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authorBjörn Stenberg <bjorn@haxx.se>2008-10-10 13:12:28 +0000
committerBjörn Stenberg <bjorn@haxx.se>2008-10-10 13:12:28 +0000
commite76c69f3e4b9075db979145a60157d8cd968f537 (patch)
tree84ac8d57c687d1fdd10d8f205aa83a048edfd875 /apps
parent98fa3913f9618a09269e9ab39abb9a53274f5676 (diff)
downloadrockbox-e76c69f3e4b9075db979145a60157d8cd968f537.tar.gz
rockbox-e76c69f3e4b9075db979145a60157d8cd968f537.tar.bz2
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Moved id3.c, mpeg.c and replaygain.c from firmware/ to apps/. This is the first step in separating the generic metadata code and the id3-specific code.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18760 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps')
-rw-r--r--apps/SOURCES4
-rw-r--r--apps/id3.c1353
-rw-r--r--apps/mpeg.c2873
-rw-r--r--apps/replaygain.c457
4 files changed, 4687 insertions, 0 deletions
diff --git a/apps/SOURCES b/apps/SOURCES
index eb09797719..7622b416f7 100644
--- a/apps/SOURCES
+++ b/apps/SOURCES
@@ -7,6 +7,7 @@ abrepeat.c
bookmark.c
debug_menu.c
filetypes.c
+id3.c
language.c
main.c
menu.c
@@ -17,6 +18,9 @@ menus/theme_menu.c
menus/eq_menu.c
buffering.c
voice_thread.c
+replaygain.c
+#else /* !SWCODEC */
+mpeg.c
#endif
menus/main_menu.c
menus/playback_menu.c
diff --git a/apps/id3.c b/apps/id3.c
new file mode 100644
index 0000000000..c1541e30df
--- /dev/null
+++ b/apps/id3.c
@@ -0,0 +1,1353 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2002 by Daniel Stenberg
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+/*
+ * Parts of this code has been stolen from the Ample project and was written
+ * by David H�deman. It has since been extended and enhanced pretty much by
+ * all sorts of friendly Rockbox people.
+ *
+ */
+
+ /* tagResolver and associated code copyright 2003 Thomas Paul Diffenbach
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+#include <stdbool.h>
+#include <stddef.h>
+#include <ctype.h>
+#include "config.h"
+#include "file.h"
+#include "logf.h"
+
+#include "id3.h"
+#include "mp3data.h"
+#include "system.h"
+#include "replaygain.h"
+#include "rbunicode.h"
+
+/** Database of audio formats **/
+const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
+{
+ /* Unknown file format */
+ [AFMT_UNKNOWN] =
+ AFMT_ENTRY("???", NULL, NULL, NULL ),
+
+ /* MPEG Audio layer 1 */
+ [AFMT_MPA_L1] =
+ AFMT_ENTRY("MP1", "mpa", NULL, "mp1\0" ),
+ /* MPEG Audio layer 2 */
+ [AFMT_MPA_L2] =
+ AFMT_ENTRY("MP2", "mpa", NULL, "mpa\0mp2\0" ),
+ /* MPEG Audio layer 3 */
+ [AFMT_MPA_L3] =
+ AFMT_ENTRY("MP3", "mpa", "mp3_enc", "mp3\0" ),
+
+#if CONFIG_CODEC == SWCODEC
+ /* Audio Interchange File Format */
+ [AFMT_AIFF] =
+ AFMT_ENTRY("AIFF", "aiff", "aiff_enc", "aiff\0aif\0"),
+ /* Uncompressed PCM in a WAV file */
+ [AFMT_PCM_WAV] =
+ AFMT_ENTRY("WAV", "wav", "wav_enc", "wav\0" ),
+ /* Ogg Vorbis */
+ [AFMT_OGG_VORBIS] =
+ AFMT_ENTRY("Ogg", "vorbis", NULL, "ogg\0" ),
+ /* FLAC */
+ [AFMT_FLAC] =
+ AFMT_ENTRY("FLAC", "flac", NULL, "flac\0" ),
+ /* Musepack */
+ [AFMT_MPC] =
+ AFMT_ENTRY("MPC", "mpc", NULL, "mpc\0" ),
+ /* A/52 (aka AC3) audio */
+ [AFMT_A52] =
+ AFMT_ENTRY("AC3", "a52", NULL, "a52\0ac3\0" ),
+ /* WavPack */
+ [AFMT_WAVPACK] =
+ AFMT_ENTRY("WV", "wavpack", "wavpack_enc", "wv\0" ),
+ /* Apple Lossless Audio Codec */
+ [AFMT_ALAC] =
+ AFMT_ENTRY("ALAC", "alac", NULL, "m4a\0m4b\0" ),
+ /* Advanced Audio Coding in M4A container */
+ [AFMT_AAC] =
+ AFMT_ENTRY("AAC", "aac", NULL, "mp4\0" ),
+ /* Shorten */
+ [AFMT_SHN] =
+ AFMT_ENTRY("SHN", "shorten", NULL, "shn\0" ),
+ /* SID File Format */
+ [AFMT_SID] =
+ AFMT_ENTRY("SID", "sid", NULL, "sid\0" ),
+ /* ADX File Format */
+ [AFMT_ADX] =
+ AFMT_ENTRY("ADX", "adx", NULL, "adx\0" ),
+ /* NESM (NES Sound Format) */
+ [AFMT_NSF] =
+ AFMT_ENTRY("NSF", "nsf", NULL, "nsf\0nsfe\0" ),
+ /* Speex File Format */
+ [AFMT_SPEEX] =
+ AFMT_ENTRY("Speex","speex", NULL, "spx\0" ),
+ /* SPC700 Save State */
+ [AFMT_SPC] =
+ AFMT_ENTRY("SPC", "spc", NULL, "spc\0" ),
+ /* APE (Monkey's Audio) */
+ [AFMT_APE] =
+ AFMT_ENTRY("APE", "ape", NULL, "ape\0mac\0" ),
+ /* WMA (WMAV1/V2 in ASF) */
+ [AFMT_WMA] =
+ AFMT_ENTRY("WMA", "wma", NULL, "wma\0wmv\0asf\0" ),
+ /* Amiga MOD File */
+ [AFMT_MOD] =
+ AFMT_ENTRY("MOD", "mod", NULL, "mod\0" ),
+ /* Amiga SAP File */
+ [AFMT_SAP] =
+ AFMT_ENTRY("SAP", "asap", NULL, "sap\0" ),
+#endif
+};
+
+#if CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING)
+/* get REC_FORMAT_* corresponding AFMT_* */
+const int rec_format_afmt[REC_NUM_FORMATS] =
+{
+ /* give AFMT_UNKNOWN by default */
+ [0 ... REC_NUM_FORMATS-1] = AFMT_UNKNOWN,
+ /* add new entries below this line */
+ [REC_FORMAT_AIFF] = AFMT_AIFF,
+ [REC_FORMAT_MPA_L3] = AFMT_MPA_L3,
+ [REC_FORMAT_WAVPACK] = AFMT_WAVPACK,
+ [REC_FORMAT_PCM_WAV] = AFMT_PCM_WAV,
+};
+
+/* get AFMT_* corresponding REC_FORMAT_* */
+const int afmt_rec_format[AFMT_NUM_CODECS] =
+{
+ /* give -1 by default */
+ [0 ... AFMT_NUM_CODECS-1] = -1,
+ /* add new entries below this line */
+ [AFMT_AIFF] = REC_FORMAT_AIFF,
+ [AFMT_MPA_L3] = REC_FORMAT_MPA_L3,
+ [AFMT_WAVPACK] = REC_FORMAT_WAVPACK,
+ [AFMT_PCM_WAV] = REC_FORMAT_PCM_WAV,
+};
+#endif /* CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING) */
+/****/
+
+static unsigned long unsync(unsigned long b0,
+ unsigned long b1,
+ unsigned long b2,
+ unsigned long b3)
+{
+ return (((long)(b0 & 0x7F) << (3*7)) |
+ ((long)(b1 & 0x7F) << (2*7)) |
+ ((long)(b2 & 0x7F) << (1*7)) |
+ ((long)(b3 & 0x7F) << (0*7)));
+}
+
+static const char* const genres[] = {
+ "Blues", "Classic Rock", "Country", "Dance", "Disco", "Funk", "Grunge",
+ "Hip-Hop", "Jazz", "Metal", "New Age", "Oldies", "Other", "Pop", "R&B",
+ "Rap", "Reggae", "Rock", "Techno", "Industrial", "Alternative", "Ska",
+ "Death Metal", "Pranks", "Soundtrack", "Euro-Techno", "Ambient", "Trip-Hop",
+ "Vocal", "Jazz+Funk", "Fusion", "Trance", "Classical", "Instrumental",
+ "Acid", "House", "Game", "Sound Clip", "Gospel", "Noise", "AlternRock",
+ "Bass", "Soul", "Punk", "Space", "Meditative", "Instrumental Pop",
+ "Instrumental Rock", "Ethnic", "Gothic", "Darkwave", "Techno-Industrial",
+ "Electronic", "Pop-Folk", "Eurodance", "Dream", "Southern Rock", "Comedy",
+ "Cult", "Gangsta", "Top 40", "Christian Rap", "Pop/Funk", "Jungle",
+ "Native American", "Cabaret", "New Wave", "Psychadelic", "Rave",
+ "Showtunes", "Trailer", "Lo-Fi", "Tribal", "Acid Punk", "Acid Jazz",
+ "Polka", "Retro", "Musical", "Rock & Roll", "Hard Rock",
+
+ /* winamp extensions */
+ "Folk", "Folk-Rock", "National Folk", "Swing", "Fast Fusion", "Bebob",
+ "Latin", "Revival", "Celtic", "Bluegrass", "Avantgarde", "Gothic Rock",
+ "Progressive Rock", "Psychedelic Rock", "Symphonic Rock", "Slow Rock",
+ "Big Band", "Chorus", "Easy Listening", "Acoustic", "Humour", "Speech",
+ "Chanson", "Opera", "Chamber Music", "Sonata", "Symphony", "Booty Bass",
+ "Primus", "Porn Groove", "Satire", "Slow Jam", "Club", "Tango", "Samba",
+ "Folklore", "Ballad", "Power Ballad", "Rhythmic Soul", "Freestyle",
+ "Duet", "Punk Rock", "Drum Solo", "A capella", "Euro-House", "Dance Hall",
+ "Goa", "Drum & Bass", "Club-House", "Hardcore", "Terror", "Indie",
+ "BritPop", "Negerpunk", "Polsk Punk", "Beat", "Christian Gangsta Rap",
+ "Heavy Metal", "Black Metal", "Crossover", "Contemporary Christian",
+ "Christian Rock", "Merengue", "Salsa", "Thrash Metal", "Anime", "Jpop",
+ "Synthpop"
+};
+
+char* id3_get_num_genre(unsigned int genre_num)
+{
+ if (genre_num < sizeof(genres)/sizeof(char*))
+ return (char*)genres[genre_num];
+ return NULL;
+}
+
+/* True if the string is from the "genres" array */
+static bool id3_is_genre_string(const char *string)
+{
+ return ( string >= genres[0] &&
+ string <= genres[sizeof(genres)/sizeof(char*) - 1] );
+}
+
+/*
+ HOW TO ADD ADDITIONAL ID3 VERSION 2 TAGS
+ Code and comments by Thomas Paul Diffenbach
+
+ To add another ID3v2 Tag, do the following:
+ 1. add a char* named for the tag to struct mp3entry in id3.h,
+ (I (tpd) prefer to use char* rather than ints, even for what seems like
+ numerical values, for cases where a number won't do, e.g.,
+ YEAR: "circa 1765", "1790/1977" (composed/performed), "28 Feb 1969"
+ TRACK: "1/12", "1 of 12", GENRE: "Freeform genre name"
+ Text is more flexible, and as the main use of id3 data is to
+ display it, converting it to an int just means reconverting to
+ display it, at a runtime cost.)
+
+ 2. If any special processing beyond copying the tag value from the Id3
+ block to the struct mp3entry is rrequired (such as converting to an
+ int), write a function to perform this special processing.
+
+ This function's prototype must match that of
+ typedef tagPostProcessFunc, that is it must be:
+ int func( struct mp3entry*, char* tag, int bufferpos )
+ the first argument is a pointer to the current mp3entry structure the
+ second argument is a pointer to the null terminated string value of the
+ tag found the third argument is the offset of the next free byte in the
+ mp3entry's buffer your function should return the corrected offset; if
+ you don't lengthen or shorten the tag string, you can return the third
+ argument unchanged.
+
+ Unless you have a good reason no to, make the function static.
+ TO JUST COPY THE TAG NO SPECIAL PROCESSING FUNCTION IS NEEDED.
+
+ 3. add one or more entries to the tagList array, using the format:
+ char* ID3 Tag symbolic name -- see the ID3 specification for these,
+ sizeof() that name minus 1,
+ offsetof( struct mp3entry, variable_name_in_struct_mp3entry ),
+ pointer to your special processing function or NULL
+ if you need no special processing
+ flag indicating if this tag is binary or textual
+ Many ID3 symbolic names come in more than one form. You can add both
+ forms, each referencing the same variable in struct mp3entry.
+ If both forms are present, the last found will be used.
+ Note that the offset can be zero, in which case no entry will be set
+ in the mp3entry struct; the frame is still read into the buffer and
+ the special processing function is called (several times, if there
+ are several frames with the same name).
+
+ 4. Alternately, use the TAG_LIST_ENTRY macro with
+ ID3 tag symbolic name,
+ variable in struct mp3entry,
+ special processing function address
+
+ 5. Add code to wps-display.c function get_tag to assign a printf-like
+ format specifier for the tag */
+
+/* Structure for ID3 Tag extraction information */
+struct tag_resolver {
+ const char* tag;
+ int tag_length;
+ size_t offset;
+ int (*ppFunc)(struct mp3entry*, char* tag, int bufferpos);
+ bool binary;
+};
+
+static bool global_ff_found;
+
+static int unsynchronize(char* tag, int len, bool *ff_found)
+{
+ int i;
+ unsigned char c;
+ unsigned char *rp, *wp;
+
+ wp = rp = (unsigned char *)tag;
+
+ rp = (unsigned char *)tag;
+ for(i = 0;i < len;i++) {
+ /* Read the next byte and write it back, but don't increment the
+ write pointer */
+ c = *rp++;
+ *wp = c;
+ if(*ff_found) {
+ /* Increment the write pointer if it isn't an unsynch pattern */
+ if(c != 0)
+ wp++;
+ *ff_found = false;
+ } else {
+ if(c == 0xff)
+ *ff_found = true;
+ wp++;
+ }
+ }
+ return (long)wp - (long)tag;
+}
+
+static int unsynchronize_frame(char* tag, int len)
+{
+ bool ff_found = false;
+
+ return unsynchronize(tag, len, &ff_found);
+}
+
+static int read_unsynched(int fd, void *buf, int len)
+{
+ int i;
+ int rc;
+ int remaining = len;
+ char *wp;
+ char *rp;
+
+ wp = buf;
+
+ while(remaining) {
+ rp = wp;
+ rc = read(fd, rp, remaining);
+ if(rc <= 0)
+ return rc;
+
+ i = unsynchronize(wp, remaining, &global_ff_found);
+ remaining -= i;
+ wp += i;
+ }
+
+ return len;
+}
+
+static int skip_unsynched(int fd, int len)
+{
+ int rc;
+ int remaining = len;
+ int rlen;
+ char buf[32];
+
+ while(remaining) {
+ rlen = MIN(sizeof(buf), (unsigned int)remaining);
+ rc = read(fd, buf, rlen);
+ if(rc <= 0)
+ return rc;
+
+ remaining -= unsynchronize(buf, rlen, &global_ff_found);
+ }
+
+ return len;
+}
+
+/* parse numeric value from string */
+static int parsetracknum( struct mp3entry* entry, char* tag, int bufferpos )
+{
+ entry->tracknum = atoi( tag );
+ return bufferpos;
+}
+
+/* parse numeric value from string */
+static int parsediscnum( struct mp3entry* entry, char* tag, int bufferpos )
+{
+ entry->discnum = atoi( tag );
+ return bufferpos;
+}
+
+/* parse numeric value from string */
+static int parseyearnum( struct mp3entry* entry, char* tag, int bufferpos )
+{
+ entry->year = atoi( tag );
+ return bufferpos;
+}
+
+/* parse numeric genre from string, version 2.2 and 2.3 */
+static int parsegenre( struct mp3entry* entry, char* tag, int bufferpos )
+{
+ if(entry->id3version >= ID3_VER_2_4) {
+ /* In version 2.4 and up, there are no parentheses, and the genre frame
+ is a list of strings, either numbers or text. */
+
+ /* Is it a number? */
+ if(isdigit(tag[0])) {
+ entry->genre_string = id3_get_num_genre(atoi( tag ));
+ return tag - entry->id3v2buf;
+ } else {
+ entry->genre_string = tag;
+ return bufferpos;
+ }
+ } else {
+ if( tag[0] == '(' && tag[1] != '(' ) {
+ entry->genre_string = id3_get_num_genre(atoi( tag + 1 ));
+ return tag - entry->id3v2buf;
+ }
+ else {
+ entry->genre_string = tag;
+ return bufferpos;
+ }
+ }
+}
+
+#if CONFIG_CODEC == SWCODEC
+/* parse user defined text, looking for replaygain information. */
+static int parseuser( struct mp3entry* entry, char* tag, int bufferpos )
+{
+ char* value = NULL;
+ int desc_len = strlen(tag);
+ int value_len = 0;
+
+ if ((tag - entry->id3v2buf + desc_len + 2) < bufferpos) {
+ /* At least part of the value was read, so we can safely try to
+ * parse it
+ */
+ value = tag + desc_len + 1;
+ value_len = parse_replaygain(tag, value, entry, tag,
+ bufferpos - (tag - entry->id3v2buf));
+ }
+
+ return tag - entry->id3v2buf + value_len;
+}
+
+/* parse RVA2 binary data and convert to replaygain information. */
+static int parserva2( struct mp3entry* entry, char* tag, int bufferpos )
+{
+ int desc_len = strlen(tag);
+ int start_pos = tag - entry->id3v2buf;
+ int end_pos = start_pos + desc_len + 5;
+ int value_len = 0;
+ unsigned char* value = tag + desc_len + 1;
+
+ /* Only parse RVA2 replaygain tags if tag version == 2.4 and channel
+ * type is master volume.
