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rockbox/apps/playback.c
Dave Chapman a1d75d3c79 Calculate accurate Ogg duration and bitrate 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6687 a1c6a512-1295-4272-9138-f99709370657
2005-06-12 12:46:36 +00:00

1626 lines
43 KiB
C
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <ctype.h>
#include "system.h"
#include "thread.h"
#include "file.h"
#include "lcd.h"
#include "font.h"
#include "backlight.h"
#include "button.h"
#include "kernel.h"
#include "tree.h"
#include "debug.h"
#include "sprintf.h"
#include "settings.h"
#include "plugin.h"
#include "wps.h"
#include "wps-display.h"
#include "audio.h"
#include "logf.h"
#include "mp3_playback.h"
#include "mp3data.h"
#include "usb.h"
#include "status.h"
#include "main_menu.h"
#include "ata.h"
#include "screens.h"
#include "playlist.h"
#include "playback.h"
#include "pcm_playback.h"
#include "buffer.h"
#ifdef HAVE_LCD_BITMAP
#include "icons.h"
#include "peakmeter.h"
#include "action.h"
#endif
#include "lang.h"
#include "bookmark.h"
#include "misc.h"
#include "sound.h"
static volatile bool playing;
static volatile bool paused;
#define CODEC_VORBIS "/.rockbox/codecs/codecvorbis.rock";
#define CODEC_MPA_L3 "/.rockbox/codecs/codecmpa.rock";
#define CODEC_FLAC "/.rockbox/codecs/codecflac.rock";
#define CODEC_WAV "/.rockbox/codecs/codecwav.rock";
#define CODEC_A52 "/.rockbox/codecs/codeca52.rock";
#define CODEC_MPC "/.rockbox/codecs/codecmpc.rock";
#define AUDIO_DEFAULT_WATERMARK (1024*256)
#define AUDIO_DEFAULT_FILECHUNK (1024*32)
#define AUDIO_PLAY 1
#define AUDIO_STOP 2
#define AUDIO_PAUSE 3
#define AUDIO_RESUME 4
#define AUDIO_NEXT 5
#define AUDIO_PREV 6
#define AUDIO_FF_REWIND 7
#define AUDIO_FLUSH_RELOAD 8
#define AUDIO_CODEC_DONE 9
#define CODEC_LOAD 1
#define CODEC_LOAD_DISK 2
/* As defined in plugins/lib/xxx2wav.h */
#define MALLOC_BUFSIZE (512*1024)
#define GUARD_BUFSIZE (8*1024)
/* TODO:
Handle playlist_peek in mpeg.c
Track changing
*/
extern bool audio_is_initialized;
/* Buffer control thread. */
static struct event_queue audio_queue;
static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
static const char audio_thread_name[] = "audio";
/* Codec thread. */
static struct event_queue codec_queue;
static long codec_stack[(DEFAULT_STACK_SIZE + 0x2500)/sizeof(long)] __attribute__ ((section(".idata")));
static const char codec_thread_name[] = "codec";
/* Is file buffer currently being refilled? */
static volatile bool filling;
/* Ring buffer where tracks and codecs are loaded. */
char *codecbuf;
/* Total size of the ring buffer. */
int codecbuflen;
/* Bytes available in the buffer. */
static volatile int codecbufused;
/* Ring buffer read and write indexes. */
static volatile int buf_ridx;
static volatile int buf_widx;
#define MAX_TRACK 10
struct track_info {
struct mp3entry id3;
struct mp3info mp3data;
char *codecbuf;
size_t codecsize;
int codectype;
volatile char *filebuf;
off_t filerem;
off_t filesize;
off_t filepos;
volatile int available;
bool taginfo_ready;
int playlist_offset;
};
/* Track information (count in file buffer, read/write indexes for
track ring structure. */
static int track_count;
static volatile int track_ridx;
static volatile int track_widx;
static bool track_changed;
/* Partially loaded song's file handle to continue buffering later. */
static int current_fd;
/* Information about how many bytes left on the buffer re-fill run. */
static size_t fill_bytesleft;
/* Track info structure about songs in the file buffer. */
static struct track_info tracks[MAX_TRACK];
/* Pointer to track info structure about current song playing. */
static volatile struct track_info *cur_ti;
/* Codec API including function callbacks. */
static struct codec_api ci;
/* When we change a song and buffer is not in filling state, this
variable keeps information about whether to go a next/previous track. */
static int new_track;
/* Configuration */
static int conf_bufferlimit;
static int conf_watermark;
static int conf_filechunk;
static bool v1first = false;
