4db3e89652
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30403 a1c6a512-1295-4272-9138-f99709370657
3844 lines
110 KiB
C
3844 lines
110 KiB
C
/***************************************************************************
|
|
* __________ __ ___.
|
|
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
|
|
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
|
|
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
|
|
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
|
|
* \/ \/ \/ \/ \/
|
|
* $Id$
|
|
*
|
|
* Copyright (C) 2005-2007 Miika Pekkarinen
|
|
* Copyright (C) 2007-2008 Nicolas Pennequin
|
|
* Copyright (C) 2011 Michael Sevakis
|
|
*
|
|
* 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 "config.h"
|
|
#include "system.h"
|
|
#include "kernel.h"
|
|
#include "panic.h"
|
|
#include "core_alloc.h"
|
|
#include "sound.h"
|
|
#include "ata.h"
|
|
#include "usb.h"
|
|
#include "codecs.h"
|
|
#include "codec_thread.h"
|
|
#include "voice_thread.h"
|
|
#include "metadata.h"
|
|
#include "cuesheet.h"
|
|
#include "buffering.h"
|
|
#include "talk.h"
|
|
#include "playlist.h"
|
|
#include "abrepeat.h"
|
|
#include "pcmbuf.h"
|
|
#include "playback.h"
|
|
#include "misc.h"
|
|
#include "settings.h"
|
|
|
|
#ifdef HAVE_TAGCACHE
|
|
#include "tagcache.h"
|
|
#endif
|
|
|
|
#ifdef AUDIO_HAVE_RECORDING
|
|
#include "pcm_record.h"
|
|
#endif
|
|
|
|
#ifdef HAVE_LCD_BITMAP
|
|
#ifdef HAVE_ALBUMART
|
|
#include "albumart.h"
|
|
#endif
|
|
#endif
|
|
|
|
/* TODO: The audio thread really is doing multitasking of acting like a
|
|
consumer and producer of tracks. It may be advantageous to better
|
|
logically separate the two functions. I won't go that far just yet. */
|
|
|
|
/* Internal support for voice playback */
|
|
#define PLAYBACK_VOICE
|
|
|
|
#if CONFIG_PLATFORM & PLATFORM_NATIVE
|
|
/* Application builds don't support direct code loading */
|
|
#define HAVE_CODEC_BUFFERING
|
|
#endif
|
|
|
|
/* Amount of guess-space to allow for codecs that must hunt and peck
|
|
* for their correct seek target, 32k seems a good size */
|
|
#define AUDIO_REBUFFER_GUESS_SIZE (1024*32)
|
|
|
|
/* Define LOGF_ENABLE to enable logf output in this file */
|
|
/* #define LOGF_ENABLE */
|
|
#include "logf.h"
|
|
|
|
/* Macros to enable logf for queues
|
|
logging on SYS_TIMEOUT can be disabled */
|
|
#ifdef SIMULATOR
|
|
/* Define this for logf output of all queuing except SYS_TIMEOUT */
|
|
#define PLAYBACK_LOGQUEUES
|
|
/* Define this to logf SYS_TIMEOUT messages */
|
|
/*#define PLAYBACK_LOGQUEUES_SYS_TIMEOUT*/
|
|
#endif
|
|
|
|
#ifdef PLAYBACK_LOGQUEUES
|
|
#define LOGFQUEUE logf
|
|
#else
|
|
#define LOGFQUEUE(...)
|
|
#endif
|
|
|
|
#ifdef PLAYBACK_LOGQUEUES_SYS_TIMEOUT
|
|
#define LOGFQUEUE_SYS_TIMEOUT logf
|
|
#else
|
|
#define LOGFQUEUE_SYS_TIMEOUT(...)
|
|
#endif
|
|
|
|
/* Variables are commented with the threads that use them:
|
|
* A=audio, C=codec, O=other. A suffix of "-" indicates that the variable is
|
|
* read but not updated on that thread. Audio is the only user unless otherwise
|
|
* specified.
