output]\n\n", programName);
+#elif defined(JACK_INPUT)
+ fprintf (stderr, "USAGE: %s [options] (<infile>|-j <jackname>) [output]\n\n", programName);
+ fprintf (stderr, "VLC input not compiled in\n");
+#elif defined(VLC_INPUT)
+ fprintf (stderr, "USAGE: %s [options] (<infile>|-V <libvlc url>) [output]\n\n", programName);
+ fprintf (stderr, "JACK input not compiled in\n");
+#else
+ fprintf (stderr, "USAGE: %s [options] <infile> [output]\n\n", programName);
+ fprintf (stderr, "Neither JACK nor libVLC input compiled in\n");
+#endif
+ fprintf (stderr, "Try \"%s -h\" for more information.\n", programName);
+ exit (0);
+}
+
+/************************************************************************
+ *
+ * parse_args
+ *
+ * PURPOSE: Sets encoding parameters to the specifications of the
+ * command line. Default settings are used for parameters
+ * not specified in the command line.
+ *
+ * SEMANTICS: The command line is parsed according to the following
+ * syntax:
+ *
+ * -j turns on JACK input
+ * -V turns on libVLC input
+ * -L turns on audio level display
+ * -m is followed by the mode
+ * -y is followed by the psychoacoustic model number
+ * -s is followed by the sampling rate
+ * -b is followed by the total bitrate, irrespective of the mode
+ * -d is followed by the emphasis flag
+ * -c is followed by the copyright/no_copyright flag
+ * -o is followed by the original/not_original flag
+ * -e is followed by the error_protection on/off flag
+ * -f turns off psy model (fast mode)
+ * -q <i> only calculate psy model every ith frame
+ * -a downmix from stereo to mono
+ * -r turn off padding bits in frames.
+ * -x force byte swapping of input
+ * -g swap the channels on an input file
+ * -t talkativity. how verbose should the program be. 0 = no messages.
+ *
+ * If the input file is in AIFF format, the sampling frequency is read
+ * from the AIFF header.
+ *
+ * The input and output filenames are read into #inpath# and #outpath#.
+ *
+ ************************************************************************/
+
+void parse_args (int argc, char **argv, frame_info * frame, int *psy,
+ unsigned long *num_samples, char inPath[MAX_NAME_SIZE],
+ char outPath[MAX_NAME_SIZE], char **mot_file, char **icy_file)
+{
+ FLOAT srate;
+ int brate;
+ frame_header *header = frame->header;
+ int err = 0, i = 0;
+ long samplerate = 0;
+
+ /* preset defaults */
+ inPath[0] = '\0';
+ outPath[0] = '\0';
+ header->lay = DFLT_LAY;
+ switch (DFLT_MOD) {
+ case 's':
+ header->mode = MPG_MD_STEREO;
+ header->mode_ext = 0;
+ break;
+ case 'd':
+ header->mode = MPG_MD_DUAL_CHANNEL;
+ header->mode_ext = 0;
+ break;
+ /* in j-stereo mode, no default header->mode_ext was defined, gave error..
+ now default = 2 added by MFC 14 Dec 1999. */
+ case 'j':
+ header->mode = MPG_MD_JOINT_STEREO;
+ header->mode_ext = 2;
+ break;
+ case 'm':
+ header->mode = MPG_MD_MONO;
+ header->mode_ext = 0;
+ break;
+ default:
+ fprintf (stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD);
+ abort ();
+ }
+ *psy = DFLT_PSY;
+ if ((header->sampling_frequency =
+ SmpFrqIndex ((long) (1000 * DFLT_SFQ), &header->version)) < 0) {
+ fprintf (stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ);
+ abort ();
+ }
+ header->bitrate_index = 14;
+ brate = 0;
+ switch (DFLT_EMP) {
+ case 'n':
+ header->emphasis = 0;
+ break;
+ case '5':
+ header->emphasis = 1;
+ break;
+ case 'c':
+ header->emphasis = 3;
+ break;
