Remove useless libtoolame-dab files

1 files changed, 0 insertions, 689 deletions

diff --git a/libtoolame-dab/audio_read.c b/libtoolame-dab/audio_read.c
deleted file mode 100644
index 3649ef4..0000000
--- a/libtoolame-dab/audio_read.c
+++ /dev/null
@@ -1,689 +0,0 @@
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include "common.h"
-#include "encoder.h"
-#include "options.h"
-#include "portableio.h"
-#if defined(JACK_INPUT)
-#include <pthread.h>
-#include <jack/jack.h>
-#include <jack/ringbuffer.h>
-#endif
-#include "audio_read.h"
-#include "vlc_input.h"
-
-#if defined(JACK_INPUT)
-jack_port_t *input_port_left;
-jack_port_t *input_port_right;
-jack_client_t *client;
-pthread_mutex_t encode_thread_lock = PTHREAD_MUTEX_INITIALIZER;
-pthread_cond_t  data_ready = PTHREAD_COND_INITIALIZER;
-
-/* shutdown can tell get_audio to stop */
-typedef struct _thread_info {
-    volatile int connected;
-} jack_thread_info_t;
-
-jack_thread_info_t thread_info;
-const size_t sample_size = sizeof(jack_default_audio_sample_t);
-
-#define DEFAULT_RB_SIZE 16384       /* ringbuffer size in frames */
-jack_ringbuffer_t *rb;
-
-/* setup_jack()
- *
- * PURPOSE:  connect to jack, setup the ports, the ringbuffer
- *
- * frame_header is needed (fill information about sampling rate)
- */
-
-void setup_jack(frame_header *header, const char* jackname) {
-    const char *client_name = jackname;
-    const char *server_name = NULL;
-    jack_options_t options = JackNullOption;
-    jack_status_t status;
-
-    /* open a client connection to the JACK server */
-
-    client = jack_client_open(client_name, options, &status, server_name);
-    if (client == NULL) {
-        fprintf(stderr, "jack_client_open() failed, "
-                "status = 0x%2.0x\n", status);
-        if (status & JackServerFailed) {
-            fprintf(stderr, "Unable to connect to JACK server\n");
-        }
-        exit(1);
-    }
-    if (status & JackServerStarted) {
-        fprintf(stderr, "JACK server started\n");
-    }
-    if (status & JackNameNotUnique) {
-        client_name = jack_get_client_name(client);
-        fprintf(stderr, "unique name `%s' assigned\n", client_name);
-    }
-
-    thread_info.connected = 1;
-
-    /* tell the JACK server to call `process()' whenever
-       there is work to be done.
-       */
-
-    jack_set_process_callback(client, process, &thread_info);
-
-    /* tell the JACK server to call `jack_shutdown()' if
-       it ever shuts down, either entirely, or if it
-       just decides to stop calling us.
-       */
-
-    jack_on_shutdown(client, jack_shutdown, &thread_info);
-
-    /* display the current sample rate. 
-    */
-
-    printf ("engine sample rate: %" PRIu32 "\n",
-            jack_get_sample_rate(client));
-
-    if ((header->sampling_frequency = SmpFrqIndex((long) jack_get_sample_rate(client), &header->version)) < 0) {
-        fprintf (stderr, "invalid sample rate\n");
-        exit(1);
-    }
-
-    /* create two ports */
-
-    input_port_left = jack_port_register(client, "input0",
-            JACK_DEFAULT_AUDIO_TYPE,
-            JackPortIsInput, 0);
-    input_port_right = jack_port_register(client, "input1",
-            JACK_DEFAULT_AUDIO_TYPE,
-            JackPortIsInput, 0);
-
-    if ((input_port_left == NULL) || (input_port_right == NULL)) {
-        fprintf(stderr, "no more JACK ports available\n");
-        exit(1);
-    }
-
-
-    /* setup the ringbuffer */
-    rb = jack_ringbuffer_create(2 * sample_size * DEFAULT_RB_SIZE);
-    fprintf(stderr, "jack sample_size: %zu\n", sample_size);
-
-
-    /* take the mutex */
-    pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS, NULL);
-    pthread_mutex_lock(&encode_thread_lock);
-
-    /* Tell the JACK server that we are ready to roll.  Our
-     * process() callback will start running now. */
-
-    if (jack_activate(client)) {
-        fprintf (stderr, "cannot activate client");
-        exit(1);
-    }
-}
-
-/**
- * The process callback for this JACK application is called in a
- * special realtime thread once for each audio cycle.
