Nyaaori/rockbox
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
rockbox/apps/metadata.c
Dave Chapman 2bf9be1c02 Patch #1352575 - Shorten codec from the ffmpeg project. Rockbox implementation by Mark Arigo. 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7814 a1c6a512-1295-4272-9138-f99709370657
2005-11-11 19:45:36 +00:00

1533 lines
43 KiB
C
Raw Blame History

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <ctype.h>
#include <inttypes.h>
#include "metadata.h"
#include "mp3_playback.h"
#include "logf.h"
#include "atoi.h"
#include "replaygain.h"
#include "debug.h"
#include "system.h"
enum tagtype { TAGTYPE_APE = 1, TAGTYPE_VORBIS };
#define APETAG_HEADER_LENGTH 32
#define APETAG_HEADER_FORMAT "8LLLL"
#define APETAG_ITEM_HEADER_FORMAT "LL"
#define APETAG_ITEM_TYPE_MASK 3
#define TAG_NAME_LENGTH 32
#define TAG_VALUE_LENGTH 128
struct apetag_header
{
char id[8];
long version;
long length;
long item_count;
long flags;
char reserved[8];
};
struct apetag_item_header
{
long length;
long flags;
};
struct format_list
{
char format;
char extension[5];
};
static const struct format_list formats[] =
{
{ AFMT_MPA_L1, "mp1" },
{ AFMT_MPA_L2, "mp2" },
{ AFMT_MPA_L2, "mpa" },
{ AFMT_MPA_L3, "mp3" },
{ AFMT_OGG_VORBIS, "ogg" },
{ AFMT_PCM_WAV, "wav" },
{ AFMT_FLAC, "flac" },
{ AFMT_MPC, "mpc" },
{ AFMT_A52, "a52" },
{ AFMT_A52, "ac3" },
{ AFMT_WAVPACK, "wv" },
{ AFMT_ALAC, "m4a" },
{ AFMT_AAC, "mp4" },
{ AFMT_SHN, "shn" },
};
static const unsigned short a52_bitrates[] =
{
32, 40, 48, 56, 64, 80, 96, 112, 128, 160,
192, 224, 256, 320, 384, 448, 512, 576, 640
};
/* Only store frame sizes for 44.1KHz - others are simply multiples
of the bitrate */
static const unsigned short a52_441framesizes[] =
{
69 * 2, 70 * 2, 87 * 2, 88 * 2, 104 * 2, 105 * 2, 121 * 2,
122 * 2, 139 * 2, 140 * 2, 174 * 2, 175 * 2, 208 * 2, 209 * 2,
243 * 2, 244 * 2, 278 * 2, 279 * 2, 348 * 2, 349 * 2, 417 * 2,
418 * 2, 487 * 2, 488 * 2, 557 * 2, 558 * 2, 696 * 2, 697 * 2,
835 * 2, 836 * 2, 975 * 2, 976 * 2, 1114 * 2, 1115 * 2, 1253 * 2,
1254 * 2, 1393 * 2, 1394 * 2
};
static const long wavpack_sample_rates [] =
{
6000, 8000, 9600, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000, 192000
};
/* Read a string from the file. Read up to size bytes, or, if eos != -1,
* until the eos character is found (eos is not stored in buf, unless it is
* nil). Writes up to buf_size chars to buf, always terminating with a nil.
* Returns number of chars read or -1 on read error.
*/
static long read_string(int fd, char* buf, long buf_size, int eos, long size)
{
long read_bytes = 0;
char c;
while (size != 0)
{
if (read(fd, &c, 1) != 1)
{
read_bytes = -1;
break;
}
read_bytes++;
size--;
if ((eos != -1) && (eos == (unsigned char) c))
{
break;
}
if (buf_size > 1)
{
*buf++ = c;
buf_size--;
}
}
*buf = 0;
return read_bytes;
}
/* Convert a little-endian structure to native format using a format string.
* Does nothing on a little-endian machine.
*/
static void convert_endian(void *data, const char *format)
{
while (*format)
{
switch (*format)
{
case 'L':
{
long* d = (long*) data;
*d = letoh32(*d);
data = d + 1;
}
break;
case 'S':
{
short* d = (short*) data;
*d = letoh16(*d);
data = d + 1;
}
break;
default:
if (isdigit(*format))
{
data = ((char*) data) + *format - '0';
}
break;
}
format++;
}
}
/* read_uint32be() - read an unsigned integer from a big-endian
(e.g. Quicktime) file. This is used by the .m4a parser
*/
#ifdef ROCKBOX_BIG_ENDIAN
#define read_uint32be(fd,buf) read((fd),(buf),4)
#else
int read_uint32be(int fd, unsigned int* buf) {
char tmp;
char* p=(char*)buf;
size_t n;
n=read(fd,p,4);
if (n==4) {
tmp=p[0];
p[0]=p[3];
p[3]=tmp;
tmp=p[2];
p[2]=p[1];
p[1]=tmp;
}
return(n);
}
#endif
/* Read an unaligned 32-bit little endian long from buffer. */
static unsigned long get_long(void* buf)
{
unsigned char* p = (unsigned char*) buf;
return p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24);
}
/* Convert an UTF-8 string to Latin-1, overwriting the old string (the new
* string is never longer than the original, so this is safe). Non-latin-1
* chars are replaced with '?'.