+ */
+ if (entry->id3version == ID3_VER_2_4 && end_pos < bufferpos
+ && *value++ == 1) {
+ long gain = 0;
+ long peak = 0;
+ long peakbits;
+ long peakbytes;
+ bool album = false;
+
+ /* The RVA2 specification is unclear on some things (id string and
+ * peak volume), but this matches how Quod Libet use them.
+ */
+
+ gain = (int16_t) ((value[0] << 8) | value[1]);
+ value += 2;
+ peakbits = *value++;
+ peakbytes = (peakbits + 7) / 8;
+
+ /* Only use the topmost 24 bits for peak volume */
+ if (peakbytes > 3) {
+ peakbytes = 3;
+ }
+
+ /* Make sure the peak bits were read */
+ if (end_pos + peakbytes < bufferpos) {
+ long shift = ((8 - (peakbits & 7)) & 7) + (3 - peakbytes) * 8;
+
+ for ( ; peakbytes; peakbytes--) {
+ peak <<= 8;
+ peak += *value++;
+ }
+
+ peak <<= shift;
+
+ if (peakbits > 24) {
+ peak += *value >> (8 - shift);
+ }
+ }
+
+ if (strcasecmp(tag, "album") == 0) {
+ album = true;
+ } else if (strcasecmp(tag, "track") != 0) {
+ /* Only accept non-track values if we don't have any previous
+ * value.
+ */
+ if (entry->track_gain != 0) {
+ return start_pos;
+ }
+ }
+
+ value_len = parse_replaygain_int(album, gain, peak * 2, entry,
+ tag, sizeof(entry->id3v2buf) - start_pos);
+ }
+
+ return start_pos + value_len;
+}
+#endif
+
+static int parsembtid( struct mp3entry* entry, char* tag, int bufferpos )
+{
+ char* value = NULL;
+ int desc_len = strlen(tag);
+ /*DEBUGF("MBID len: %d\n", desc_len);*/
+ int value_len = 0;
+
+ if ((tag - entry->id3v2buf + desc_len + 2) < bufferpos)
+ {
+ value = tag + desc_len + 1;
+
+ if (strcasecmp(tag, "http://musicbrainz.org") == 0)
+ {
+ /* Musicbrainz track IDs are always 36 chars long plus null */
+ value_len = 37;
+
+ entry->mb_track_id = value;
+
+ /*DEBUGF("ENTRY: %s LEN: %d\n", entry->mb_track_id, strlen(entry->mb_track_id));*/
+ }
+ }
+
+ return tag - entry->id3v2buf + value_len;
+}
+
+static const struct tag_resolver taglist[] = {
+ { "TPE1", 4, offsetof(struct mp3entry, artist), NULL, false },
+ { "TP1", 3, offsetof(struct mp3entry, artist), NULL, false },
+ { "TIT2", 4, offsetof(struct mp3entry, title), NULL, false },
+ { "TT2", 3, offsetof(struct mp3entry, title), NULL, false },
+ { "TALB", 4, offsetof(struct mp3entry, album), NULL, false },
+ { "TAL", 3, offsetof(struct mp3entry, album), NULL, false },
+ { "TRK", 3, offsetof(struct mp3entry, track_string), &parsetracknum, false },
+ { "TPOS", 4, offsetof(struct mp3entry, disc_string), &parsediscnum, false },
+ { "TRCK", 4, offsetof(struct mp3entry, track_string), &parsetracknum, false },
+ { "TDRC", 4, offsetof(struct mp3entry, year_string), &parseyearnum, false },
+ { "TYER", 4, offsetof(struct mp3entry, year_string), &parseyearnum, false },
+ { "TYE", 3, offsetof(struct mp3entry, year_string), &parseyearnum, false },
+ { "TCOM", 4, offsetof(struct mp3entry, composer), NULL, false },
+ { "TPE2", 4, offsetof(struct mp3entry, albumartist), NULL, false },
+ { "TP2", 3, offsetof(struct mp3entry, albumartist), NULL, false },
+ { "TIT1", 4, offsetof(struct mp3entry, grouping), NULL, false },
+ { "TT1", 3, offsetof(struct mp3entry, grouping), NULL, false },
+ { "COMM", 4, offsetof(struct mp3entry, comment), NULL, false },
+ { "TCON", 4, offsetof(struct mp3entry, genre_string), &parsegenre, false },
+ { "TCO", 3, offsetof(struct mp3entry, genre_string), &parsegenre, false },
+#if CONFIG_CODEC == SWCODEC
+ { "TXXX", 4, 0, &parseuser, false },
+ { "RVA2", 4, 0, &parserva2, true },
+#endif
+ { "UFID", 4, 0, &parsembtid, false },
+};
+
+#define TAGLIST_SIZE ((int)(sizeof(taglist) / sizeof(taglist[0])))
+
+/* Get the length of an ID3 string in the given encoding. Returns the length
+ * in bytes, including end nil, or -1 if the encoding is unknown.
+ */
+static int unicode_len(char encoding, const void* string)
+{
+ int len = 0;
+
+ if (encoding == 0x01 || encoding == 0x02) {
+ char first;
+ const char *s = string;
+ /* string might be unaligned, so using short* can crash on ARM and SH1 */
+ do {
+ first = *s++;
+ } while ((first | *s++) != 0);
+
+ len = s - (const char*) string;
+ } else {
+ len = strlen((char*) string) + 1;
+ }
+
+ return len;
+}
+
+/* Checks to see if the passed in string is a 16-bit wide Unicode v2
+ string. If it is, we convert it to a UTF-8 string. If it's not unicode,
+ we convert from the default codepage */
+static int unicode_munge(char* string, char* utf8buf, int *len) {
+ long tmp;
+ bool le = false;
+ int i = 0;
+ unsigned char *str = (unsigned char *)string;
+ int templen = 0;
+ unsigned char* utf8 = (unsigned char *)utf8buf;
+
+ switch (str[0]) {
+ case 0x00: /* Type 0x00 is ordinary ISO 8859-1 */
+ str++;
+ (*len)--;
+ utf8 = iso_decode(str, utf8, -1, *len);
+ *utf8 = 0;
+ *len = (unsigned long)utf8 - (unsigned long)utf8buf;
+ break;
+
+ case 0x01: /* Unicode with or without BOM */
+ case 0x02:
+ (*len)--;
+ str++;
+
+ /* Handle frames with more than one string
+ (needed for TXXX frames).*/
+ do {
+ tmp = bytes2int(0, 0, str[0], str[1]);
+
+ /* Now check if there is a BOM
+ (zero-width non-breaking space, 0xfeff)
+ and if it is in little or big endian format */
+ if(tmp == 0xfffe) { /* Little endian? */
+ le = true;
+ str += 2;
+ (*len)-=2;
+ } else if(tmp == 0xfeff) { /* Big endian? */
+ str += 2;
+ (*len)-=2;
+ } else
+ /* If there is no BOM (which is a specification violation),
+ let's try to guess it. If one of the bytes is 0x00, it is
+ probably the most significant one. */
+ if(str[1] == 0)
+ le = true;
+
+ do {
+ if(le)
+ utf8 = utf16LEdecode(str, utf8, 1);
+ else
+ utf8 = utf16BEdecode(str, utf8, 1);
+
+ str+=2;
+ i += 2;
+ } while((str[0] || str[1]) && (i < *len));
+
+ *utf8++ = 0; /* Terminate the string */
+ templen += (strlen(&utf8buf[templen]) + 1);
+ str += 2;
+ i+=2;
+ } while(i < *len);
+ *len = templen - 1;
+ break;
+
+ case 0x03: /* UTF-8 encoded string */
+ for(i=0; i < *len; i++)
+ utf8[i] = str[i+1];
+ (*len)--;
+ break;
+
+ default: /* Plain old string */
+ utf8 = iso_decode(str, utf8, -1, *len);
+ *utf8 = 0;
+ *len = (unsigned long)utf8 - (unsigned long)utf8buf;
+ break;
+ }
+ return 0;
+}
+
+/*
+ * Sets the title of an MP3 entry based on its ID3v1 tag.
+ *
+ * Arguments: file - the MP3 file to scen for a ID3v1 tag
+ * entry - the entry to set the title in
+ *
+ * Returns: true if a title was found and created, else false
+ */
+static bool setid3v1title(int fd, struct mp3entry *entry)
+{
+ unsigned char buffer[128];
+ static const char offsets[] = {3, 33, 63, 97, 93, 125, 127};
+ int i, j;
+ unsigned char* utf8;
+
+ if (-1 == lseek(fd, -128, SEEK_END))
+ return false;
+
+ if (read(fd, buffer, sizeof buffer) != sizeof buffer)
+ return false;
+
+ if (strncmp((char *)buffer, "TAG", 3))
+ return false;
+
+ entry->id3v1len = 128;
+ entry->id3version = ID3_VER_1_0;
+
+ for (i=0; i < (int)sizeof offsets; i++) {
+ unsigned char* ptr = (unsigned char *)buffer + offsets[i];
+
+ switch(i) {
+ case 0:
+ case 1:
+ case 2:
+ /* kill trailing space in strings */
+ for (j=29; j && (ptr[j]==0 || ptr[j]==' '); j--)
+ ptr[j] = 0;
+ /* convert string to utf8 */
+ utf8 = (unsigned char *)entry->id3v1buf[i];
+ utf8 = iso_decode(ptr, utf8, -1, 30);
+ /* make sure string is terminated */
+ *utf8 = 0;
+ break;
+
+ case 3:
+ /* kill trailing space in strings */
+ for (j=27; j && (ptr[j]==0 || ptr[j]==' '); j--)
+ ptr[j] = 0;
+ /* convert string to utf8 */
+ utf8 = (unsigned char *)entry->id3v1buf[3];
+ utf8 = iso_decode(ptr, utf8, -1, 28);
+ /* make sure string is terminated */
+ *utf8 = 0;
+ break;
+
+ case 4:
+ ptr[4] = 0;
+ entry->year = atoi((char *)ptr);
+ break;
+
+ case 5:
+ /* id3v1.1 uses last two bytes of comment field for track
+ number: first must be 0 and second is track num */
+ if (!ptr[0] && ptr[1]) {
+ entry->tracknum = ptr[1];
+ entry->id3version = ID3_VER_1_1;
+ }
+ break;
+
+ case 6:
+ /* genre */
+ entry->genre_string = id3_get_num_genre(ptr[0]);
+ break;
+ }
+ }
+
+ entry->title = entry->id3v1buf[0];
+ entry->artist = entry->id3v1buf[1];
+ entry->album = entry->id3v1buf[2];
+ entry->comment = entry->id3v1buf[3];
+
+ return true;
+}
+
+
+/*
+ * Sets the title of an MP3 entry based on its ID3v2 tag.
+ *
+ * Arguments: file - the MP3 file to scan for a ID3v2 tag
+ * entry - the entry to set the title in
+ *
+ * Returns: true if a title was found and created, else false
+ */
+static void setid3v2title(int fd, struct mp3entry *entry)
+{
+ int minframesize;
+ int size;
+ long bufferpos = 0, totframelen, framelen;
+ char header[10];
+ char tmp[4];
+ unsigned char version;
+ char *buffer = entry->id3v2buf;
+ int bytesread = 0;
+ int buffersize = sizeof(entry->id3v2buf);
+ unsigned char global_flags;
+ int flags;
+ int skip;
+ bool global_unsynch = false;
+ bool unsynch = false;
+ int i, j;
+ int rc;
+
+ global_ff_found = false;
+
+ /* Bail out if the tag is shorter than 10 bytes */
+ if(entry->id3v2len < 10)
+ return;
+
+ /* Read the ID3 tag version from the header */
+ lseek(fd, 0, SEEK_SET);
+ if(10 != read(fd, header, 10))
+ return;
+
+ /* Get the total ID3 tag size */
+ size = entry->id3v2len - 10;
+
+ version = header[3];
+ switch ( version ) {
+ case 2:
+ version = ID3_VER_2_2;
+ minframesize = 8;
+ break;
+
+ case 3:
+ version = ID3_VER_2_3;
+ minframesize = 12;
+ break;
+
+ case 4:
+ version = ID3_VER_2_4;
+ minframesize = 12;
+ break;
+
+ default:
+ /* unsupported id3 version */
+ return;
+ }
+ entry->id3version = version;
+ entry->tracknum = entry->year = entry->discnum = 0;
+ entry->title = entry->artist = entry->album = NULL; /* FIXME incomplete */
+
+ global_flags = header[5];
+
+ /* Skip the extended header if it is present */
+ if(global_flags & 0x40) {
+ if(version == ID3_VER_2_3) {
+ if(10 != read(fd, header, 10))
+ return;
+ /* The 2.3 extended header size doesn't include the header size
+ field itself. Also, it is not unsynched. */
+ framelen =
+ bytes2int(header[0], header[1], header[2], header[3]) + 4;
+
+ /* Skip the rest of the header */
+ lseek(fd, framelen - 10, SEEK_CUR);
+ }
+
+ if(version >= ID3_VER_2_4) {
+ if(4 != read(fd, header, 4))
+ return;
+
+ /* The 2.4 extended header size does include the entire header,
+ so here we can just skip it. This header is unsynched. */
+ framelen = unsync(header[0], header[1],
+ header[2], header[3]);
+
+ lseek(fd, framelen - 4, SEEK_CUR);
+ }
+ }
+
+ /* Is unsynchronization applied? */
+ if(global_flags & 0x80) {
+ global_unsynch = true;
+ }
+
+ /*
+ * We must have at least minframesize bytes left for the
+ * remaining frames to be interesting
+ */
+ while (size >= minframesize && bufferpos < buffersize - 1) {
+ flags = 0;
+
+ /* Read frame header and check length */
+ if(version >= ID3_VER_2_3) {
+ if(global_unsynch && version <= ID3_VER_2_3)
+ rc = read_unsynched(fd, header, 10);
+ else
+ rc = read(fd, header, 10);
+ if(rc != 10)
+ return;
+ /* Adjust for the 10 bytes we read */
+ size -= 10;
+
+ flags = bytes2int(0, 0, header[8], header[9]);
+
+ if (version >= ID3_VER_2_4) {
+ framelen = unsync(header[4], header[5],
+ header[6], header[7]);
+ } else {
+ /* version .3 files don't use synchsafe ints for
+ * size */
+ framelen = bytes2int(header[4], header[5],
+ header[6], header[7]);
+ }
+ } else {
+ if(6 != read(fd, header, 6))
+ return;
+ /* Adjust for the 6 bytes we read */
+ size -= 6;
+
+ framelen = bytes2int(0, header[3], header[4], header[5]);
+ }
+
+ logf("framelen = %ld", framelen);
+ if(framelen == 0){
+ if (header[0] == 0 && header[1] == 0 && header[2] == 0)
+ return;
+ else
+ continue;
+ }
+
+ unsynch = false;
+
+ if(flags)
+ {
+ skip = 0;
+
+ if (version >= ID3_VER_2_4) {
+ if(flags & 0x0040) { /* Grouping identity */
+ lseek(fd, 1, SEEK_CUR); /* Skip 1 byte */
+ framelen--;
+ }
+ } else {
+ if(flags & 0x0020) { /* Grouping identity */
+ lseek(fd, 1, SEEK_CUR); /* Skip 1 byte */
+ framelen--;
+ }
+ }
+
+ if(flags & 0x000c) /* Compression or encryption */
+ {
+ /* Skip it */
+ size -= framelen;
+ lseek(fd, framelen, SEEK_CUR);
+ continue;
+ }
+
+ if(flags & 0x0002) /* Unsynchronization */
+ unsynch = true;
+
+ if (version >= ID3_VER_2_4) {
+ if(flags & 0x0001) { /* Data length indicator */
+ if(4 != read(fd, tmp, 4))
+ return;
+
+ /* We don't need the data length */
+ framelen -= 4;
+ }
+ }
+ }
+
+ /* Keep track of the remaining frame size */
+ totframelen = framelen;
+
+ /* If the frame is larger than the remaining buffer space we try
+ to read as much as would fit in the buffer */
+ if(framelen >= buffersize - bufferpos)
+ framelen = buffersize - bufferpos - 1;
+
+ logf("id3v2 frame: %.4s", header);
+
+ /* Check for certain frame headers
+
+ 'size' is the amount of frame bytes remaining. We decrement it by
+ the amount of bytes we read. If we fail to read as many bytes as
+ we expect, we assume that we can't read from this file, and bail
+ out.
+
+ For each frame. we will iterate over the list of supported tags,
+ and read the tag into entry's buffer. All tags will be kept as
+ strings, for cases where a number won't do, e.g., YEAR: "circa
+ 1765", "1790/1977" (composed/performed), "28 Feb 1969" TRACK:
+ "1/12", "1 of 12", GENRE: "Freeform genre name" Text is more
+ flexible, and as the main use of id3 data is to display it,
+ converting it to an int just means reconverting to display it, at a
+ runtime cost.
+
+ For tags that the current code does convert to ints, a post
+ processing function will be called via a pointer to function. */
+
+ for (i=0; i<TAGLIST_SIZE; i++) {
+ const struct tag_resolver* tr = &taglist[i];
+ char** ptag = tr->offset ? (char**) (((char*)entry) + tr->offset)
+ : NULL;
+ char* tag;
+
+ /* Only ID3_VER_2_2 uses frames with three-character names. */
+ if (((version == ID3_VER_2_2) && (tr->tag_length != 3))
+ || ((version > ID3_VER_2_2) && (tr->tag_length != 4))) {
+ continue;
+ }
+
+ /* Note that parser functions sometimes set *ptag to NULL, so
+ * the "!*ptag" check here doesn't always have the desired
+ * effect. Should the parser functions (parsegenre in
+ * particular) be updated to handle the case of being called
+ * multiple times, or should the "*ptag" check be removed?