- *
- * It fills the ringbuffer
- */
-int process(jack_nframes_t nframes, void *arg) {
-    int i;
-    int samp;
-    //jack_thread_info_t *info = (jack_thread_info_t *) arg;
-
-    jack_default_audio_sample_t *in_left, *in_right;
-    in_left = jack_port_get_buffer(input_port_left, nframes);
-    in_right = jack_port_get_buffer(input_port_right, nframes);
-
-    /* Sndfile requires interleaved data.  It is simpler here to
-     * just queue interleaved samples to a single ringbuffer. */
-    //fprintf(stderr, "process()\n");
-    for (i = 0; i < nframes; i++) {
-        /*
-           jack_ringbuffer_write(rb, (void *)(in_left + i), sample_size);
-           jack_ringbuffer_write(rb, (void *)(in_right + i), sample_size);
-        */
-        /* convert to shorts, then insert into ringbuffer */
-        samp = lrintf(in_left[i] * 1.0 * 0x7FFF);
-        jack_ringbuffer_write(rb, (char*)&samp, 2); 
-        samp = lrintf(in_right[i] * 1.0 * 0x7FFF);
-        jack_ringbuffer_write(rb, (char*)&samp, 2); 
-        
-    }
-    //fprintf(stderr, "PROCESS()\n");
-
-    /* tell read_samples that we've got new data */
-    pthread_cond_signal(&data_ready);
-
-    return 0;
-}
-
-/**
- * JACK calls this shutdown_callback if the server ever shuts down or
- * decides to disconnect the client.
- */
-void jack_shutdown(void *arg)
-{
-    jack_thread_info_t *info = (jack_thread_info_t *) arg;
-
-    info->connected = 0;
-    /* tell read_samples to move on */
-
-    pthread_cond_signal(&data_ready);
-}
-#endif // defined(JACK_INPUT)
-
-/************************************************************************
- *
- * read_samples()
- *
- * PURPOSE:  reads the PCM samples from a file to the buffer
- *
- *  SEMANTICS:
- * Reads #samples_read# number of shorts from #musicin# filepointer
- * into #sample_buffer[]#.  Returns the number of samples read.
- *
- ************************************************************************/
-
-unsigned long read_samples (music_in_t* musicin, short sample_buffer[2304],
-        unsigned long num_samples, unsigned long frame_size)
-{
-    unsigned long samples_read;
-    static unsigned long samples_to_read;
-    static char init = TRUE;
-
-    void* jack_sample_buffer;
-
-    if (init) {
-        samples_to_read = num_samples;
-        init = FALSE;
-    }
-    if (samples_to_read >= frame_size)
-        samples_read = frame_size;
-    else
-        samples_read = samples_to_read;
-
-    if (0) { }
-#if defined(JACK_INPUT)
-    else if (glopts.input_select == INPUT_SELECT_JACK) {
-        int f = 2;
-        while (jack_ringbuffer_read_space(rb) < f * samples_read) {
-            /* wait until process() signals more data */
-            pthread_cond_wait(&data_ready, &encode_thread_lock);
-
-            if (thread_info.connected == 0) {
-                pthread_mutex_unlock(&encode_thread_lock);
-                jack_client_close(client);
-                jack_ringbuffer_free(rb);
-                return 0;
-            }
-        }
-
-        jack_sample_buffer = malloc(f * (int)samples_read);
-        int bytes_read = jack_ringbuffer_read(rb, jack_sample_buffer, f * (int)samples_read);
-        //fprintf(stderr, " read_bytes / f = %d, should be %d\n", (int)bytes_read/f, samples_read);
-        samples_read = bytes_read / f;
-        if (bytes_read % f != 0) {
-            fprintf(stderr, "cannot divide bytes_read by f: %d mod f = %d", bytes_read, bytes_read % f);
-        }
-        //fprintf(stderr, " #%d(%d) \n", (int)bytes_read, samples_read);
-        //f2les_array(jack_sample_buffer, sample_buffer, samples_read, 1);
-
-        memcpy(sample_buffer, jack_sample_buffer, bytes_read);
-
-        free(jack_sample_buffer);
-
-    }
-#endif // defined(JACK_INPUT)
-    else if (glopts.input_select == INPUT_SELECT_WAV) {
-        if ((samples_read =
-                    fread (sample_buffer, sizeof (short), (int) samples_read,
-                        musicin->wav_input)) == 0)
-            fprintf (stderr, "Hit end of WAV audio data\n");
-    }
-    else if (glopts.input_select == INPUT_SELECT_VLC) {
-#if defined(VLC_INPUT)
