Use libtoolame-dab in dabplus-enc

7 files changed, 337 insertions, 1169 deletions

diff --git a/libtoolame-dab.sym b/libtoolame-dab.sym
index 9fa2b97..2e8a1f1 100644
--- a/libtoolame-dab.sym
+++ b/libtoolame-dab.sym
@@ -1 +1,8 @@
+toolame_init
+toolame_enable_downmix_stereo
+toolame_enable_byteswap
+toolame_set_channel_mode
+toolame_set_psy_model
+toolame_set_bitrate
+toolame_set_pad
 toolame_encode_frame
diff --git a/libtoolame-dab/bitstream.c b/libtoolame-dab/bitstream.c
index 3417c89..4346d60 100644
--- a/libtoolame-dab/bitstream.c
+++ b/libtoolame-dab/bitstream.c
@@ -127,23 +127,11 @@ void empty_buffer (Bit_stream_struc * bs, int minimum)
 
 
 /* open the device to write the bit stream into it */
-void open_bit_stream_w (Bit_stream_struc * bs, char *bs_filenam, int size)
+void open_bit_stream_w (Bit_stream_struc * bs, int size)
 {
     bs->zmq_sock = NULL;
 
-    if (bs_filenam[0] == '-')
-        bs->pt = stdout;
-    else if (strncmp(bs_filenam, "tcp://", 4) == 0) {
-        if (zmqoutput_open(bs, bs_filenam) != 0) {
-            fprintf(stderr, "Could not initialise ZMQ\n");
-            exit(1);
-        }
-        bs->pt = NULL; // we're not using file output
-    }
-    else if ((bs->pt = fopen (bs_filenam, "wb")) == NULL) {
-        fprintf (stderr, "Could not create \"%s\".\n", bs_filenam);
-        exit (1);
-    }
+    bs->pt = NULL; // we're not using file output
     alloc_buffer (bs, size);
     bs->buf_byte_idx = size - 1;
     bs->buf_bit_idx = 8;
diff --git a/libtoolame-dab/bitstream.h b/libtoolame-dab/bitstream.h
index f7759d8..017a12d 100644
--- a/libtoolame-dab/bitstream.h
+++ b/libtoolame-dab/bitstream.h
@@ -1,6 +1,6 @@
 int refill_buffer (Bit_stream_struc *);
 void empty_buffer (Bit_stream_struc *, int);
-void open_bit_stream_w (Bit_stream_struc *, char *, int);
+void open_bit_stream_w (Bit_stream_struc *, int);
 void close_bit_stream_w (Bit_stream_struc *);
 void alloc_buffer (Bit_stream_struc *, int);
 void desalloc_buffer (Bit_stream_struc *);
diff --git a/libtoolame-dab/toolame.c b/libtoolame-dab/toolame.c
index 09eb69e..a626552 100644
--- a/libtoolame-dab/toolame.c
+++ b/libtoolame-dab/toolame.c
@@ -1,15 +1,9 @@
 #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"
@@ -24,20 +18,15 @@
 #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 smr_dump(double smr[2][SBLIMIT], int nch);
+
 void global_init (void)
 {
     glopts.usepsy = TRUE;
@@ -127,18 +116,17 @@ static FLOAT smrdef[2][32];
 static unsigned int scfsi[2][SBLIMIT];
 static unsigned int bit_alloc[2][SBLIMIT];
 
-static uint8_t* xpad_data;
+static char* mot_file = NULL;
+static char* icy_file = NULL;
 
 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");
@@ -151,12 +139,115 @@ int toolame_init(void)
     memset ((char *) scfsi, 0, sizeof (scfsi));
     memset ((char *) bit_alloc, 0, sizeof (bit_alloc));
 