|
|
*/
|
|
|
|
/** Miscellaneous **/
|
|
bool audio_is_initialized = false; /* (A,O-) */
|
|
extern struct codec_api ci; /* (A,C) */
|
|
|
|
/** Possible arrangements of the main buffer **/
|
|
static enum audio_buffer_state
|
|
{
|
|
AUDIOBUF_STATE_TRASHED = -1, /* trashed; must be reset */
|
|
AUDIOBUF_STATE_INITIALIZED = 0, /* voice+audio OR audio-only */
|
|
AUDIOBUF_STATE_VOICED_ONLY = 1, /* voice-only */
|
|
} buffer_state = AUDIOBUF_STATE_TRASHED; /* (A,O) */
|
|
|
|
/** Main state control **/
|
|
static bool ff_rw_mode SHAREDBSS_ATTR = false; /* Pre-ff-rewind mode (A,O-) */
|
|
|
|
enum play_status
|
|
{
|
|
PLAY_STOPPED = 0,
|
|
PLAY_PLAYING = AUDIO_STATUS_PLAY,
|
|
PLAY_PAUSED = AUDIO_STATUS_PLAY | AUDIO_STATUS_PAUSE,
|
|
} play_status = PLAY_STOPPED;
|
|
|
|
/* Sizeable things that only need exist during playback and not when stopped */
|
|
static struct audio_scratch_memory
|
|
{
|
|
struct mp3entry codec_id3; /* (A,C) */
|
|
struct mp3entry unbuffered_id3;
|
|
struct cuesheet *curr_cue; /* Will follow this structure */
|
|
} * audio_scratch_memory = NULL;
|
|
|
|
/* These are used to store the current, next and optionally the peek-ahead
|
|
* mp3entry's - this guarantees that the pointer returned by audio_current/
|
|
* next_track will be valid for the full duration of the currently playing
|
|
* track */
|
|
enum audio_id3_types
|
|
{
|
|
/* These are allocated statically */
|
|
PLAYING_ID3 = 0,
|
|
NEXTTRACK_ID3,
|
|
#ifdef AUDIO_FAST_SKIP_PREVIEW
|
|
/* The real playing metadata must has to be protected since it contains
|
|
critical info for other features */
|
|
PLAYING_PEEK_ID3,
|
|
#endif
|
|
ID3_TYPE_NUM_STATIC,
|
|
/* These go in the scratch memory */
|
|
UNBUFFERED_ID3 = ID3_TYPE_NUM_STATIC,
|
|
CODEC_ID3,
|
|
};
|
|
static struct mp3entry static_id3_entries[ID3_TYPE_NUM_STATIC]; /* (A,O) */
|
|
|
|
/* Peeking functions can yield and mess us up */
|
|
static struct mutex id3_mutex SHAREDBSS_ATTR; /* (A,O)*/
|
|
|
|
|
|
/** For Scrobbler support **/
|
|
|
|
/* Previous track elapsed time */
|
|
static unsigned long prev_track_elapsed = 0; /* (A,O-) */
|
|
|
|
|
|
/** For album art support **/
|
|
#define MAX_MULTIPLE_AA SKINNABLE_SCREENS_COUNT
|
|
#ifdef HAVE_ALBUMART
|
|
|
|
static struct albumart_slot
|
|
{
|
|
struct dim dim; /* Holds width, height of the albumart */
|
|
int used; /* Counter; increments if something uses it */
|
|
} albumart_slots[MAX_MULTIPLE_AA]; /* (A,O) */
|
|
|
|
#define FOREACH_ALBUMART(i) for(i = 0;i < MAX_MULTIPLE_AA; i++)
|
|
#endif /* HAVE_ALBUMART */
|
|
|
|
|
|
/** Information used for tracking buffer fills **/
|
|
|
|
/* Buffer and thread state tracking */
|
|
static enum filling_state
|
|
{
|
|
STATE_IDLE = 0, /* audio is stopped: nothing to do */
|
|
STATE_FILLING, /* adding tracks to the buffer */
|
|
STATE_FULL, /* can't add any more tracks */
|
|
STATE_END_OF_PLAYLIST, /* all remaining tracks have been added */
|
|
STATE_FINISHED, /* all remaining tracks are fully buffered */
|
|
STATE_ENDING, /* audio playback is ending */
|
|
STATE_ENDED, /* audio playback is done */
|
|
#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
|
|
STATE_USB, /* USB mode, ignore most messages */
|
|
#endif
|
|
} filling = STATE_IDLE;
|
|
|
|
/* Track info - holds information about each track in the buffer */
|
|
struct track_info
|
|
{
|
|
/* In per-track allocated order: */
|
|
int id3_hid; /* Metadata handle ID */
|
|
int cuesheet_hid; /* Parsed cuesheet handle ID */
|
|
#ifdef HAVE_ALBUMART
|
|
int aa_hid[MAX_MULTIPLE_AA];/* Album art handle IDs */
|
|
#endif
|
|
#ifdef HAVE_CODEC_BUFFERING
|
|
int codec_hid; /* Buffered codec handle ID */
|
|
#endif
|
|
int audio_hid; /* Main audio data handle ID */
|
|
size_t filesize; /* File total length on disk
|
|
TODO: This should be stored
|
|
in the handle or the
|
|
id3 and would use less
|
|
ram */
|
|
};
|
|
|
|
/* Track list - holds info about all buffered tracks */