static void mp3_set_elapsed(struct mp3entry* id3);
int mp3_get_file_pos(void);
#ifdef SIMULATOR
bool audiobuffer_insert_sim(char *buf, size_t length)
{
(void)buf;
(void)length;
return true;
}
#endif
void* get_codec_memory_callback(size_t *size)
{
*size = MALLOC_BUFSIZE;
return &audiobuf[0];
}
void codec_set_elapsed_callback(unsigned int value)
{
unsigned int latency;
#ifndef SIMULATOR
latency = audiobuffer_get_latency();
#else
latency = 0;
#endif
if (value < latency) {
cur_ti->id3.elapsed = 0;
} else if (value - latency > cur_ti->id3.elapsed
|| value - latency < cur_ti->id3.elapsed - 2) {
cur_ti->id3.elapsed = value - latency;
}
}
size_t codec_filebuf_callback(void *ptr, size_t size)
{
char *buf = (char *)ptr;
int copy_n;
int part_n;
if (ci.stop_codec || !playing)
return 0;
copy_n = MIN((off_t)size, (off_t)cur_ti->available + cur_ti->filerem);
while (copy_n > cur_ti->available) {
yield();
if (ci.stop_codec)
return 0;
}
if (copy_n == 0)
return 0;
part_n = MIN(copy_n, codecbuflen - buf_ridx);
memcpy(buf, &codecbuf[buf_ridx], part_n);
if (part_n < copy_n) {
memcpy(&buf[part_n], &codecbuf[0], copy_n - part_n);
}
buf_ridx += copy_n;
if (buf_ridx >= codecbuflen)
buf_ridx -= codecbuflen;
ci.curpos += copy_n;
cur_ti->available -= copy_n;
codecbufused -= copy_n;
return copy_n;
}
void* codec_request_buffer_callback(size_t *realsize, size_t reqsize)
{
size_t part_n;
if (ci.stop_codec || !playing) {
*realsize = 0;
return NULL;
}
*realsize = MIN((off_t)reqsize, (off_t)cur_ti->available + cur_ti->filerem);
if (*realsize == 0) {
return NULL;
}
while ((int)*realsize > cur_ti->available) {
yield();
if (ci.stop_codec) {
*realsize = 0;
return NULL;
}
}
part_n = MIN((int)*realsize, codecbuflen - buf_ridx);
if (part_n < *realsize) {
part_n += GUARD_BUFSIZE;
if (part_n < *realsize)
*realsize = part_n;
memcpy(&codecbuf[codecbuflen], &codecbuf[0], *realsize -
(codecbuflen - buf_ridx));
}
return (char *)&codecbuf[buf_ridx];
}
void codec_advance_buffer_callback(size_t amount)
{
if ((int)amount > cur_ti->available + cur_ti->filerem)
amount = cur_ti->available + cur_ti->filerem;
if ((int)amount > cur_ti->available) {
codecbufused = 0;
buf_ridx = buf_widx;
cur_ti->available = 0;
while ((int)amount < cur_ti->available && !ci.stop_codec)
yield();
} else {
cur_ti->available -= amount;
codecbufused -= amount;
buf_ridx += amount;
if (buf_ridx >= codecbuflen)
buf_ridx -= codecbuflen;
}
ci.curpos += amount;
cur_ti->id3.offset = ci.curpos;
}
void codec_advance_buffer_loc_callback(void *ptr)
{
size_t amount;
amount = (int)ptr - (int)&codecbuf[buf_ridx];
codec_advance_buffer_callback(amount);
}
off_t codec_mp3_get_filepos_callback(int newtime)
{
off_t newpos;
cur_ti->id3.elapsed = newtime;
newpos = mp3_get_file_pos();
return newpos;
}
bool codec_seek_buffer_callback(off_t newpos)
{
int difference;
if (newpos < 0)
newpos = 0;
if (newpos >= cur_ti->filesize)
newpos = cur_ti->filesize - 1;
difference = newpos - ci.curpos;
if (difference >= 0) {
logf("seek: +%d", difference);
codec_advance_buffer_callback(difference);
#ifndef SIMULATOR
pcm_play_stop();
#endif
return true;
}
difference = -difference;
if (ci.curpos - difference < 0)
difference = ci.curpos;
if (codecbufused + difference > codecbuflen) {
/* We need to reload the song. FIX THIS! */
return false;
}
logf("seek: -%d", difference);
codecbufused += difference;
cur_ti->available += difference;
buf_ridx -= difference;
if (buf_ridx < 0)
buf_ridx = codecbuflen + buf_ridx;
ci.curpos -= difference;
#ifndef SIMULATOR
pcm_play_stop();
#endif
return true;
}
void codec_configure_callback(int setting, void *value)
{
switch (setting) {
case CODEC_SET_FILEBUF_WATERMARK:
conf_watermark = (unsigned int)value;
break;
case CODEC_SET_FILEBUF_CHUNKSIZE:
conf_filechunk = (unsigned int)value;
break;
case CODEC_SET_FILEBUF_LIMIT:
conf_bufferlimit = (unsigned int)value;
break;
default:
logf("Illegal key: %d", setting);
}
}
/* Simple file type probing by looking filename extension. */
int probe_file_format(const char *filename)
{
char *suffix;
suffix = strrchr(filename, '.');