-    xpad_data = NULL;
+    global_init();
+
+    header.extension = 0;
+    header.version = MPEG_AUDIO_ID;	/* Default: MPEG-1 */
+    header.copyright = 0;
+    header.original = 0;
+    header.error_protection = FALSE;
+    header.dab_extension = 1;
+    header.lay = DFLT_LAY;
+
+    model = DFLT_PSY;
 
     return 0;
 }
 
-int toolame_encode_frame(short buffer[2][1152])
+int toolame_enable_downmix_stereo(void)
+{
+    glopts.downmix = TRUE;
+    header.mode = MPG_MD_MONO;
+    header.mode_ext = 0;
+    return 0;
+}
+
+int toolame_enable_byteswap(void)
+{
+    glopts.byteswap = TRUE;
+    return 0;
+}
+
+int toolame_set_channel_mode(const char mode)
+{
+    switch (mode) {
+        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, "libtoolame-dab: Bad mode %c\n", mode);
+            return 1;
+    }
+    return 0;
+}
+
+int toolame_set_psy_model(int new_model)
+{
+    if (new_model < 0 || new_model > 3) {
+        fprintf(stderr, "libtoolame-dab: Invalid PSY model %d\n", new_model);
+        return 1;
+    }
+    model = new_model;
+    return 0;
+}
+
+int toolame_set_bitrate(int brate)
+{
+    int err = 0;
+
+    /* 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;
+    }
+
+    open_bit_stream_w(&bs, BUFFER_SIZE);
+
+    return err;
+}
+
+int toolame_set_pad(int pad_len)
+{
+    header.dab_length = pad_len;
+    if (header.dab_length <= 0) {
+        fprintf(stderr, "Invalid XPAD length specified\n");
+        return 1;
+    }
+
+    return 0;
+}
+
+int toolame_encode_frame(
+        short buffer[2][1152],
+        unsigned char *xpad_data,
+        unsigned char *output_buffer)
 {
     extern int minimum;
     const int nch = frame.nch;
@@ -414,1034 +505,6 @@ int toolame_encode_frame(short buffer[2][1152])
     }
 }
 
-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;
 
diff --git a/libtoolame-dab/toolame.h b/libtoolame-dab/toolame.h
index 4289ab5..d7f8198 100644
--- a/libtoolame-dab/toolame.h
+++ b/libtoolame-dab/toolame.h
@@ -1,16 +1,32 @@
+#ifndef __TOOLAME_H_
+#define __TOOLAME_H_
 
-void global_init (void);
-void proginfo (void);
-void short_usage (void);
+/* All exported functions shown here return zero
+ * on success */
 
-void obtain_parameters (frame_info *, int *, unsigned long *,
-			       char[MAX_NAME_SIZE], char[MAX_NAME_SIZE]);
-void parse_args (int, char **, frame_info *, int *, unsigned long *,
-			char[MAX_NAME_SIZE], char[MAX_NAME_SIZE], char**, char**);
-void print_config (frame_info *, int *,
-			  char[MAX_NAME_SIZE], char[MAX_NAME_SIZE]);
-void usage (void);
+/* Initialise toolame encoding library. */
+int toolame_init(void);
 
+int toolame_enable_downmix_stereo(void);
+int toolame_enable_byteswap(void);
 
-void smr_dump(double smr[2][SBLIMIT], int nch);
+/* Set channel mode. Allowed values:
+ * s, d, j , and m
+ */
+int toolame_set_channel_mode(const char mode);
+
+/* Valid PSY models: 0 to 3 */
+int toolame_set_psy_model(int new_model);
+
+int toolame_set_bitrate(int brate);
+
+/* Enable PAD insertion from the specified file with length */
+int toolame_set_pad(int pad_len);
+
+int toolame_encode_frame(
+        short buffer[2][1152],
+        unsigned char* xpad_data,
+        unsigned char *output_buffer);
+
+#endif // __TOOLAME_H_
 
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp
index d0130fd..8abc82a 100644
--- a/src/dabplus-enc.cpp
+++ b/src/dabplus-enc.cpp
@@ -46,16 +46,23 @@ extern "C" {
 
 extern "C" {
 #include <fec.h>
+#include "libtoolame-dab/toolame.h"
 }
 
+// Enumerate which encoder we can use
+enum class encoder_selection_t {
+    fdk_dabplus,
+    toolame_dab
+};
+
 using namespace std;
 