|
|
#if MEMORYSIZE >= 32
|
|
#define TRACK_LIST_LEN 128 /* Must be 2^int(+n) */
|
|
#elif MEMORYSIZE >= 16
|
|
#define TRACK_LIST_LEN 64
|
|
#elif MEMORYSIZE >= 8
|
|
#define TRACK_LIST_LEN 32
|
|
#else
|
|
#define TRACK_LIST_LEN 16
|
|
#endif
|
|
|
|
#define TRACK_LIST_MASK (TRACK_LIST_LEN-1)
|
|
|
|
static struct
|
|
{
|
|
/* read, write and current are maintained unwrapped, limited only by the
|
|
unsigned int range and wrap-safe comparisons are used */
|
|
|
|
/* NOTE: there appears to be a bug in arm-elf-eabi-gcc 4.4.4 for ARMv4 where
|
|
if 'end' follows 'start' in this structure, track_list_count performs
|
|
'start - end' rather than 'end - start', giving negative count values...
|
|
so leave it this way for now! */
|
|
unsigned int end; /* Next open position */
|
|
unsigned int start; /* First track in list */
|
|
unsigned int current; /* Currently decoding track */
|
|
struct track_info tracks[TRACK_LIST_LEN]; /* Buffered track information */
|
|
} track_list; /* (A, O-) */
|
|
|
|
|
|
/* Playlist steps from playlist position to next track to be buffered */
|
|
static int playlist_peek_offset = 0;
|
|
|
|
/* Metadata handle of track load in progress (meaning all handles have not
|
|
yet been opened for the track, id3 always exists or the track does not)
|
|
|
|
Tracks are keyed by their metadata handles if track list pointers are
|
|
insufficient to make comparisons */
|
|
static int in_progress_id3_hid = ERR_HANDLE_NOT_FOUND;
|
|
|
|
#ifdef HAVE_DISK_STORAGE
|
|
/* Buffer margin A.K.A. anti-skip buffer (in seconds) */
|
|
static size_t buffer_margin = 5;
|
|
#endif
|
|
|
|
/* Values returned for track loading */
|
|
enum track_load_status
|
|
{
|
|
LOAD_TRACK_ERR_START_CODEC = -6,
|
|
LOAD_TRACK_ERR_FINISH_FAILED = -5,
|
|
LOAD_TRACK_ERR_FINISH_FULL = -4,
|
|
LOAD_TRACK_ERR_BUSY = -3,
|
|
LOAD_TRACK_ERR_NO_MORE = -2,
|
|
LOAD_TRACK_ERR_FAILED = -1,
|
|
LOAD_TRACK_OK = 0,
|
|
LOAD_TRACK_READY = 1,
|
|
};
|
|
|
|
/** Track change controls **/
|
|
|
|
/* What sort of skip is pending globally? */
|
|
enum track_skip_type
|
|
{
|
|
/* Relative to what user is intended to see: */
|
|
/* Codec: +0, Track List: +0, Playlist: +0 */
|
|
TRACK_SKIP_NONE = 0, /* no track skip */
|
|
/* Codec: +1, Track List: +1, Playlist: +0 */
|
|
TRACK_SKIP_AUTO, /* codec-initiated skip */
|
|
/* Codec: +1, Track List: +1, Playlist: +1 */
|
|
TRACK_SKIP_AUTO_NEW_PLAYLIST, /* codec-initiated skip is new playlist */
|
|
/* Codec: xx, Track List: +0, Playlist: +0 */
|
|
TRACK_SKIP_AUTO_END_PLAYLIST, /* codec-initiated end of playlist */
|
|
/* Manual skip: Never pends */
|
|
TRACK_SKIP_MANUAL, /* manual track skip */
|
|
/* Manual skip: Never pends */
|
|
TRACK_SKIP_DIR_CHANGE, /* manual directory skip */
|
|
} skip_pending = TRACK_SKIP_NONE;
|
|
|
|
/* Note about TRACK_SKIP_AUTO_NEW_PLAYLIST:
|
|
Fixing playlist code to be able to peek into the first song of
|
|
the next playlist would fix any issues and this wouldn't need
|
|
to be a special case since pre-advancing the playlist would be
|
|
unneeded - it could be much more like TRACK_SKIP_AUTO and all
|
|
actions that require reversal during an in-progress transition
|
|
would work as expected */
|
|
|
|
/* Used to indicate status for the events. Must be separate to satisfy all
|
|
clients so the correct metadata is read when sending the change events
|
|
and also so that it is read correctly outside the events. */
|
|
static bool automatic_skip = false; /* (A, O-) */
|
|
|
|
/* Pending manual track skip offset */
|
|
static int skip_offset = 0; /* (A, O) */
|
|
|
|
/* Track change notification */
|
|
static struct
|
|
{
|
|
unsigned int in; /* Number of pcmbuf posts (audio isr) */
|
|
unsigned int out; /* Number of times audio has read the difference */
|
|
} track_change = { 0, 0 };
|
|
|
|
/** Codec status **/
|
|
/* Did the codec notify us it finished while we were paused or while still
|
|
in an automatic transition?
