Introduced extraction of .au (but not its usage within cdrskin)

This commit is contained in:
Thomas Schmitt 2006-10-24 07:50:31 +00:00
parent f78cc40041
commit 0d26beb2ed
5 changed files with 148 additions and 45 deletions

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@ -926,7 +926,7 @@ int Cdrtrack_extract_audio(struct CdrtracK *track, int *fd, off_t *xtr_size,
#ifdef Cdrskin_libburn_has_audioxtR #ifdef Cdrskin_libburn_has_audioxtR
struct libdax_audioxtr *xtr= NULL; struct libdax_audioxtr *xtr= NULL;
char *fmt,*fmt_info; char *fmt,*fmt_info;
int num_channels,sample_rate,bits_per_sample,ret; int num_channels,sample_rate,bits_per_sample,msb_first,ret;
#endif #endif
*fd= -1; *fd= -1;
@ -952,7 +952,7 @@ int Cdrtrack_extract_audio(struct CdrtracK *track, int *fd, off_t *xtr_size,
if(ret<=0) if(ret<=0)
return(ret); return(ret);
libdax_audioxtr_get_id(xtr,&fmt,&fmt_info, libdax_audioxtr_get_id(xtr,&fmt,&fmt_info,
&num_channels,&sample_rate,&bits_per_sample,0); &num_channels,&sample_rate,&bits_per_sample,&msb_first,0);
if(strcmp(fmt,".wav")!=0 || if(strcmp(fmt,".wav")!=0 ||
num_channels!=2 || sample_rate!=44100 || bits_per_sample!=16) { num_channels!=2 || sample_rate!=44100 || bits_per_sample!=16) {
fprintf(stderr,"cdrskin: ( %s )\n",fmt_info); fprintf(stderr,"cdrskin: ( %s )\n",fmt_info);

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@ -1 +1 @@
#define Cdrskin_timestamP "2006.10.23.134719" #define Cdrskin_timestamP "2006.10.24.075039"

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@ -41,11 +41,16 @@ int libdax_audioxtr_new(struct libdax_audioxtr **xtr, char *path, int flag)
o->data_size= 0; o->data_size= 0;
o->extract_count= 0; o->extract_count= 0;
o->wav_num_channels= 0; o->num_channels= 0;
o->wav_sample_rate= 0; o->sample_rate= 0;
o->wav_bits_per_sample= 0; o->bits_per_sample= 0;
o->msb_first= 0;
o->wav_subchunk2_size= 0; o->wav_subchunk2_size= 0;
o->au_data_location= 0;
o->au_data_size= 0xffffffff;
ret= libdax_audioxtr_open(o,0); ret= libdax_audioxtr_open(o,0);
if(ret<=0) if(ret<=0)
{ret= -2*(ret<0); goto failure;} {ret= -2*(ret<0); goto failure;}
@ -108,14 +113,19 @@ static int libdax_audioxtr_open(struct libdax_audioxtr *o, int flag)
} }
static int libdax_audioxtr_identify(struct libdax_audioxtr *o, int flag) static int libdax_audioxtr_identify_wav(struct libdax_audioxtr *o, int flag)
{ {
int ret; int ret;
char buf[45]; char buf[45];
/* currently this only works for MS WAVE files .wav */ /* check wether this is a MS WAVE file .wav */
/* info used: http://ccrma.stanford.edu/courses/422/projects/WaveFormat/ */ /* info used: http://ccrma.stanford.edu/courses/422/projects/WaveFormat/ */
if(o->fd!=0) {
ret= lseek(o->fd,0,SEEK_SET);
if(ret==-1)
return(0);
}
ret= read(o->fd, buf, 44); ret= read(o->fd, buf, 44);
if(ret<44) if(ret<44)
return(0); return(0);
@ -133,25 +143,97 @@ static int libdax_audioxtr_identify(struct libdax_audioxtr *o, int flag)
return(0); return(0);
strcpy(o->fmt,".wav"); strcpy(o->fmt,".wav");
o->wav_num_channels= libdax_audioxtr_to_int(o,(unsigned char *) buf+22,2,0); o->msb_first= 0;