 void usage(const char* name) {
     fprintf(stderr,
     "dabplus-enc %s is a HE-AACv2 encoder for DAB+\n"
-    "based on fdk-aac-dabplus that can read from"
-    "JACK, ALSA or a file source\n"
-    "and encode to a ZeroMQ output for ODR-DabMux.\n"
+    "based on fdk-aac-dabplus and a Toolame-based MPEG\n"
+    "encoder for DAB that can read from JACK, ALSA or\n"
+    "a file source and encode to a ZeroMQ output for ODR-DabMux.\n"
     "(Experimental!)It can also use libvlc as an input.\n"
     "\n"
     "The -D option enables experimental sound card clock drift compensation.\n"
@@ -110,6 +117,7 @@ void usage(const char* name) {
     "   Drift compensation\n"
     "     -D, --drift-comp                     Enable ALSA/VLC sound card drift compensation.\n"
     "   Encoder parameters:\n"
+    "     -a, --dab                            Encode in DAB and not in DAB+.\n"
     "     -b, --bitrate={ 8, 16, ..., 192 }    Output bitrate in kbps. Must be a multiple of 8.\n"
     "     -A, --no-afterburner                 Disable AAC encoder quality increaser.\n"
     "     -c, --channels={ 1, 2 }              Nb of input channels (default: 2).\n"
@@ -240,9 +248,11 @@ int prepare_aac_encoder(
 
 int main(int argc, char *argv[])
 {
-    int subchannel_index = 8; //64kbps subchannel
+    int bitrate = 64; //64kbps subchannel
     int ch=0;
 
+    encoder_selection_t selected_encoder = encoder_selection_t::fdk_dabplus;
+
     // For the ALSA input
     const char *alsa_device = NULL;
 
@@ -314,7 +324,7 @@ int main(int argc, char *argv[])
         {"vlc-uri",        required_argument,  0, 'v'},
         {"write-icy-text", required_argument,  0, 'w'},
         {"aaclc",          no_argument,        0,  0 },
-        {"afterburner",    no_argument,        0, 'a'},
+        {"dab",            no_argument,        0, 'a'},
         {"drift-comp",     no_argument,        0, 'D'},
         {"fifo-silence",   no_argument,        0,  3 },
         {"help",           no_argument,        0, 'h'},
@@ -361,13 +371,13 @@ int main(int argc, char *argv[])
             inFifoSilence = true;
             break;
         case 'a':
-            fprintf(stderr, "Warning, -a option does not exist anymore!\n");
+            selected_encoder = encoder_selection_t::toolame_dab;
             break;
         case 'A':
             afterburner = false;
             break;
         case 'b':
-            subchannel_index = atoi(optarg) / 8;
+            bitrate = atoi(optarg);
             break;
         case 'c':
             channels = atoi(optarg);
@@ -464,15 +474,25 @@ int main(int argc, char *argv[])
         return 1;
     }
 
-    if (subchannel_index < 1 || subchannel_index > 24) {
-        fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. Try other bitrate.\n",
-                subchannel_index);
-        return 1;
-    }
+    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+        int subchannel_index = bitrate / 8;
 
-    if ( ! (sample_rate == 32000 || sample_rate == 48000)) {
-        fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n");
-        return 1;
+        if (subchannel_index < 1 || subchannel_index > 24) {
+            fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. Try other bitrate.\n",
+                    subchannel_index);
+            return 1;
+        }
+
+        if ( ! (sample_rate == 32000 || sample_rate == 48000)) {
+            fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n");
+            return 1;
+        }
+    }
+    else if (selected_encoder == encoder_selection_t::toolame_dab) {
+        if ( ! (sample_rate == 24000 || sample_rate == 48000)) {
+            fprintf(stderr, "Invalid sample rate. Possible values are: 24000, 48000.\n");
+            return 1;
+        }
     }
 
     if (padlen < 0) {
@@ -554,12 +574,46 @@ int main(int argc, char *argv[])
     }
 
 
+    std::vector<uint8_t> input_buf;
+
     HANDLE_AACENCODER encoder;
 