*/
static void convert_utf8(char* utf8)
{
char* dest = utf8;
long code = 0;
unsigned char c;
int tail = 0;
int size = 0;
while ((c = *utf8++) != 0)
{
if ((tail <= 0) && ((c <= 0x7f) || (c >= 0xc2)))
{
/* Start of new character. */
if (c < 0x80)
{
size = 1;
}
else if (c < 0xe0)
{
size = 2;
c &= 0x1f;
}
else if (c < 0xf0)
{
size = 3;
c &= 0x0f;
}
else if (c < 0xf5)
{
size = 4;
c &= 0x07;
}
else
{
/* Invalid size. */
size = 0;
}
code = c;
tail = size - 1;
}
else if ((tail > 0) && ((c & 0xc0) == 0x80))
{
/* Valid continuation character. */
code = (code << 6) | (c & 0x3f);
tail--;
if (tail == 0)
{
if (((size == 2) && (code < 0x80))
|| ((size == 3) && (code < 0x800))
|| ((size == 4) && (code < 0x10000)))
{
/* Invalid encoding. */
code = 0;
}
}
}
else
{
tail = -1;
}
if ((tail == 0) && (code > 0))
{
*dest++ = (code <= 0xff) ? (char) (code & 0xff) : '?';
}
}
*dest = 0;
}
/* Read a string tag from an M4A file */
void read_m4a_tag_string(int fd, int len,char** bufptr,size_t* bytes_remaining, char** dest)
{
int data_length;
if (bytes_remaining==0) {
lseek(fd,len,SEEK_CUR); /* Skip everything */
} else {
/* Skip the data tag header - maybe we should parse it properly? */
lseek(fd,16,SEEK_CUR);
len-=16;
*dest=*bufptr;
if ((size_t)len+1 > *bytes_remaining) {
read(fd,*bufptr,*bytes_remaining-1);
lseek(fd,len-(*bytes_remaining-1),SEEK_CUR);
*bufptr+=(*bytes_remaining-1);
} else {
read(fd,*bufptr,len);
*bufptr+=len;
}
**bufptr=(char)0;
convert_utf8(*dest);
data_length = strlen(*dest)+1;
*bufptr=(*dest)+data_length;
*bytes_remaining-=data_length;
}
}
/* Parse the tag (the name-value pair) and fill id3 and buffer accordingly.
* String values to keep are written to buf. Returns number of bytes written
* to buf (including end nil).
*/
static long parse_tag(const char* name, char* value, struct mp3entry* id3,
char* buf, long buf_remaining, enum tagtype type)
{
long len = 0;
char** p;
if ((((strcasecmp(name, "track") == 0) && (type == TAGTYPE_APE)))
|| ((strcasecmp(name, "tracknumber") == 0) && (type == TAGTYPE_VORBIS)))
{
id3->tracknum = atoi(value);
p = &(id3->track_string);
}
else if (((strcasecmp(name, "year") == 0) && (type == TAGTYPE_APE))
|| ((strcasecmp(name, "date") == 0) && (type == TAGTYPE_VORBIS)))
{
/* Date can be in more any format in a Vorbis tag, so don't try to
* parse it.
*/
if (type != TAGTYPE_VORBIS)
{
id3->year = atoi(value);
}
p = &(id3->year_string);
}
else if (strcasecmp(name, "title") == 0)
{
p = &(id3->title);
}
else if (strcasecmp(name, "artist") == 0)
{
p = &(id3->artist);
}
else if (strcasecmp(name, "album") == 0)
{
p = &(id3->album);
}
else if (strcasecmp(name, "genre") == 0)
{
p = &(id3->genre_string);
}
else if (strcasecmp(name, "composer") == 0)
{
p = &(id3->composer);
}
else
{
len = parse_replaygain(name, value, id3, buf, buf_remaining);
p = NULL;
}
if (p)
{
len = strlen(value);
len = MIN(len, buf_remaining - 1);
if (len > 0)
{
strncpy(buf, value, len);
buf[len] = 0;
*p = buf;
len++;
}
else
{
len = 0;
}
}
return len;
}
/* Read the items in an APEV2 tag. Only looks for a tag at the end of a
* file. Returns true if a tag was found and fully read, false otherwise.