+ */
+ if( (!ptag || !*ptag) && !memcmp( header, tr->tag, tr->tag_length ) ) {
+
+ /* found a tag matching one in tagList, and not yet filled */
+ tag = buffer + bufferpos;
+
+ if(global_unsynch && version <= ID3_VER_2_3)
+ bytesread = read_unsynched(fd, tag, framelen);
+ else
+ bytesread = read(fd, tag, framelen);
+
+ if( bytesread != framelen )
+ return;
+
+ size -= bytesread;
+
+ if(unsynch || (global_unsynch && version >= ID3_VER_2_4))
+ bytesread = unsynchronize_frame(tag, bytesread);
+
+ /* the COMM frame has a 3 char field to hold an ISO-639-1
+ * language string and an optional short description;
+ * remove them so unicode_munge can work correctly
+ */
+
+ if(!memcmp( header, "COMM", 4 )) {
+ int offset;
+ /* ignore comments with iTunes 7 soundcheck/gapless data */
+ if(!strncmp(tag+4, "iTun", 4))
+ break;
+ offset = 3 + unicode_len(*tag, tag + 4);
+ if(bytesread > offset) {
+ bytesread -= offset;
+ memmove(tag + 1, tag + 1 + offset, bytesread - 1);
+ }
+ }
+
+ /* Attempt to parse Unicode string only if the tag contents
+ aren't binary */
+ if(!tr->binary) {
+ /* UTF-8 could potentially be 3 times larger */
+ /* so we need to create a new buffer */
+ char utf8buf[(3 * bytesread) + 1];
+
+ unicode_munge( tag, utf8buf, &bytesread );
+
+ if(bytesread >= buffersize - bufferpos)
+ bytesread = buffersize - bufferpos - 1;
+
+ for (j = 0; j < bytesread; j++)
+ tag[j] = utf8buf[j];
+
+ /* remove trailing spaces */
+ while ( bytesread > 0 && isspace(tag[bytesread-1]))
+ bytesread--;
+ }
+
+ tag[bytesread] = 0;
+ bufferpos += bytesread + 1;
+
+ if (ptag)
+ *ptag = tag;
+
+ if( tr->ppFunc )
+ bufferpos = tr->ppFunc(entry, tag, bufferpos);
+
+ /* Seek to the next frame */
+ if(framelen < totframelen)
+ lseek(fd, totframelen - framelen, SEEK_CUR);
+ break;
+ }
+ }
+
+ if( i == TAGLIST_SIZE ) {
+ /* no tag in tagList was found, or it was a repeat.
+ skip it using the total size */
+
+ if(global_unsynch && version <= ID3_VER_2_3) {
+ size -= skip_unsynched(fd, totframelen);
+ } else {
+ size -= totframelen;
+ if( lseek(fd, totframelen, SEEK_CUR) == -1 )
+ return;
+ }
+ }
+ }
+}
+
+/*
+ * Calculates the size of the ID3v2 tag.
+ *
+ * Arguments: file - the file to search for a tag.
+ *
+ * Returns: the size of the tag or 0 if none was found
+ */
+int getid3v2len(int fd)
+{
+ char buf[6];
+ int offset;
+
+ /* Make sure file has a ID3 tag */
+ if((-1 == lseek(fd, 0, SEEK_SET)) ||
+ (read(fd, buf, 6) != 6) ||
+ (strncmp(buf, "ID3", strlen("ID3")) != 0))
+ offset = 0;
+
+ /* Now check what the ID3v2 size field says */
+ else
+ if(read(fd, buf, 4) != 4)
+ offset = 0;
+ else
+ offset = unsync(buf[0], buf[1], buf[2], buf[3]) + 10;
+
+ logf("ID3V2 Length: 0x%x", offset);
+ return offset;
+}
+
+/*
+ * Calculates the length (in milliseconds) of an MP3 file.
+ *
+ * Modified to only use integers.
+ *
+ * Arguments: file - the file to calculate the length upon
+ * entry - the entry to update with the length
+ *
+ * Returns: the song length in milliseconds,
+ * 0 means that it couldn't be calculated
+ */
+static int getsonglength(int fd, struct mp3entry *entry)
+{
+ unsigned long filetime = 0;
+ struct mp3info info;
+ long bytecount;
+
+ /* Start searching after ID3v2 header */
+ if(-1 == lseek(fd, entry->id3v2len, SEEK_SET))
+ return 0;
+
+ bytecount = get_mp3file_info(fd, &info);
+
+ logf("Space between ID3V2 tag and first audio frame: 0x%lx bytes",
+ bytecount);
+
+ if(bytecount < 0)
+ return -1;
+
+ bytecount += entry->id3v2len;
+
+ /* Validate byte count, in case the file has been edited without
+ * updating the header.
+ */
+ if (info.byte_count)
+ {
+ const unsigned long expected = entry->filesize - entry->id3v1len
+ - entry->id3v2len;
+ const unsigned long diff = MAX(10240, info.byte_count / 20);
+
+ if ((info.byte_count > expected + diff)
+ || (info.byte_count < expected - diff))
+ {
+ logf("Note: info.byte_count differs from expected value by "
+ "%ld bytes", labs((long) (expected - info.byte_count)));
+ info.byte_count = 0;
+ info.frame_count = 0;
+ info.file_time = 0;
+ info.enc_padding = 0;
+
+ /* Even if the bitrate was based on "known bad" values, it
+ * should still be better for VBR files than using the bitrate
+ * of the first audio frame.
+ */
+ }
+ }
+
+ entry->bitrate = info.bitrate;
+ entry->frequency = info.frequency;
+ entry->version = info.version;
+ entry->layer = info.layer;
+ switch(entry->layer) {
+#if CONFIG_CODEC==SWCODEC
+ case 0:
+ entry->codectype=AFMT_MPA_L1;
+ break;
+#endif
+ case 1:
+ entry->codectype=AFMT_MPA_L2;
+ break;
+ case 2:
+ entry->codectype=AFMT_MPA_L3;
+ break;
+ }
+
+ /* If the file time hasn't been established, this may be a fixed
+ rate MP3, so just use the default formula */
+
+ filetime = info.file_time;
+
+ if(filetime == 0)
+ {
+ /* Prevent a division by zero */
+ if (info.bitrate < 8)
+ filetime = 0;
+ else
+ filetime = (entry->filesize - bytecount) / (info.bitrate / 8);
+ /* bitrate is in kbps so this delivers milliseconds. Doing bitrate / 8
+ * instead of filesize * 8 is exact, because mpeg audio bitrates are
+ * always multiples of 8, and it avoids overflows. */
+ }
+
+ entry->frame_count = info.frame_count;
+
+ entry->vbr = info.is_vbr;
+ entry->has_toc = info.has_toc;
+
+#if CONFIG_CODEC==SWCODEC
+ entry->lead_trim = info.enc_delay;
+ entry->tail_trim = info.enc_padding;
+#endif
+
+ memcpy(entry->toc, info.toc, sizeof(info.toc));
+
+ entry->vbr_header_pos = info.vbr_header_pos;
+
+ /* Update the seek point for the first playable frame */
+ entry->first_frame_offset = bytecount;
+ logf("First frame is at %lx", entry->first_frame_offset);
+
+ return filetime;
+}
+
+/*
+ * Checks all relevant information (such as ID3v1 tag, ID3v2 tag, length etc)
+ * about an MP3 file and updates it's entry accordingly.
+ *
+ Note, that this returns true for successful, false for error! */
+bool get_mp3_metadata(int fd, struct mp3entry *entry, const char *filename)
+{
+#if CONFIG_CODEC != SWCODEC
+ memset(entry, 0, sizeof(struct mp3entry));
+#endif
+
+ strncpy(entry->path, filename, sizeof(entry->path));
+
+ entry->title = NULL;
+ entry->filesize = filesize(fd);
+ entry->id3v2len = getid3v2len(fd);
+ entry->tracknum = 0;
+ entry->discnum = 0;
+
+ if (entry->id3v2len)
+ setid3v2title(fd, entry);
+ int len = getsonglength(fd, entry);
+ if (len < 0)
+ return false;
+ entry->length = len;
+
+ /* Subtract the meta information from the file size to get
+ the true size of the MP3 stream */
+ entry->filesize -= entry->first_frame_offset;
+
+ /* only seek to end of file if no id3v2 tags were found */
+ if (!entry->id3v2len) {
+ setid3v1title(fd, entry);
+ }
+
+ if(!entry->length || (entry->filesize < 8 ))
+ /* no song length or less than 8 bytes is hereby considered to be an
+ invalid mp3 and won't be played by us! */
+ return false;
+
+ return true;
+}
+
+/* Note, that this returns false for successful, true for error! */
+bool mp3info(struct mp3entry *entry, const char *filename)
+{
+ int fd;
+ bool result;
+
+ fd = open(filename, O_RDONLY);
+ if (fd < 0)
+ return true;
+
+ result = !get_mp3_metadata(fd, entry, filename);
+
+ close(fd);
+
+ return result;
+}
+
+void adjust_mp3entry(struct mp3entry *entry, void *dest, const void *orig)
+{
+ long offset;
+ if (orig > dest)
+ offset = - ((size_t)orig - (size_t)dest);
+ else
+ offset = (size_t)dest - (size_t)orig;
+
+ if (entry->title)
+ entry->title += offset;
+ if (entry->artist)
+ entry->artist += offset;
+ if (entry->album)
+ entry->album += offset;
+ if (entry->genre_string && !id3_is_genre_string(entry->genre_string))
+ /* Don't adjust that if it points to an entry of the "genres" array */
+ entry->genre_string += offset;
+ if (entry->track_string)
+ entry->track_string += offset;
+ if (entry->disc_string)
+ entry->disc_string += offset;
+ if (entry->year_string)
+ entry->year_string += offset;
+ if (entry->composer)
+ entry->composer += offset;
+ if (entry->comment)
+ entry->comment += offset;
+ if (entry->albumartist)
+ entry->albumartist += offset;
+ if (entry->grouping)
+ entry->grouping += offset;
+#if CONFIG_CODEC == SWCODEC
+ if (entry->track_gain_string)
+ entry->track_gain_string += offset;
+ if (entry->album_gain_string)
+ entry->album_gain_string += offset;
+#endif
+ if (entry->mb_track_id)
+ entry->mb_track_id += offset;
+}
+
+void copy_mp3entry(struct mp3entry *dest, const struct mp3entry *orig)
+{
+ memcpy(dest, orig, sizeof(struct mp3entry));
+ adjust_mp3entry(dest, dest, orig);
+}
+
+#ifdef DEBUG_STANDALONE
+
+char *secs2str(int ms)
+{
+ static char buffer[32];
+ int secs = ms/1000;
+ ms %= 1000;
+ snprintf(buffer, sizeof(buffer), "%d:%02d.%d", secs/60, secs%60, ms/100);
+ return buffer;
+}
+
+int main(int argc, char **argv)
+{
+ int i;
+ for(i=1; i<argc; i++) {
+ struct mp3entry mp3;
+ mp3.album = "Bogus";
+ if(mp3info(&mp3, argv[i], false)) {
+ printf("Failed to get %s\n", argv[i]);
+ return 0;
+ }
+
+ printf("****** File: %s\n"
+ " Title: %s\n"
+ " Artist: %s\n"
+ " Album: %s\n"
+ " Genre: %s (%d) \n"
+ " Composer: %s\n"
+ " Year: %s (%d)\n"
+ " Track: %s (%d)\n"
+ " Length: %s / %d s\n"
+ " Bitrate: %d\n"
+ " Frequency: %d\n",
+ argv[i],
+ mp3.title?mp3.title:"<blank>",
+ mp3.artist?mp3.artist:"<blank>",
+ mp3.album?mp3.album:"<blank>",
+ mp3.genre_string?mp3.genre_string:"<blank>",
+ mp3.genre,
+ mp3.composer?mp3.composer:"<blank>",
+ mp3.year_string?mp3.year_string:"<blank>",
+ mp3.year,
+ mp3.track_string?mp3.track_string:"<blank>",
+ mp3.tracknum,
+ secs2str(mp3.length),
+ mp3.length/1000,
+ mp3.bitrate,
+ mp3.frequency);
+ }
+
+ return 0;
+}
+
+#endif
diff --git a/apps/mpeg.c b/apps/mpeg.c
new file mode 100644
index 0000000000..2c65e46060
--- /dev/null
+++ b/apps/mpeg.c
@@ -0,0 +1,2873 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2002 by Linus Nielsen Feltzing
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+#include <stdbool.h>
+#include <stdlib.h>
+#include "config.h"
+
+#if CONFIG_CODEC != SWCODEC
+
+#include "debug.h"
+#include "panic.h"
+#include "id3.h"
+#include "mpeg.h"
+#include "audio.h"
+#include "ata.h"
+#include "string.h"
+#include <kernel.h>
+#include "thread.h"
+#include "errno.h"
+#include "mp3data.h"
+#include "buffer.h"
+#include "mp3_playback.h"
+#include "sound.h"
+#include "bitswap.h"
+#include "events.h"
+#ifndef SIMULATOR
+#include "i2c.h"
+#include "mas.h"
+#include "system.h"
+#include "usb.h"
+#include "file.h"
+#include "hwcompat.h"
+#endif /* !SIMULATOR */
+#ifdef HAVE_LCD_BITMAP
+#include "lcd.h"
+#endif
+
+#ifndef SIMULATOR
+extern unsigned long mas_version_code;
+#endif
+
+#if CONFIG_CODEC == MAS3587F
+extern enum /* from mp3_playback.c */
+{
+ MPEG_DECODER,
+ MPEG_ENCODER
+} mpeg_mode;
+#endif /* CONFIG_CODEC == MAS3587F */
+
+extern char* playlist_peek(int steps);
+extern bool playlist_check(int steps);
+extern int playlist_next(int steps);
+extern int playlist_amount(void);
+extern int playlist_update_resume_info(const struct mp3entry* id3);
+
+#define MPEG_PLAY 1
+#define MPEG_STOP 2
+#define MPEG_PAUSE 3
+#define MPEG_RESUME 4
+#define MPEG_NEXT 5
+#define MPEG_PREV 6
+#define MPEG_FF_REWIND 7
+#define MPEG_FLUSH_RELOAD 8
+#define MPEG_RECORD 9
+#define MPEG_INIT_RECORDING 10
+#define MPEG_INIT_PLAYBACK 11
+#define MPEG_NEW_FILE 12
+#define MPEG_PAUSE_RECORDING 13
+#define MPEG_RESUME_RECORDING 14
+#define MPEG_NEED_DATA 100
+#define MPEG_TRACK_CHANGE 101
+#define MPEG_SAVE_DATA 102
+#define MPEG_STOP_DONE 103
+#define MPEG_PRERECORDING_TICK 104
+
+/* indicator for MPEG_NEED_DATA */
+#define GENERATE_UNBUFFER_EVENTS 1
+
+/* list of tracks in memory */
+#define MAX_TRACK_ENTRIES (1<<4) /* Must be power of 2 */
+#define MAX_TRACK_ENTRIES_MASK (MAX_TRACK_ENTRIES - 1)
+
+struct trackdata
+{
+ struct mp3entry id3;
+ int mempos;
+ int load_ahead_index;
+};
+
+static struct trackdata trackdata[MAX_TRACK_ENTRIES];
+
+static unsigned int current_track_counter = 0;
+static unsigned int last_track_counter = 0;
+
+/* Play time of the previous track */
+unsigned long prev_track_elapsed;
+
+#ifndef SIMULATOR
+static int track_read_idx = 0;
+static int track_write_idx = 0;
+#endif /* !SIMULATOR */
+
+/* Cuesheet callback */
+static bool (*cuesheet_callback)(const char *filename) = NULL;
+
+static const char mpeg_thread_name[] = "mpeg";
+static unsigned int mpeg_errno;
+
+static bool playing = false; /* We are playing an MP3 stream */
+static bool is_playing = false; /* We are (attempting to) playing MP3 files */
+static bool paused; /* playback is paused */
+
+#ifdef SIMULATOR
+static char mpeg_stack[DEFAULT_STACK_SIZE];
+static struct mp3entry taginfo;
+
+#else /* !SIMULATOR */
+static struct event_queue mpeg_queue;
+static long mpeg_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
+
+static int audiobuflen;
+static int audiobuf_write;
+static int audiobuf_swapwrite;
+static int audiobuf_read;
+
+static int mpeg_file;
+
+static bool play_pending; /* We are about to start playing */
+static bool play_pending_track_change; /* When starting play we're starting a new file */
+static bool filling; /* We are filling the buffer with data from disk */
+static bool dma_underrun; /* True when the DMA has stopped because of
+ slow disk reading (read error, shaking) */
+static bool mpeg_stop_done;
+
+static int last_dma_tick = 0;
+static int last_dma_chunk_size;
+
+static long low_watermark; /* Dynamic low watermark level */
+static long low_watermark_margin = 0; /* Extra time in seconds for watermark */
+static long lowest_watermark_level; /* Debug value to observe the buffer
+ usage */
+#if CONFIG_CODEC == MAS3587F
+static char recording_filename[MAX_PATH]; /* argument to thread */
+static char delayed_filename[MAX_PATH]; /* internal copy of above */
+
+static char xing_buffer[MAX_XING_HEADER_SIZE];
+
+static bool init_recording_done;
+static bool init_playback_done;
+static bool prerecording; /* True if prerecording is enabled */
+static bool is_prerecording; /* True if we are prerecording */
+static bool is_recording; /* We are recording */
+
+static enum {
+ NOT_SAVING = 0, /* reasons to save data, sorted by importance */
+ BUFFER_FULL,
+ NEW_FILE,
+ STOP_RECORDING
+} saving_status;
+
+static int rec_frequency_index; /* For create_xing_header() calls */
+static int rec_version_index; /* For create_xing_header() calls */
+
+struct prerecord_info {
+ int mempos;
+ unsigned long framecount;
+};
+
+static struct prerecord_info prerecord_buffer[MPEG_MAX_PRERECORD_SECONDS];
+static int prerecord_index; /* Current index in the prerecord buffer */
+static int prerecording_max_seconds; /* Max number of seconds to store */
+static int prerecord_count; /* Number of seconds in the prerecord buffer */
+static int prerecord_timeout; /* The tick count of the next prerecord data
+ store */
+
+unsigned long record_start_time; /* Value of current_tick when recording
+ was started */
+unsigned long pause_start_time; /* Value of current_tick when pause was
+ started */
+static unsigned long last_rec_time;