-    if (prepare_aac_encoder(&encoder, subchannel_index, channels,
-                sample_rate, afterburner, &aot) != 0) {
-        fprintf(stderr, "Encoder preparation failed\n");
-        return 1;
+    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+        int subchannel_index = bitrate / 8;
+        if (prepare_aac_encoder(&encoder, subchannel_index, channels,
+                    sample_rate, afterburner, &aot) != 0) {
+            fprintf(stderr, "Encoder preparation failed\n");
+            return 1;
+        }
+
+        if (aacEncInfo(encoder, &info) != AACENC_OK) {
+            fprintf(stderr, "Unable to get the encoder info\n");
+            return 1;
+        }
+
+        // Each DAB+ frame will need input_size audio bytes
+        const int input_size = channels * BYTES_PER_SAMPLE * info.frameLength;
+        fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n",
+                info.frameLength,
+                input_size);
+
+        input_buf.resize(input_size);
+    }
+    else if (selected_encoder == encoder_selection_t::toolame_dab) {
+        int err = toolame_init();
+
+        if (err == 0) {
+            toolame_set_bitrate(bitrate);
+        }
+
+        if (err) {
+            fprintf(stderr, "libtoolame-dab init failed: %d\n", err);
+            return err;
+        }
+
+        // TODO int toolame_set_pad(int pad_len);
+
+        input_buf.resize(2 * 1152);
     }
 
     /* We assume that we need to call the encoder
@@ -567,24 +621,13 @@ int main(int argc, char *argv[])
      * frame. This information is used when the alsa drift compensation
      * is active
      */
-    const int enc_calls_per_output =
-        (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000;
-
+    int enc_calls_per_output = 1; // Valid for libtoolame-dab
 
-    if (aacEncInfo(encoder, &info) != AACENC_OK) {
-        fprintf(stderr, "Unable to get the encoder info\n");
-        return 1;
+    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+        enc_calls_per_output = (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000;
     }
 
-    // Each DAB+ frame will need input_size audio bytes
-    const int input_size = channels * BYTES_PER_SAMPLE * info.frameLength;
-    fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n",
-            info.frameLength,
-            input_size);
-
-    uint8_t input_buf[input_size];
-
-    int max_size = 8*input_size + NUM_SAMPLES_PER_CALL;
+    int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL;
 
     SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);
 
@@ -661,11 +704,22 @@ int main(int argc, char *argv[])
         }
     }
 
-    int outbuf_size = subchannel_index*120;
-    uint8_t zmqframebuf[ZMQ_HEADER_SIZE + 24*120];
-    zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf;
+    int outbuf_size;
+    std::vector<uint8_t> zmqframebuf;
+    std::vector<uint8_t> outbuf;
+    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+        outbuf_size = bitrate/8*120;
+        outbuf.resize(24*120);
+        zmqframebuf.resize(ZMQ_HEADER_SIZE + 24*120);
+    }
+    else if (selected_encoder == encoder_selection_t::toolame_dab) {
+        outbuf_size = 3 * bitrate;
+        outbuf.resize(outbuf_size);
+        zmqframebuf.resize(ZMQ_HEADER_SIZE + outbuf_size);
+    }
+
+    zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)&zmqframebuf[0];
 
-    uint8_t outbuf[24*120];
 
     unsigned char pad_buf[padlen + 1];
 
@@ -686,8 +740,6 @@ int main(int argc, char *argv[])
         int out_identifier = OUT_BITSTREAM_DATA;
 
         AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
-        AACENC_InArgs in_args = { 0 };
-        AACENC_OutArgs out_args = { 0 };
         void *in_ptr[2], *out_ptr;
         int in_size[2], in_elem_size[2];
         int out_size, out_elem_size;
@@ -750,20 +802,20 @@ int main(int argc, char *argv[])
         }
 
         // -------------- Read Data
-        memset(outbuf, 0x00, outbuf_size);
-        memset(input_buf, 0x00, input_size);
+        memset(&outbuf[0], 0x00, outbuf_size);
+        memset(&input_buf[0], 0x00, input_buf.size());
 
         ssize_t read;
         if (infile) {
-            read = file_in.read(input_buf, input_size);
+            read = file_in.read(&input_buf[0], input_buf.size());
             if (read < 0) {
                 break;
             }
-            else if (read != input_size) {
+            else if (read != input_buf.size()) {
                 if (inFifoSilence && file_in.eof()) {
-                    memset(input_buf, 0, input_size);
-                    read = input_size;
-                    usleep((long)input_size * 1000000 /
+                    memset(&input_buf[0], 0, input_buf.size());
+                    read = input_buf.size();
+                    usleep((long)input_buf.size() * 1000000 /
                             (BYTES_PER_SAMPLE * channels * sample_rate));
                 }
                 else {
@@ -786,9 +838,9 @@ int main(int argc, char *argv[])
                 }
 
                 size_t overruns;
-                read = queue.pop(input_buf, input_size, &overruns); // returns bytes
+                read = queue.pop(input_buf, input_buf.size(), &overruns); // returns bytes
 
