/*************************************************************************** * __________ __ ___. * 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 #include #include #include #include #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" }, #if CONFIG_CODEC == SWCODEC { 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" }, { AFMT_AIFF, "aif" }, { AFMT_AIFF, "aiff" }, { AFMT_SID, "sid" }, { AFMT_ADX, "adx" }, #endif }; #if CONFIG_CODEC == SWCODEC 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); } /* 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; 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; } 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; } 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, r; 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. */ /* A page is always < 64 kB */ if (lseek(fd, -(MIN(64 * 1024, id3->filesize)), SEEK_END) < 0) { return false; } remaining = 0; while (!eof) { 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 - 3)) { 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]); /* If this page is very small the beginning of the next * header could be in buffer. Jump near end of this header * and continue */ i += 27; } else { break; } } else { i++; } } if (i < remaining) { /* Move the remaining bytes to start of buffer. * Reuse var 'segments' as it is no longer needed */ segments = 0; while (i < remaining) { buf[segments++] = buf[i++]; } remaining = segments; } else { /* Discard the rest of the buffer */ 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; /* get RIFF chunk header */ if ((lseek(fd, 0, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, 12)) < 12)) { return false; } if ((memcmp(buf, "RIFF",4) != 0) || (memcmp(&buf[8], "WAVE", 4) !=0 )) { return false; } /* iterate over WAVE chunks until 'data' chunk */ while (true) { /* get chunk header */ if ((read_bytes = read(fd, buf, 8)) < 8) return false; /* chunkSize */ i = get_long(&buf[4]); if (memcmp(buf, "fmt ", 4) == 0) { /* get rest of chunk */ if ((read_bytes = read(fd, buf, 16)) < 16) return false; i -= 16; /* skipping wFormatTag */ /* wChannels */ channels = buf[2] | (buf[3] << 8); /* dwSamplesPerSec */ id3->frequency = get_long(&buf[4]); /* dwAvgBytesPerSec */ id3->bitrate = (get_long(&buf[8]) * 8) / 1000; /* skipping wBlockAlign */ /* wBitsPerSample */ bitspersample = buf[14] | (buf[15] << 8); } else if (memcmp(buf, "data", 4) == 0) { numbytes = i; break; } else if (memcmp(buf, "fact", 4) == 0) { /* dwSampleLength */ if (i >= 4) { /* get rest of chunk */ if ((read_bytes = read(fd, buf, 2)) < 2) return false; i -= 2; totalsamples = get_long(buf); } } /* seek to next chunk (even chunk sizes must be padded) */ if (i & 0x01) i++; if(lseek(fd, i, SEEK_CUR) < 0) return false; } 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 (n < 4) return false; 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; if (memcmp(&buf[i],"alac",4)==0) { id3->codectype=AFMT_ALAC; } else if (memcmp(&buf[i],"mp4a",4)==0) { id3->codectype=AFMT_AAC; } else { logf("Not an ALAC or 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 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 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 { header[0] = letoh32(header[0]); unsigned int streamversion = (header[0] >> 11) & 0x03FF; if (streamversion != 4 && streamversion != 5 && streamversion != 6) return false; id3->frequency = 44100; id3->track_gain = 0; id3->track_peak = 0; id3->album_gain = 0; id3->album_peak = 0; if (streamversion >= 5) samples = (uint64_t)header[1]*1152; // 32 bit else samples = (uint64_t)(header[1] >> 16)*1152; // 16 bit samples -= 576; if (streamversion < 6) samples -= 1152; } id3->vbr = true; /* 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; } static bool get_sid_metadata(int fd, struct mp3entry* id3) { /* Use the trackname part of the id3 structure as a temporary buffer */ unsigned char* buf = id3->path; int read_bytes; char *p; if ((lseek(fd, 0, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, sizeof(id3->path))) < 44)) { return false; } if ((memcmp(buf, "PSID",4) != 0)) { return false; } p = id3->id3v2buf; /* Copy Title */ strcpy(p, &buf[0x16]); id3->title = p; p += strlen(p)+1; /* Copy Artist */ strcpy(p, &buf[0x36]); id3->artist = p; p += strlen(p)+1; id3->bitrate = 706; id3->frequency = 44100; /* New idea as posted by Marco Alanen (ravon): * Set the songlength in seconds to the number of subsongs * so every second represents a subsong. * Users can then skip the current subsong by seeking */ id3->length = (buf[0xf]-1)*1000; id3->vbr = false; id3->filesize = filesize(fd); return true; } static bool get_adx_metadata(int fd, struct mp3entry* id3) { /* Use the trackname part of the id3 structure as a temporary buffer */ unsigned char * buf = id3->path; int chanstart, channels, read_bytes; int looping = 0, start_adr = 0, end_adr = 0; /* try to get the basic header */ if ((lseek(fd, 0, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, 0x38)) < 0x38)) { DEBUGF("lseek or read failed\n"); return false; } /* ADX starts with 0x80 */ if (buf[0] != 0x80) { DEBUGF("get_adx_metadata: wrong first byte %c\n",buf[0]); return false; } /* check for a reasonable offset */ chanstart = ((buf[2] << 8) | buf[3]) + 4; if (chanstart > 4096) { DEBUGF("get_adx_metadata: bad chanstart %i\n", chanstart); return