*/
static bool read_ape_tags(int fd, struct mp3entry* id3)
{
struct apetag_header header;
if ((lseek(fd, -APETAG_HEADER_LENGTH, SEEK_END) < 0)
|| (read(fd, &header, APETAG_HEADER_LENGTH) != APETAG_HEADER_LENGTH)
|| (memcmp(header.id, "APETAGEX", sizeof(header.id))))
{
return false;
}
convert_endian(&header, APETAG_HEADER_FORMAT);
id3->genre = 0xff;
if ((header.version == 2000) && (header.item_count > 0)
&& (header.length > APETAG_HEADER_LENGTH))
{
char *buf = id3->id3v2buf;
unsigned int buf_remaining = sizeof(id3->id3v2buf)
+ sizeof(id3->id3v1buf);
unsigned int tag_remaining = header.length - APETAG_HEADER_LENGTH;
int i;
if (lseek(fd, -header.length, SEEK_END) < 0)
{
return false;
}
for (i = 0; i < header.item_count; i++)
{
struct apetag_item_header item;
char name[TAG_NAME_LENGTH];
char value[TAG_VALUE_LENGTH];
long r;
if (tag_remaining < sizeof(item))
{
break;
}
if (read(fd, &item, sizeof(item)) < (long) sizeof(item))
{
return false;
}
convert_endian(&item, APETAG_ITEM_HEADER_FORMAT);
tag_remaining -= sizeof(item);
r = read_string(fd, name, sizeof(name), 0, tag_remaining);
if (r == -1)
{
return false;
}
tag_remaining -= r + item.length;
if ((item.flags & APETAG_ITEM_TYPE_MASK) == 0)
{
long len;
if (read_string(fd, value, sizeof(value), -1, item.length)
!= item.length)
{
return false;
}
convert_utf8(value);
len = parse_tag(name, value, id3, buf, buf_remaining,
TAGTYPE_APE);
buf += len;
buf_remaining -= len;
}
else
{
if (lseek(fd, item.length, SEEK_CUR) < 0)
{
return false;
}
}
}
}
return true;
}
/* Read the items in a Vorbis comment packet. Returns true the items were
* fully read, false otherwise.
*/
static bool read_vorbis_tags(int fd, struct mp3entry *id3,
long tag_remaining)
{
char *buf = id3->id3v2buf;
long comment_count;
long len;
int buf_remaining = sizeof(id3->id3v2buf) + sizeof(id3->id3v1buf);
int i;
id3->genre = 255;
if (read(fd, &len, sizeof(len)) < (long) sizeof(len))
{
return false;
}
convert_endian(&len, "L");
if ((lseek(fd, len, SEEK_CUR) < 0)
|| (read(fd, &comment_count, sizeof(comment_count))
< (long) sizeof(comment_count)))
{
return false;
}
convert_endian(&comment_count, "L");
tag_remaining -= len + sizeof(len) + sizeof(comment_count);
if (tag_remaining <= 0)
{
return true;
}
for (i = 0; i < comment_count; i++)
{
char name[TAG_NAME_LENGTH];
char value[TAG_VALUE_LENGTH];
long read_len;
if (tag_remaining < 4)
{
break;
}
if (read(fd, &len, sizeof(len)) < (long) sizeof(len))
{
return false;
}
convert_endian(&len, "L");
tag_remaining -= 4;
/* Quit if we've passed the end of the page */
if (tag_remaining < len)
{
break;
}
tag_remaining -= len;
read_len = read_string(fd, name, sizeof(name), '=', len);
if (read_len < 0)
{
return false;
}
len -= read_len;
if (read_string(fd, value, sizeof(value), -1, len) < 0)
{
return false;
}
convert_utf8(value);
len = parse_tag(name, value, id3, buf, buf_remaining,
TAGTYPE_VORBIS);
buf += len;
buf_remaining -= len;
}
/* Skip to the end of the block */
if (tag_remaining)
{
if (lseek(fd, tag_remaining, SEEK_CUR) < 0)
{
return false;
}
}
return true;
}
/* Skip an ID3v2 tag if it can be found. We assume the tag is located at the
* start of the file, which should be true in all cases where we need to skip it.
* Returns true if successfully skipped or not skipped, and false if
* something went wrong while skipping.
*/
static bool skip_id3v2(int fd, struct mp3entry *id3)
{
char buf[4];
read(fd, buf, 4);
if (memcmp(buf, "ID3", 3) == 0)
{
/* We have found an ID3v2 tag at the start of the file - find its
length and then skip it. */
if ((id3->first_frame_offset = getid3v2len(fd)) == 0)
return false;
if ((lseek(fd, id3->first_frame_offset, SEEK_SET) < 0))
return false;
return true;
} else {
lseek(fd, 0, SEEK_SET);
id3->first_frame_offset = 0;
return true;
}
}
/* A simple parser to read vital metadata from an Ogg Vorbis file. Returns
* false if metadata needed by the Vorbis codec couldn't be read.
*/
static bool get_vorbis_metadata(int fd, struct mp3entry* id3)
{
/* An Ogg File is split into pages, each starting with the string
* "OggS". Each page has a timestamp (in PCM samples) referred to as
* the "granule position".
*
* An Ogg Vorbis has the following structure:
* 1) Identification header (containing samplerate, numchannels, etc)
* 2) Comment header - containing the Vorbis Comments
* 3) Setup header - containing codec setup information
* 4) Many audio packets...
*/
/* Use the path name of the id3 structure as a temporary buffer. */
unsigned char* buf = id3->path;
long comment_size;
long remaining = 0;
long last_serial = 0;
long serial;
int segments;
int i;
bool eof = false;
if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, buf, 58) < 4))
{
return false;
}
if ((memcmp(buf, "OggS", 4) != 0) || (memcmp(&buf[29], "vorbis", 6) != 0))
{
return false;
}
/* We need to ensure the serial number from this page is the same as the
* one from the last page (since we only support a single bitstream).