+ default:
+ fprintf (stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP);
+ abort ();
+ }
+ header->copyright = 0;
+ header->original = 0;
+ header->error_protection = FALSE;
+ header->dab_extension = 0;
+
+ glopts.input_select = INPUT_SELECT_WAV;
+
+ /* process args */
+ while (++i < argc && err == 0) {
+ char c, *token, *arg, *nextArg;
+ int argUsed;
+
+ token = argv[i];
+ if (*token++ == '-') {
+ if (i + 1 < argc)
+ nextArg = argv[i + 1];
+ else
+ nextArg = "";
+ argUsed = 0;
+ if (!*token) {
+ /* The user wants to use stdin and/or stdout. */
+ if (inPath[0] == '\0')
+ strncpy (inPath, argv[i], MAX_NAME_SIZE);
+ else if (outPath[0] == '\0')
+ strncpy (outPath, argv[i], MAX_NAME_SIZE);
+ }
+ while ((c = *token++)) {
+ if (*token /* NumericQ(token) */ )
+ arg = token;
+ else
+ arg = nextArg;
+ switch (c) {
+ case 'm':
+ argUsed = 1;
+ if (*arg == 's') {
+ header->mode = MPG_MD_STEREO;
+ header->mode_ext = 0;
+ } else if (*arg == 'd') {
+ header->mode = MPG_MD_DUAL_CHANNEL;
+ header->mode_ext = 0;
+ } else if (*arg == 'j') {
+ header->mode = MPG_MD_JOINT_STEREO;
+ } else if (*arg == 'm') {
+ header->mode = MPG_MD_MONO;
+ header->mode_ext = 0;
+ } else {
+ fprintf (stderr, "%s: -m mode must be s/d/j/m not %s\n",
+ programName, arg);
+ err = 1;
+ }
+ break;
+ case 'y':
+ *psy = atoi (arg);
+ argUsed = 1;
+ break;
+
+ case 'L':
+ glopts.show_level = 1;
+ break;
+
+ case 's':
+ argUsed = 1;
+ srate = atof (arg);
+ /* samplerate = rint( 1000.0 * srate ); $A */
+ samplerate = (long) ((1000.0 * srate) + 0.5);
+ if ((header->sampling_frequency =
+ SmpFrqIndex ((long) samplerate, &header->version)) < 0)
+ err = 1;
+ break;
+
+ case 'j':
+ glopts.input_select = INPUT_SELECT_JACK;
+ break;
+
+ case 'b':
+ argUsed = 1;
+ brate = atoi (arg);
+ break;
+ case 'd':
+ argUsed = 1;
+ if (*arg == 'n')
+ header->emphasis = 0;
+ else if (*arg == '5')
+ header->emphasis = 1;
+ else if (*arg == 'c')
+ header->emphasis = 3;
+ else {
+ fprintf (stderr, "%s: -d emp must be n/5/c not %s\n", programName,
+ arg);
+ err = 1;
+ }
+ break;
+ case 'P':
+ argUsed = 1;
+ *mot_file = arg;
+ break;
+ case 'p':
+ argUsed = 1;
+ header->dab_length = atoi(arg);
+ break;
+ case 'c':
+ header->copyright = 1;
+ break;
+ case 'o':
+ header->original = 1;
+ break;
+ case 'e':
+ header->error_protection = TRUE;
+ break;
+ case 'r':
+ glopts.usepadbit = FALSE;
+ header->padding = 0;
+ break;
+ case 'q':
+ argUsed = 1;
+ glopts.quickmode = TRUE;
+ glopts.usepsy = TRUE;
+ glopts.quickcount = atoi (arg);
+ if (glopts.quickcount == 0) {
+ /* just don't use psy model */
+ glopts.usepsy = FALSE;
+ glopts.quickcount = FALSE;
+ }
+ break;
+ case 'a':
+ glopts.downmix = TRUE;
+ header->mode = MPG_MD_MONO;
+ header->mode_ext = 0;
+ break;
+ case 'x':
+ glopts.byteswap = TRUE;
+ break;
+ case 'v':
+ argUsed = 1;
+ glopts.vbr = TRUE;
+ glopts.vbrlevel = atof (arg);
+ glopts.usepadbit = FALSE; /* don't use padding for VBR */
+ header->padding = 0;
+ /* MFC Feb 2003: in VBR mode, joint stereo doesn't make
+ any sense at the moment, as there are no noisy subbands
+ according to bits_for_nonoise in vbr mode */
+ header->mode = MPG_MD_STEREO; /* force stereo mode */
+ header->mode_ext = 0;
+ break;
+ case 'V':
+ glopts.input_select = INPUT_SELECT_VLC;
+ break;
+ case 'W':
+ argUsed = 1;
+ *icy_file = arg;
+ break;
+ case 'l':
+ argUsed = 1;
+ glopts.athlevel = atof(arg);
+ break;
+ case 'h':
+ usage ();