+static unsigned long num_rec_bytes;
+static unsigned long last_rec_bytes;
+static unsigned long frame_count_start;
+static unsigned long frame_count_end;
+static unsigned long saved_header = 0;
+
+/* Shadow MAS registers */
+unsigned long shadow_encoder_control = 0;
+#endif /* CONFIG_CODEC == MAS3587F */
+
+#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
+unsigned long shadow_io_control_main = 0;
+unsigned long shadow_soft_mute = 0;
+unsigned shadow_codec_reg0;
+#endif /* (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F) */
+
+#ifdef HAVE_RECORDING
+static const unsigned char empty_id3_header[] =
+{
+ 'I', 'D', '3', 0x03, 0x00, 0x00,
+ 0x00, 0x00, 0x1f, 0x76 /* Size is 4096 minus 10 bytes for the header */
+};
+#endif /* HAVE_RECORDING */
+
+
+static int get_unplayed_space(void);
+static int get_playable_space(void);
+static int get_unswapped_space(void);
+#endif /* !SIMULATOR */
+
+#if (CONFIG_CODEC == MAS3587F) && !defined(SIMULATOR)
+static void init_recording(void);
+static void prepend_header(void);
+static void update_header(void);
+static void start_prerecording(void);
+static void start_recording(void);
+static void stop_recording(void);
+static int get_unsaved_space(void);
+static void pause_recording(void);
+static void resume_recording(void);
+#endif /* (CONFIG_CODEC == MAS3587F) && !defined(SIMULATOR) */
+
+
+#ifndef SIMULATOR
+static int num_tracks_in_memory(void)
+{
+ return (track_write_idx - track_read_idx) & MAX_TRACK_ENTRIES_MASK;
+}
+
+#ifdef DEBUG_TAGS
+static void debug_tags(void)
+{
+ int i;
+
+ for(i = 0;i < MAX_TRACK_ENTRIES;i++)
+ {
+ DEBUGF("%d - %s\n", i, trackdata[i].id3.path);
+ }
+ DEBUGF("read: %d, write :%d\n", track_read_idx, track_write_idx);
+ DEBUGF("num_tracks_in_memory: %d\n", num_tracks_in_memory());
+}
+#else /* !DEBUG_TAGS */
+#define debug_tags()
+#endif /* !DEBUG_TAGS */
+
+static void remove_current_tag(void)
+{
+ if(num_tracks_in_memory() > 0)
+ {
+ /* First move the index, so nobody tries to access the tag */
+ track_read_idx = (track_read_idx+1) & MAX_TRACK_ENTRIES_MASK;
+ debug_tags();
+ }
+ else
+ {
+ DEBUGF("remove_current_tag: no tracks to remove\n");
+ }
+}
+
+static void remove_all_non_current_tags(void)
+{
+ track_write_idx = (track_read_idx+1) & MAX_TRACK_ENTRIES_MASK;
+ debug_tags();
+}
+
+static void remove_all_tags(void)
+{
+ track_write_idx = track_read_idx;
+
+ debug_tags();
+}
+
+static struct trackdata *get_trackdata(int offset)
+{
+ if(offset >= num_tracks_in_memory())
+ return NULL;
+ else
+ return &trackdata[(track_read_idx + offset) & MAX_TRACK_ENTRIES_MASK];
+}
+#endif /* !SIMULATOR */
+
+/***********************************************************************/
+/* audio event handling */
+
+#define MAX_EVENT_HANDLERS 10
+struct event_handlers_table
+{
+ AUDIO_EVENT_HANDLER handler;
+ unsigned short mask;
+};
+static struct event_handlers_table event_handlers[MAX_EVENT_HANDLERS];
+static int event_handlers_count = 0;
+
+void audio_register_event_handler(AUDIO_EVENT_HANDLER handler, unsigned short mask)
+{
+ if (event_handlers_count < MAX_EVENT_HANDLERS)
+ {
+ event_handlers[event_handlers_count].handler = handler;
+ event_handlers[event_handlers_count].mask = mask;
+ event_handlers_count++;
+ }
+}
+
+/* dispatch calls each handler in the order registered and returns after some
+ handler actually handles the event (the event is assumed to no longer be valid
+ after this, due to the handler changing some condition); returns true if someone
+ handled the event, which is expected to cause the caller to skip its own handling
+ of the event */
+#ifndef SIMULATOR
+static bool audio_dispatch_event(unsigned short event, unsigned long data)
+{
+ int i = 0;
+ for(i=0; i < event_handlers_count; i++)
+ {
+ if ( event_handlers[i].mask & event )
+ {
+ int rc = event_handlers[i].handler(event, data);
+ if ( rc == AUDIO_EVENT_RC_HANDLED )
+ return true;
+ }
+ }
+ return false;
+}
+#endif
+
+/***********************************************************************/
+
+static void set_elapsed(struct mp3entry* id3)
+{
+ if ( id3->vbr ) {
+ if ( id3->has_toc ) {
+ /* calculate elapsed time using TOC */
+ int i;
+ unsigned int remainder, plen, relpos, nextpos;
+
+ /* find wich percent we're at */
+ for (i=0; i<100; i++ )
+ {
+ if ( id3->offset < id3->toc[i] * (id3->filesize / 256) )
+ {
+ break;
+ }
+ }
+
+ i--;
+ if (i < 0)
+ i = 0;
+
+ relpos = id3->toc[i];
+
+ if (i < 99)
+ {
+ nextpos = id3->toc[i+1];
+ }
+ else
+ {
+ nextpos = 256;
+ }
+
+ remainder = id3->offset - (relpos * (id3->filesize / 256));
+
+ /* set time for this percent (divide before multiply to prevent
+ overflow on long files. loss of precision is negligible on
+ short files) */
+ id3->elapsed = i * (id3->length / 100);
+
+ /* calculate remainder time */
+ plen = (nextpos - relpos) * (id3->filesize / 256);
+ id3->elapsed += (((remainder * 100) / plen) *
+ (id3->length / 10000));
+ }
+ else {
+ /* no TOC exists. set a rough estimate using average bitrate */
+ int tpk = id3->length / (id3->filesize / 1024);
+ id3->elapsed = id3->offset / 1024 * tpk;
+ }
+ }
+ else
+ /* constant bitrate, use exact calculation */
+ id3->elapsed = id3->offset / (id3->bitrate / 8);
+}
+
+int audio_get_file_pos(void)
+{
+ int pos = -1;
+ struct mp3entry *id3 = audio_current_track();
+
+ if (id3->vbr)
+ {
+ if (id3->has_toc)
+ {
+ /* Use the TOC to find the new position */
+ unsigned int percent, remainder;
+ int curtoc, nexttoc, plen;
+
+ percent = (id3->elapsed*100)/id3->length;
+ if (percent > 99)
+ percent = 99;
+
+ curtoc = id3->toc[percent];
+
+ if (percent < 99)
+ nexttoc = id3->toc[percent+1];
+ else
+ nexttoc = 256;
+
+ pos = (id3->filesize/256)*curtoc;
+
+ /* Use the remainder to get a more accurate position */
+ remainder = (id3->elapsed*100)%id3->length;
+ remainder = (remainder*100)/id3->length;
+ plen = (nexttoc - curtoc)*(id3->filesize/256);
+ pos += (plen/100)*remainder;
+ }
+ else
+ {
+ /* No TOC exists, estimate the new position */
+ pos = (id3->filesize / (id3->length / 1000)) *
+ (id3->elapsed / 1000);
+ }
+ }
+ else if (id3->bitrate)
+ pos = id3->elapsed * (id3->bitrate / 8);
+ else
+ {
+ return -1;
+ }
+
+ if (pos >= (int)(id3->filesize - id3->id3v1len))
+ {
+ /* Don't seek right to the end of the file so that we can
+ transition properly to the next song */
+ pos = id3->filesize - id3->id3v1len - 1;
+ }
+ else if (pos < (int)id3->first_frame_offset)
+ {
+ /* skip past id3v2 tag and other leading garbage */
+ pos = id3->first_frame_offset;
+ }
+ return pos;
+}
+
+unsigned long mpeg_get_last_header(void)
+{
+#ifdef SIMULATOR
+ return 0;
+#else /* !SIMULATOR */
+ unsigned long tmp[2];
+
+ /* Read the frame data from the MAS and reconstruct it with the
+ frame sync and all */
+ mas_readmem(MAS_BANK_D0, MAS_D0_MPEG_STATUS_1, tmp, 2);
+ return 0xffe00000 | ((tmp[0] & 0x7c00) << 6) | (tmp[1] & 0xffff);
+#endif /* !SIMULATOR */
+}
+
+void audio_set_cuesheet_callback(bool (*handler)(const char *filename))
+{
+ cuesheet_callback = handler;
+}
+
+#ifndef SIMULATOR
+/* Send callback events to notify about removing old tracks. */
+static void generate_unbuffer_events(void)
+{
+ int i;
+ int numentries = MAX_TRACK_ENTRIES - num_tracks_in_memory();
+ int cur_idx = track_write_idx;
+
+ for (i = 0; i < numentries; i++)
+ {
+ /* Send an event to notify that track has finished. */
+ send_event(PLAYBACK_EVENT_TRACK_FINISH, &trackdata[cur_idx].id3);
+ cur_idx = (cur_idx + 1) & MAX_TRACK_ENTRIES_MASK;
+ }
+}
+
+/* Send callback events to notify about new tracks. */
+static void generate_postbuffer_events(void)
+{
+ int i;
+ int numentries = num_tracks_in_memory();
+ int cur_idx = track_read_idx;
+
+ for (i = 0; i < numentries; i++)
+ {
+ send_event(PLAYBACK_EVENT_TRACK_BUFFER, &trackdata[cur_idx].id3);
+ cur_idx = (cur_idx + 1) & MAX_TRACK_ENTRIES_MASK;
+ }
+}
+
+static void recalculate_watermark(int bitrate)
+{
+ int bytes_per_sec;
+ int time = ata_spinup_time;
+
+ /* A bitrate of 0 probably means empty VBR header. We play safe
+ and set a high threshold */
+ if(bitrate == 0)
+ bitrate = 320;
+
+ bytes_per_sec = bitrate * 1000 / 8;
+
+ if(time)
+ {
+ /* No drive spins up faster than 3.5s */
+ if(time < 350)
+ time = 350;
+
+ time = time * 3;
+ low_watermark = ((low_watermark_margin * HZ + time) *
+ bytes_per_sec) / HZ;
+ }
+ else
+ {
+ low_watermark = MPEG_LOW_WATER;
+ }
+}
+
+#ifdef HAVE_DISK_STORAGE
+void audio_set_buffer_margin(int seconds)
+{
+ low_watermark_margin = seconds;
+}
+#endif
+
+void audio_get_debugdata(struct audio_debug *dbgdata)
+{
+ dbgdata->audiobuflen = audiobuflen;
+ dbgdata->audiobuf_write = audiobuf_write;
+ dbgdata->audiobuf_swapwrite = audiobuf_swapwrite;
+ dbgdata->audiobuf_read = audiobuf_read;
+
+ dbgdata->last_dma_chunk_size = last_dma_chunk_size;
+
+#if CONFIG_CPU == SH7034
+ dbgdata->dma_on = (SCR0 & 0x80) != 0;
+#endif
+ dbgdata->playing = playing;
+ dbgdata->play_pending = play_pending;
+ dbgdata->is_playing = is_playing;
+ dbgdata->filling = filling;
+ dbgdata->dma_underrun = dma_underrun;
+
+ dbgdata->unplayed_space = get_unplayed_space();
+ dbgdata->playable_space = get_playable_space();
+ dbgdata->unswapped_space = get_unswapped_space();
+
+ dbgdata->low_watermark_level = low_watermark;
+ dbgdata->lowest_watermark_level = lowest_watermark_level;
+}
+
+#ifdef DEBUG
+static void dbg_timer_start(void)
+{
+ /* We are using timer 2 */
+
+ TSTR &= ~0x04; /* Stop the timer */
+ TSNC &= ~0x04; /* No synchronization */
+ TMDR &= ~0x44; /* Operate normally */
+
+ TCNT2 = 0; /* Start counting at 0 */
+ TCR2 = 0x03; /* Sysclock/8 */
+
+ TSTR |= 0x04; /* Start timer 2 */
+}
+
+static int dbg_cnt2us(unsigned int cnt)
+{
+ return (cnt * 10000) / (FREQ/800);
+}
+#endif /* DEBUG */
+
+static int get_unplayed_space(void)
+{
+ int space = audiobuf_write - audiobuf_read;
+ if (space < 0)
+ space += audiobuflen;
+ return space;
+}
+
+static int get_playable_space(void)
+{
+ int space = audiobuf_swapwrite - audiobuf_read;
+ if (space < 0)
+ space += audiobuflen;
+ return space;
+}
+
+static int get_unplayed_space_current_song(void)
+{
+ int space;
+
+ if (num_tracks_in_memory() > 1)
+ {
+ space = get_trackdata(1)->mempos - audiobuf_read;
+ }
+ else
+ {
+ space = audiobuf_write - audiobuf_read;
+ }
+
+ if (space < 0)
+ space += audiobuflen;
+
+ return space;
+}
+
+static int get_unswapped_space(void)
+{
+ int space = audiobuf_write - audiobuf_swapwrite;
+ if (space < 0)
+ space += audiobuflen;
+ return space;
+}
+
+#if CONFIG_CODEC == MAS3587F
+static int get_unsaved_space(void)
+{
+ int space = audiobuf_write - audiobuf_read;
+ if (space < 0)
+ space += audiobuflen;
+ return space;
+}
+
+static void drain_dma_buffer(void)
+{
+ while (PBDRH & 0x40)
+ {
+ xor_b(0x08, &PADRH);
+
+ while (PBDRH & 0x80);
+
+ xor_b(0x08, &PADRH);
+
+ while (!(PBDRH & 0x80));
+ }
+}
+
+#ifdef DEBUG
+static long timing_info_index = 0;
+static long timing_info[1024];
+#endif /* DEBUG */
+
+void rec_tick (void) __attribute__ ((section (".icode")));
+void rec_tick(void)
+{
+ int i;
+ int delay;
+ char data;
+
+ if(is_recording && (PBDRH & 0x40))
+ {
+#ifdef DEBUG
+ timing_info[timing_info_index++] = current_tick;
+ TCNT2 = 0;
+#endif /* DEBUG */
+ /* Note: Although this loop is run in interrupt context, further
+ * optimisation will do no good. The MAS would then deliver bad
+ * frames occasionally, as observed in extended experiments. */
+ i = 0;
+ while (PBDRH & 0x40) /* We try to read as long as EOD is high */
+ {
+ xor_b(0x08, &PADRH); /* Set PR active, independent of polarity */
+
+ delay = 100;
+ while (PBDRH & 0x80) /* Wait until /RTW becomes active */
+ {
+ if (--delay <= 0) /* Bail out if we have to wait too long */
+ { /* i.e. the MAS doesn't want to talk to us */
+ xor_b(0x08, &PADRH); /* Set PR inactive */
+ goto transfer_end; /* and get out of here */
+ }
+ }
+
+ data = *(unsigned char *)0x04000000; /* read data byte */
+
+ xor_b(0x08, &PADRH); /* Set PR inactive */
+
+ audiobuf[audiobuf_write++] = data;
+
+ if (audiobuf_write >= audiobuflen)
+ audiobuf_write = 0;
+
+ i++;
+ }
+ transfer_end:
+
+#ifdef DEBUG
+ timing_info[timing_info_index++] = TCNT2 + (i << 16);
+ timing_info_index &= 0x3ff;
+#endif /* DEBUG */
+
+ num_rec_bytes += i;
+
+ if(is_prerecording)
+ {
+ if(TIME_AFTER(current_tick, prerecord_timeout))
+ {
+ prerecord_timeout = current_tick + HZ;
+ queue_post(&mpeg_queue, MPEG_PRERECORDING_TICK, 0);
+ }
+ }
+ else
+ {
+ /* Signal to save the data if we are running out of buffer
+ space */
+ if (audiobuflen - get_unsaved_space() < MPEG_RECORDING_LOW_WATER
+ && saving_status == NOT_SAVING)
+ {
+ saving_status = BUFFER_FULL;
+ queue_post(&mpeg_queue, MPEG_SAVE_DATA, 0);
+ }
+ }
+ }
+}
+#endif /* CONFIG_CODEC == MAS3587F */
+
+void playback_tick(void)
+{
+ struct trackdata *ptd = get_trackdata(0);
+ if(ptd)
+ {
+ ptd->id3.elapsed += (current_tick - last_dma_tick) * 1000 / HZ;
+ last_dma_tick = current_tick;
+ audio_dispatch_event(AUDIO_EVENT_POS_REPORT,
+ (unsigned long)ptd->id3.elapsed);
+ }
+}
+
+static void reset_mp3_buffer(void)
+{
+ audiobuf_read = 0;
+ audiobuf_write = 0;
+ audiobuf_swapwrite = 0;
+ lowest_watermark_level = audiobuflen;
+}
+
+ /* DMA transfer end interrupt callback */
+static void transfer_end(unsigned char** ppbuf, size_t* psize)
+{
+ if(playing && !paused)
+ {
+ int unplayed_space_left;
+ int space_until_end_of_buffer;
+ int track_offset = 1;
+ struct trackdata *track;
+
+ audiobuf_read += last_dma_chunk_size;
+ if(audiobuf_read >= audiobuflen)
+ audiobuf_read = 0;
+
+ /* First, check if we are on a track boundary */
+ if (num_tracks_in_memory() > 1)
+ {
+ if (audiobuf_read == get_trackdata(track_offset)->mempos)
+ {
+ if ( ! audio_dispatch_event(AUDIO_EVENT_END_OF_TRACK, 0) )
+ {
+ queue_post(&mpeg_queue, MPEG_TRACK_CHANGE, 0);
+ track_offset++;
+ }
+ }
+ }
+
+ unplayed_space_left = get_unplayed_space();
+
+ space_until_end_of_buffer = audiobuflen - audiobuf_read;
+
+ if(!filling && unplayed_space_left < low_watermark)
+ {
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, GENERATE_UNBUFFER_EVENTS);
+ }
+
+ if(unplayed_space_left)
+ {
+ last_dma_chunk_size = MIN(0x2000, unplayed_space_left);
+ last_dma_chunk_size = MIN(last_dma_chunk_size,
+ space_until_end_of_buffer);
+
+ /* several tracks loaded? */
+ track = get_trackdata(track_offset);
+ if(track)
+ {
+ /* will we move across the track boundary? */
+ if (( audiobuf_read < track->mempos ) &&
+ ((audiobuf_read+last_dma_chunk_size) >
+ track->mempos ))
+ {
+ /* Make sure that we end exactly on the boundary */
+ last_dma_chunk_size = track->mempos - audiobuf_read;
+ }
+ }
+
+ *psize = last_dma_chunk_size & 0xffff;
+ *ppbuf = audiobuf + audiobuf_read;
+ track = get_trackdata(0);
+ if(track)
+ track->id3.offset += last_dma_chunk_size;
+
+ /* Update the watermark debug level */
+ if(unplayed_space_left < lowest_watermark_level)
+ lowest_watermark_level = unplayed_space_left;
+ }
+ else
+ {
+ /* Check if the end of data is because of a hard disk error.