*/
serial = get_long(&buf[14]);
id3->frequency = get_long(&buf[40]);
id3->filesize = filesize(fd);
/* Comments are in second Ogg page */
if (lseek(fd, 58, SEEK_SET) < 0)
{
return false;
}
/* Minimum header length for Ogg pages is 27. */
if (read(fd, buf, 27) < 27)
{
return false;
}
if (memcmp(buf, "OggS", 4) !=0 )
{
return false;
}
segments = buf[26];
/* read in segment table */
if (read(fd, buf, segments) < segments)
{
return false;
}
/* The second packet in a vorbis stream is the comment packet. It *may*
* extend beyond the second page, but usually does not. Here we find the
* length of the comment packet (or the rest of the page if the comment
* packet extends to the third page).
*/
for (i = 0; i < segments; i++)
{
remaining += buf[i];
/* The last segment of a packet is always < 255 bytes */
if (buf[i] < 255)
{
break;
}
}
/* Now read in packet header (type and id string) */
if (read(fd, buf, 7) < 7)
{
return false;
}
comment_size = remaining;
remaining -= 7;
/* The first byte of a packet is the packet type; comment packets are
* type 3.
*/
if ((buf[0] != 3) || (memcmp(buf + 1, "vorbis", 6) !=0))
{
return false;
}
/* Failure to read the tags isn't fatal. */
read_vorbis_tags(fd, id3, remaining);
/* We now need to search for the last page in the file - identified by
* by ('O','g','g','S',0) and retrieve totalsamples.
*/
if (lseek(fd, -64 * 1024, SEEK_END) < 0) /* A page is always < 64 kB */
{
return false;
}
remaining = 0;
while (!eof)
{
long r = read(fd, &buf[remaining], MAX_PATH - remaining);
if (r <= 0)
{
eof = true;
}
else
{
remaining += r;
}
/* Inefficient (but simple) search */
i = 0;
while (i < (remaining - 5))
{
if ((buf[i] == 'O') && (memcmp(&buf[i], "OggS", 4) == 0))
{
if (i < (remaining - 17))
{
/* Note that this only reads the low 32 bits of a
* 64 bit value.
*/
id3->samples = get_long(&buf[i + 6]);
last_serial = get_long(&buf[i + 14]);
/* We can discard the rest of the buffer */
remaining = 0;
}
else
{
break;
}
}
else
{
i++;
}
}
if (i < (remaining - 5))
{
/* Move OggS to start of buffer. */
while (i >0)
{
buf[i--] = buf[remaining--];
}
}
else
{
remaining = 0;
}
}
/* This file has mutiple vorbis bitstreams (or is corrupt). */
/* FIXME we should display an error here. */
if (serial != last_serial)
{
logf("serialno mismatch");
logf("%ld", serial);
logf("%ld", last_serial);
return false;
}
id3->length = (id3->samples / id3->frequency) * 1000;
id3->bitrate = (((int64_t) id3->filesize - comment_size) * 8) / id3->length;
id3->vbr = true;
return true;
}
static bool get_flac_metadata(int fd, struct mp3entry* id3)
{
/* A simple parser to read vital metadata from a FLAC file - length,
* frequency, bitrate etc. This code should either be moved to a
* seperate file, or discarded in favour of the libFLAC code.
* The FLAC stream specification can be found at
* http://flac.sourceforge.net/format.html#stream
*/
/* Use the trackname part of the id3 structure as a temporary buffer */
unsigned char* buf = id3->path;
bool rc = false;
if (!skip_id3v2(fd, id3) || (read(fd, buf, 4) < 4))
{
return rc;
}
if (memcmp(buf, "fLaC", 4) != 0)
{
return rc;
}
while (true)
{
long i;
if (read(fd, buf, 4) < 0)
{
return rc;
}
/* The length of the block */
i = (buf[1] << 16) | (buf[2] << 8) | buf[3];
if ((buf[0] & 0x7f) == 0) /* 0 is the STREAMINFO block */
{
unsigned long totalsamples;
/* FIXME: Don't trust the value of i */
if (read(fd, buf, i) < 0)
{
return rc;
}
id3->vbr = true; /* All FLAC files are VBR */
id3->filesize = filesize(fd);
id3->frequency = (buf[10] << 12) | (buf[11] << 4)
| ((buf[12] & 0xf0) >> 4);
rc = true; /* Got vital metadata */
/* totalsamples is a 36-bit field, but we assume <= 32 bits are used */
totalsamples = (buf[14] << 24) | (buf[15] << 16)
| (buf[16] << 8) | buf[17];
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
id3->length = (totalsamples / id3->frequency) * 1000;
id3->bitrate = (id3->filesize * 8) / id3->length;
}
else if ((buf[0] & 0x7f) == 4) /* 4 is the VORBIS_COMMENT block */
{
/* The next i bytes of the file contain the VORBIS COMMENTS. */
if (!read_vorbis_tags(fd, id3, i))
