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authorMatthias P. Braendli <matthias.braendli@mpb.li>2016-02-15 04:34:28 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2016-02-15 04:34:28 +0100
commit5671aa4f0f5536de2fc58b6f89f01947f4f8f1c1 (patch)
treeb1bddbfac8bd69c733faf734803e7f1842285e22 /libtoolame-dab/audio_read.c
parentba346d2469facf500cbcaa9cf9117ce04ea0b6da (diff)
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Remove useless libtoolame-dab files
Diffstat (limited to 'libtoolame-dab/audio_read.c')
-rw-r--r--libtoolame-dab/audio_read.c689
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);
- }
-}