+ break;
+ case 'g':
+ glopts.channelswap = TRUE;
+ break;
+ case 't':
+ argUsed = 1;
+ glopts.verbosity = atoi (arg);
+ break;
+ default:
+ fprintf (stderr, "%s: unrec option %c\n", programName, c);
+ err = 1;
+ break;
+ }
+ if (argUsed) {
+ if (arg == token)
+ token = ""; /* no more from token */
+ else
+ ++i; /* skip arg we used */
+ arg = "";
+ argUsed = 0;
+ }
+ }
+ } else {
+ if (inPath[0] == '\0')
+ strcpy (inPath, argv[i]);
+ else if (outPath[0] == '\0')
+ strcpy (outPath, argv[i]);
+ else {
+ fprintf (stderr, "%s: excess arg %s\n", programName, argv[i]);
+ err = 1;
+ }
+ }
+ }
+
+ /* Always enable DAB mode */
+ header->error_protection = TRUE;
+ header->dab_extension = 4;
+ header->padding = 0;
+ glopts.dab = TRUE;
+
+ if (err)
+ usage (); /* If err has occured, then call usage() */
+
+ if (glopts.input_select != INPUT_SELECT_JACK && inPath[0] == '\0')
+ usage (); /* If not in jack-mode and no file specified, then call usage() */
+
+ if (outPath[0] == '\0') {
+ /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */
+ new_ext (inPath, DFLT_EXT, outPath);
+ }
+
+ if (glopts.input_select == INPUT_SELECT_JACK) {
+#if defined(JACK_INPUT)
+ musicin.jack_name = inPath;
+ *num_samples = MAX_U_32_NUM;
+
+ setup_jack(header, musicin.jack_name);
+#else
+ fprintf(stderr, "JACK input not compiled in\n");
+ exit(1);
+#endif
+ }
+ else if (glopts.input_select == INPUT_SELECT_WAV) {
+ if (!strcmp (inPath, "-")) {
+ musicin.wav_input = stdin; /* read from stdin */
+ *num_samples = MAX_U_32_NUM;
+ } else {
+ if ((musicin.wav_input = fopen (inPath, "rb")) == NULL) {
+ fprintf (stderr, "Could not find \"%s\".\n", inPath);
+ exit (1);
+ }
+ parse_input_file (musicin.wav_input, inPath, header, num_samples);
+ }
+ }
+ else if (glopts.input_select == INPUT_SELECT_VLC) {
+ if (samplerate == 0) {
+ fprintf (stderr, "Samplerate not specified\n");
+ exit (1);
+ }
+ *num_samples = MAX_U_32_NUM;
+ int channels = (header->mode == MPG_MD_MONO) ? 1 : 2;
+#if defined(VLC_INPUT)
+ if (vlc_in_prepare(glopts.verbosity, samplerate, inPath, channels, *icy_file) != 0) {
+ fprintf(stderr, "VLC initialisation failed\n");
+ exit(1);
+ }
+#else
+ fprintf(stderr, "VLC input not compiled in\n");
+ exit(1);
+#endif
+ }
+ else {
+ fprintf(stderr, "INVALID INPUT\n");
+ exit(1);
+ }
+
+
+ /* check for a valid bitrate */
+ if (brate == 0)
+ brate = bitrate[header->version][10];
+
+ /* Check to see we have a sane value for the bitrate for this version */
+ if ((header->bitrate_index = BitrateIndex (brate, header->version)) < 0)
+ err = 1;
+
+ if (header->dab_extension) {
+ /* in 48 kHz (= MPEG-1) */
+ /* if the bit rate per channel is less then 56 kbit/s, we have 2 scf-crc */
+ /* else we have 4 scf-crc */
+ /* in 24 kHz (= MPEG-2), we have 4 scf-crc */
+ if (header->version == MPEG_AUDIO_ID && (brate / (header->mode == MPG_MD_MONO ? 1 : 2) < 56))
+ header->dab_extension = 2;
+ }
+
+ bs.zmq_framesize = 3 * brate;
+
+ /* All options are hunky dory, open the input audio file and
+ return to the main drag */
+ open_bit_stream_w (&bs, outPath, BUFFER_SIZE);
+}
+
+
+void smr_dump(double smr[2][SBLIMIT], int nch) {
+ int ch, sb;
+
+ fprintf(stdout,"SMR:");
+ for (ch = 0;ch<nch; ch++) {
+ if (ch==1)
+ fprintf(stdout," ");
+ for (sb=0;sb<SBLIMIT;sb++)
+ fprintf(stdout,"%3.0f ",smr[ch][sb]);
+ fprintf(stdout,"\n");
+ }
+}
+
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-11 06:33:44 +0000