|
|
|
|
If paused, it is necessary to defer a codec-initiated skip until resuming
|
|
or else the track will move forward while not playing audio!
|
|
|
|
If in-progress, skips should not build-up ahead of where the WPS is when
|
|
really short tracks finish decoding.
|
|
|
|
If it is forgotten, it will be missed altogether and playback will just sit
|
|
there looking stupid and comatose until the user does something */
|
|
static bool codec_skip_pending = false;
|
|
static int codec_skip_status;
|
|
static bool codec_seeking = false; /* Codec seeking ack expected? */
|
|
static unsigned int position_key = 0;
|
|
|
|
/* Event queues */
|
|
static struct event_queue audio_queue SHAREDBSS_ATTR;
|
|
|
|
/* Audio thread */
|
|
static struct queue_sender_list audio_queue_sender_list SHAREDBSS_ATTR;
|
|
static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
|
|
static const char audio_thread_name[] = "audio";
|
|
static unsigned int audio_thread_id = 0;
|
|
|
|
/* Forward declarations */
|
|
enum audio_start_playback_flags
|
|
{
|
|
AUDIO_START_RESTART = 0x1, /* "Restart" playback (flush _all_ tracks) */
|
|
AUDIO_START_NEWBUF = 0x2, /* Mark the audiobuffer as invalid */
|
|
};
|
|
|
|
static void audio_start_playback(size_t offset, unsigned int flags);
|
|
static void audio_stop_playback(void);
|
|
static void buffer_event_buffer_low_callback(void *data);
|
|
static void buffer_event_rebuffer_callback(void *data);
|
|
static void buffer_event_finished_callback(void *data);
|
|
void audio_pcmbuf_sync_position(void);
|
|
|
|
|
|
/**************************************/
|
|
|
|
/** --- audio_queue helpers --- **/
|
|
static void audio_queue_post(long id, intptr_t data)
|
|
{
|
|
queue_post(&audio_queue, id, data);
|
|
}
|
|
|
|
static intptr_t audio_queue_send(long id, intptr_t data)
|
|
{
|
|
return queue_send(&audio_queue, id, data);
|
|
}
|
|
|
|
|
|
/** --- MP3Entry --- **/
|
|
|
|
/* Does the mp3entry have enough info for us to use it? */
|
|
static struct mp3entry * valid_mp3entry(const struct mp3entry *id3)
|
|
{
|
|
return id3 && (id3->length != 0 || id3->filesize != 0) &&
|
|
id3->codectype != AFMT_UNKNOWN ? (struct mp3entry *)id3 : NULL;
|
|
}
|
|
|
|
/* Return a pointer to an mp3entry on the buffer, as it is */
|
|
static struct mp3entry * bufgetid3(int handle_id)
|
|
{
|
|
if (handle_id < 0)
|
|
return NULL;
|
|
|
|
struct mp3entry *id3;
|
|
ssize_t ret = bufgetdata(handle_id, 0, (void *)&id3);
|
|
|
|
if (ret != sizeof(struct mp3entry))
|
|
return NULL;
|
|
|
|
return id3;
|
|
}
|
|
|
|
/* Read an mp3entry from the buffer, adjusted */
|
|
static bool bufreadid3(int handle_id, struct mp3entry *id3out)
|
|
{
|
|
struct mp3entry *id3 = bufgetid3(handle_id);
|
|
|
|
if (id3)
|
|
{
|
|
copy_mp3entry(id3out, id3);
|
|
return true;
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
/* Lock the id3 mutex */
|
|
static void id3_mutex_lock(void)
|
|
{
|
|
mutex_lock(&id3_mutex);
|
|
}
|
|
|
|
/* Unlock the id3 mutex */
|
|
static void id3_mutex_unlock(void)
|
|
{
|
|
mutex_unlock(&id3_mutex);
|
|
}
|
|
|
|
/* Return one of the collection of mp3entry pointers - collect them all here */
|
|
static inline struct mp3entry * id3_get(enum audio_id3_types id3_num)
|
|
{
|
|
switch (id3_num)
|
|
{
|
|