-        ssize_t bytes_read = vlc_in_read(sample_buffer, sizeof(short) * (int)samples_read);
-        if (bytes_read == -1) {
-            fprintf (stderr, "VLC input error\n");
-            samples_read = 0;
-        }
-        else {
-            samples_read = bytes_read / sizeof(short);
-        }
-#else
-        samples_read = 0;
-#endif
-    }
-
-    /*
-       Samples are big-endian. If this is a little-endian machine
-       we must swap
-       */
-    if (NativeByteOrder == order_unknown) {
-        NativeByteOrder = DetermineByteOrder ();
-        if (NativeByteOrder == order_unknown) {
-            fprintf (stderr, "byte order not determined\n");
-            exit (1);
-        }
-    }
-    if (NativeByteOrder != order_littleEndian || (glopts.byteswap == TRUE))
-        SwapBytesInWords (sample_buffer, samples_read);
-
-    if (num_samples != MAX_U_32_NUM)
-        samples_to_read -= samples_read;
-
-    if (samples_read < frame_size && samples_read > 0) {
-        /* fill out frame with zeros */
-        for (; samples_read < frame_size; sample_buffer[samples_read++] = 0);
-        samples_to_read = 0;
-        samples_read = frame_size;
-    }
-    return (samples_read);
-}
-
-/************************************************************************
- *
- * get_audio()
- *
- * PURPOSE:  reads a frame of audio data from a file to the buffer,
- *   aligns the data for future processing, and separates the
- *   left and right channels
- *
- *
- ************************************************************************/
-    unsigned long
-get_audio (music_in_t* musicin, short buffer[2][1152], unsigned long num_samples,
-        int nch, frame_header *header)
-{
-    int j;
-    short insamp[2304];
-    unsigned long samples_read;
-
-    if (nch == 2) {     /* stereo */
-        samples_read =
-            read_samples (musicin, insamp, num_samples, (unsigned long) 2304);
-        if (glopts.channelswap == TRUE) {
-            for (j = 0; j < 1152; j++) {
-                buffer[1][j] = insamp[2 * j];
-                buffer[0][j] = insamp[2 * j + 1];
-            }
-        } else {
-            for (j = 0; j < 1152; j++) {
-                buffer[0][j] = insamp[2 * j];
-                buffer[1][j] = insamp[2 * j + 1];
-            }
-        }
-    } else if (glopts.downmix == TRUE) {
-        samples_read =
-            read_samples (musicin, insamp, num_samples, (unsigned long) 2304);
-        for (j = 0; j < 1152; j++) {
-            buffer[0][j] = 0.5 * (insamp[2 * j] + insamp[2 * j + 1]);
-        }
-    } else {            /* mono */
-        samples_read =
-            read_samples (musicin, insamp, num_samples, (unsigned long) 1152);
-        for (j = 0; j < 1152; j++) {
-            buffer[0][j] = insamp[j];
-            /* buffer[1][j] = 0;  don't bother zeroing this buffer. MFC Nov 99 */
-        }
-    }
-    return (samples_read);
-}
-
-
-/*****************************************************************************
- *
- *  Routines to determine byte order and swap bytes
- *
- *****************************************************************************/
-
-enum byte_order DetermineByteOrder (void)
-{
-    char s[sizeof (long) + 1];
-    union {
-        long longval;
-        char charval[sizeof (long)];
-    } probe;
-    probe.longval = 0x41424344L;  /* ABCD in ASCII */
-    strncpy (s, probe.charval, sizeof (long));
-    s[sizeof (long)] = '\0';
-    /* fprintf( stderr, "byte order is %s\n", s ); */
-    if (strcmp (s, "ABCD") == 0)
-        return order_bigEndian;
-    else if (strcmp (s, "DCBA") == 0)
-        return order_littleEndian;
-    else
-        return order_unknown;
-}
-
-void SwapBytesInWords (short *loc, int words)
-{
-    int i;
-    short thisval;
-    char *dst, *src;
-    src = (char *) &thisval;
-    for (i = 0; i < words; i++) {
-        thisval = *loc;
-        dst = (char *) loc++;
-        dst[0] = src[1];
-        dst[1] = src[0];
-    }
-}
-
-/*****************************************************************************
- *
- *  Read Audio Interchange File Format (AIFF) headers.