false; } /* check for a workable number of channels */ channels = buf[7]; if (channels != 1 && channels != 2) { DEBUGF("get_adx_metadata: bad channel count %i\n",channels); return false; } id3->frequency = (buf[8] << 24) | (buf[9] << 16) | (buf[10] << 8) | buf[11]; /* 32 samples per 18 bytes */ id3->bitrate = id3->frequency * channels * 18 * 8 / 32 / 1000; id3->length = ((unsigned long) (buf[12] << 24) | (buf[13] << 16) | (buf[14] << 8) | buf[15]) / id3->frequency * 1000; id3->vbr = false; id3->filesize = filesize(fd); /* get loop info */ if (!memcmp(buf+0x10,"\x01\xF4\x03\x00",4)) { /* Soul Calibur 2 style (type 03) */ DEBUGF("get_adx_metadata: type 03 found\n"); /* check if header is too small for loop data */ if (chanstart-6 < 0x2c) looping=0; else { looping = (buf[0x18]) || (buf[0x19]) || (buf[0x1a]) || (buf[0x1b]); end_adr = (buf[0x28]<<24) | (buf[0x29]<<16) | (buf[0x2a]<<8) | (buf[0x2b]); start_adr = ( (buf[0x1c]<<24) | (buf[0x1d]<<16) | (buf[0x1e]<<8) | (buf[0x1f]) )/32*channels*18+chanstart; } } else if (!memcmp(buf+0x10,"\x01\xF4\x04\x00",4)) { /* Standard (type 04) */ DEBUGF("get_adx_metadata: type 04 found\n"); /* check if header is too small for loop data */ if (chanstart-6 < 0x38) looping=0; else { looping = (buf[0x24]) || (buf[0x25]) || (buf[0x26]) || (buf[0x27]); end_adr = (buf[0x34]<<24) | (buf[0x35]<<16) | (buf[0x36]<<8) | buf[0x37]; start_adr = ( (buf[0x28]<<24) | (buf[0x29]<<16) | (buf[0x2a]<<8) | (buf[0x2b]) )/32*channels*18+chanstart; } } else { DEBUGF("get_adx_metadata: error, couldn't determine ADX type\n"); return false; } if (looping) { /* 2 loops, 10 second fade */ id3->length = (start_adr-chanstart + 2*(end_adr-start_adr)) *8 / id3->bitrate + 10000; } /* try to get the channel header */ if ((lseek(fd, chanstart-6, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, 6)) < 6)) { return false; } /* check channel header */ if (memcmp(buf, "(c)CRI", 6) != 0) return false; return true; } #endif /* CONFIG_CODEC == SWCODEC */ static bool get_aiff_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 numChannels = 0; unsigned long numSampleFrames = 0; unsigned long sampleSize = 0; unsigned long sampleRate = 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, "FORM",4) != 0) || (memcmp(&buf[8], "AIFF", 4) !=0 )) { return false; } buf += 12; read_bytes -= 12; while ((numbytes == 0) && (read_bytes >= 8)) { /* chunkSize */ i = ((buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7]); if (memcmp(buf, "COMM", 4) == 0) { /* numChannels */ numChannels = ((buf[8]<<8)|buf[9]); /* numSampleFrames */ numSampleFrames =((buf[10]<<24)|(buf[11]<<16)|(buf[12]<<8)|buf[13]); /* sampleSize */ sampleSize = ((buf[14]<<8)|buf[15]); /* sampleRate */ sampleRate = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21]); sampleRate = sampleRate >> (16+14-buf[17]); /* save format infos */ id3->bitrate = (sampleSize * numChannels * sampleRate) / 1000; id3->frequency = sampleRate; id3->length = (numSampleFrames / id3->frequency) * 1000; id3->vbr = false; /* AIFF files are CBR */ id3->filesize = filesize(fd); } else if (memcmp(buf, "SSND", 4) == 0) { numbytes = i - 8; } if (i & 0x01) { i++; /* odd chunk sizes must be padded */ } buf += i + 8; read_bytes -= i + 8; } if ((numbytes == 0) || (numChannels == 0)) { return false; } 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) { #if CONFIG_CODEC == SWCODEC unsigned char* buf; unsigned long totalsamples; int i; #endif /* Take our best guess at the codec type based on file extension */ track->id3.codectype = probe_file_format(trackname); /* Load codec specific track tag information and confirm the codec type. */ 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; #if CONFIG_CODEC == SWCODEC 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 * now works with self-extrating WavPack files and also will fail on * WavPack files containing floating-point audio data (although these * should be possible to play in theory). */ /* Use the trackname part of the id3 structure as a temporary buffer */ buf = track->id3.path; for (i = 0; i < 256; ++i) { /* at every 256 bytes into file, try to read a WavPack header */ if ((lseek(fd, i * 256, SEEK_SET) < 0) || (read(fd, buf, 32) < 32)) { return false; } /* if valid WavPack 4 header version & not floating data, break */ if (memcmp (buf, "wvpk", 4) == 0 && buf [9] == 4 && (buf [8] >= 2 && buf [8] <= 0x10) && !(buf [24] & 0x80)) { break; } } if (i == 256) { 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->id3.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; case AFMT_SHN: track->id3.vbr = true; track->id3.filesize = filesize(fd); if (!skip_id3v2(fd, &(track->id3))) { return false; } /* TODO: read the id3v2 header if it exists */ break; case AFMT_SID: if (!get_sid_metadata(fd, &(track->id3))) { return false; } break; case AFMT_ADX: if (!get_adx_metadata(fd, &(track->id3))) { DEBUGF("get_adx_metadata error\n"); return false; } break; #endif /* CONFIG_CODEC == SWCODEC */ case AFMT_AIFF: if (!get_aiff_metadata(fd, &(track->id3))) { return false; } break; default: /* If we don't know how to read the metadata, assume we can't play the file */ return false; break; } /* We have successfully read the metadata from the file */ lseek(fd, 0, SEEK_SET); strncpy(track->id3.path, trackname, sizeof(track->id3.path)); track->taginfo_ready = true; return true; }