-                if (read != input_size) {
+                if (read != input_buf.size()) {
                     status |= STATUS_UNDERRUN;
                 }
 
@@ -799,12 +851,12 @@ int main(int argc, char *argv[])
             else {
                 vlc_in = &vlc_in_direct;
 
-                read = vlc_in_direct.read(input_buf, input_size);
+                read = vlc_in_direct.read(input_buf, input_buf.size());
                 if (read < 0) {
                     fprintf(stderr, "Detected fault in VLC input!\n");
                     break;
                 }
-                else if (read != input_size) {
+                else if (read != input_buf.size()) {
                     fprintf(stderr, "Short VLC read !\n");
                     break;
                 }
@@ -823,9 +875,9 @@ int main(int argc, char *argv[])
             }
 
             size_t overruns;
-            read = queue.pop(input_buf, input_size, &overruns); // returns bytes
+            read = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
 
-            if (read != input_size) {
+            if (read != input_buf.size()) {
                 status |= STATUS_UNDERRUN;
             }
 
@@ -834,11 +886,11 @@ int main(int argc, char *argv[])
             }
         }
         else {
-            read = alsa_in_direct.read(input_buf, input_size);
+            read = alsa_in_direct.read(&input_buf[0], input_buf.size());
             if (read < 0) {
                 break;
             }
-            else if (read != input_size) {
+            else if (read != input_buf.size()) {
                 fprintf(stderr, "Short alsa read !\n");
             }
         }
@@ -869,50 +921,80 @@ int main(int argc, char *argv[])
             measured_silence_ms = 0;
         }
 
-        // -------------- AAC Encoding
-
-        int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
-
-
-        in_ptr[0] = input_buf;
-        in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
-        in_size[0] = read;
-        in_size[1] = calculated_padlen;
-        in_elem_size[0] = BYTES_PER_SAMPLE;
-        in_elem_size[1] = sizeof(uint8_t);
-        in_args.numInSamples = input_size/BYTES_PER_SAMPLE;
-        in_args.numAncBytes = calculated_padlen;
+        int numOutBytes = 0;
+        if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+            AACENC_InArgs in_args = { 0 };
+            AACENC_OutArgs out_args = { 0 };
+            // -------------- AAC Encoding
+
+            const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
+            const int subchannel_index = bitrate / 8;
+
+            in_ptr[0] = &input_buf[0];
+            in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
+            in_size[0] = read;
+            in_size[1] = calculated_padlen;
+            in_elem_size[0] = BYTES_PER_SAMPLE;
+            in_elem_size[1] = sizeof(uint8_t);
+            in_args.numInSamples = input_buf.size()/BYTES_PER_SAMPLE;
+            in_args.numAncBytes = calculated_padlen;
+
+            in_buf.bufs = (void**)&in_ptr;
+            in_buf.bufferIdentifiers = in_identifier;
+            in_buf.bufSizes = in_size;
+            in_buf.bufElSizes = in_elem_size;
+
+            out_ptr = &outbuf[0];
+            out_size = outbuf.size();
+            out_elem_size = 1;
+            out_buf.numBufs = 1;
+            out_buf.bufs = &out_ptr;
+            out_buf.bufferIdentifiers = &out_identifier;
+            out_buf.bufSizes = &out_size;
+            out_buf.bufElSizes = &out_elem_size;
+
+            AACENC_ERROR err;
+            if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args))
+                    != AACENC_OK) {
+                if (err == AACENC_ENCODE_EOF) {
+                    fprintf(stderr, "encoder error: EOF reached\n");
+                    break;
+                }
+                fprintf(stderr, "Encoding failed (%d)\n", err);
+                retval = 3;
+                break;
+            }
+            calls++;
 
-        in_buf.bufs = (void**)&in_ptr;
-        in_buf.bufferIdentifiers = in_identifier;
-        in_buf.bufSizes = in_size;
-        in_buf.bufElSizes = in_elem_size;
+            numOutBytes = out_args.numOutBytes;
+        }
+        else if (selected_encoder == encoder_selection_t::toolame_dab) {
+            const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
+            uint8_t *xpad_data = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
 