o->wav_sample_rate= libdax_audioxtr_to_int(o,(unsigned char *) buf+24,4,0); o->num_channels= libdax_audioxtr_to_int(o,(unsigned char *) buf+22,2,0);
o->wav_bits_per_sample= libdax_audioxtr_to_int(o,(unsigned char *)buf+34,2,0); o->sample_rate= libdax_audioxtr_to_int(o,(unsigned char *) buf+24,4,0);
o->bits_per_sample= libdax_audioxtr_to_int(o,(unsigned char *)buf+34,2,0);
sprintf(o->fmt_info, sprintf(o->fmt_info,
".wav , num_channels=%d , sample_rate=%d , bits_per_sample=%d", ".wav , num_channels=%d , sample_rate=%d , bits_per_sample=%d",
o->wav_num_channels,o->wav_sample_rate,o->wav_bits_per_sample); o->num_channels,o->sample_rate,o->bits_per_sample);
o->wav_subchunk2_size= libdax_audioxtr_to_int(o,(unsigned char *)buf+40,4,0); o->wav_subchunk2_size= libdax_audioxtr_to_int(o,(unsigned char *)buf+40,4,0);
o->data_size= o->wav_subchunk2_size; o->data_size= o->wav_subchunk2_size;
return(1); return(1);
} }
static int libdax_audioxtr_identify_au(struct libdax_audioxtr *o, int flag)
{
int ret,encoding;
char buf[24];
/* Check wether this is a Sun Audio, .au file */
/* info used: http://ccrma.stanford.edu/courses/422/projects/WaveFormat/ */
if(o->fd!=0) {
ret= lseek(o->fd,0,SEEK_SET);
if(ret==-1)
return(0);
}
ret= read(o->fd, buf, 24);
if(ret<24)
return(0);
if(strncmp(buf,".snd",4)!=0)
return(0);
strcpy(o->fmt,".au");
o->msb_first= 1;
o->au_data_location= libdax_audioxtr_to_int(o,(unsigned char *)buf+4,4,1);
o->au_data_size= libdax_audioxtr_to_int(o,(unsigned char *)buf+8,4,1);
encoding= libdax_audioxtr_to_int(o,(unsigned char *)buf+12,4,1);
if(encoding==2)
o->bits_per_sample= 8;
else if(encoding==3)
o->bits_per_sample= 16;
else if(encoding==4)
o->bits_per_sample= 24;
else if(encoding==5)
o->bits_per_sample= 32;
else
o->bits_per_sample= -encoding;
o->sample_rate= libdax_audioxtr_to_int(o,(unsigned char *)buf+16,4,1);
o->num_channels= libdax_audioxtr_to_int(o,(unsigned char *)buf+20,4,1);
if(o->au_data_size!=0xffffffff)
o->data_size= o->au_data_size;
else
o->data_size= 0;
sprintf(o->fmt_info,
".au , num_channels=%d , sample_rate=%d , bits_per_sample=%d",
o->num_channels,o->sample_rate,o->bits_per_sample);
/* <<< for testing only */;
return(1);
return(o->bits_per_sample>0); /* Audio format must be linear PCM */
}
static int libdax_audioxtr_identify(struct libdax_audioxtr *o, int flag)
{
int ret;
ret= libdax_audioxtr_identify_wav(o, 0);
if(ret!=0)
return(ret);
if(o->fd==0) /* cannot rewind stdin */
return(0);
ret= libdax_audioxtr_identify_au(o, 0);
if(ret!=0)
return(ret);
return(0);
}
/* @param flag bit0=msb_first */
static unsigned libdax_audioxtr_to_int(struct libdax_audioxtr *o, static unsigned libdax_audioxtr_to_int(struct libdax_audioxtr *o,
unsigned char *bytes, int len, int flag) unsigned char *bytes, int len, int flag)
{ {
unsigned int ret= 0; unsigned int ret= 0;
int i; int i;
if(flag&1)
for(i= 0; i<len; i++)
ret= ret*256+bytes[i];
else
for(i= len-1; i>=0; i--) for(i= len-1; i>=0; i--)
ret= ret*256+bytes[i]; ret= ret*256+bytes[i];
return(ret); return(ret);