case UNBUFFERED_ID3:
|
|
return &audio_scratch_memory->unbuffered_id3;
|
|
case CODEC_ID3:
|
|
return &audio_scratch_memory->codec_id3;
|
|
default:
|
|
return &static_id3_entries[id3_num];
|
|
}
|
|
}
|
|
|
|
/* Copy an mp3entry into one of the mp3 entries */
|
|
static void id3_write(enum audio_id3_types id3_num,
|
|
const struct mp3entry *id3_src)
|
|
{
|
|
struct mp3entry *dest_id3 = id3_get(id3_num);
|
|
|
|
if (id3_src)
|
|
copy_mp3entry(dest_id3, id3_src);
|
|
else
|
|
wipe_mp3entry(dest_id3);
|
|
}
|
|
|
|
/* Call id3_write "safely" because peek aheads can yield, even if the fast
|
|
preview isn't enabled */
|
|
static void id3_write_locked(enum audio_id3_types id3_num,
|
|
const struct mp3entry *id3_src)
|
|
{
|
|
id3_mutex_lock();
|
|
id3_write(id3_num, id3_src);
|
|
id3_mutex_unlock();
|
|
}
|
|
|
|
|
|
/** --- Track info --- **/
|
|
|
|
/* Close a handle and mark it invalid */
|
|
static void track_info_close_handle(int *hid_p)
|
|
{
|
|
int hid = *hid_p;
|
|
|
|
/* bufclose returns true if the handle is not found, or if it is closed
|
|
* successfully, so these checks are safe on non-existant handles */
|
|
if (hid >= 0)
|
|
bufclose(hid);
|
|
|
|
/* Always reset to "no handle" in case it was something else */
|
|
*hid_p = ERR_HANDLE_NOT_FOUND;
|
|
}
|
|
|
|
/* Close all handles in a struct track_info and clear it */
|
|
static void track_info_close(struct track_info *info)
|
|
{
|
|
/* Close them in the order they are allocated on the buffer to speed up
|
|
the handle searching */
|
|
track_info_close_handle(&info->id3_hid);
|
|
track_info_close_handle(&info->cuesheet_hid);
|
|
#ifdef HAVE_ALBUMART
|
|
int i;
|
|
FOREACH_ALBUMART(i)
|
|
track_info_close_handle(&info->aa_hid[i]);
|
|
#endif
|
|
#ifdef HAVE_CODEC_BUFFERING
|
|
track_info_close_handle(&info->codec_hid);
|
|
#endif
|
|
track_info_close_handle(&info->audio_hid);
|
|
info->filesize = 0;
|
|
}
|
|
|
|
/* Invalidate all members to initial values - does not close handles */
|
|
static void track_info_wipe(struct track_info * info)
|
|
{
|
|
info->id3_hid = ERR_HANDLE_NOT_FOUND;
|
|
info->cuesheet_hid = ERR_HANDLE_NOT_FOUND;
|
|
#ifdef HAVE_ALBUMART
|
|
int i;
|
|
FOREACH_ALBUMART(i)
|
|
info->aa_hid[i] = ERR_HANDLE_NOT_FOUND;
|
|
#endif
|
|
#ifdef HAVE_CODEC_BUFFERING
|
|
info->codec_hid = ERR_HANDLE_NOT_FOUND;
|
|
#endif
|
|
info->audio_hid = ERR_HANDLE_NOT_FOUND;
|
|
info->filesize = 0;
|
|
}
|
|
|
|
|
|
/** --- Track list --- **/
|
|
|
|
/* Initialize the track list */
|
|
static void track_list_init(void)
|
|
{
|
|
int i;
|
|
for (i = 0; i < TRACK_LIST_LEN; i++)
|
|
track_info_wipe(&track_list.tracks[i]);
|
|
|
|
track_list.start = track_list.end = track_list.current;
|
|
}
|
|
|
|
/* Return number of items allocated in the list */
|
|
static unsigned int track_list_count(void)
|
|
{
|
|
return track_list.end - track_list.start;
|
|
}
|
|
|
|
/* Return true if the list is empty */
|
|
static inline bool track_list_empty(void)
|
|
{
|
|
return track_list.end == track_list.start;
|
|
}
|
|
|
|
/* Returns true if the list is holding the maximum number of items */
|
|
static bool track_list_full(void)
|
|
{
|
|
return track_list.end - track_list.start >= TRACK_LIST_LEN;
|
|
}
|
|
|
|
/* Test if the index is within the allocated range */