+ If there is an open file handle, we are still playing music.
+ If not, the last file has been loaded, and the file handle is
+ closed. */
+ if(mpeg_file >= 0)
+ {
+ /* Update the watermark debug level */
+ if(unplayed_space_left < lowest_watermark_level)
+ lowest_watermark_level = unplayed_space_left;
+
+ DEBUGF("DMA underrun.\n");
+ dma_underrun = true;
+ }
+ else
+ {
+ if ( ! audio_dispatch_event(AUDIO_EVENT_END_OF_TRACK, 0) )
+ {
+ DEBUGF("No more MP3 data. Stopping.\n");
+ queue_post(&mpeg_queue, MPEG_TRACK_CHANGE, 0);
+ playing = false;
+ }
+ }
+ *psize = 0; /* no more transfer */
+ }
+ }
+}
+
+static struct trackdata *add_track_to_tag_list(const char *filename)
+{
+ struct trackdata *track;
+
+ if(num_tracks_in_memory() >= MAX_TRACK_ENTRIES)
+ {
+ DEBUGF("Tag memory is full\n");
+ return NULL;
+ }
+
+ track = &trackdata[track_write_idx];
+
+ /* grab id3 tag of new file and
+ remember where in memory it starts */
+ if(mp3info(&track->id3, filename))
+ {
+ DEBUGF("Bad mp3\n");
+ return NULL;
+ }
+ track->mempos = audiobuf_write;
+ track->id3.elapsed = 0;
+#ifdef HAVE_LCD_BITMAP
+ if (track->id3.title)
+ lcd_getstringsize(track->id3.title, NULL, NULL);
+ if (track->id3.artist)
+ lcd_getstringsize(track->id3.artist, NULL, NULL);
+ if (track->id3.album)
+ lcd_getstringsize(track->id3.album, NULL, NULL);
+#endif
+ if (cuesheet_callback)
+ if (cuesheet_callback(filename))
+ track->id3.cuesheet_type = 1;
+
+ track_write_idx = (track_write_idx+1) & MAX_TRACK_ENTRIES_MASK;
+ debug_tags();
+ return track;
+}
+
+static int new_file(int steps)
+{
+ int max_steps = playlist_amount();
+ int start = 0;
+ int i;
+ struct trackdata *track;
+
+ /* Find out how many steps to advance. The load_ahead_index field tells
+ us how many playlist entries it had to skip to get to a valid one.
+ We add those together to find out where to start. */
+ if(steps > 0 && num_tracks_in_memory() > 1)
+ {
+ /* Begin with the song after the currently playing one */
+ i = 1;
+ while((track = get_trackdata(i++)))
+ {
+ start += track->load_ahead_index;
+ }
+ }
+
+ do {
+ char *trackname;
+
+ trackname = playlist_peek( start + steps );
+ if ( !trackname )
+ return -1;
+
+ DEBUGF("Loading %s\n", trackname);
+
+ mpeg_file = open(trackname, O_RDONLY);
+ if(mpeg_file < 0) {
+ DEBUGF("Couldn't open file: %s\n",trackname);
+ if(steps < 0)
+ steps--;
+ else
+ steps++;
+ }
+ else
+ {
+ struct trackdata *track = add_track_to_tag_list(trackname);
+
+ if(!track)
+ {
+ /* Bad mp3 file */
+ if(steps < 0)
+ steps--;
+ else
+ steps++;
+ close(mpeg_file);
+ mpeg_file = -1;
+ }
+ else
+ {
+ /* skip past id3v2 tag */
+ lseek(mpeg_file,
+ track->id3.first_frame_offset,
+ SEEK_SET);
+ track->id3.index = steps;
+ track->load_ahead_index = steps;
+ track->id3.offset = 0;
+
+ if(track->id3.vbr)
+ /* Average bitrate * 1.5 */
+ recalculate_watermark(
+ (track->id3.bitrate * 3) / 2);
+ else
+ recalculate_watermark(
+ track->id3.bitrate);
+
+ }
+ }
+
+ /* Bail out if no file could be opened */
+ if(abs(steps) > max_steps)
+ return -1;
+ } while ( mpeg_file < 0 );
+
+ return 0;
+}
+
+static void stop_playing(void)
+{
+ struct trackdata *track;
+
+ /* Stop the current stream */
+ mp3_play_stop();
+ playing = false;
+ filling = false;
+
+ track = get_trackdata(0);
+ if (track != NULL)
+ prev_track_elapsed = track->id3.elapsed;
+
+ if(mpeg_file >= 0)
+ close(mpeg_file);
+ mpeg_file = -1;
+ remove_all_tags();
+ generate_unbuffer_events();
+ reset_mp3_buffer();
+}
+
+static void end_current_track(void) {
+ struct trackdata *track;
+
+ play_pending = false;
+ playing = false;
+ mp3_play_pause(false);
+
+ track = get_trackdata(0);
+ if (track != NULL)
+ prev_track_elapsed = track->id3.elapsed;
+
+ reset_mp3_buffer();
+ remove_all_tags();
+ generate_unbuffer_events();
+
+ if(mpeg_file >= 0)
+ close(mpeg_file);
+}
+
+/* Is this a really the end of playback or is a new playlist starting */
+static void check_playlist_end(int direction)
+{
+ /* Use the largest possible step size to account for skipped tracks */
+ int steps = playlist_amount();
+
+ if (direction < 0)
+ steps = -steps;
+
+ if (playlist_next(steps) < 0)
+ is_playing = false;
+}
+
+static void update_playlist(void)
+{
+ if (num_tracks_in_memory() > 0)
+ {
+ struct trackdata *track = get_trackdata(0);
+ track->id3.index = playlist_next(track->id3.index);
+ }
+ else
+ {
+ /* End of playlist? */
+ check_playlist_end(1);
+ }
+
+ playlist_update_resume_info(audio_current_track());
+}
+
+static void track_change(void)
+{
+ DEBUGF("Track change\n");
+
+ struct trackdata *track = get_trackdata(0);
+ prev_track_elapsed = track->id3.elapsed;
+
+#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
+ /* Reset the AVC */
+ sound_set_avc(-1);
+#endif /* (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F) */
+
+ if (num_tracks_in_memory() > 0)
+ {
+ remove_current_tag();
+ send_event(PLAYBACK_EVENT_TRACK_CHANGE, audio_current_track());
+ update_playlist();
+ }
+
+ current_track_counter++;
+}
+
+unsigned long audio_prev_elapsed(void)
+{
+ return prev_track_elapsed;
+}
+
+#ifdef DEBUG
+void hexdump(const unsigned char *buf, int len)
+{
+ int i;
+
+ for(i = 0;i < len;i++)
+ {
+ if(i && (i & 15) == 0)
+ {
+ DEBUGF("\n");
+ }
+ DEBUGF("%02x ", buf[i]);
+ }
+ DEBUGF("\n");
+}
+#endif /* DEBUG */
+
+static void start_playback_if_ready(void)
+{
+ int playable_space;
+
+ playable_space = audiobuf_swapwrite - audiobuf_read;
+ if(playable_space < 0)
+ playable_space += audiobuflen;
+
+ /* See if we have started playing yet. If not, do it. */
+ if(play_pending || dma_underrun)
+ {
+ /* If the filling has stopped, and we still haven't reached
+ the watermark, the file must be smaller than the
+ watermark. We must still play it. */
+ if((playable_space >= MPEG_PLAY_PENDING_THRESHOLD) ||
+ !filling || dma_underrun)
+ {
+ DEBUGF("P\n");
+ if (play_pending) /* don't do this when recovering from DMA underrun */
+ {
+ generate_postbuffer_events(); /* signal first track as buffered */
+ if (play_pending_track_change)
+ {
+ play_pending_track_change = false;
+ send_event(PLAYBACK_EVENT_TRACK_CHANGE, audio_current_track());
+ }
+ play_pending = false;
+ }
+ playing = true;
+
+ last_dma_chunk_size = MIN(0x2000, get_unplayed_space_current_song());
+ mp3_play_data(audiobuf + audiobuf_read, last_dma_chunk_size, transfer_end);
+ dma_underrun = false;
+
+ if (!paused)
+ {
+ last_dma_tick = current_tick;
+ mp3_play_pause(true);
+ }
+
+ /* Tell ourselves that we need more data */
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+ }
+ }
+}
+
+static bool swap_one_chunk(void)
+{
+ int free_space_left;
+ int amount_to_swap;
+
+ free_space_left = get_unswapped_space();
+
+ if(free_space_left == 0 && !play_pending)
+ return false;
+
+ /* Swap in larger chunks when the user is waiting for the playback
+ to start, or when there is dangerously little playable data left */
+ if(play_pending)
+ amount_to_swap = MIN(MPEG_PLAY_PENDING_SWAPSIZE, free_space_left);
+ else
+ {
+ if(get_playable_space() < low_watermark)
+ amount_to_swap = MIN(MPEG_LOW_WATER_SWAP_CHUNKSIZE,
+ free_space_left);
+ else
+ amount_to_swap = MIN(MPEG_SWAP_CHUNKSIZE, free_space_left);
+ }
+
+ if(audiobuf_write < audiobuf_swapwrite)
+ amount_to_swap = MIN(audiobuflen - audiobuf_swapwrite,
+ amount_to_swap);
+ else
+ amount_to_swap = MIN(audiobuf_write - audiobuf_swapwrite,
+ amount_to_swap);
+
+ bitswap(audiobuf + audiobuf_swapwrite, amount_to_swap);
+
+ audiobuf_swapwrite += amount_to_swap;
+ if(audiobuf_swapwrite >= audiobuflen)
+ {
+ audiobuf_swapwrite = 0;
+ }
+
+ return true;
+}
+
+static void mpeg_thread(void)
+{
+ static int pause_tick = 0;
+ static unsigned int pause_track = 0;
+ struct queue_event ev;
+ int len;
+ int free_space_left;
+ int unplayed_space_left;
+ int amount_to_read;
+ int t1, t2;
+ int start_offset;
+#if CONFIG_CODEC == MAS3587F
+ int amount_to_save;
+ int save_endpos = 0;
+ int rc;
+ int level;
+ long offset;
+#endif /* CONFIG_CODEC == MAS3587F */
+
+ is_playing = false;
+ play_pending = false;
+ playing = false;
+ mpeg_file = -1;
+
+ while(1)
+ {
+#if CONFIG_CODEC == MAS3587F
+ if(mpeg_mode == MPEG_DECODER)
+ {
+#endif /* CONFIG_CODEC == MAS3587F */
+ yield();
+
+ /* Swap if necessary, and don't block on the queue_wait() */
+ if(swap_one_chunk())
+ {
+ queue_wait_w_tmo(&mpeg_queue, &ev, 0);
+ }
+ else if (playing)
+ {
+ /* periodically update resume info */
+ queue_wait_w_tmo(&mpeg_queue, &ev, HZ/2);
+ }
+ else
+ {
+ DEBUGF("S R:%x W:%x SW:%x\n",
+ audiobuf_read, audiobuf_write, audiobuf_swapwrite);
+ queue_wait(&mpeg_queue, &ev);
+ }
+
+ start_playback_if_ready();
+
+ switch(ev.id)
+ {
+ case MPEG_PLAY:
+ DEBUGF("MPEG_PLAY\n");
+
+#if CONFIG_TUNER
+ /* Silence the A/D input, it may be on because the radio
+ may be playing */
+ mas_codec_writereg(6, 0x0000);
+#endif /* CONFIG_TUNER */
+
+ /* Stop the current stream */
+ paused = false;
+ end_current_track();
+
+ if ( new_file(0) == -1 )
+ {
+ is_playing = false;
+ track_change();
+ break;
+ }
+
+ start_offset = (int)ev.data;
+
+ /* mid-song resume? */
+ if (start_offset) {
+ struct mp3entry* id3 = &get_trackdata(0)->id3;
+ lseek(mpeg_file, start_offset, SEEK_SET);
+ id3->offset = start_offset;
+ set_elapsed(id3);
+ }
+ else {
+ /* skip past id3v2 tag */
+ lseek(mpeg_file,
+ get_trackdata(0)->id3.first_frame_offset,
+ SEEK_SET);
+
+ }
+
+ /* Make it read more data */
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+
+ /* Tell the file loading code that we want to start playing
+ as soon as we have some data */
+ play_pending = true;
+ play_pending_track_change = true;
+
+ update_playlist();
+ current_track_counter++;
+ break;
+
+ case MPEG_STOP:
+ DEBUGF("MPEG_STOP\n");
+ is_playing = false;
+ paused = false;
+
+ if (playing)
+ playlist_update_resume_info(audio_current_track());
+
+ stop_playing();
+ mpeg_stop_done = true;
+ break;
+
+ case MPEG_PAUSE:
+ DEBUGF("MPEG_PAUSE\n");
+ /* Stop the current stream */
+ if (playing)
+ playlist_update_resume_info(audio_current_track());
+ paused = true;
+ playing = false;
+ pause_tick = current_tick;
+ pause_track = current_track_counter;
+ mp3_play_pause(false);
+ break;
+
+ case MPEG_RESUME:
+ DEBUGF("MPEG_RESUME\n");
+ /* Continue the current stream */
+ paused = false;
+ if (!play_pending)
+ {
+ playing = true;
+ if ( current_track_counter == pause_track )
+ last_dma_tick += current_tick - pause_tick;
+ else
+ last_dma_tick = current_tick;
+ pause_tick = 0;
+ mp3_play_pause(true);
+ }
+ break;
+
+ case MPEG_NEXT:
+ DEBUGF("MPEG_NEXT\n");
+ /* is next track in ram? */
+ if ( num_tracks_in_memory() > 1 ) {
+ int unplayed_space_left, unswapped_space_left;
+
+ /* stop the current stream */
+ play_pending = false;
+ playing = false;
+ mp3_play_pause(false);
+
+ track_change();
+ audiobuf_read = get_trackdata(0)->mempos;
+ last_dma_chunk_size = MIN(0x2000, get_unplayed_space_current_song());
+ mp3_play_data(audiobuf + audiobuf_read, last_dma_chunk_size, transfer_end);
+ dma_underrun = false;
+ last_dma_tick = current_tick;
+
+ unplayed_space_left = get_unplayed_space();
+ unswapped_space_left = get_unswapped_space();
+
+ /* should we start reading more data? */
+ if(!filling && (unplayed_space_left < low_watermark)) {
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, GENERATE_UNBUFFER_EVENTS);
+ play_pending = true;
+ } else if(unswapped_space_left &&
+ unswapped_space_left > unplayed_space_left) {
+ /* Stop swapping the data from the previous file */
+ audiobuf_swapwrite = audiobuf_read;
+ play_pending = true;
+ } else {
+ playing = true;
+ if (!paused)
+ mp3_play_pause(true);
+ }
+ }
+ else {
+ if (!playlist_check(1))
+ break;
+
+ /* stop the current stream */
+ end_current_track();
+
+ if (new_file(1) < 0) {
+ DEBUGF("No more files to play\n");
+ filling = false;
+
+ check_playlist_end(1);
+ current_track_counter++;
+ } else {
+ /* Make it read more data */
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+
+ /* Tell the file loading code that we want
+ to start playing as soon as we have some data */
+ play_pending = true;
+ play_pending_track_change = true;
+
+ update_playlist();
+ current_track_counter++;
+ }
+ }
+ break;
+
+ case MPEG_PREV: {
+ DEBUGF("MPEG_PREV\n");
+
+ if (!playlist_check(-1))
+ break;
+
+ /* stop the current stream */
+ end_current_track();
+
+ /* Open the next file */
+ if (new_file(-1) < 0) {
+ DEBUGF("No more files to play\n");
+ filling = false;
+
+ check_playlist_end(-1);
+ current_track_counter++;
+ } else {
+ /* Make it read more data */
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+
+ /* Tell the file loading code that we want to
+ start playing as soon as we have some data */
+ play_pending = true;
+ play_pending_track_change = true;
+
+ update_playlist();
+ current_track_counter++;
+ }
+ break;
+ }
+
+ case MPEG_FF_REWIND: {
+ struct mp3entry *id3 = audio_current_track();
+ unsigned int oldtime = id3->elapsed;
+ unsigned int newtime = (unsigned int)ev.data;
+ int curpos, newpos, diffpos;
+ DEBUGF("MPEG_FF_REWIND\n");
+
+ id3->elapsed = newtime;
+
+ newpos = audio_get_file_pos();
+ if(newpos < 0)
+ {
+ id3->elapsed = oldtime;
+ break;
+ }
+
+ if (mpeg_file >= 0)
+ curpos = lseek(mpeg_file, 0, SEEK_CUR);
+ else
+ curpos = id3->filesize;
+
+ if (num_tracks_in_memory() > 1)
+ {
+ /* We have started loading other tracks that need to be
+ accounted for */
+ struct trackdata *track;
+ int i = 0;
+
+ while((track = get_trackdata(i++)))
+ {
+ curpos += track->id3.filesize;
+ }
+ }
+
+ diffpos = curpos - newpos;
+
+ if(!filling && diffpos >= 0 && diffpos < audiobuflen)
+ {
+ int unplayed_space_left, unswapped_space_left;
+
+ /* We are changing to a position that's already in
+ memory, so we just move the DMA read pointer. */
+ audiobuf_read = audiobuf_write - diffpos;
+ if (audiobuf_read < 0)
+ {
+ audiobuf_read += audiobuflen;
+ }
+
+ unplayed_space_left = get_unplayed_space();
+ unswapped_space_left = get_unswapped_space();
+
+ /* If unswapped_space_left is larger than
+ unplayed_space_left, it means that the swapwrite pointer
+ hasn't yet advanced up to the new location of the read
+ pointer. We just move it, there is no need to swap
+ data that won't be played anyway. */
+
+ if (unswapped_space_left > unplayed_space_left)
+ {
+ DEBUGF("Moved swapwrite\n");
+ audiobuf_swapwrite = audiobuf_read;
+ play_pending = true;
+ }
+
+ if (mpeg_file>=0 && unplayed_space_left < low_watermark)
+ {
+ /* We need to load more data before starting */
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, GENERATE_UNBUFFER_EVENTS);
+ play_pending = true;
+ }
+ else
+ {
+ /* resume will start at new position */
+ last_dma_chunk_size =
+ MIN(0x2000, get_unplayed_space_current_song());
+ mp3_play_data(audiobuf + audiobuf_read,
+ last_dma_chunk_size, transfer_end);
+ dma_underrun = false;
+ }
+ }
+ else
+ {
+ /* Move to the new position in the file and start
+ loading data */
+ reset_mp3_buffer();
+
+ if (num_tracks_in_memory() > 1)
+ {
+ /* We have to reload the current track */
+ close(mpeg_file);
+ remove_all_non_current_tags();
+ generate_unbuffer_events();
+ mpeg_file = -1;