-        out_ptr = outbuf;
-        out_size = sizeof(outbuf);
-        out_elem_size = 1;
-        out_buf.numBufs = 1;
-        out_buf.bufs = &out_ptr;
-        out_buf.bufferIdentifiers = &out_identifier;
-        out_buf.bufSizes = &out_size;
-        out_buf.bufElSizes = &out_elem_size;
+            short input_buffers[2][1152];
 
-        AACENC_ERROR err;
-        if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args))
-                != AACENC_OK) {
-            if (err == AACENC_ENCODE_EOF) {
-                fprintf(stderr, "encoder error: EOF reached\n");
-                break;
+            if (channels == 1) {
+                memcpy(input_buffers[0], &input_buf[0], 1152);
             }
-            fprintf(stderr, "Encoding failed (%d)\n", err);
-            retval = 3;
-            break;
+            else if (channels == 2) {
+                for (int ch = 0; ch < 2; ch++) {
+                    for (int i = 0; i < 1152; i++) {
+                        input_buffers[ch][i] = input_buf[2*i + ch];
+                    }
+                }
+            }
+            else {
+                fprintf(stderr, "INTERNAL ERROR! invalid number of channels\n");
+            }
+
+            toolame_encode_frame(input_buffers, xpad_data, &outbuf[0]);
         }
-        calls++;
 
         /* Check if the encoder has generated output data */
-        if (out_args.numOutBytes != 0)
-        {
+        if (numOutBytes != 0 and
+            selected_encoder == encoder_selection_t::fdk_dabplus) {
+
             // Our timing code depends on this
             if (calls != enc_calls_per_output) {
                 fprintf(stderr, "INTERNAL ERROR! calls=%d"
@@ -925,6 +1007,7 @@ int main(int argc, char *argv[])
             int row, col;
             unsigned char buf_to_rs_enc[110];
             unsigned char rs_enc[10];
+            const int subchannel_index = bitrate / 8;
             for(row=0; row < subchannel_index; row++) {
                 for(col=0;col < 110; col++) {
                     buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
@@ -937,25 +1020,32 @@ int main(int argc, char *argv[])
                     assert(subchannel_index * col + row < outbuf_size);
                 }
             }
+        }
 
+        if (numOutBytes != 0) {
             if (out_fh) {
-                fwrite(outbuf, 1, outbuf_size, out_fh);
+                fwrite(&outbuf[0], 1, outbuf_size, out_fh);
             }
             else {
                 // ------------ ZeroMQ transmit
                 try {
                     zmq_frame_header->version = 1;
-                    zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
+                    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+                        zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
+                    }
+                    else if (selected_encoder == encoder_selection_t::toolame_dab) {
+                        zmq_frame_header->encoder = ZMQ_ENCODER_TOOLAME;
+                    }
                     zmq_frame_header->datasize = outbuf_size;
                     zmq_frame_header->audiolevel_left = peak_left;
                     zmq_frame_header->audiolevel_right = peak_right;
 
-                    assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= NUMOF(zmqframebuf));
+                    assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= zmqframebuf.size());
 
                     memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
-                            outbuf, outbuf_size);
+                            &outbuf[0], outbuf_size);
 
-                    zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header),
+                    zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header),
                             ZMQ_DONTWAIT);
                 }
                 catch (zmq::error_t& e) {
@@ -970,7 +1060,10 @@ int main(int argc, char *argv[])
                     break;
                 }
             }
+        }
 
+        if (numOutBytes != 0)
+        {
             if (show_level) {
                 if (channels == 1) {
                     fprintf(stderr, "\rIn: [%-6s] %1s %1s %1s",
diff --git a/src/utils.h b/src/utils.h
index c75935f..a0ab1ae 100644
--- a/src/utils.h
+++ b/src/utils.h
@@ -35,6 +35,7 @@ struct zmq_frame_header_t
 } __attribute__ ((packed));
 
 #define ZMQ_ENCODER_FDK 1
+#define ZMQ_ENCODER_TOOLAME 2
 
 #define ZMQ_HEADER_SIZE sizeof(struct zmq_frame_header_t)