|
|
static bool track_list_in_range(int pos)
|
|
{
|
|
return (int)(pos - track_list.start) >= 0 &&
|
|
(int)(pos - track_list.end) < 0;
|
|
}
|
|
|
|
static struct track_info * track_list_entry(int pos)
|
|
{
|
|
return &track_list.tracks[pos & TRACK_LIST_MASK];
|
|
}
|
|
|
|
/* Return the info of the last allocation plus an offset, NULL if result is
|
|
out of bounds */
|
|
static struct track_info * track_list_last(int offset)
|
|
{
|
|
/* Last is before the end since the end isn't inclusive */
|
|
unsigned int pos = track_list.end + offset - 1;
|
|
|
|
if (!track_list_in_range(pos))
|
|
return NULL;
|
|
|
|
return track_list_entry(pos);
|
|
}
|
|
|
|
/* Allocate space at the end for another track if not full */
|
|
static struct track_info * track_list_alloc_track(void)
|
|
{
|
|
if (track_list_full())
|
|
return NULL;
|
|
|
|
return track_list_entry(track_list.end++);
|
|
}
|
|
|
|
/* Remove the last track entry allocated in order to support backing out
|
|
of a track load */
|
|
static void track_list_unalloc_track(void)
|
|
{
|
|
if (track_list_empty())
|
|
return;
|
|
|
|
track_list.end--;
|
|
|
|
if (track_list.current == track_list.end &&
|
|
track_list.current != track_list.start)
|
|
{
|
|
/* Current _must_ remain within bounds */
|
|
track_list.current--;
|
|
}
|
|
}
|
|
|
|
/* Return current track plus an offset, NULL if result is out of bounds */
|
|
static struct track_info * track_list_current(int offset)
|
|
{
|
|
unsigned int pos = track_list.current + offset;
|
|
|
|
if (!track_list_in_range(pos))
|
|
return NULL;
|
|
|
|
return track_list_entry(pos);
|
|
}
|
|
|
|
/* Return current based upon what's intended that the user sees - not
|
|
necessarily where decoding is taking place */
|
|
static struct track_info * track_list_user_current(int offset)
|
|
{
|
|
if (skip_pending == TRACK_SKIP_AUTO ||
|
|
skip_pending == TRACK_SKIP_AUTO_NEW_PLAYLIST)
|
|
{
|
|
offset--;
|
|
}
|
|
|
|
return track_list_current(offset);
|
|
}
|
|
|
|
/* Advance current track by an offset, return false if result is out of
|
|
bounds */
|
|
static struct track_info * track_list_advance_current(int offset)
|
|
{
|
|
unsigned int pos = track_list.current + offset;
|
|
|
|
if (!track_list_in_range(pos))
|
|
return NULL;
|
|
|
|
track_list.current = pos;
|
|
return track_list_entry(pos);
|
|
}
|
|
|
|
/* Clear tracks in the list, optionally preserving the current track -
|
|
returns 'false' if the operation was changed */
|
|
enum track_clear_action
|
|
{
|
|
TRACK_LIST_CLEAR_ALL = 0, /* Clear all tracks */
|
|
TRACK_LIST_KEEP_CURRENT, /* Keep current only; clear before + after */
|
|
TRACK_LIST_KEEP_NEW /* Keep current and those that follow */
|
|
};
|
|
|
|
static void track_list_clear(enum track_clear_action action)
|
|
{
|
|
logf("%s(%d)", __func__, (int)action);
|
|
|
|
/* Don't care now since rebuffering is imminent */
|
|
buf_set_watermark(0);
|
|
|
|
if (action != TRACK_LIST_CLEAR_ALL)
|
|
{
|
|
struct track_info *cur = track_list_current(0);
|
|
|
|
if (!cur || cur->id3_hid < 0)
|
|
action = TRACK_LIST_CLEAR_ALL; /* Nothing worthwhile keeping */
|
|
}
|
|
|
|
/* Noone should see this progressing */
|
|
int start = track_list.start;
|
|
int current = track_list.current;
|
|
int end = track_list.end;