+ }
+
+ if (mpeg_file < 0)
+ {
+ mpeg_file = open(id3->path, O_RDONLY);
+ if (mpeg_file < 0)
+ {
+ id3->elapsed = oldtime;
+ break;
+ }
+ }
+
+ if(-1 == lseek(mpeg_file, newpos, SEEK_SET))
+ {
+ id3->elapsed = oldtime;
+ break;
+ }
+
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+
+ /* Tell the file loading code that we want to start playing
+ as soon as we have some data */
+ play_pending = true;
+ }
+
+ id3->offset = newpos;
+
+ break;
+ }
+
+ case MPEG_FLUSH_RELOAD: {
+ int numtracks = num_tracks_in_memory();
+ bool reload_track = false;
+
+ if (numtracks > 1)
+ {
+ /* Reset the buffer */
+ audiobuf_write = get_trackdata(1)->mempos;
+
+ /* Reset swapwrite unless we're still swapping current
+ track */
+ if (get_unplayed_space() <= get_playable_space())
+ audiobuf_swapwrite = audiobuf_write;
+
+ close(mpeg_file);
+ remove_all_non_current_tags();
+ generate_unbuffer_events();
+ mpeg_file = -1;
+ reload_track = true;
+ }
+ else if (numtracks == 1 && mpeg_file < 0)
+ {
+ reload_track = true;
+ }
+
+ if(reload_track && new_file(1) >= 0)
+ {
+ /* Tell ourselves that we want more data */
+ filling = true;
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+ }
+
+ break;
+ }
+
+ case MPEG_NEED_DATA:
+ free_space_left = audiobuf_read - audiobuf_write;
+
+ /* We interpret 0 as "empty buffer" */
+ if(free_space_left <= 0)
+ free_space_left += audiobuflen;
+
+ unplayed_space_left = audiobuflen - free_space_left;
+
+ /* Make sure that we don't fill the entire buffer */
+ free_space_left -= MPEG_HIGH_WATER;
+
+ if (ev.data == GENERATE_UNBUFFER_EVENTS)
+ generate_unbuffer_events();
+
+ /* do we have any more buffer space to fill? */
+ if(free_space_left <= 0)
+ {
+ DEBUGF("0\n");
+ filling = false;
+ generate_postbuffer_events();
+ ata_sleep();
+ break;
+ }
+
+ /* Read small chunks while we are below the low water mark */
+ if(unplayed_space_left < low_watermark)
+ amount_to_read = MIN(MPEG_LOW_WATER_CHUNKSIZE,
+ free_space_left);
+ else
+ amount_to_read = free_space_left;
+
+ /* Don't read more than until the end of the buffer */
+ amount_to_read = MIN(audiobuflen - audiobuf_write,
+ amount_to_read);
+#ifdef HAVE_MMC /* MMC is slow, so don't read too large chunks */
+ amount_to_read = MIN(0x40000, amount_to_read);
+#elif MEM == 8
+ amount_to_read = MIN(0x100000, amount_to_read);
+#endif
+
+ /* Read as much mpeg data as we can fit in the buffer */
+ if(mpeg_file >= 0)
+ {
+ DEBUGF("R\n");
+ t1 = current_tick;
+ len = read(mpeg_file, audiobuf + audiobuf_write,
+ amount_to_read);
+ if(len > 0)
+ {
+ t2 = current_tick;
+ DEBUGF("time: %d\n", t2 - t1);
+ DEBUGF("R: %x\n", len);
+
+ /* Now make sure that we don't feed the MAS with ID3V1
+ data */
+ if (len < amount_to_read)
+ {
+ int i;
+ static const unsigned char tag[] = "TAG";
+ int taglen = 128;
+ int tagptr = audiobuf_write + len - 128;
+
+ /* Really rare case: entire potential tag wasn't
+ read in this call AND audiobuf_write < 128 */
+ if (tagptr < 0)
+ tagptr += audiobuflen;
+
+ for(i = 0;i < 3;i++)
+ {
+ if(tagptr >= audiobuflen)
+ tagptr -= audiobuflen;
+
+ if(audiobuf[tagptr] != tag[i])
+ {
+ taglen = 0;
+ break;
+ }
+
+ tagptr++;
+ }
+
+ if(taglen)
+ {
+ /* Skip id3v1 tag */
+ DEBUGF("Skipping ID3v1 tag\n");
+ len -= taglen;
+
+ /* In the very rare case when the entire tag
+ wasn't read in this read() len will be < 0.
+ Take care of this when changing the write
+ pointer. */
+ }
+ }
+
+ audiobuf_write += len;
+
+ if (audiobuf_write < 0)
+ audiobuf_write += audiobuflen;
+
+ if(audiobuf_write >= audiobuflen)
+ {
+ audiobuf_write = 0;
+ DEBUGF("W\n");
+ }
+
+ /* Tell ourselves that we want more data */
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+ }
+ else
+ {
+ if(len < 0)
+ {
+ DEBUGF("MPEG read error\n");
+ }
+
+ close(mpeg_file);
+ mpeg_file = -1;
+
+ if(new_file(1) < 0)
+ {
+ /* No more data to play */
+ DEBUGF("No more files to play\n");
+ filling = false;
+ }
+ else
+ {
+ /* Tell ourselves that we want more data */
+ queue_post(&mpeg_queue, MPEG_NEED_DATA, 0);
+ }
+ }
+ }
+ break;
+
+ case MPEG_TRACK_CHANGE:
+ track_change();
+ break;
+
+#ifndef USB_NONE
+ case SYS_USB_CONNECTED:
+ is_playing = false;
+ paused = false;
+ stop_playing();
+
+ /* Tell the USB thread that we are safe */
+ DEBUGF("mpeg_thread got SYS_USB_CONNECTED\n");
+ usb_acknowledge(SYS_USB_CONNECTED_ACK);
+
+ /* Wait until the USB cable is extracted again */
+ usb_wait_for_disconnect(&mpeg_queue);
+ break;
+#endif /* !USB_NONE */
+
+#if CONFIG_CODEC == MAS3587F
+ case MPEG_INIT_RECORDING:
+ init_recording();
+ init_recording_done = true;
+ break;
+#endif /* CONFIG_CODEC == MAS3587F */
+
+ case SYS_TIMEOUT:
+ if (playing)
+ playlist_update_resume_info(audio_current_track());
+ break;
+ }
+#if CONFIG_CODEC == MAS3587F
+ }
+ else
+ {
+ queue_wait(&mpeg_queue, &ev);
+ switch(ev.id)
+ {
+ case MPEG_RECORD:
+ if (is_prerecording)
+ {
+ int startpos;
+
+ /* Go back prerecord_count seconds in the buffer */
+ startpos = prerecord_index - prerecord_count;
+ if(startpos < 0)
+ startpos += prerecording_max_seconds;
+
+ /* Read the position data from the prerecord buffer */
+ frame_count_start = prerecord_buffer[startpos].framecount;
+ startpos = prerecord_buffer[startpos].mempos;
+
+ DEBUGF("Start looking at address %x (%x)\n",
+ audiobuf+startpos, startpos);
+
+ saved_header = mpeg_get_last_header();
+
+ mem_find_next_frame(startpos, &offset, 1800,
+ saved_header);
+
+ audiobuf_read = startpos + offset;
+ if(audiobuf_read >= audiobuflen)
+ audiobuf_read -= audiobuflen;
+
+ DEBUGF("New audiobuf_read address: %x (%x)\n",
+ audiobuf+audiobuf_read, audiobuf_read);
+
+ level = disable_irq_save();
+ num_rec_bytes = get_unsaved_space();
+ restore_irq(level);
+ }
+ else
+ {
+ frame_count_start = 0;
+ num_rec_bytes = 0;
+ audiobuf_read = MPEG_RESERVED_HEADER_SPACE;
+ audiobuf_write = MPEG_RESERVED_HEADER_SPACE;
+ }
+
+ prepend_header();
+ DEBUGF("Recording...\n");
+ start_recording();
+
+ /* Wait until at least one frame is encoded and get the
+ frame header, for later use by the Xing header
+ generation */
+ sleep(HZ/5);
+ saved_header = mpeg_get_last_header();
+
+ /* delayed until buffer is saved, don't open yet */
+ strcpy(delayed_filename, recording_filename);
+ mpeg_file = -1;
+
+ break;
+
+ case MPEG_STOP:
+ DEBUGF("MPEG_STOP\n");
+
+ stop_recording();
+
+ /* Save the remaining data in the buffer */
+ save_endpos = audiobuf_write;
+ saving_status = STOP_RECORDING;
+ queue_post(&mpeg_queue, MPEG_SAVE_DATA, 0);
+ break;
+
+ case MPEG_STOP_DONE:
+ DEBUGF("MPEG_STOP_DONE\n");
+
+ if (mpeg_file >= 0)
+ close(mpeg_file);
+ mpeg_file = -1;
+
+ update_header();
+#ifdef DEBUG1
+ {
+ int i;
+ for(i = 0;i < 512;i++)
+ {
+ DEBUGF("%d - %d us (%d bytes)\n",
+ timing_info[i*2],
+ (timing_info[i*2+1] & 0xffff) *
+ 10000 / 13824,
+ timing_info[i*2+1] >> 16);
+ }
+ }
+#endif /* DEBUG1 */
+
+ if (prerecording)
+ {
+ start_prerecording();
+ }
+ mpeg_stop_done = true;
+ break;
+
+ case MPEG_NEW_FILE:
+ /* Bail out when a more important save is happening */
+ if (saving_status > NEW_FILE)
+ break;
+
+ /* Make sure we have at least one complete frame
+ in the buffer. If we haven't recorded a single
+ frame within 200ms, the MAS is probably not recording
+ anything, and we bail out. */
+ amount_to_save = get_unsaved_space();
+ if (amount_to_save < 1800)
+ {
+ sleep(HZ/5);
+ amount_to_save = get_unsaved_space();
+ }
+
+ mas_readmem(MAS_BANK_D0, MAS_D0_MPEG_FRAME_COUNT,
+ &frame_count_end, 1);
+
+ last_rec_time = current_tick - record_start_time;
+ record_start_time = current_tick;
+ if (paused)
+ pause_start_time = record_start_time;
+
+ /* capture all values at one point */
+ level = disable_irq_save();
+ save_endpos = audiobuf_write;
+ last_rec_bytes = num_rec_bytes;
+ num_rec_bytes = 0;
+ restore_irq(level);
+
+ if (amount_to_save >= 1800)
+ {
+ /* Now find a frame boundary to split at */
+ save_endpos -= 1800;
+ if (save_endpos < 0)
+ save_endpos += audiobuflen;
+
+ rc = mem_find_next_frame(save_endpos, &offset, 1800,
+ saved_header);
+ if (!rc) /* No header found, save whole buffer */
+ offset = 1800;
+
+ save_endpos += offset;
+ if (save_endpos >= audiobuflen)
+ save_endpos -= audiobuflen;
+
+ last_rec_bytes += offset - 1800;
+ level = disable_irq_save();
+ num_rec_bytes += 1800 - offset;
+ restore_irq(level);
+ }
+
+ saving_status = NEW_FILE;
+ queue_post(&mpeg_queue, MPEG_SAVE_DATA, 0);
+ break;
+
+ case MPEG_SAVE_DATA:
+ if (saving_status == BUFFER_FULL)
+ save_endpos = audiobuf_write;
+
+ if (mpeg_file < 0) /* delayed file open */
+ {
+ mpeg_file = open(delayed_filename, O_WRONLY|O_CREAT);
+
+ if (mpeg_file < 0)
+ panicf("recfile: %d", mpeg_file);
+ }
+
+ amount_to_save = save_endpos - audiobuf_read;
+ if (amount_to_save < 0)
+ amount_to_save += audiobuflen;
+
+ amount_to_save = MIN(amount_to_save,
+ audiobuflen - audiobuf_read);
+#ifdef HAVE_MMC /* MMC is slow, so don't save too large chunks at once */
+ amount_to_save = MIN(0x40000, amount_to_save);
+#elif MEM == 8
+ amount_to_save = MIN(0x100000, amount_to_save);
+#endif
+ rc = write(mpeg_file, audiobuf + audiobuf_read,
+ amount_to_save);
+ if (rc < 0)
+ {
+ if (errno == ENOSPC)
+ {
+ mpeg_errno = AUDIOERR_DISK_FULL;
+ stop_recording();
+ queue_post(&mpeg_queue, MPEG_STOP_DONE, 0);
+ /* will close the file */
+ break;
+ }
+ else
+ panicf("rec wrt: %d", rc);
+ }
+
+ audiobuf_read += amount_to_save;
+ if (audiobuf_read >= audiobuflen)
+ audiobuf_read = 0;
+
+ if (audiobuf_read == save_endpos) /* all saved */
+ {
+ switch (saving_status)
+ {
+ case BUFFER_FULL:
+ rc = fsync(mpeg_file);
+ if (rc < 0)
+ panicf("rec fls: %d", rc);
+ ata_sleep();
+ break;
+
+ case NEW_FILE:
+ /* Close the current file */
+ rc = close(mpeg_file);
+ if (rc < 0)
+ panicf("rec cls: %d", rc);
+ mpeg_file = -1;
+ update_header();
+ ata_sleep();
+
+ /* copy new filename */
+ strcpy(delayed_filename, recording_filename);
+ prepend_header();
+ frame_count_start = frame_count_end;
+ break;
+
+ case STOP_RECORDING:
+ queue_post(&mpeg_queue, MPEG_STOP_DONE, 0);
+ /* will close the file */
+ break;
+
+ default:
+ break;
+ }
+ saving_status = NOT_SAVING;
+ }
+ else /* tell ourselves to save the next chunk */
+ queue_post(&mpeg_queue, MPEG_SAVE_DATA, 0);
+
+ break;
+
+ case MPEG_PRERECORDING_TICK:
+ if(!is_prerecording)
+ break;
+
+ /* Store the write pointer every second */
+ prerecord_buffer[prerecord_index].mempos = audiobuf_write;
+ mas_readmem(MAS_BANK_D0, MAS_D0_MPEG_FRAME_COUNT,
+ &prerecord_buffer[prerecord_index].framecount, 1);
+
+ /* Wrap if necessary */
+ if(++prerecord_index == prerecording_max_seconds)
+ prerecord_index = 0;
+
+ /* Update the number of seconds recorded */
+ if(prerecord_count < prerecording_max_seconds)
+ prerecord_count++;
+ break;
+
+ case MPEG_INIT_PLAYBACK:
+ /* Stop the prerecording */
+ stop_recording();
+ reset_mp3_buffer();
+ mp3_play_init();
+ init_playback_done = true;
+ break;
+
+ case MPEG_PAUSE_RECORDING:
+ pause_recording();
+ break;
+
+ case MPEG_RESUME_RECORDING:
+ resume_recording();
+ break;
+
+ case SYS_USB_CONNECTED:
+ /* We can safely go to USB mode if no recording
+ is taking place */
+ if((!is_recording || is_prerecording) && mpeg_stop_done)
+ {
+ /* Even if we aren't recording, we still call this
+ function, to put the MAS in monitoring mode,
+ to save power. */
+ stop_recording();
+
+ /* Tell the USB thread that we are safe */
+ DEBUGF("mpeg_thread got SYS_USB_CONNECTED\n");
+ usb_acknowledge(SYS_USB_CONNECTED_ACK);
+
+ /* Wait until the USB cable is extracted again */
+ usb_wait_for_disconnect(&mpeg_queue);
+ }
+ break;
+ }
+ }
+#endif /* CONFIG_CODEC == MAS3587F */
+ }
+}
+#endif /* !SIMULATOR */
+
+struct mp3entry* audio_current_track()
+{
+#ifdef SIMULATOR
+ return &taginfo;
+#else /* !SIMULATOR */
+ if(num_tracks_in_memory())
+ return &get_trackdata(0)->id3;
+ else
+ return NULL;
+#endif /* !SIMULATOR */
+}
+
+struct mp3entry* audio_next_track()
+{
+#ifdef SIMULATOR
+ return &taginfo;
+#else /* !SIMULATOR */
+ if(num_tracks_in_memory() > 1)
+ return &get_trackdata(1)->id3;
+ else
+ return NULL;
+#endif /* !SIMULATOR */
+}
+
+bool audio_has_changed_track(void)
+{
+ if(last_track_counter != current_track_counter)
+ {
+ last_track_counter = current_track_counter;
+ return true;
+ }
+ return false;
+}
+
+#if CONFIG_CODEC == MAS3587F
+#ifndef SIMULATOR
+void audio_init_playback(void)
+{
+ init_playback_done = false;
+ queue_post(&mpeg_queue, MPEG_INIT_PLAYBACK, 0);
+
+ while(!init_playback_done)
+ sleep(1);
+}
+
+
+/****************************************************************************
+ * Recording functions
+ ***************************************************************************/
+void audio_init_recording(unsigned int buffer_offset)
+{
+ buffer_offset = buffer_offset;
+ init_recording_done = false;
+ queue_post(&mpeg_queue, MPEG_INIT_RECORDING, 0);
+
+ while(!init_recording_done)
+ sleep(1);
+}
+
+static void init_recording(void)
+{
+ unsigned long val;
+ int rc;
+
+ /* Disable IRQ6 */
+ IPRB &= 0xff0f;
+
+ stop_playing();
+ is_playing = false;
+ paused = false;
+
+ /* Init the recording variables */
+ is_recording = false;
+ is_prerecording = false;
+
+ mpeg_stop_done = true;
+
+ mas_reset();
+
+ /* Enable the audio CODEC and the DSP core, max analog voltage range */
+ rc = mas_direct_config_write(MAS_CONTROL, 0x8c00);
+ if(rc < 0)
+ panicf("mas_ctrl_w: %d", rc);
+
+ /* Stop the current application */
+ val = 0;
+ mas_writemem(MAS_BANK_D0, MAS_D0_APP_SELECT, &val, 1);
+ do
+ {
+ mas_readmem(MAS_BANK_D0, MAS_D0_APP_RUNNING, &val, 1);
+ } while(val);
+
+ /* Perform black magic as described by the data sheet */
+ if((mas_version_code & 0x0fff) == 0x0102)
+ {
+ DEBUGF("Performing MAS black magic for B2 version\n");
+ mas_writereg(0xa3, 0x98);
+ mas_writereg(0x94, 0xfffff);
+ val = 0;
+ mas_writemem(MAS_BANK_D1, 0, &val, 1);
+ mas_writereg(0xa3, 0x90);
+ }
+
+ /* Enable A/D Converters */
+ shadow_codec_reg0 = 0xcccd;
+ mas_codec_writereg(0x0, shadow_codec_reg0);
+
+ /* Copy left channel to right (mono mode) */
+ mas_codec_writereg(8, 0x8000);
+
+ /* ADC scale 0%, DSP scale 100%
+ We use the DSP output for monitoring, because it works with all
+ sources including S/PDIF */
+ mas_codec_writereg(6, 0x0000);
+ mas_codec_writereg(7, 0x4000);
+
+ /* No mute */
+ shadow_soft_mute = 0;
+ mas_writemem(MAS_BANK_D0, MAS_D0_SOFT_MUTE, &shadow_soft_mute, 1);
+
+#ifdef HAVE_SPDIF_OUT
+ val = 0x09; /* Disable SDO and SDI, low impedance S/PDIF outputs */
+#else
+ val = 0x2d; /* Disable SDO and SDI, disable S/PDIF output */
+#endif
+ mas_writemem(MAS_BANK_D0, MAS_D0_INTERFACE_CONTROL, &val, 1);
+
+ /* Set Demand mode, monitoring OFF and validate all settings */
+ shadow_io_control_main = 0x125;
+ mas_writemem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &shadow_io_control_main, 1);
+
+ /* Start the encoder application */
+ val = 0x40;
+ mas_writemem(MAS_BANK_D0, MAS_D0_APP_SELECT, &val, 1);
+ do
+ {
+ mas_readmem(MAS_BANK_D0, MAS_D0_APP_RUNNING, &val, 1);
+ } while(!(val & 0x40));
+
+ /* We have started the recording application with monitoring OFF.