- *
- *****************************************************************************/
-
-int aiff_read_headers (FILE * file_ptr, IFF_AIFF * aiff_ptr)
-{
-    int chunkSize, subSize, sound_position;
-
-    if (fseek (file_ptr, 0, SEEK_SET) != 0)
-        return -1;
-
-    if (Read32BitsHighLow (file_ptr) != IFF_ID_FORM)
-        return -1;
-
-    chunkSize = Read32BitsHighLow (file_ptr);
-
-    if (Read32BitsHighLow (file_ptr) != IFF_ID_AIFF)
-        return -1;
-
-    sound_position = 0;
-    while (chunkSize > 0) {
-        chunkSize -= 4;
-        switch (Read32BitsHighLow (file_ptr)) {
-
-            case IFF_ID_COMM:
-                chunkSize -= subSize = Read32BitsHighLow (file_ptr);
-                aiff_ptr->numChannels = Read16BitsHighLow (file_ptr);
-                subSize -= 2;
-                aiff_ptr->numSampleFrames = Read32BitsHighLow (file_ptr);
-                subSize -= 4;
-                aiff_ptr->sampleSize = Read16BitsHighLow (file_ptr);
-                subSize -= 2;
-                aiff_ptr->sampleRate = ReadIeeeExtendedHighLow (file_ptr);
-                subSize -= 10;
-                while (subSize > 0) {
-                    getc (file_ptr);
-                    subSize -= 1;
-                }
-                break;
-
-            case IFF_ID_SSND:
-                chunkSize -= subSize = Read32BitsHighLow (file_ptr);
-                aiff_ptr->blkAlgn.offset = Read32BitsHighLow (file_ptr);
-                subSize -= 4;
-                aiff_ptr->blkAlgn.blockSize = Read32BitsHighLow (file_ptr);
-                subSize -= 4;
-                sound_position = ftell (file_ptr) + aiff_ptr->blkAlgn.offset;
-                if (fseek (file_ptr, (long) subSize, SEEK_CUR) != 0)
-                    return -1;
-                aiff_ptr->sampleType = IFF_ID_SSND;
-                break;
-
-            default:
-                chunkSize -= subSize = Read32BitsHighLow (file_ptr);
-                while (subSize > 0) {
-                    getc (file_ptr);
-                    subSize -= 1;
-                }
-                break;
-        }
-    }
-    return sound_position;
-}
-
-/*****************************************************************************
- *
- *  Seek past some Audio Interchange File Format (AIFF) headers to sound data.
- *
- *****************************************************************************/
-
-int aiff_seek_to_sound_data (FILE * file_ptr)
-{
-    if (fseek
-            (file_ptr, AIFF_FORM_HEADER_SIZE + AIFF_SSND_HEADER_SIZE,
-             SEEK_SET) != 0)
-        return (-1);
-    return (0);
-}
-
-/************************************************************
- *   parse_input_file()
- *   Determine the type of sound file. (stdin, wav, aiff, raw pcm)
- *   Determine Sampling Frequency
- *             number of samples
- *             whether the new sample is stereo or mono. 
- *
- *   If file is coming from /dev/stdin assume it is raw PCM. (it's what I use. YMMV)
- *
- *  This is just a hacked together function. The aiff parsing comes from the ISO code.
- *  The WAV code comes from Nick Burch
- *  The ugly /dev/stdin hack comes from me.