{
return rc;
}
}
else
{
if (buf[0] & 0x80)
{
/* If we have reached the last metadata block, abort. */
break;
}
else
{
/* Skip to next metadata block */
if (lseek(fd, i, SEEK_CUR) < 0)
{
return rc;
}
}
}
}
return true;
}
static bool get_wave_metadata(int fd, struct mp3entry* id3)
{
/* Use the trackname part of the id3 structure as a temporary buffer */
unsigned char* buf = id3->path;
unsigned long totalsamples = 0;
unsigned long channels = 0;
unsigned long bitspersample = 0;
unsigned long numbytes = 0;
int read_bytes;
int i;
if ((lseek(fd, 0, SEEK_SET) < 0)
|| ((read_bytes = read(fd, buf, sizeof(id3->path))) < 44))
{
return false;
}
if ((memcmp(buf, "RIFF",4) != 0)
|| (memcmp(&buf[8], "WAVE", 4) !=0 ))
{
return false;
}
buf += 12;
read_bytes -= 12;
while ((numbytes == 0) && (read_bytes >= 8))
{
/* chunkSize */
i = get_long(&buf[4]);
if (memcmp(buf, "fmt ", 4) == 0)
{
/* skipping wFormatTag */
/* wChannels */
channels = buf[10] | (buf[11] << 8);
/* dwSamplesPerSec */
id3->frequency = get_long(&buf[12]);
/* dwAvgBytesPerSec */
id3->bitrate = (get_long(&buf[16]) * 8) / 1000;
/* skipping wBlockAlign */
/* wBitsPerSample */
bitspersample = buf[22] | (buf[23] << 8);
}
else if (memcmp(buf, "data", 4) == 0)
{
numbytes = i;
}
else if (memcmp(buf, "fact", 4) == 0)
{
/* dwSampleLength */
if (i >= 4)
{
totalsamples = get_long(&buf[8]);
}
}
/* go to next chunk (even chunk sizes must be padded) */
if (i & 0x01)
{
i++;
}
buf += i + 8;
read_bytes -= i + 8;
}
if ((numbytes == 0) || (channels == 0))
{
return false;
}
if (totalsamples == 0)
{
/* for PCM only */
totalsamples = numbytes
/ ((((bitspersample - 1) / 8) + 1) * channels);
}
id3->vbr = false; /* All WAV files are CBR */
id3->filesize = filesize(fd);
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
id3->length = (totalsamples / id3->frequency) * 1000;
return true;
}
static bool get_m4a_metadata(int fd, struct mp3entry* id3)
{
unsigned char* buf;
unsigned long totalsamples;
int i,j,k;
size_t n;
size_t bytes_remaining;
char* id3buf;
unsigned int compressedsize;
unsigned int sample_count;
unsigned int sample_duration;
int numentries;
int entry_size;
int size_remaining;
int chunk_len;
unsigned char chunk_id[4];
int sub_chunk_len;
unsigned char sub_chunk_id[4];
/* A simple parser to read vital metadata from an ALAC file.
This parser also works for AAC files - they are both stored in
a Quicktime M4A container. */
/* Use the trackname part of the id3 structure as a temporary buffer */
buf=id3->path;
lseek(fd, 0, SEEK_SET);
totalsamples=0;
compressedsize=0;
/* read the chunks - we stop when we find the mdat chunk and set compressedsize */
while (compressedsize==0) {
n=read_uint32be(fd,&chunk_len);
// This means it was a 64-bit file, so we have problems.
if (chunk_len == 1) {
logf("need 64bit support\n");
return false;
}
n=read(fd,&chunk_id,4);
if (memcmp(&chunk_id,"ftyp",4)==0) {
/* Check for M4A type */
n=read(fd,&chunk_id,4);
if ((memcmp(&chunk_id,"M4A ",4)!=0) &&
(memcmp(&chunk_id,"mp42",4)!=0)) {
logf("Not an M4A file, aborting\n");
return false;
}
/* Skip rest of chunk */
lseek(fd, chunk_len - 8 - 4, SEEK_CUR); /* FIXME not 8 */
} else if (memcmp(&chunk_id,"moov",4)==0) {
size_remaining=chunk_len - 8; /* FIXME not 8 */
while (size_remaining > 0) {
n=read_uint32be(fd,&sub_chunk_len);
if ((sub_chunk_len < 1) || (sub_chunk_len > size_remaining)) {
logf("Strange sub_chunk_len value inside moov: %d (remaining: %d)\n",sub_chunk_len,size_remaining);
return false;
}
n=read(fd,&sub_chunk_id,4);
size_remaining-=8;
if (memcmp(&sub_chunk_id,"mvhd",4)==0) {
/* We don't need anything from here - skip */
lseek(fd, sub_chunk_len - 8, SEEK_CUR); /* FIXME not 8 */
size_remaining-=(sub_chunk_len-8);
} else if (memcmp(&sub_chunk_id,"udta",4)==0) {
/* The udta chunk contains the metadata - track, artist, album etc.
The format appears to be:
udta
meta
hdlr
ilst
.nam
[rest of tags]
free
NOTE: This code was written by examination of some .m4a files
produced by iTunes v4.9 - it may not therefore be 100%
compliant with all streams. But it should fail gracefully.