+ This is because we want to record at least one frame to fill the DMA
+ buffer, because the silly MAS will not negate EOD until at least one
+ DMA transfer has taken place.
+ Now let's wait for some data to be encoded. */
+ sleep(HZ/5);
+
+ /* Now set it to Monitoring mode as default, saves power */
+ shadow_io_control_main = 0x525;
+ mas_writemem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &shadow_io_control_main, 1);
+
+ /* Wait until the DSP has accepted the settings */
+ do
+ {
+ mas_readmem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &val,1);
+ } while(val & 1);
+
+ drain_dma_buffer();
+ mpeg_mode = MPEG_ENCODER;
+
+ DEBUGF("MAS Recording application started\n");
+
+ /* At this point, all settings are the reset MAS defaults, next thing is to
+ call mpeg_set_recording_options(). */
+}
+
+void audio_record(const char *filename)
+{
+ mpeg_errno = 0;
+
+ strncpy(recording_filename, filename, MAX_PATH - 1);
+ recording_filename[MAX_PATH - 1] = 0;
+
+ queue_post(&mpeg_queue, MPEG_RECORD, 0);
+}
+
+void audio_pause_recording(void)
+{
+ queue_post(&mpeg_queue, MPEG_PAUSE_RECORDING, 0);
+}
+
+void audio_resume_recording(void)
+{
+ queue_post(&mpeg_queue, MPEG_RESUME_RECORDING, 0);
+}
+
+static void prepend_header(void)
+{
+ int startpos;
+ unsigned i;
+
+ /* Make room for header */
+ audiobuf_read -= MPEG_RESERVED_HEADER_SPACE;
+ if(audiobuf_read < 0)
+ {
+ /* Clear the bottom half */
+ memset(audiobuf, 0, audiobuf_read + MPEG_RESERVED_HEADER_SPACE);
+
+ /* And the top half */
+ audiobuf_read += audiobuflen;
+ memset(audiobuf + audiobuf_read, 0, audiobuflen - audiobuf_read);
+ }
+ else
+ {
+ memset(audiobuf + audiobuf_read, 0, MPEG_RESERVED_HEADER_SPACE);
+ }
+ /* Copy the empty ID3 header */
+ startpos = audiobuf_read;
+ for(i = 0; i < sizeof(empty_id3_header); i++)
+ {
+ audiobuf[startpos++] = empty_id3_header[i];
+ if(startpos == audiobuflen)
+ startpos = 0;
+ }
+}
+
+static void update_header(void)
+{
+ int fd, framelen;
+ unsigned long frames;
+
+ if (last_rec_bytes > 0)
+ {
+ /* Create the Xing header */
+ fd = open(delayed_filename, O_RDWR);
+ if (fd < 0)
+ panicf("rec upd: %d (%s)", fd, recording_filename);
+
+ frames = frame_count_end - frame_count_start;
+ /* If the number of recorded frames has reached 0x7ffff,
+ we can no longer trust it */
+ if (frame_count_end == 0x7ffff)
+ frames = 0;
+
+ /* saved_header is saved right before stopping the MAS */
+ framelen = create_xing_header(fd, 0, last_rec_bytes, xing_buffer,
+ frames, last_rec_time * (1000/HZ),
+ saved_header, NULL, false);
+
+ lseek(fd, MPEG_RESERVED_HEADER_SPACE - framelen, SEEK_SET);
+ write(fd, xing_buffer, framelen);
+ close(fd);
+ }
+}
+
+static void start_prerecording(void)
+{
+ unsigned long val;
+
+ DEBUGF("Starting prerecording\n");
+
+ prerecord_index = 0;
+ prerecord_count = 0;
+ prerecord_timeout = current_tick + HZ;
+ memset(prerecord_buffer, 0, sizeof(prerecord_buffer));
+ reset_mp3_buffer();
+
+ is_prerecording = true;
+
+ /* Stop monitoring and start the encoder */
+ shadow_io_control_main &= ~(1 << 10);
+ mas_writemem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &shadow_io_control_main, 1);
+ DEBUGF("mas_writemem(MAS_BANK_D0, IO_CONTROL_MAIN, %x)\n", shadow_io_control_main);
+
+ /* Wait until the DSP has accepted the settings */
+ do
+ {
+ mas_readmem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &val,1);
+ } while(val & 1);
+
+ is_recording = true;
+ saving_status = NOT_SAVING;
+
+ demand_irq_enable(true);
+}
+
+static void start_recording(void)
+{
+ unsigned long val;
+
+ if(is_prerecording)
+ {
+ /* This will make the IRQ handler start recording
+ for real, i.e send MPEG_SAVE_DATA messages when
+ the buffer is full */
+ is_prerecording = false;
+ }
+ else
+ {
+ /* If prerecording is off, we need to stop the monitoring
+ and start the encoder */
+ shadow_io_control_main &= ~(1 << 10);
+ mas_writemem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &shadow_io_control_main, 1);
+ DEBUGF("mas_writemem(MAS_BANK_D0, IO_CONTROL_MAIN, %x)\n", shadow_io_control_main);
+
+ /* Wait until the DSP has accepted the settings */
+ do
+ {
+ mas_readmem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &val,1);
+ } while(val & 1);
+ }
+
+ is_recording = true;
+ saving_status = NOT_SAVING;
+ paused = false;
+
+ /* Store the current time */
+ if(prerecording)
+ record_start_time = current_tick - prerecord_count * HZ;
+ else
+ record_start_time = current_tick;
+
+ pause_start_time = 0;
+
+ demand_irq_enable(true);
+}
+
+static void pause_recording(void)
+{
+ pause_start_time = current_tick;
+
+ /* Set the pause bit */
+ shadow_soft_mute |= 2;
+ mas_writemem(MAS_BANK_D0, MAS_D0_SOFT_MUTE, &shadow_soft_mute, 1);
+
+ paused = true;
+}
+
+static void resume_recording(void)
+{
+ paused = false;
+
+ /* Clear the pause bit */
+ shadow_soft_mute &= ~2;
+ mas_writemem(MAS_BANK_D0, MAS_D0_SOFT_MUTE, &shadow_soft_mute, 1);
+
+ /* Compensate for the time we have been paused */
+ if(pause_start_time)
+ {
+ record_start_time =
+ current_tick - (pause_start_time - record_start_time);
+ pause_start_time = 0;
+ }
+}
+
+static void stop_recording(void)
+{
+ unsigned long val;
+
+ /* Let it finish the last frame */
+ if(!paused)
+ pause_recording();
+ sleep(HZ/5);
+
+ demand_irq_enable(false);
+
+ is_recording = false;
+ is_prerecording = false;
+
+ last_rec_bytes = num_rec_bytes;
+ mas_readmem(MAS_BANK_D0, MAS_D0_MPEG_FRAME_COUNT, &frame_count_end, 1);
+ last_rec_time = current_tick - record_start_time;
+
+ /* Start monitoring */
+ shadow_io_control_main |= (1 << 10);
+ mas_writemem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &shadow_io_control_main, 1);
+ DEBUGF("mas_writemem(MAS_BANK_D0, IO_CONTROL_MAIN, %x)\n", shadow_io_control_main);
+
+ /* Wait until the DSP has accepted the settings */
+ do
+ {
+ mas_readmem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &val,1);
+ } while(val & 1);
+
+ resume_recording();
+}
+
+void audio_set_recording_options(struct audio_recording_options *options)
+{
+ bool is_mpeg1;
+
+ is_mpeg1 = (options->rec_frequency < 3)?true:false;
+
+ rec_version_index = is_mpeg1?3:2;
+ rec_frequency_index = options->rec_frequency % 3;
+
+ shadow_encoder_control = (options->rec_quality << 17) |
+ (rec_frequency_index << 10) |
+ ((is_mpeg1?1:0) << 9) |
+ (((options->rec_channels * 2 + 1) & 3) << 6) |
+ (1 << 5) /* MS-stereo */ |
+ (1 << 2) /* Is an original */;
+ mas_writemem(MAS_BANK_D0, MAS_D0_ENCODER_CONTROL, &shadow_encoder_control,1);
+
+ DEBUGF("mas_writemem(MAS_BANK_D0, ENCODER_CONTROL, %x)\n", shadow_encoder_control);
+
+ shadow_soft_mute = options->rec_editable?4:0;
+ mas_writemem(MAS_BANK_D0, MAS_D0_SOFT_MUTE, &shadow_soft_mute,1);
+
+ DEBUGF("mas_writemem(MAS_BANK_D0, SOFT_MUTE, %x)\n", shadow_soft_mute);
+
+ shadow_io_control_main = ((1 << 10) | /* Monitoring ON */
+ ((options->rec_source < 2)?1:2) << 8) | /* Input select */
+ (1 << 5) | /* SDO strobe invert */
+ ((is_mpeg1?0:1) << 3) |
+ (1 << 2) | /* Inverted SIBC clock signal */
+ 1; /* Validate */
+ mas_writemem(MAS_BANK_D0, MAS_D0_IO_CONTROL_MAIN, &shadow_io_control_main,1);
+
+ DEBUGF("mas_writemem(MAS_BANK_D0, IO_CONTROL_MAIN, %x)\n", shadow_io_control_main);
+
+ if(options->rec_source == AUDIO_SRC_MIC)
+ {
+ /* Copy left channel to right (mono mode) */
+ mas_codec_writereg(8, 0x8000);
+ }
+ else
+ {
+ /* Stereo input mode */
+ mas_codec_writereg(8, 0);
+ }
+
+ prerecording_max_seconds = options->rec_prerecord_time;
+ if(prerecording_max_seconds)
+ {
+ prerecording = true;
+ start_prerecording();
+ }
+ else
+ {
+ prerecording = false;
+ is_prerecording = false;
+ is_recording = false;
+ }
+}
+
+/* If use_mic is true, the left gain is used */
+void audio_set_recording_gain(int left, int right, int type)
+{
+ /* Enable both left and right A/D */
+ shadow_codec_reg0 = (left << 12) |
+ (right << 8) |
+ (left << 4) |
+ (type==AUDIO_GAIN_MIC?0x0008:0) | /* Connect left A/D to mic */
+ 0x0007;
+ mas_codec_writereg(0x0, shadow_codec_reg0);
+}
+
+#if CONFIG_TUNER & S1A0903X01
+/* Get the (unpitched) MAS PLL frequency, for avoiding FM interference with the
+ * Samsung tuner. Zero means unknown. Currently handles recording from analog
+ * input only. */
+int mpeg_get_mas_pllfreq(void)
+{
+ if (mpeg_mode != MPEG_ENCODER)
+ return 0;
+
+ if (rec_frequency_index == 0) /* 44.1 kHz / 22.05 kHz */
+ return 22579000;
+ else
+ return 24576000;
+}
+#endif /* CONFIG_TUNER & S1A0903X01 */
+
+/* try to make some kind of beep, also in recording mode */
+void audio_beep(int duration)
+{
+ long starttick = current_tick;
+ do
+ { /* toggle bit 0 of codec register 0, toggling the DAC off & on.