if (suffix == NULL)
return AFMT_UNKNOWN;
suffix += 1;
if (!strcasecmp("mp1", suffix))
return AFMT_MPA_L1;
else if (!strcasecmp("mp2", suffix))
return AFMT_MPA_L2;
else if (!strcasecmp("mpa", suffix))
return AFMT_MPA_L2;
else if (!strcasecmp("mp3", suffix))
return AFMT_MPA_L3;
else if (!strcasecmp("ogg", suffix))
return AFMT_OGG_VORBIS;
else if (!strcasecmp("wav", suffix))
return AFMT_PCM_WAV;
else if (!strcasecmp("flac", suffix))
return AFMT_FLAC;
else if (!strcasecmp("mpc", suffix))
return AFMT_MPC;
else if (!strcasecmp("aac", suffix))
return AFMT_AAC;
else if (!strcasecmp("ape", suffix))
return AFMT_APE;
else if (!strcasecmp("wma", suffix))
return AFMT_WMA;
else if ((!strcasecmp("a52", suffix)) || (!strcasecmp("ac3", suffix)))
return AFMT_A52;
else if (!strcasecmp("rm", suffix))
return AFMT_REAL;
return AFMT_UNKNOWN;
}
void yield_codecs(void)
{
yield();
#ifndef SIMULATOR
if (!pcm_is_playing())
sleep(5);
while (pcm_is_lowdata() && !ci.stop_codec &&
playing && queue_empty(&audio_queue))
yield();
#endif
}
void audio_fill_file_buffer(void)
{
size_t i;
int rc;
logf("Filling buffer...");
i = 0;
while ((off_t)i < tracks[track_widx].filerem) {
/* Give codecs some processing time. */
yield_codecs();
if (!queue_empty(&audio_queue)) {
logf("Filling interrupted");
close(current_fd);
current_fd = -1;
return ;
}
if (fill_bytesleft < MIN((unsigned int)conf_filechunk,
tracks[track_widx].filerem - i))
break ;
rc = MIN(conf_filechunk, codecbuflen - buf_widx);
rc = read(current_fd, &codecbuf[buf_widx], rc);
if (rc <= 0) {
tracks[track_widx].filerem = 0;
break ;
}
buf_widx += rc;
if (buf_widx >= codecbuflen)
buf_widx -= codecbuflen;
i += rc;
tracks[track_widx].available += rc;
fill_bytesleft -= rc;
}
tracks[track_widx].filerem -= i;
codecbufused += i;
tracks[track_widx].filepos += i;
logf("Done:%d", tracks[track_widx].available);
if (tracks[track_widx].filerem == 0) {
if (++track_widx == MAX_TRACK)
track_widx = 0;
tracks[track_widx].filerem = 0;
close(current_fd);
current_fd = -1;
}
}
bool loadcodec(const char *trackname, bool start_play)
{
char msgbuf[80];
off_t size;
int filetype;
int fd;
int i, rc;
const char *codec_path;
int copy_n;
int prev_track;
filetype = probe_file_format(trackname);
switch (filetype) {
case AFMT_OGG_VORBIS:
logf("Codec: Vorbis");
codec_path = CODEC_VORBIS;
break;
case AFMT_MPA_L2:
case AFMT_MPA_L3:
logf("Codec: MPA L2/L3");
codec_path = CODEC_MPA_L3;
break;
case AFMT_PCM_WAV:
logf("Codec: PCM WAV");
codec_path = CODEC_WAV;
break;
case AFMT_FLAC:
logf("Codec: FLAC");
codec_path = CODEC_FLAC;
break;
case AFMT_A52:
logf("Codec: A52");
codec_path = CODEC_A52;
break;
case AFMT_MPC:
logf("Codec: Musepack");
codec_path = CODEC_MPC;
break;
default:
logf("Codec: Unsupported");
snprintf(msgbuf, sizeof(msgbuf)-1, "No codec for: %s", trackname);
splash(HZ*2, true, msgbuf);
codec_path = NULL;
}
tracks[track_widx].codectype = filetype;
tracks[track_widx].codecsize = 0;
if (codec_path == NULL)
return false;
if (!start_play) {
prev_track = track_widx - 1;
if (prev_track < 0)
prev_track = MAX_TRACK-1;
if (track_count > 0 && filetype == tracks[prev_track].codectype) {
logf("Reusing prev. codec");
return true;
}
} else {
/* Load the codec directly from disk and save some memory. */
cur_ti = &tracks[track_widx];
ci.filesize = cur_ti->filesize;
ci.id3 = (struct mp3entry *)&cur_ti->id3;
ci.mp3data = (struct mp3info *)&cur_ti->mp3data;
ci.taginfo_ready = (bool *)&cur_ti->taginfo_ready;
ci.curpos = 0;
playing = true;
logf("Starting codec");
queue_post(&codec_queue, CODEC_LOAD_DISK, (void *)codec_path);
return true;
}
fd = open(codec_path, O_RDONLY);
if (fd < 0) {
logf("Codec doesn't exist!");
snprintf(msgbuf, sizeof(msgbuf)-1, "Couldn't load codec: %s", codec_path);
splash(HZ*2, true, msgbuf);
return false;
}
size = filesize(fd);
if ((off_t)fill_bytesleft < size + conf_watermark) {
logf("Not enough space");
close(fd);
return false;
}
i = 0;
while (i < size) {
yield_codecs();
if (!queue_empty(&audio_queue)) {
logf("Buffering interrupted");
close(fd);
return false;
}