*/
j=(sub_chunk_len-8);
size_remaining-=j;
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
j-=8;
if (memcmp(&sub_chunk_id,"meta",4)==0) {
lseek(fd, 4, SEEK_CUR);
j-=4;
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
j-=8;
if (memcmp(&sub_chunk_id,"hdlr",4)==0) {
lseek(fd, sub_chunk_len - 8, SEEK_CUR);
j-=(sub_chunk_len - 8);
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
j-=8;
if (memcmp(&sub_chunk_id,"ilst",4)==0) {
/* Here are the actual tags. We use the id3v2 300-byte buffer
to store the string data */
bytes_remaining=sizeof(id3->id3v2buf);
id3->genre=255; /* Not every track is the Blues */
id3buf=id3->id3v2buf;
k=sub_chunk_len-8;
j-=k;
while (k > 0) {
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
k-=8;
if (memcmp(sub_chunk_id,"\251nam",4)==0) {
read_m4a_tag_string(fd,sub_chunk_len-8,&id3buf,&bytes_remaining,&id3->title);
} else if (memcmp(sub_chunk_id,"\251ART",4)==0) {
read_m4a_tag_string(fd,sub_chunk_len-8,&id3buf,&bytes_remaining,&id3->artist);
} else if (memcmp(sub_chunk_id,"\251alb",4)==0) {
read_m4a_tag_string(fd,sub_chunk_len-8,&id3buf,&bytes_remaining,&id3->album);
} else if (memcmp(sub_chunk_id,"\251gen",4)==0) {
read_m4a_tag_string(fd,sub_chunk_len-8,&id3buf,&bytes_remaining,&id3->genre_string);
} else if (memcmp(sub_chunk_id,"\251day",4)==0) {
read_m4a_tag_string(fd,sub_chunk_len-8,&id3buf,&bytes_remaining,&id3->year_string);
} else if (memcmp(sub_chunk_id,"trkn",4)==0) {
if (sub_chunk_len==0x20) {
read(fd,buf,sub_chunk_len-8);
id3->tracknum=buf[19];
} else {
lseek(fd, sub_chunk_len-8,SEEK_CUR);
}
} else {
lseek(fd, sub_chunk_len-8,SEEK_CUR);
}
k-=(sub_chunk_len-8);
}
}
}
}
/* Skip any remaining data in udta chunk */
lseek(fd, j, SEEK_CUR);
} else if (memcmp(&sub_chunk_id,"trak",4)==0) {
/* Format of trak chunk:
tkhd
mdia
mdhd
hdlr
minf
smhd
dinf
stbl
stsd - Samplerate, Samplesize, Numchannels
stts - time_to_sample array - RLE'd table containing duration of each block
stsz - sample_byte_size array - ?Size in bytes of each compressed block
stsc - Seek table related?
stco - Seek table related?
*/
/* Skip tkhd - not needed */
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
if (memcmp(&sub_chunk_id,"tkhd",4)!=0) {
logf("Expecting tkhd\n");
return false;
}
lseek(fd, sub_chunk_len - 8, SEEK_CUR); /* FIXME not 8 */
size_remaining-=sub_chunk_len;
/* Process mdia - skipping possible edts */
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
if (memcmp(&sub_chunk_id,"edts",4)==0) {
lseek(fd, sub_chunk_len - 8, SEEK_CUR); /* FIXME not 8 */
size_remaining-=sub_chunk_len;
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
}
if (memcmp(&sub_chunk_id,"mdia",4)!=0) {
logf("Expecting mdia\n");
return false;
}
size_remaining-=sub_chunk_len;
j=sub_chunk_len-8;
while (j > 0) {
n=read_uint32be(fd,&sub_chunk_len);
n=read(fd,&sub_chunk_id,4);
j-=4;
if (memcmp(&sub_chunk_id,"minf",4)==0) {
j=sub_chunk_len-8;
} else if (memcmp(&sub_chunk_id,"stbl",4)==0) {
j=sub_chunk_len-8;
} else if (memcmp(&sub_chunk_id,"stsd",4)==0) {
n=read(fd,buf,sub_chunk_len-8);
j-=sub_chunk_len;
i=0;
/* Skip version and flags */
i+=4;
numentries=(buf[i]<<24)|(buf[i+1]<<16)|(buf[i+2]<<8)|buf[i+3];
i+=4;
if (numentries!=1) {
logf("ERROR: Expecting only one entry in stsd\n");
}
entry_size=(buf[i]<<24)|(buf[i+1]<<16)|(buf[i+2]<<8)|buf[i+3];
i+=4;
/* Check the codec type - 'alac' for ALAC, 'mp4a' for AAC */
if ((id3->codectype==AFMT_ALAC) &&
(memcmp(&buf[i],"alac",4)!=0)) {
logf("Not an ALAC file\n");
return false;
}
if ((id3->codectype==AFMT_AAC) &&
(memcmp(&buf[i],"mp4a",4)!=0)) {
logf("Not a MP4 AAC file\n");
return false;
}
//numchannels=(buf[i+20]<<8)|buf[i+21]; /* Not used - assume Stereo */
//samplesize=(buf[i+22]<<8)|buf[i+23]; /* Not used - assume 16-bit */
/* Samplerate is 32-bit fixed point, but this works for < 65536 Hz */
id3->frequency=(buf[i+28]<<8)|buf[i+29];
} else if (memcmp(&sub_chunk_id,"stts",4)==0) {
j-=sub_chunk_len;
i=8;
n=read(fd,buf,8);
i+=8;
numentries=(buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7];
for (k=0;k<numentries;k++) {
n=read_uint32be(fd,&sample_count);
n=read_uint32be(fd,&sample_duration);
totalsamples+=sample_count*sample_duration;
i+=8;
}
if (i > 0) lseek(fd, sub_chunk_len - i, SEEK_CUR);
} else if (memcmp(&sub_chunk_id,"stsz",4)==0) {