|
|
|
|
track_list.start = current;
|
|
|
|
switch (action)
|
|
{
|
|
case TRACK_LIST_CLEAR_ALL:
|
|
/* Result: .start = .current, .end = .current */
|
|
track_list.end = current;
|
|
break;
|
|
|
|
case TRACK_LIST_KEEP_CURRENT:
|
|
/* Result: .start = .current, .end = .current + 1 */
|
|
track_list.end = current + 1;
|
|
break;
|
|
|
|
case TRACK_LIST_KEEP_NEW:
|
|
/* Result: .start = .current, .end = .end */
|
|
end = current;
|
|
break;
|
|
}
|
|
|
|
/* Close all open handles in the range except the for the current track
|
|
if preserving that */
|
|
while (start != end)
|
|
{
|
|
if (action != TRACK_LIST_KEEP_CURRENT || start != current)
|
|
{
|
|
struct track_info *info =
|
|
&track_list.tracks[start & TRACK_LIST_MASK];
|
|
|
|
/* If this is the in-progress load, abort it */
|
|
if (in_progress_id3_hid >= 0 &&
|
|
info->id3_hid == in_progress_id3_hid)
|
|
{
|
|
in_progress_id3_hid = ERR_HANDLE_NOT_FOUND;
|
|
}
|
|
|
|
track_info_close(info);
|
|
}
|
|
|
|
start++;
|
|
}
|
|
}
|
|
|
|
|
|
/** --- Audio buffer -- **/
|
|
|
|
/* What size is needed for the scratch buffer? */
|
|
static size_t scratch_mem_size(void)
|
|
{
|
|
size_t size = sizeof (struct audio_scratch_memory);
|
|
|
|
if (global_settings.cuesheet)
|
|
size += sizeof (struct cuesheet);
|
|
|
|
return size;
|
|
}
|
|
|
|
/* Initialize the memory area where data is stored that is only used when
|
|
playing audio and anything depending upon it */
|
|
static void scratch_mem_init(void *mem)
|
|
{
|
|
audio_scratch_memory = (struct audio_scratch_memory *)mem;
|
|
id3_write_locked(UNBUFFERED_ID3, NULL);
|
|
id3_write(CODEC_ID3, NULL);
|
|
ci.id3 = id3_get(CODEC_ID3);
|
|
audio_scratch_memory->curr_cue = NULL;
|
|
|
|
if (global_settings.cuesheet)
|
|
{
|
|
audio_scratch_memory->curr_cue =
|
|
SKIPBYTES((struct cuesheet *)audio_scratch_memory,
|
|
sizeof (struct audio_scratch_memory));
|
|
}
|
|
}
|
|
|
|
static int audiobuf_handle;
|
|
static size_t filebuflen;
|
|
|
|
size_t audio_buffer_available(void)
|
|
{
|
|
if (audiobuf_handle > 0) /* if allocated return what we got */
|
|
return filebuflen;
|
|
return core_available();
|
|
}
|
|
|
|
/* Set up the audio buffer for playback
|
|
* filebuflen must be pre-initialized with the maximum size */
|
|
static void audio_reset_buffer_noalloc(void* filebuf)
|
|
{
|
|
/*
|
|
* Layout audio buffer as follows:
|
|
* [[|TALK]|SCRATCH|BUFFERING|PCM|[VOICE|]]
|
|
*/
|
|
|
|
/* see audio_get_recording_buffer if this is modified */
|
|
logf("%s()", __func__);
|
|
|
|
/* If the setup of anything allocated before the file buffer is
|
|
changed, do check the adjustments after the buffer_alloc call
|
|
as it will likely be affected and need sliding over */
|
|
|
|
/* Initially set up file buffer as all space available */
|
|
size_t allocsize;
|
|
|
|
/* Subtract whatever voice needs */
|
|
allocsize = talkbuf_init(filebuf);
|
|
allocsize = ALIGN_UP(allocsize, sizeof (intptr_t));
|
|
if (allocsize > filebuflen)
|
|
goto bufpanic;
|
|
|
|
filebuf += allocsize;
|
|
filebuflen -= allocsize;
|
|
|
|
if (talk_voice_required())
|
|
{
|
|
/* Need a space for voice PCM output */
|
|
allocsize = voicebuf_init(filebuf + filebuflen);
|
|
|
|
allocsize = ALIGN_UP(allocsize, sizeof (intptr_t));
|
|