- *                                                  MFC Dec 99
- **************************************************************/
-    void
-parse_input_file (FILE * musicin, char inPath[MAX_NAME_SIZE], frame_header *header,
-        unsigned long *num_samples)
-{
-
-    IFF_AIFF pcm_aiff_data;
-    long soundPosition;
-
-    unsigned char wave_header_buffer[40]; //HH fixed
-    int wave_header_read = 0;
-    int wave_header_stereo = -1;
-    int wave_header_16bit = -1;
-    unsigned long samplerate;
-
-    /*************************** STDIN ********************************/
-    /* check if we're reading from stdin. Assume it's a raw PCM file. */
-    /* Of course, you could be piping a WAV file into stdin. Not done in this code */
-    /* this code is probably very dodgy and was written to suit my needs. MFC Dec 99 */
-    if ((strcmp (inPath, "/dev/stdin") == 0)) {
-        fprintf (stderr, "Reading from stdin\n");
-        fprintf (stderr, "Remember to set samplerate with '-s'.\n");
-        *num_samples = MAX_U_32_NUM;    /* huge sound file */
-        return;
-    }
-
-    if (fseek (musicin, 0L, SEEK_SET) == -1) {
-        fprintf (stderr, "Input is not seekable, assuming pipe with raw PCM\n");
-        fprintf (stderr, "Remember to set samplerate with '-s'.\n");
-        *num_samples = MAX_U_32_NUM;    /* huge sound file */
-        return;
-    }
-
-    /****************************  AIFF ********************************/
-    if ((soundPosition = aiff_read_headers (musicin, &pcm_aiff_data)) != -1) {
-        fprintf (stderr, ">>> Using Audio IFF sound file headers\n");
-        aiff_check (inPath, &pcm_aiff_data, &header->version);
-        if (fseek (musicin, soundPosition, SEEK_SET) != 0) {
-            fprintf (stderr, "Could not seek to PCM sound data in \"%s\".\n",
-                    inPath);
-            exit (1);
-        }
-        fprintf (stderr, "Parsing AIFF audio file \n");
-        header->sampling_frequency =
-            SmpFrqIndex ((long) pcm_aiff_data.sampleRate, &header->version);
-        fprintf (stderr, ">>> %f Hz sampling frequency selected\n",
-                pcm_aiff_data.sampleRate);
-
-        /* Determine number of samples in sound file */
-        *num_samples = pcm_aiff_data.numChannels * pcm_aiff_data.numSampleFrames;
-
-        if (pcm_aiff_data.numChannels == 1) {
-            header->mode = MPG_MD_MONO;
-            header->mode_ext = 0;
-        }
-        return;
-    }
-
-    /**************************** WAVE *********************************/
-    /*   Nick Burch <The_Leveller@newmail.net> */
-    /*********************************/
-    /* Wave File Headers:   (Dec)    */
-    /* 8-11 = "WAVE"                 */
-    /* 22 = Stereo / Mono            */
-    /*       01 = mono, 02 = stereo  */
-    /* 24 = Sampling Frequency       */
-    /* 32 = Data Rate                */
-    /*       01 = x1 (8bit Mono)     */
-    /*       02 = x2 (8bit Stereo or */
-    /*                16bit Mono)    */
-    /*       04 = x4 (16bit Stereo)  */
-    /*********************************/
-
-    fseek (musicin, 0, SEEK_SET);
-    fread (wave_header_buffer, 1, 40, musicin);
-
-    if (wave_header_buffer[8] == 'W' && wave_header_buffer[9] == 'A'
-            && wave_header_buffer[10] == 'V' && wave_header_buffer[11] == 'E') {
-        fprintf (stderr, "Parsing Wave File Header\n");
-        if (NativeByteOrder == order_unknown) {
-            NativeByteOrder = DetermineByteOrder ();
-            if (NativeByteOrder == order_unknown) {
-                fprintf (stderr, "byte order not determined\n");
-                exit (1);
-            }
-        }
-        if (NativeByteOrder == order_littleEndian) {
-            samplerate = wave_header_buffer[24] +
-                (wave_header_buffer[25] << 8) +
-                (wave_header_buffer[26] << 16) +
-                (wave_header_buffer[27] << 24);
-        } else {
-            samplerate = wave_header_buffer[27] +
-                (wave_header_buffer[26] << 8) +
-                (wave_header_buffer[25] << 16) +
-                (wave_header_buffer[24] << 24);
-        }
-        /* Wave File */
-        wave_header_read = 1;
-        switch (samplerate) {
-            case 44100:
-            case 48000:
-            case 32000:
-            case 24000:
-            case 22050:
-            case 16000:
-                fprintf (stderr, ">>> %ld Hz sampling freq selected\n", samplerate);
-                break;
-            default:
-                /* Unknown Unsupported Frequency */
-                fprintf (stderr, ">>> Unknown samp freq %ld Hz in Wave Header\n",
-                        samplerate);
-                fprintf (stderr, ">>> Default 44.1 kHz samp freq selected\n");
-                samplerate = 44100;
-        }
-
-        if ((header->sampling_frequency =
-                    SmpFrqIndex ((long) samplerate, &header->version)) < 0) {
-            fprintf (stderr, "invalid sample rate\n");
-            exit (0);
-        }
-
-        if ((long) wave_header_buffer[22] == 1) {
-            fprintf (stderr, ">>> Input Wave File is Mono\n");
-            wave_header_stereo = 0;
-            header->mode = MPG_MD_MONO;
-            header->mode_ext = 0;
-        }
-        if ((long) wave_header_buffer[22] == 2) {
-            fprintf (stderr, ">>> Input Wave File is Stereo\n");
-            wave_header_stereo = 1;
-        }
-        if ((long) wave_header_buffer[32] == 1) {
-            fprintf (stderr, ">>> Input Wave File is 8 Bit\n");
-            wave_header_16bit = 0;
-            fprintf (stderr, "Input File must be 16 Bit! Please Re-sample");
-            exit (1);
-        }
-        if ((long) wave_header_buffer[32] == 2) {
-            if (wave_header_stereo == 1) {
-                fprintf (stderr, ">>> Input Wave File is 8 Bit\n");
-                wave_header_16bit = 0;
-                fprintf (stderr, "Input File must be 16 Bit! Please Re-sample");
-                exit (1);
-            } else {
-                /* fprintf(stderr,  ">>> Input Wave File is 16 Bit\n" ); */
-                wave_header_16bit = 1;
-            }
-        }
-        if ((long) wave_header_buffer[32] == 4) {
-            /* fprintf(stderr,  ">>> Input Wave File is 16 Bit\n" ); */
-            wave_header_16bit = 1;
-        }
-        /* should probably use the wave header to determine size here FIXME MFC Feb 2003 */
-        *num_samples = MAX_U_32_NUM;
-        if (fseek (musicin, 44, SEEK_SET) != 0) {   /* there's a way of calculating the size of the
-                                                       wave header. i'll just jump 44 to start with */
-            fprintf (stderr, "Could not seek to PCM sound data in \"%s\".\n",
-                    inPath);
-            exit (1);
-        }
-        return;
-    }
-
-    /*************************** PCM **************************/
-    fprintf (stderr, "No header found. Assuming Raw PCM sound file\n");
-    /* Raw PCM. No header. Reset the input file to read from the start */
-    fseek (musicin, 0, SEEK_SET);
-    /* Assume it is a huge sound file since there's no real info available */
-    /* FIXME: Could always fstat the file? Probably not worth it. MFC Feb 2003 */
-    *num_samples = MAX_U_32_NUM;
-}
-
-
-
-/************************************************************************
- *
- * aiff_check
- *
- * PURPOSE:  Checks AIFF header information to make sure it is valid.
- *           Exits if not.
- *
- ************************************************************************/
-
-void aiff_check (char *file_name, IFF_AIFF * pcm_aiff_data, int *version)
-{
-    if (pcm_aiff_data->sampleType != IFF_ID_SSND) {
-        fprintf (stderr, "Sound data is not PCM in \"%s\".\n", file_name);
-        exit (1);
-    }
-
-    if (SmpFrqIndex ((long) pcm_aiff_data->sampleRate, version) < 0) {
-        fprintf (stderr, "in \"%s\".\n", file_name);
-        exit (1);
-    }
-
-    if (pcm_aiff_data->sampleSize != sizeof (short) * BITS_IN_A_BYTE) {
-        fprintf (stderr, "Sound data is not %zu bits in \"%s\".\n",
-                sizeof (short) * BITS_IN_A_BYTE, file_name);
-        exit (1);
-    }
-
-    if (pcm_aiff_data->numChannels != MONO
-            && pcm_aiff_data->numChannels != STEREO) {
-        fprintf (stderr, "Sound data is not mono or stereo in \"%s\".\n",
-                file_name);
-        exit (1);
-    }
-
-    if (pcm_aiff_data->blkAlgn.blockSize != 0) {
-        fprintf (stderr, "Block size is not %d bytes in \"%s\".\n", 0, file_name);
-        exit (1);
-    }
-
-    if (pcm_aiff_data->blkAlgn.offset != 0) {
-        fprintf (stderr, "Block offset is not %d bytes in \"%s\".\n", 0,
-                file_name);
-        exit (1);
-    }
-}