copy_n = MIN(conf_filechunk, codecbuflen - buf_widx);
rc = read(fd, &codecbuf[buf_widx], copy_n);
if (rc < 0)
return false;
buf_widx += rc;
if (buf_widx >= codecbuflen)
buf_widx -= codecbuflen;
i += rc;
}
close(fd);
logf("Done: %dB", i);
codecbufused += size;
fill_bytesleft -= size;
tracks[track_widx].codecsize = size;
return true;
}
bool audio_load_track(int offset, bool start_play, int peek_offset)
{
char *trackname;
int fd;
off_t size;
int rc, i;
int copy_n;
/* Used by the metadata parsers */
unsigned long totalsamples,bytespersample,channels,bitspersample,numbytes;
unsigned char* buf;
int j,eof;
if (track_count >= MAX_TRACK)
return false;
trackname = playlist_peek(peek_offset);
if (!trackname) {
return false;
}
fd = open(trackname, O_RDONLY);
if (fd < 0)
return false;
size = filesize(fd);
tracks[track_widx].filerem = size;
tracks[track_widx].filesize = size;
tracks[track_widx].filepos = 0;
tracks[track_widx].available = 0;
tracks[track_widx].taginfo_ready = false;
tracks[track_widx].playlist_offset = offset;
/* Load the codec */
if (buf_widx >= codecbuflen)
buf_widx -= codecbuflen;
/* Set default values */
if (start_play) {
conf_bufferlimit = 0;
conf_watermark = AUDIO_DEFAULT_WATERMARK;
conf_filechunk = AUDIO_DEFAULT_FILECHUNK;
}
tracks[track_widx].codecbuf = &codecbuf[buf_widx];
if (!loadcodec(trackname, start_play)) {
close(fd);
return false;
}
tracks[track_widx].filebuf = &codecbuf[buf_widx];
//logf("%s", trackname);
logf("Buffering track:%d/%d", track_widx, track_ridx);
if (!queue_empty(&audio_queue)) {
logf("Interrupted!");
ci.stop_codec = true;
close(fd);
return false;
}
/* Load codec specific track tag information. */
switch (tracks[track_widx].codectype) {
case AFMT_MPA_L2:
case AFMT_MPA_L3:
/* Should check the return value. */
mp3info(&tracks[track_widx].id3, trackname, v1first);
lseek(fd, 0, SEEK_SET);
/* This is too slow to execute on some files. */
// get_mp3file_info(fd, &tracks[track_widx].mp3data);
if (offset) {
lseek(fd, offset, SEEK_SET);
tracks[track_widx].id3.offset = offset;
mp3_set_elapsed(&tracks[track_widx].id3);
tracks[track_widx].filepos = offset;
tracks[track_widx].filerem = tracks[track_widx].filesize - offset;
ci.curpos = offset;
} else {
lseek(fd, 0, SEEK_SET);
}
/*
logf("T:%s", tracks[track_widx].id3.title);
logf("L:%d", tracks[track_widx].id3.length);
logf("O:%d", tracks[track_widx].id3.first_frame_offset);
logf("F:%d", tracks[track_widx].id3.frequency);
*/
tracks[track_widx].taginfo_ready = true;
break ;
case AFMT_PCM_WAV:
/* Use the trackname part of the id3 structure as a temporary buffer */
buf=tracks[track_widx].id3.path;
lseek(fd, 0, SEEK_SET);
rc = read(fd, buf, 44);
if (rc < 44) {
close(fd);
return false;
}
if ((memcmp(buf,"RIFF",4)!=0) ||
(memcmp(&buf[8],"WAVEfmt",7)!=0)) {
logf("%s is not a WAV file\n",trackname);
close(fd);
return(false);
}
/* FIX: Correctly parse WAV header - we assume canonical
44-byte header */
bitspersample=buf[34];
channels=buf[22];
if ((bitspersample!=16) || (channels != 2)) {
logf("Unsupported WAV file - %d bitspersample, %d channels\n",
bitspersample,channels);
close(fd);
return(false);
}
bytespersample=((bitspersample/8)*channels);
numbytes=(buf[40]|(buf[41]<<8)|(buf[42]<<16)|(buf[43]<<24));
totalsamples=numbytes/bytespersample;
tracks[track_widx].id3.vbr=false; /* All WAV files are CBR */
tracks[track_widx].id3.filesize=filesize(fd);
tracks[track_widx].id3.frequency=buf[24]|(buf[25]<<8)|(buf[26]<<16)|(buf[27]<<24);
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
tracks[track_widx].id3.length=(totalsamples/tracks[track_widx].id3.frequency)*1000;
tracks[track_widx].id3.bitrate=(tracks[track_widx].id3.frequency*bytespersample)/(1000/8);
lseek(fd, 0, SEEK_SET);
strncpy(tracks[track_widx].id3.path,trackname,sizeof(tracks[track_widx].id3.path));
tracks[track_widx].taginfo_ready = true;
break;
case AFMT_FLAC:
/* A simple parser to read vital metadata from a FLAC file - length, frequency, bitrate etc. */
/* This code should either be moved to a seperate file, or discarded in favour of the libFLAC code */