@ -162,12 +244,18 @@ static int libdax_audioxtr_init_reading(struct libdax_audioxtr *o, int flag)
{ {
int ret; int ret;
/* currently this only works for MS WAVE files .wav */;
/* currently this only works for MS WAVE files .wav and Sun .au*/;
if(o->fd==0) /* stdin: hope no read came after libdax_audioxtr_identify() */ if(o->fd==0) /* stdin: hope no read came after libdax_audioxtr_identify() */
return(1); return(1);
o->extract_count= 0; o->extract_count= 0;
if(strcmp(o->fmt,".wav")==0)
ret= lseek(o->fd,44,SEEK_SET); ret= lseek(o->fd,44,SEEK_SET);
else if(strcmp(o->fmt,".au")==0)
ret= lseek(o->fd,o->au_data_location,SEEK_SET);
else
ret= -1;
if(ret==-1) if(ret==-1)
return(0); return(0);
@ -178,28 +266,22 @@ static int libdax_audioxtr_init_reading(struct libdax_audioxtr *o, int flag)
int libdax_audioxtr_get_id(struct libdax_audioxtr *o, int libdax_audioxtr_get_id(struct libdax_audioxtr *o,
char **fmt, char **fmt_info, char **fmt, char **fmt_info,
int *num_channels, int *sample_rate, int *bits_per_sample, int *num_channels, int *sample_rate, int *bits_per_sample,
int flag) int *msb_first, int flag)
{ {
*fmt= o->fmt; *fmt= o->fmt;
*fmt_info= o->fmt_info; *fmt_info= o->fmt_info;
if(strcmp(o->fmt,".wav")==0) { *num_channels= o->num_channels;
*num_channels= o->wav_num_channels; *sample_rate= o->sample_rate;
*sample_rate= o->wav_sample_rate; *bits_per_sample= o->bits_per_sample;
*bits_per_sample= o->wav_bits_per_sample; *msb_first= o->msb_first;
} else
*num_channels= *sample_rate= *bits_per_sample= 0;
return(1); return(1);
} }
int libdax_audioxtr_get_size(struct libdax_audioxtr *o, off_t *size, int flag) int libdax_audioxtr_get_size(struct libdax_audioxtr *o, off_t *size, int flag)
{ {
if(strcmp(o->fmt,".wav")==0) { *size= o->data_size;
*size= o->wav_subchunk2_size;
return(1); return(1);
}
*size= 0;
return(0);
} }
@ -226,7 +308,7 @@ int libdax_audioxtr_detach_fd(struct libdax_audioxtr *o, int *fd, int flag)
{ {
if(o->fd<0) if(o->fd<0)
return(-1); return(-1);
if(strcmp(o->fmt,".wav")!=0) if(strcmp(o->fmt,".wav")!=0 && strcmp(o->fmt,".au")!=0)
return(0); return(0);
if(flag&1) { if(flag&1) {
*fd= o->fd; *fd= o->fd;

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@ -46,18 +46,19 @@ int libdax_audioxtr_new(struct libdax_audioxtr **xtr, char *path, int flag);
/** Obtain identification parameters of opened audio source. /** Obtain identification parameters of opened audio source.
@param xtr Opaque handle to extractor @param xtr Opaque handle to extractor
@param fmt Gets pointed to the audio file format id text: ".wav" @param fmt Gets pointed to the audio file format id text: ".wav" , ".au"
@param fmt_info Gets pointed to a format info text telling parameters @param fmt_info Gets pointed to a format info text telling parameters
@param num_channels e.g. 1=mono, 2=stereo, etc @param num_channels e.g. 1=mono, 2=stereo, etc
@param sample_rate e.g. 11025, 44100 @param sample_rate e.g. 11025, 44100
@param bits_per_sample e.g. 8= 8 bits per sample, 16= 16 bits ... @param bits_per_sample e.g. 8= 8 bits per sample, 16= 16 bits ...