j-=sub_chunk_len;
i=8;
n=read(fd,buf,8);
i+=8;
numentries=(buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7];
for (k=0;k<numentries;k++) {
n=read_uint32be(fd,&sample_count);
n=read_uint32be(fd,&sample_duration);
totalsamples+=sample_count*sample_duration;
i+=8;
}
if (i > 0) lseek(fd, sub_chunk_len - i, SEEK_CUR);
} else {
lseek(fd, sub_chunk_len - 8, SEEK_CUR); /* FIXME not 8 */
j-=sub_chunk_len;
}
}
} else {
logf("Unexpected sub_chunk_id inside moov: %c%c%c%c\n",
sub_chunk_id[0],sub_chunk_id[1],sub_chunk_id[2],sub_chunk_id[3]);
return false;
}
}
} else if (memcmp(&chunk_id,"mdat",4)==0) {
/* once we hit mdat we stop reading and return.
* this is on the assumption that there is no furhter interesting
* stuff in the stream. if there is stuff will fail (:()).
* But we need the read pointer to be at the mdat stuff
* for the decoder. And we don't want to rely on fseek/ftell,
* as they may not always be avilable */
lseek(fd, chunk_len - 8, SEEK_CUR); /* FIXME not 8 */
compressedsize=chunk_len-8;
} else if (memcmp(&chunk_id,"free",4)==0) {
/* these following atoms can be skipped !!!! */
lseek(fd, chunk_len - 8, SEEK_CUR); /* FIXME not 8 */
} else {
logf("(top) unknown chunk id: %c%c%c%c\n", chunk_id[0],chunk_id[1],chunk_id[2],chunk_id[3]);
return false;
}
}
id3->vbr=true; /* All ALAC files are VBR */
id3->filesize=filesize(fd);
id3->samples=totalsamples;
id3->length=(10*totalsamples)/(id3->frequency/100);
id3->bitrate=(compressedsize*8)/id3->length;;
return true;
}
static bool get_musepack_metadata(int fd, struct mp3entry *id3)
{
const int32_t sfreqs_sv7[4] = { 44100, 48000, 37800, 32000 };
uint32_t header[8];
uint64_t samples = 0;
int i;
if (!skip_id3v2(fd, id3))
return false;
if (read(fd, header, 4*8) != 4*8) return false;
/* Musepack files are little endian, might need swapping */
for (i = 1; i < 8; i++)
header[i] = letoh32(header[i]);
if (!memcmp(header, "MP+", 3)) { /* Compare to sig "MP+" */
unsigned int streamversion;
header[0] = letoh32(header[0]);
streamversion = (header[0] >> 24) & 15;
if (streamversion >= 8) {
return false; /* SV8 or higher don't exist yet, so no support */
} else if (streamversion == 7) {
unsigned int gapless = (header[5] >> 31) & 0x0001;
unsigned int last_frame_samples = (header[5] >> 20) & 0x07ff;
int track_gain, album_gain;
unsigned int bufused;
id3->frequency = sfreqs_sv7[(header[2] >> 16) & 0x0003];
samples = (uint64_t)header[1]*1152; /* 1152 is mpc frame size */
if (gapless)
samples -= 1152 - last_frame_samples;
else
samples -= 481; /* Musepack subband synth filter delay */
/* Extract ReplayGain data from header */
track_gain = (int16_t)((header[3] >> 16) & 0xffff);
id3->track_gain = get_replaygain_int(track_gain);
id3->track_peak = ((uint16_t)(header[3] & 0xffff)) << 9;
album_gain = (int16_t)((header[4] >> 16) & 0xffff);
id3->album_gain = get_replaygain_int(album_gain);
id3->album_peak = ((uint16_t)(header[4] & 0xffff)) << 9;
/* Write replaygain values to strings for use in id3 screen. We use
the XING header as buffer space since Musepack files shouldn't
need to use it in any other way */
id3->track_gain_string = id3->toc;
bufused = snprintf(id3->track_gain_string, 45,
"%d.%d dB", track_gain/100, abs(track_gain)%100);
id3->album_gain_string = id3->toc + bufused + 1;
bufused = snprintf(id3->album_gain_string, 45,
"%d.%d dB", album_gain/100, abs(album_gain)%100);
}
} else {
/* There's no certain way to detect these pre-sv7 streams, apparently */
/* TODO: add sv6 parsing here */
return false;
}
/* Estimate bitrate, we should probably subtract the various header sizes
here for super-accurate results */
id3->length = samples/id3->frequency*1000;
id3->filesize = filesize(fd);
id3->bitrate = id3->filesize*8/id3->length;
return true;
}
/* Simple file type probing by looking at the filename extension. */
unsigned int probe_file_format(const char *filename)
{
char *suffix;
unsigned int i;
suffix = strrchr(filename, '.');
if (suffix == NULL)
{
return AFMT_UNKNOWN;
}
suffix += 1;
for (i = 0; i < sizeof(formats) / sizeof(formats[0]); i++)
{
if (strcasecmp(suffix, formats[i].extension) == 0)
{
return formats[i].format;
}
}
return AFMT_UNKNOWN;
}
/* Get metadata for track - return false if parsing showed problems with the
* file that would prevent playback.