/* The FLAC stream specification can be found at http://flac.sourceforge.net/format.html#stream */
/* Use the trackname part of the id3 structure as a temporary buffer */
buf=tracks[track_widx].id3.path;
lseek(fd, 0, SEEK_SET);
rc = read(fd, buf, 4);
if (rc < 4) {
close(fd);
return false;
}
if (memcmp(buf,"fLaC",4)!=0) {
logf("%s is not a FLAC file\n",trackname);
close(fd);
return(false);
}
while (1) {
rc = read(fd, buf, 4);
i = (buf[1]<<16)|(buf[2]<<8)|buf[3]; /* The length of the block */
if ((buf[0]&0x7f)==0) { /* 0 is the STREAMINFO block */
rc = read(fd, buf, i); /* FIXME: Don't trust the value of i */
if (rc < 0) {
close(fd);
return false;
}
tracks[track_widx].id3.vbr=true; /* All FLAC files are VBR */
tracks[track_widx].id3.filesize=filesize(fd);
tracks[track_widx].id3.frequency=(buf[10]<<12)|(buf[11]<<4)|((buf[12]&0xf0)>>4);
/* NOT NEEDED: bitspersample=(((buf[12]&0x01)<<4)|((buf[13]&0xf0)>>4))+1; */
/* totalsamples is a 36-bit field, but we assume <= 32 bits are used */
totalsamples=(buf[14]<<24)|(buf[15]<<16)|(buf[16]<<8)|buf[17];
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
tracks[track_widx].id3.length=(totalsamples/tracks[track_widx].id3.frequency)*1000;
tracks[track_widx].id3.bitrate=(filesize(fd)*8)/tracks[track_widx].id3.length;
} else if ((buf[0]&0x7f)==4) { /* 4 is the VORBIS_COMMENT block */
/* The next i bytes of the file contain the VORBIS COMMENTS - just skip them for now. */
lseek(fd, i, SEEK_CUR);
} else {
if (buf[0]&0x80) { /* If we have reached the last metadata block, abort. */
break;
} else {
lseek(fd, i, SEEK_CUR); /* Skip to next metadata block */
}
}
}
lseek(fd, 0, SEEK_SET);
strncpy(tracks[track_widx].id3.path,trackname,sizeof(tracks[track_widx].id3.path));
tracks[track_widx].taginfo_ready = true;
break;
case AFMT_OGG_VORBIS:
/* A simple parser to read vital metadata from an Ogg Vorbis file */
/* An Ogg File is split into pages, each starting with the string
"OggS". Each page has a timestamp (in PCM samples) referred to as
the "granule position".
An Ogg Vorbis has the following structure:
1) Identification header (containing samplerate, numchannels, etc)
2) Comment header - containing the Vorbis Comments
3) Setup header - containing codec setup information
4) Many audio packets...
*/
/* Use the trackname part of the id3 structure as a temporary buffer */
buf=tracks[track_widx].id3.path;
lseek(fd, 0, SEEK_SET);
rc = read(fd, buf, 58);
if (rc < 4) {
close(fd);
return false;
}
if ((memcmp(buf,"OggS",4)!=0) || (memcmp(&buf[29],"vorbis",6)!=0)) {
logf("%s is not an Ogg Vorbis file\n",trackname);
close(fd);
return(false);
}
/* Ogg stores integers in little-endian format. */
tracks[track_widx].id3.filesize=filesize(fd);
tracks[track_widx].id3.frequency=buf[40]|(buf[41]<<8)|(buf[42]<<16)|(buf[43]<<24);
channels=buf[39];
/* We now need to search for the last page in the file - identified by
by ('O','g','g','S',0) and retrieve totalsamples */
lseek(fd, -32*1024, SEEK_END);
eof=0;
j=0; /* The number of bytes currently in buffer */
i=0;
totalsamples=0;
while (!eof) {
rc = read(fd, &buf[j], MAX_PATH-j);
if (rc <= 0) {
eof=1;
} else {
j+=rc;
}
/* Inefficient (but simple) search */
i=0;
while (i < (j-5)) {
if (memcmp(&buf[i],"OggS",5)==0) {
if (i < (j-10)) {
totalsamples=(buf[i+6])|(buf[i+7]<<8)|(buf[i+8]<<16)|(buf[i+9]<<24);
j=0; /* We can discard the rest of the buffer */
} else {
break;
}
} else {
i++;
}
}
if (i < (j-5)) {
/* Move OggS to start of buffer */
while(i>0) buf[i--]=buf[j--];
} else {
j=0;
}
}
tracks[track_widx].id3.length=(totalsamples/tracks[track_widx].id3.frequency)*1000;
/* The following calculation should use datasize, not filesize (i.e. exclude comments etc) */
tracks[track_widx].id3.bitrate=(filesize(fd)*8)/tracks[track_widx].id3.length;
tracks[track_widx].id3.vbr=true;
lseek(fd, 0, SEEK_SET);
strncpy(tracks[track_widx].id3.path,trackname,sizeof(tracks[track_widx].id3.path));
tracks[track_widx].taginfo_ready = true;
break;
/* If we don't know how to read the metadata, just store the filename */
default:
strncpy(tracks[track_widx].id3.path,trackname,sizeof(tracks[track_widx].id3.path));
tracks[track_widx].taginfo_ready = true;
break;
}
track_changed = true;
track_count++;
i = tracks[track_widx].filepos;
while (i < size) {
/* Give codecs some processing time to prevent glitches. */
yield_codecs();
/* Limit buffering size at first run. */
if (conf_bufferlimit && (int)fill_bytesleft >= conf_bufferlimit) {
fill_bytesleft = conf_bufferlimit;
conf_bufferlimit = 0;
}
if (!queue_empty(&audio_queue)) {
logf("Buffering interrupted");
close(fd);
return false;
}
if (fill_bytesleft == 0)
break ;
copy_n = MIN(conf_filechunk, codecbuflen - buf_widx);
copy_n = MIN(size - i, copy_n);
copy_n = MIN((int)fill_bytesleft, copy_n);
rc = read(fd, &codecbuf[buf_widx], copy_n);
if (rc < 0) {
logf("File error!");
close(fd);
return false;
}
buf_widx += rc;
if (buf_widx >= codecbuflen)
buf_widx -= codecbuflen;
i += rc;
tracks[track_widx].available += rc;
tracks[track_widx].filerem -= rc;
codecbufused += rc;
fill_bytesleft -= rc;
}
tracks[track_widx].filepos = i;
/* Leave the file handle open for faster buffer refill. */
if (tracks[track_widx].filerem != 0) {
current_fd = fd;
logf("Partially buf:%d", tracks[track_widx].available);
return false;
} else {
logf("Completely buf.");
close(fd);
current_fd = -1;
if (++track_widx >= MAX_TRACK) {
track_widx = 0;
}
tracks[track_widx].filerem = 0;
}
return true;
}
void audio_insert_tracks(int offset, bool start_playing, int peek_offset)
{
fill_bytesleft = codecbuflen - codecbufused;
filling = true;
while (audio_load_track(offset, start_playing, peek_offset)) {
start_playing = false;
offset = 0;
peek_offset++;
}
filling = false;
}
void audio_play_start(int offset)
{
memset(&tracks, 0, sizeof(struct track_info) * MAX_TRACK);
sound_set(SOUND_VOLUME, global_settings.volume);
track_count = 0;
track_widx = 0;
track_ridx = 0;
buf_ridx = 0;
buf_widx = 0;
codecbufused = 0;
#ifndef SIMULATOR
pcm_set_boost_mode(true);
#endif
audio_insert_tracks(offset, true, 0);
#ifndef SIMULATOR
pcm_set_boost_mode(false);
ata_sleep();
#endif
}
void audio_check_buffer(void)
{
int i;
int cur_idx;
/* Fill buffer as full as possible for cross-fader. */
#ifndef SIMULATOR
if (pcm_is_crossfade_enabled() && cur_ti->id3.length > 0
&& cur_ti->id3.length - cur_ti->id3.elapsed < 20000 && playing)
pcm_set_boost_mode(true);
#endif
/* Start buffer filling as necessary. */
if (codecbufused > conf_watermark || !queue_empty(&audio_queue)
|| !playing || ci.stop_codec || ci.reload_codec)
return ;
filling = true;
#ifndef SIMULATOR
pcm_set_boost_mode(true);
#endif
fill_bytesleft = codecbuflen - codecbufused;
/* Calculate real track count after throwing away old tracks. */
cur_idx = track_ridx;
for (i = 0; i < track_count; i++) {
if (cur_idx == track_widx)
break ;
if (++cur_idx >= MAX_TRACK)
cur_idx = 0;
}
track_count = i;
if (tracks[cur_idx].filerem != 0)
track_count++;
/* Mark all other entries null. */
cur_idx = track_widx;
for (i = 0; i < MAX_TRACK - track_count; i++) {
if (++cur_idx >= MAX_TRACK)
cur_idx = 0;
tracks[cur_idx].filesize = 0;
tracks[cur_idx].available = 0;
}
/* Try to load remainings of the file. */
if (tracks[track_widx].filerem > 0)
audio_fill_file_buffer();
/* Load new files to fill the entire buffer. */
if (tracks[track_widx].filerem == 0)
audio_insert_tracks(0, false, 1);
#ifndef SIMULATOR
pcm_set_boost_mode(false);
if (playing)
ata_sleep();
#endif
filling = false;
}
void audio_update_trackinfo(void)
{
if (new_track >= 0) {
buf_ridx += cur_ti->available;
codecbufused -= cur_ti->available;
cur_ti = &tracks[track_ridx];
buf_ridx += cur_ti->codecsize;
codecbufused -= cur_ti->codecsize;
if (buf_ridx >= codecbuflen)
buf_ridx -= codecbuflen;
#ifndef SIMULATOR
pcm_crossfade_start();
if (!filling)
pcm_set_boost_mode(false);
#endif
} else {
buf_ridx -= ci.curpos + cur_ti->codecsize;
codecbufused += ci.curpos + cur_ti->codecsize;
cur_ti->available = cur_ti->filesize;
cur_ti = &tracks[track_ridx];
buf_ridx -= cur_ti->filesize;
codecbufused += cur_ti->filesize;
cur_ti->available = cur_ti->filesize;
if (buf_ridx < 0)