+ * While this is still audible even without an external signal,
+ * it doesn't affect the (pre-)recording. */
+ mas_codec_writereg(0, shadow_codec_reg0 ^ 1);
+ mas_codec_writereg(0, shadow_codec_reg0);
+ yield();
+ }
+ while (current_tick - starttick < duration);
+}
+
+void audio_new_file(const char *filename)
+{
+ mpeg_errno = 0;
+
+ strncpy(recording_filename, filename, MAX_PATH - 1);
+ recording_filename[MAX_PATH - 1] = 0;
+
+ queue_post(&mpeg_queue, MPEG_NEW_FILE, 0);
+}
+
+unsigned long audio_recorded_time(void)
+{
+ if(is_prerecording)
+ return prerecord_count * HZ;
+
+ if(is_recording)
+ {
+ if(paused)
+ return pause_start_time - record_start_time;
+ else
+ return current_tick - record_start_time;
+ }
+
+ return 0;
+}
+
+unsigned long audio_num_recorded_bytes(void)
+{
+ int num_bytes;
+ int index;
+
+ if(is_recording)
+ {
+ if(is_prerecording)
+ {
+ index = prerecord_index - prerecord_count;
+ if(index < 0)
+ index += prerecording_max_seconds;
+
+ num_bytes = audiobuf_write - prerecord_buffer[index].mempos;
+ if(num_bytes < 0)
+ num_bytes += audiobuflen;
+
+ return num_bytes;;
+ }
+ else
+ return num_rec_bytes;
+ }
+ else
+ return 0;
+}
+
+#else /* SIMULATOR */
+
+/* dummies coming up */
+
+void audio_init_playback(void)
+{
+ /* a dummy */
+}
+unsigned long audio_recorded_time(void)
+{
+ /* a dummy */
+ return 0;
+}
+void audio_beep(int duration)
+{
+ /* a dummy */
+ (void)duration;
+}
+void audio_pause_recording(void)
+{
+ /* a dummy */
+}
+void audio_resume_recording(void)
+{
+ /* a dummy */
+}
+unsigned long audio_num_recorded_bytes(void)
+{
+ /* a dummy */
+ return 0;
+}
+void audio_record(const char *filename)
+{
+ /* a dummy */
+ (void)filename;
+}
+void audio_new_file(const char *filename)
+{
+ /* a dummy */
+ (void)filename;
+}
+
+void audio_set_recording_gain(int left, int right, int type)
+{
+ /* a dummy */
+ (void)left;
+ (void)right;
+ (void)type;
+}
+void audio_init_recording(unsigned int buffer_offset)
+{
+ /* a dummy */
+ (void)buffer_offset;
+}
+void audio_set_recording_options(struct audio_recording_options *options)
+{
+ /* a dummy */
+ (void)options;
+}
+#endif /* SIMULATOR */
+#endif /* CONFIG_CODEC == MAS3587F */
+
+void audio_play(long offset)
+{
+#ifdef SIMULATOR
+ char* trackname;
+ int steps=0;
+
+ is_playing = true;
+
+ do {
+ trackname = playlist_peek( steps );
+ if (!trackname)
+ break;
+ if(mp3info(&taginfo, trackname)) {
+ /* bad mp3, move on */
+ if(++steps > playlist_amount())
+ break;
+ continue;
+ }
+#ifdef HAVE_MPEG_PLAY
+ real_mpeg_play(trackname);
+#endif
+ playlist_next(steps);
+ taginfo.offset = offset;
+ set_elapsed(&taginfo);
+ is_playing = true;
+ playing = true;
+ break;
+ } while(1);
+#else /* !SIMULATOR */
+ is_playing = true;
+
+ queue_post(&mpeg_queue, MPEG_PLAY, offset);
+#endif /* !SIMULATOR */
+
+ mpeg_errno = 0;
+}
+
+void audio_stop(void)
+{
+#ifndef SIMULATOR
+ if (playing)
+ {
+ struct trackdata *track = get_trackdata(0);
+ prev_track_elapsed = track->id3.elapsed;
+ }
+ mpeg_stop_done = false;
+ queue_post(&mpeg_queue, MPEG_STOP, 0);
+ while(!mpeg_stop_done)
+ yield();
+#else /* SIMULATOR */
+ paused = false;
+ is_playing = false;
+ playing = false;
+#endif /* SIMULATOR */
+}
+
+/* dummy */
+void audio_stop_recording(void)
+{
+ audio_stop();
+}
+
+void audio_pause(void)
+{
+#ifndef SIMULATOR
+ queue_post(&mpeg_queue, MPEG_PAUSE, 0);
+#else /* SIMULATOR */
+ is_playing = true;
+ playing = false;
+ paused = true;
+#endif /* SIMULATOR */
+}
+
+void audio_resume(void)
+{
+#ifndef SIMULATOR
+ queue_post(&mpeg_queue, MPEG_RESUME, 0);
+#else /* SIMULATOR */
+ is_playing = true;
+ playing = true;
+ paused = false;
+#endif /* SIMULATOR */
+}
+
+void audio_next(void)
+{
+#ifndef SIMULATOR
+ queue_remove_from_head(&mpeg_queue, MPEG_NEED_DATA);
+ queue_post(&mpeg_queue, MPEG_NEXT, 0);
+#else /* SIMULATOR */
+ char* file;
+ int steps = 1;
+ int index;
+
+ do {
+ file = playlist_peek(steps);
+ if(!file)
+ break;
+ if(mp3info(&taginfo, file)) {
+ if(++steps > playlist_amount())
+ break;
+ continue;
+ }
+ index = playlist_next(steps);
+ taginfo.index = index;
+ current_track_counter++;
+ is_playing = true;
+ playing = true;
+ break;
+ } while(1);
+#endif /* SIMULATOR */
+}
+
+void audio_prev(void)
+{
+#ifndef SIMULATOR
+ queue_remove_from_head(&mpeg_queue, MPEG_NEED_DATA);
+ queue_post(&mpeg_queue, MPEG_PREV, 0);
+#else /* SIMULATOR */
+ char* file;
+ int steps = -1;
+ int index;
+
+ do {
+ file = playlist_peek(steps);
+ if(!file)
+ break;
+ if(mp3info(&taginfo, file)) {
+ steps--;
+ continue;
+ }
+ index = playlist_next(steps);
+ taginfo.index = index;
+ current_track_counter++;
+ is_playing = true;
+ playing = true;
+ break;
+ } while(1);
+#endif /* SIMULATOR */
+}
+
+void audio_ff_rewind(long newtime)
+{
+#ifndef SIMULATOR
+ queue_post(&mpeg_queue, MPEG_FF_REWIND, newtime);
+#else /* SIMULATOR */
+ (void)newtime;
+#endif /* SIMULATOR */
+}
+
+void audio_flush_and_reload_tracks(void)
+{
+#ifndef SIMULATOR
+ queue_post(&mpeg_queue, MPEG_FLUSH_RELOAD, 0);
+#endif /* !SIMULATOR*/
+}
+
+int audio_status(void)
+{
+ int ret = 0;
+
+ if(is_playing)
+ ret |= AUDIO_STATUS_PLAY;
+
+ if(paused)
+ ret |= AUDIO_STATUS_PAUSE;
+
+#if (CONFIG_CODEC == MAS3587F) && !defined(SIMULATOR)
+ if(is_recording && !is_prerecording)
+ ret |= AUDIO_STATUS_RECORD;
+
+ if(is_prerecording)
+ ret |= AUDIO_STATUS_PRERECORD;
+#endif /* CONFIG_CODEC == MAS3587F */
+
+ if(mpeg_errno)
+ ret |= AUDIO_STATUS_ERROR;
+
+ return ret;
+}
+
+unsigned int audio_error(void)
+{
+ return mpeg_errno;
+}
+
+void audio_error_clear(void)
+{
+ mpeg_errno = 0;
+}
+
+#ifdef SIMULATOR
+static void mpeg_thread(void)
+{
+ struct mp3entry* id3;
+ while ( 1 ) {
+ if (is_playing) {
+ id3 = audio_current_track();
+ if (!paused)
+ {
+ id3->elapsed+=1000;
+ id3->offset+=1000;
+ }
+ if (id3->elapsed>=id3->length)
+ audio_next();
+ }
+ sleep(HZ);
+ }
+}
+#endif /* SIMULATOR */
+
+void audio_init(void)
+{
+ mpeg_errno = 0;
+
+#ifndef SIMULATOR
+ audiobuflen = audiobufend - audiobuf;
+ queue_init(&mpeg_queue, true);
+#endif /* !SIMULATOR */
+ create_thread(mpeg_thread, mpeg_stack,
+ sizeof(mpeg_stack), 0, mpeg_thread_name
+ IF_PRIO(, PRIORITY_SYSTEM)
+ IF_COP(, CPU));
+
+ memset(trackdata, sizeof(trackdata), 0);
+
+#if (CONFIG_CODEC == MAS3587F) && !defined(SIMULATOR)
+ if (HW_MASK & PR_ACTIVE_HIGH)
+ and_b(~0x08, &PADRH);
+ else
+ or_b(0x08, &PADRH);
+#endif /* CONFIG_CODEC == MAS3587F */
+
+#ifdef DEBUG
+#ifndef SIMULATOR
+ dbg_timer_start();
+ dbg_cnt2us(0);
+#endif /* !SIMULATOR */
+#endif /* DEBUG */
+}
+
+#endif /* CONFIG_CODEC != SWCODEC */
diff --git a/apps/replaygain.c b/apps/replaygain.c
new file mode 100644
index 0000000000..e160a1b23d
--- /dev/null
+++ b/apps/replaygain.c
@@ -0,0 +1,457 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Magnus Holmgren
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include <ctype.h>
+#include <inttypes.h>
+#include <math.h>
+#include <stdbool.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <system.h>
+#include "id3.h"
+#include "debug.h"
+#include "replaygain.h"
+
+/* The fixed point math routines (with the exception of fp_atof) are based
+ * on oMathFP by Dan Carter (http://orbisstudios.com).
+ */
+
+/* 12 bits of precision gives fairly accurate result, but still allows a
+ * compact implementation. The math code supports up to 13...
+ */
+
+#define FP_BITS (12)
+#define FP_MASK ((1 << FP_BITS) - 1)
+#define FP_ONE (1 << FP_BITS)
+#define FP_TWO (2 << FP_BITS)
+#define FP_HALF (1 << (FP_BITS - 1))
+#define FP_LN2 ( 45426 >> (16 - FP_BITS))
+#define FP_LN2_INV ( 94548 >> (16 - FP_BITS))
+#define FP_EXP_ZERO ( 10922 >> (16 - FP_BITS))
+#define FP_EXP_ONE ( -182 >> (16 - FP_BITS))
+#define FP_EXP_TWO ( 4 >> (16 - FP_BITS))
+#define FP_INF (0x7fffffff)
+#define FP_LN10 (150902 >> (16 - FP_BITS))
+
+#define FP_MAX_DIGITS (4)
+#define FP_MAX_DIGITS_INT (10000)
+
+#define FP_FAST_MUL_DIV
+
+#ifdef FP_FAST_MUL_DIV
+
+/* These macros can easily overflow, but they are good enough for our uses,
+ * and saves some code.
+ */
+#define fp_mul(x, y) (((x) * (y)) >> FP_BITS)
+#define fp_div(x, y) (((x) << FP_BITS) / (y))
+
+#else
+
+static long fp_mul(long x, long y)
+{
+ long x_neg = 0;
+ long y_neg = 0;
+ long rc;
+
+ if ((x == 0) || (y == 0))
+ {
+ return 0;
+ }
+
+ if (x < 0)
+ {
+ x_neg = 1;
+ x = -x;
+ }
+
+ if (y < 0)
+ {
+ y_neg = 1;
+ y = -y;
+ }
+
+ rc = (((x >> FP_BITS) * (y >> FP_BITS)) << FP_BITS)
+ + (((x & FP_MASK) * (y & FP_MASK)) >> FP_BITS)
+ + ((x & FP_MASK) * (y >> FP_BITS))
+ + ((x >> FP_BITS) * (y & FP_MASK));
+
+ if ((x_neg ^ y_neg) == 1)
+ {
+ rc = -rc;
+ }
+
+ return rc;
+}
+
+static long fp_div(long x, long y)
+{
+ long x_neg = 0;
+ long y_neg = 0;
+ long shifty;
+ long rc;
+ int msb = 0;
+ int lsb = 0;
+
+ if (x == 0)
+ {
+ return 0;
+ }
+
+ if (y == 0)
+ {
+ return (x < 0) ? -FP_INF : FP_INF;
+ }
+
+ if (x < 0)
+ {
+ x_neg = 1;
+ x = -x;
+ }
+
+ if (y < 0)
+ {
+ y_neg = 1;
+ y = -y;
+ }
+
+ while ((x & (1 << (30 - msb))) == 0)
+ {
+ msb++;
+ }
+
+ while ((y & (1 << lsb)) == 0)
+ {
+ lsb++;
+ }
+
+ shifty = FP_BITS - (msb + lsb);
+ rc = ((x << msb) / (y >> lsb));
+
+ if (shifty > 0)
+ {
+ rc <<= shifty;
+ }
+ else
+ {
+ rc >>= -shifty;
+ }
+
+ if ((x_neg ^ y_neg) == 1)
+ {
+ rc = -rc;
+ }
+
+ return rc;
+}
+
+#endif /* FP_FAST_MUL_DIV */
+
+static long fp_exp(long x)
+{
+ long k;
+ long z;
+ long R;
+ long xp;
+
+ if (x == 0)
+ {
+ return FP_ONE;
+ }
+
+ k = (fp_mul(abs(x), FP_LN2_INV) + FP_HALF) & ~FP_MASK;
+
+ if (x < 0)
+ {
+ k = -k;
+ }
+
+ x -= fp_mul(k, FP_LN2);
+ z = fp_mul(x, x);
+ R = FP_TWO + fp_mul(z, FP_EXP_ZERO + fp_mul(z, FP_EXP_ONE
+ + fp_mul(z, FP_EXP_TWO)));
+ xp = FP_ONE + fp_div(fp_mul(FP_TWO, x), R - x);
+
+ if (k < 0)
+ {
+ k = FP_ONE >> (-k >> FP_BITS);
+ }
+ else
+ {
+ k = FP_ONE << (k >> FP_BITS);
+ }
+
+ return fp_mul(k, xp);
+}
+
+static long fp_exp10(long x)
+{
+ if (x == 0)
+ {
+ return FP_ONE;
+ }
+
+ return fp_exp(fp_mul(FP_LN10, x));
+}
+
+static long fp_atof(const char* s, int precision)
+{
+ long int_part = 0;
+ long int_one = 1 << precision;
+ long frac_part = 0;
+ long frac_count = 0;
+ long frac_max = ((precision * 4) + 12) / 13;
+ long frac_max_int = 1;
+ long sign = 1;
+ bool point = false;
+
+ while ((*s != '\0') && isspace(*s))
+ {
+ s++;
+ }
+
+ if (*s == '-')
+ {
+ sign = -1;
+ s++;
+ }
+ else if (*s == '+')
+ {
+ s++;
+ }
+
+ while (*s != '\0')
+ {
+ if (*s == '.')
+ {
+ if (point)
+ {
+ break;
+ }
+
+ point = true;
+ }
+ else if (isdigit(*s))
+ {
+ if (point)
+ {
+ if (frac_count < frac_max)
+ {
+ frac_part = frac_part * 10 + (*s - '0');
+ frac_count++;
+ frac_max_int *= 10;
+ }
+ }
+ else
+ {
+ int_part = int_part * 10 + (*s - '0');
+ }
+ }
+ else
+ {
+ break;
+ }
+
+ s++;
+ }
+
+ while (frac_count < frac_max)
+ {
+ frac_part *= 10;
+ frac_count++;
+ frac_max_int *= 10;
+ }
+
+ return sign * ((int_part * int_one)
+ + (((int64_t) frac_part * int_one) / frac_max_int));
+}
+
+static long convert_gain(long gain)
+{
+ /* Don't allow unreasonably low or high gain changes.
+ * Our math code can't handle it properly anyway. :)
+ */
+ if (gain < (-48 * FP_ONE))
+ {
+ gain = -48 * FP_ONE;
+ }
+
+ if (gain > (17 * FP_ONE))
+ {
+ gain = 17 * FP_ONE;
+ }
+
+ gain = fp_exp10(gain / 20) << (24 - FP_BITS);
+
+ return gain;
+}
+
+/* Get the sample scale factor in Q7.24 format from a gain value. Returns 0
+ * for no gain.
+ *
+ * str Gain in dB as a string. E.g., "-3.45 dB"; the "dB" part is ignored.
+ */
+static long get_replaygain(const char* str)
+{
+ long gain = 0;
+
+ if (str)
+ {
+ gain = fp_atof(str, FP_BITS);
+ gain = convert_gain(gain);
+ }
+
+ return gain;
+}
+
+/* Get the peak volume in Q7.24 format.
+ *
+ * str Peak volume. Full scale is specified as "1.0". Returns 0 for no peak.
+ */
+static long get_replaypeak(const char* str)
+{
+ long peak = 0;
+
+ if (str)
+ {
+ peak = fp_atof(str, 24);
+ }
+
+ return peak;
+}
+
+/* Get a sample scale factor in Q7.24 format from a gain value.
+ *
+ * int_gain Gain in dB, multiplied by 100.
+ */
+long get_replaygain_int(long int_gain)
+{
+ return convert_gain(int_gain * FP_ONE / 100);
+}
+
+/* Parse a ReplayGain tag conforming to the "VorbisGain standard". If a
+ * valid tag is found, update mp3entry struct accordingly. Existing values
+ * are not overwritten. Returns number of bytes written to buffer.
+ *
+ * key Name of the tag.
+ * value Value of the tag.
+ * entry mp3entry struct to update.
+ * buffer Where to store the text for gain values (for later display).
+ * length Bytes left in buffer.
+ */
+long parse_replaygain(const char* key, const char* value,
+ struct mp3entry* entry, char* buffer, int length)
+{
+ char **p = NULL;
+
+ if (((strcasecmp(key, "replaygain_track_gain") == 0)
+ || (strcasecmp(key, "rg_radio") == 0)) && !entry->track_gain)
+ {
+ entry->track_gain = get_replaygain(value);
+ p = &(entry->track_gain_string);
+ }
+ else if (((strcasecmp(key, "replaygain_album_gain") == 0)
+ || (strcasecmp(key, "rg_audiophile") == 0)) && !entry->album_gain)
+ {
+ entry->album_gain = get_replaygain(value);
+ p = &(entry->album_gain_string);
+ }
+ else if (((strcasecmp(key, "replaygain_track_peak") == 0)
+ || (strcasecmp(key, "rg_peak") == 0)) && !entry->track_peak)
+ {
+ entry->track_peak = get_replaypeak(value);
+ }
+ else if ((strcasecmp(key, "replaygain_album_peak") == 0)
+ && !entry->album_peak)
+ {
+ entry->album_peak = get_replaypeak(value);
+ }
+
+ if (p)
+ {
+ int len = strlen(value);
+
+ len = MIN(len, length - 1);
+
+ /* A few characters just isn't interesting... */
+ if (len > 1)
+ {
+ strncpy(buffer, value, len);
+ buffer[len] = 0;
+ *p = buffer;
+ return len + 1;
+ }
+ }
+
+ return 0;
+}
+
+/* Set ReplayGain values from integers. Existing values are not overwritten.
+ * Returns number of bytes written to buffer.
+ *
+ * album If true, set album values, otherwise set track values.
+ * gain Gain value in dB, multiplied by 512. 0 for no gain.
+ * peak Peak volume in Q7.24 format, where 1.0 is full scale. 0 for no
+ * peak volume.
+ * buffer Where to store the text for gain values (for later display).
+ * length Bytes left in buffer.
+ */
+long parse_replaygain_int(bool album, long gain, long peak,
+ struct mp3entry* entry, char* buffer, int length)
+{
+ long len = 0;
+
+ if (buffer != NULL)
+ {
+ len = snprintf(buffer, length, "%d.%02d dB", gain / 512,
+ ((abs(gain) & 0x01ff) * 100 + 256) / 512);
+ len++;
+ }
+
+ if (gain != 0)
+ {
+ gain = convert_gain(gain * FP_ONE / 512);
+ }
+
+ if (album)
+ {
+ entry->album_gain = gain;
+ entry->album_gain_string = buffer;
+
+ if (peak)
+ {
+ entry->album_peak = peak;
+ }
+ }
+ else
+ {
+ entry->track_gain = gain;
+ entry->track_gain_string = buffer;
+
+ if (peak)
+ {
+ entry->track_peak = peak;
+ }
+ }
+
+ return len;
+}