Include toolame-dab as library

1 files changed, 689 insertions, 0 deletions

diff --git a/libtoolame-dab/audio_read.c b/libtoolame-dab/audio_read.c
new file mode 100644
index 0000000..3649ef4
--- /dev/null
+++ b/libtoolame-dab/audio_read.c
@@ -0,0 +1,689 @@
+#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);
+    }
+}