buf_ridx = codecbuflen + buf_ridx;
}
ci.filesize = cur_ti->filesize;
cur_ti->id3.elapsed = 0;
cur_ti->id3.offset = 0;
ci.id3 = (struct mp3entry *)&cur_ti->id3;
ci.mp3data = (struct mp3info *)&cur_ti->mp3data;
ci.curpos = 0;
ci.taginfo_ready = (bool *)&cur_ti->taginfo_ready;
track_changed = true;
}
void audio_change_track(void)
{
if (track_ridx == track_widx) {
logf("No more tracks");
playing = false;
return ;
}
if (++track_ridx >= MAX_TRACK)
track_ridx = 0;
audio_update_trackinfo();
queue_post(&codec_queue, CODEC_LOAD, 0);
}
bool codec_request_next_track_callback(void)
{
if (ci.stop_codec || !playing || !queue_empty(&audio_queue))
return false;
logf("Request new track");
/* Advance to next track. */
if (ci.reload_codec && new_track > 0) {
playlist_next(1);
if (++track_ridx == MAX_TRACK)
track_ridx = 0;
if (tracks[track_ridx].filesize == 0) {
logf("Loading from disk...");
new_track = 0;
queue_post(&audio_queue, AUDIO_PLAY, 0);
return false;
}
}
/* Advance to previous track. */
else if (ci.reload_codec && new_track < 0) {
playlist_next(-1);
if (--track_ridx < 0)
track_ridx = MAX_TRACK-1;
if (tracks[track_ridx].filesize == 0 ||
codecbufused+ci.curpos+tracks[track_ridx].filesize
/*+ (off_t)tracks[track_ridx].codecsize*/ > codecbuflen) {
logf("Loading from disk...");
new_track = 0;
queue_post(&audio_queue, AUDIO_PLAY, 0);
return false;
}
}
/* Codec requested track change (next track). */
else {
playlist_next(1);
if (++track_ridx >= MAX_TRACK)
track_ridx = 0;
if (track_ridx == track_widx && tracks[track_ridx].filerem == 0) {
if (ci.reload_codec) {
} else {
logf("No more tracks");
}
new_track = 0;
return false;
}
}
ci.reload_codec = false;
if (cur_ti->codectype != tracks[track_ridx].codectype) {
if (--track_ridx < 0)
track_ridx = MAX_TRACK-1;
logf("New codec");
new_track = 0;
return false;
}
logf("On-the-fly change");
audio_update_trackinfo();
new_track = 0;
return true;
}
void audio_thread(void)
{
struct event ev;
while (1) {
sleep(50);
audio_check_buffer();
queue_wait_w_tmo(&audio_queue, &ev, 10);
switch (ev.id) {
case AUDIO_PLAY:
ci.stop_codec = true;
ci.reload_codec = false;
ci.seek_time = 0;
#ifndef SIMULATOR
pcm_play_stop();
#endif
audio_play_start((int)ev.data);
break ;
case AUDIO_STOP:
paused = false;
#ifndef SIMULATOR
pcm_play_stop();
pcm_play_pause(true);
#endif
break ;
case AUDIO_PAUSE:
break ;
case AUDIO_RESUME:
break ;
case AUDIO_NEXT:
break ;
case AUDIO_CODEC_DONE:
//if (playing)
// audio_change_track();
break ;
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
playing = false;
ci.stop_codec = true;
logf("USB Connection");
pcm_play_stop();
pcm_play_pause(true);
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&audio_queue);
break ;
#endif
}
}
}
void codec_thread(void)
{
struct event ev;
size_t codecsize;
int status;
int wrap;
while (1) {
status = 0;
queue_wait(&codec_queue, &ev);
switch (ev.id) {
case CODEC_LOAD_DISK:
ci.stop_codec = false;
status = codec_load_file((char *)ev.data, &ci);
break ;
case CODEC_LOAD:
logf("Codec start");
codecsize = cur_ti->codecsize;
if (codecsize == 0) {
logf("Codec slot is empty!");
playing = false;
break ;
}
ci.stop_codec = false;
wrap = (int)&codecbuf[codecbuflen] - (int)cur_ti->codecbuf;
status = codec_load_ram(cur_ti->codecbuf, codecsize,
&ci, &codecbuf[0], codecbuflen);
break ;
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&codec_queue);
break ;
#endif
}
switch (ev.id) {
case CODEC_LOAD_DISK:
case CODEC_LOAD:
if (status != PLUGIN_OK) {
logf("Codec failure");
playing = false;
} else {
logf("Codec finished");
}
if (playing && !ci.stop_codec && !ci.reload_codec) {
audio_change_track();
continue ;
} else if (ci.stop_codec) {
//playing = false;
}
//queue_post(&audio_queue, AUDIO_CODEC_DONE, (void *)status);
}
}
}
struct mp3entry* audio_current_track(void)
{
logf("audio_current_track");
if (track_count > 0 && cur_ti->taginfo_ready)
return (struct mp3entry *)&cur_ti->id3;
else
return NULL;
}
struct mp3entry* audio_next_track(void)
{
int next_idx = track_ridx + 1;
if (track_count == 0)
return NULL;
if (next_idx >= MAX_TRACK)