@param msb_first Byte order of samples: 0=Intel 1=Motorola
@param flag Bitfield for control purposes (unused yet, submit 0) @param flag Bitfield for control purposes (unused yet, submit 0)
@return >0 success, <=0 failure @return >0 success, <=0 failure
*/ */
int libdax_audioxtr_get_id(struct libdax_audioxtr *xtr, int libdax_audioxtr_get_id(struct libdax_audioxtr *xtr,
char **fmt, char **fmt_info, char **fmt, char **fmt_info,
int *num_channels, int *sample_rate, int *num_channels, int *sample_rate,
int *bits_per_sample, int flag); int *bits_per_sample, int *msb_first, int flag);
/** Obtain a prediction about the extracted size based on internal information /** Obtain a prediction about the extracted size based on internal information
@ -136,7 +137,7 @@ struct libdax_audioxtr {
/* Source of the encoded audio data */ /* Source of the encoded audio data */
char path[LIBDAX_AUDIOXTR_STRLEN]; char path[LIBDAX_AUDIOXTR_STRLEN];
/* File descriptor to path */ /* File descriptor to path. Anything else than 0 must be lseek-able */
int fd; int fd;
/* Format identifier. E.g. ".wav" */ /* Format identifier. E.g. ".wav" */
@ -145,6 +146,18 @@ struct libdax_audioxtr {
/* Format parameter info text */ /* Format parameter info text */
char fmt_info[LIBDAX_AUDIOXTR_STRLEN]; char fmt_info[LIBDAX_AUDIOXTR_STRLEN];
/* 1= mono, 2= stereo, etc. */
int num_channels;
/* 8000, 44100, etc. */
int sample_rate;
/* 8 bits = 8, 16 bits = 16, etc. */
int bits_per_sample;
/* Byte order of samples: 0=Intel 1=Motorola */
int msb_first;
/* Number of bytes to extract (0= unknown/unlimited) */ /* Number of bytes to extract (0= unknown/unlimited) */
off_t data_size; off_t data_size;
@ -157,19 +170,20 @@ struct libdax_audioxtr {
/* MS WAVE Format */ /* MS WAVE Format */
/* info used: http://ccrma.stanford.edu/courses/422/projects/WaveFormat/ */ /* info used: http://ccrma.stanford.edu/courses/422/projects/WaveFormat/ */
/* 1= mono, 2= stereo, etc. */
int wav_num_channels;
/* 8000, 44100, etc. */
int wav_sample_rate;
/* 8 bits = 8, 16 bits = 16, etc. */
int wav_bits_per_sample;
/* == NumSamples * NumChannels * BitsPerSample/8 /* == NumSamples * NumChannels * BitsPerSample/8
This is the number of bytes in the data. */ This is the number of bytes in the data. */
unsigned wav_subchunk2_size; unsigned wav_subchunk2_size;
/* Sun Audio, .au */
/* info used: http://www.opengroup.org/public/pubs/external/auformat.html */
/* Number of bytes in non-payload header part */
unsigned au_data_location;
/* Number of payload bytes or 0xffffffff */
unsigned au_data_size;
}; };
@ -186,8 +200,15 @@ static int libdax_audioxtr_open(struct libdax_audioxtr *o, int flag);
*/ */
static int libdax_audioxtr_identify(struct libdax_audioxtr *o, int flag); static int libdax_audioxtr_identify(struct libdax_audioxtr *o, int flag);
/** Specialized identifier for .wav */
static int libdax_audioxtr_identify_wav(struct libdax_audioxtr *o, int flag);
/** Specialized identifier for .au */
static int libdax_audioxtr_identify_au(struct libdax_audioxtr *o, int flag);
/** Convert a byte string into a number (currently only little endian) /** Convert a byte string into a number (currently only little endian)
@param flag Bitfield for control purposes
bit0=msb_first
@return The resulting number @return The resulting number
*/ */
static unsigned libdax_audioxtr_to_int(struct libdax_audioxtr *o, static unsigned libdax_audioxtr_to_int(struct libdax_audioxtr *o,

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@ -53,7 +53,7 @@ int main(int argc, char **argv)
/* Inquired source parameters */ /* Inquired source parameters */
char *fmt, *fmt_info; char *fmt, *fmt_info;
int num_channels, sample_rate, bits_per_sample; int num_channels, sample_rate, bits_per_sample, msb_first;
off_t data_size; off_t data_size;
/* Auxiliary variables */ /* Auxiliary variables */
@ -142,7 +142,7 @@ help:;
} }
/* Obtain and print parameters of audio source */ /* Obtain and print parameters of audio source */
libdax_audioxtr_get_id(xtr, &fmt, &fmt_info, libdax_audioxtr_get_id(xtr, &fmt, &fmt_info,
&num_channels, &sample_rate, &bits_per_sample, 0); &num_channels, &sample_rate, &bits_per_sample, &msb_first, 0);
fprintf(stderr, "Detected format: %s\n", fmt_info); fprintf(stderr, "Detected format: %s\n", fmt_info);
libdax_audioxtr_get_size(xtr, &data_size, 0); libdax_audioxtr_get_size(xtr, &data_size, 0);
fprintf(stderr, "Data size : %.f bytes\n", (double) data_size); fprintf(stderr, "Data size : %.f bytes\n", (double) data_size);