*/
bool get_metadata(struct track_info* track, int fd, const char* trackname,
bool v1first)
{
unsigned char* buf;
unsigned long totalsamples;
int i;
/* Load codec specific track tag information. */
switch (track->id3.codectype)
{
case AFMT_MPA_L1:
case AFMT_MPA_L2:
case AFMT_MPA_L3:
if (mp3info(&track->id3, trackname, v1first))
{
return false;
}
break;
case AFMT_FLAC:
if (!get_flac_metadata(fd, &(track->id3)))
{
return false;
}
break;
case AFMT_MPC:
if (!get_musepack_metadata(fd, &(track->id3)))
return false;
read_ape_tags(fd, &(track->id3));
break;
case AFMT_OGG_VORBIS:
if (!get_vorbis_metadata(fd, &(track->id3)))
{
return false;
}
break;
case AFMT_PCM_WAV:
if (!get_wave_metadata(fd, &(track->id3)))
{
return false;
}
break;
case AFMT_WAVPACK:
/* A simple parser to read basic information from a WavPack file.
* This will fail on WavPack files that don't have the WavPack header
* as the first thing (i.e. self-extracting WavPack files) or WavPack
* files that have so much extra RIFF data stored in the first block
* that they don't have samples (very rare, I would think).
*/
/* Use the trackname part of the id3 structure as a temporary buffer */
buf = track->id3.path;
if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, buf, 32) < 32))
{
return false;
}
if (memcmp (buf, "wvpk", 4) != 0 || buf [9] != 4 || buf [8] < 2)
{
logf ("%s is not a WavPack file\n", trackname);
return false;
}
track->id3.vbr = true; /* All WavPack files are VBR */
track->id3.filesize = filesize (fd);
if ((buf [20] | buf [21] | buf [22] | buf [23]) &&
(buf [12] & buf [13] & buf [14] & buf [15]) != 0xff)
{
int srindx = ((buf [26] >> 7) & 1) + ((buf [27] << 1) & 14);
if (srindx == 15)
{
track->id3.frequency = 44100;
}
else
{
track->id3.frequency = wavpack_sample_rates[srindx];
}
totalsamples = get_long(&buf[12]);
track->id3.length = totalsamples / (track->id3.frequency / 100) * 10;
track->id3.bitrate = filesize (fd) / (track->id3.length / 8);
}
read_ape_tags(fd, &track->id3); /* use any apetag info we find */
break;
case AFMT_A52:
/* Use the trackname part of the id3 structure as a temporary buffer */
buf = track->id3.path;
if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, buf, 5) < 5))
{
return false;
}
if ((buf[0] != 0x0b) || (buf[1] != 0x77))
{
logf("%s is not an A52/AC3 file\n",trackname);
return false;
}
i = buf[4] & 0x3e;
if (i > 36)
{
logf("A52: Invalid frmsizecod: %d\n",i);
return false;
}
track->id3.bitrate = a52_bitrates[i >> 1];
track->id3.vbr = false;
track->id3.filesize = filesize(fd);
switch (buf[4] & 0xc0)
{
case 0x00:
track->id3.frequency = 48000;
track->id3.bytesperframe=track->id3.bitrate * 2 * 2;
break;
case 0x40:
track->id3.frequency = 44100;
track->id3.bytesperframe = a52_441framesizes[i];
break;
case 0x80:
track->id3.frequency = 32000;
track->id3.bytesperframe = track->id3.bitrate * 3 * 2;
break;
default:
logf("A52: Invalid samplerate code: 0x%02x\n", buf[4] & 0xc0);
return false;
break;
}
/* One A52 frame contains 6 blocks, each containing 256 samples */
totalsamples = (track->filesize / track->id3.bytesperframe) * 6 * 256;
track->id3.length = (totalsamples / track->id3.frequency) * 1000;
break;
case AFMT_ALAC:
case AFMT_AAC:
if (!get_m4a_metadata(fd, &(track->id3)))
{
// return false;
}
break;
/* If we don't know how to read the metadata, just store the filename */
default:
break;
}
lseek(fd, 0, SEEK_SET);
strncpy(track->id3.path, trackname, sizeof(track->id3.path));
track->taginfo_ready = true;
return true;
}
Reference in a new issue View git blame Copy permalink