next_idx = 0;
if (!tracks[next_idx].taginfo_ready)
return NULL;
//logf("audio_next_track");
return &tracks[next_idx].id3;
}
bool audio_has_changed_track(void)
{
if (track_changed && track_count > 0) {
if (!cur_ti->taginfo_ready)
return false;
track_changed = false;
return true;
}
return false;
}
void audio_play(int offset)
{
logf("audio_play");
ci.stop_codec = true;
#ifndef SIMULATOR
pcm_play_pause(true);
#endif
paused = false;
playing = true;
queue_post(&audio_queue, AUDIO_PLAY, (void *)offset);
}
void audio_stop(void)
{
logf("audio_stop");
playing = false;
paused = false;
ci.stop_codec = true;
if (current_fd >= 0) {
close(current_fd);
current_fd = -1;
}
queue_post(&audio_queue, AUDIO_STOP, 0);
#ifndef SIMULATOR
pcm_play_pause(true);
#endif
}
void audio_pause(void)
{
logf("audio_pause");
#ifndef SIMULATOR
pcm_play_pause(false);
#endif
paused = true;
//queue_post(&audio_queue, AUDIO_PAUSE, 0);
}
void audio_resume(void)
{
logf("audio_resume");
#ifndef SIMULATOR
pcm_play_pause(true);
#endif
paused = false;
//queue_post(&audio_queue, AUDIO_RESUME, 0);
}
void audio_next(void)
{
logf("audio_next");
new_track = 1;
ci.reload_codec = true;
/* Detect if disk is spinning.. */
if (filling) {
ci.stop_codec = true;
playlist_next(1);
queue_post(&audio_queue, AUDIO_PLAY, 0);
} else {
#ifndef SIMULATOR
pcm_play_stop();
#endif
}
}
void audio_prev(void)
{
logf("audio_prev");
new_track = -1;
ci.reload_codec = true;
#ifndef SIMULATOR
pcm_play_stop();
#endif
if (filling) {
ci.stop_codec = true;
playlist_next(-1);
queue_post(&audio_queue, AUDIO_PLAY, 0);
}
//queue_post(&audio_queue, AUDIO_PREV, 0);
}
void audio_ff_rewind(int newpos)
{
logf("rewind: %d", newpos);
/* Does not work yet. */
if (playing)
ci.seek_time = newpos+1;
}
void audio_flush_and_reload_tracks(void)
{
logf("flush & reload");
}
void audio_error_clear(void)
{
}
int audio_status(void)
{
int ret = 0;
if (playing)
ret |= AUDIO_STATUS_PLAY;
if (paused)
ret |= AUDIO_STATUS_PAUSE;
return ret;
}
int audio_get_file_pos(void)
{
return 0;
}
/* Copied from mpeg.c. Should be moved somewhere else. */
static void mp3_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 < (int)(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 == simple frame calculation */
id3->elapsed = id3->offset / id3->bpf * id3->tpf;
}
/* Copied from mpeg.c. Should be moved somewhere else. */
int mp3_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->bpf && id3->tpf)
pos = (id3->elapsed/id3->tpf)*id3->bpf;
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;
}
#ifndef SIMULATOR
void audio_set_buffer_margin(int seconds)
{
(void)seconds;
}
#endif
void mpeg_id3_options(bool _v1first)
{
v1first = _v1first;
}
void audio_init(void)
{
logf("audio api init");
codecbuflen = audiobufend - audiobuf - PCMBUF_SIZE
- MALLOC_BUFSIZE - GUARD_BUFSIZE;
//codecbuflen = 2*512*1024;
codecbufused = 0;
filling = false;
codecbuf = &audiobuf[MALLOC_BUFSIZE];
playing = false;
paused = false;
track_changed = false;
current_fd = -1;
logf("abuf:%0x", PCMBUF_SIZE);
logf("fbuf:%0x", codecbuflen);
logf("mbuf:%0x", MALLOC_BUFSIZE);
/* Initialize codec api. */
ci.read_filebuf = codec_filebuf_callback;
#ifndef SIMULATOR
ci.audiobuffer_insert = audiobuffer_insert;
#else
ci.audiobuffer_insert = audiobuffer_insert_sim;
#endif
ci.get_codec_memory = get_codec_memory_callback;
ci.request_buffer = codec_request_buffer_callback;
ci.advance_buffer = codec_advance_buffer_callback;
ci.advance_buffer_loc = codec_advance_buffer_loc_callback;
ci.request_next_track = codec_request_next_track_callback;
ci.mp3_get_filepos = codec_mp3_get_filepos_callback;
ci.seek_buffer = codec_seek_buffer_callback;
ci.set_elapsed = codec_set_elapsed_callback;
ci.configure = codec_configure_callback;
queue_init(&audio_queue);
queue_init(&codec_queue);
create_thread(codec_thread, codec_stack, sizeof(codec_stack),
codec_thread_name);
create_thread(audio_thread, audio_stack, sizeof(audio_stack),
audio_thread_name);
#ifndef SIMULATOR
audio_is_initialized = true;
#endif
}
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