if (allocsize > filebuflen)
|
|
goto bufpanic;
|
|
|
|
filebuflen -= allocsize;
|
|
}
|
|
|
|
/* Subtract whatever the pcm buffer says it used plus the guard buffer */
|
|
allocsize = pcmbuf_init(filebuf + filebuflen);
|
|
|
|
/* Make sure filebuflen is a pointer sized multiple after adjustment */
|
|
allocsize = ALIGN_UP(allocsize, sizeof (intptr_t));
|
|
if (allocsize > filebuflen)
|
|
goto bufpanic;
|
|
|
|
filebuflen -= allocsize;
|
|
|
|
/* Scratch memory */
|
|
allocsize = scratch_mem_size();
|
|
if (allocsize > filebuflen)
|
|
goto bufpanic;
|
|
|
|
scratch_mem_init(filebuf);
|
|
filebuf += allocsize;
|
|
filebuflen -= allocsize;
|
|
|
|
buffering_reset(filebuf, filebuflen);
|
|
|
|
/* Clear any references to the file buffer */
|
|
buffer_state = AUDIOBUF_STATE_INITIALIZED;
|
|
|
|
#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
|
|
/* Make sure everything adds up - yes, some info is a bit redundant but
|
|
aids viewing and the summation of certain variables should add up to
|
|
the location of others. */
|
|
{
|
|
logf("fbuf: %08X", (unsigned)filebuf);
|
|
logf("fbufe: %08X", (unsigned)(filebuf + filebuflen));
|
|
logf("sbuf: %08X", (unsigned)audio_scratch_memory);
|
|
logf("sbufe: %08X", (unsigned)(audio_scratch_memory + allocsize));
|
|
}
|
|
#endif
|
|
|
|
return;
|
|
|
|
bufpanic:
|
|
panicf("%s(): EOM (%zu > %zu)", __func__, allocsize, filebuflen);
|
|
}
|
|
|
|
|
|
/* Buffer must not move. */
|
|
static int shrink_callback(int handle, unsigned hints, void* start, size_t old_size)
|
|
{
|
|
long offset = audio_current_track()->offset;
|
|
int status = audio_status();
|
|
/* TODO: Do it without stopping playback, if possible */
|
|
/* don't call audio_hard_stop() as it frees this handle */
|
|
if (thread_self() == audio_thread_id)
|
|
{ /* inline case Q_AUDIO_STOP (audio_hard_stop() response
|
|
* if we're in the audio thread */
|
|
audio_stop_playback();
|
|
queue_clear(&audio_queue);
|
|
}
|
|
else
|
|
audio_queue_send(Q_AUDIO_STOP, 1);
|
|
#ifdef PLAYBACK_VOICE
|
|
voice_stop();
|
|
#endif
|
|
/* we should be free to change the buffer now */
|
|
size_t wanted_size = (hints & BUFLIB_SHRINK_SIZE_MASK);
|
|
ssize_t size = (ssize_t)old_size - wanted_size;
|
|
/* set final buffer size before calling audio_reset_buffer_noalloc() */
|
|
filebuflen = size;
|
|
switch (hints & BUFLIB_SHRINK_POS_MASK)
|
|
{
|
|
case BUFLIB_SHRINK_POS_BACK:
|
|
core_shrink(handle, start, size);
|
|
audio_reset_buffer_noalloc(start);
|
|
break;
|
|
case BUFLIB_SHRINK_POS_FRONT:
|
|
core_shrink(handle, start + wanted_size, size);
|
|
audio_reset_buffer_noalloc(start + wanted_size);
|
|
break;
|
|
}
|
|
if ((status & AUDIO_STATUS_PLAY) == AUDIO_STATUS_PLAY)
|
|
{
|
|
if (thread_self() == audio_thread_id)
|
|
audio_start_playback(offset, 0); /* inline Q_AUDIO_PLAY */
|
|
else
|
|
audio_play(offset);
|
|
}
|
|
|
|
return BUFLIB_CB_OK;
|
|
}
|
|
|
|
static struct buflib_callbacks ops = {
|
|
.move_callback = NULL,
|
|
.shrink_callback = shrink_callback,
|
|
};
|
|
|
|
static void audio_reset_buffer(void)
|
|
{
|
|
if (audiobuf_handle > 0)
|
|
{
|
|
core_free(audiobuf_handle);
|
|
audiobuf_handle = 0;
|
|
}
|
|
audiobuf_handle = core_alloc_maximum("audiobuf", &filebuflen, &ops);
|
|
unsigned char *filebuf = core_get_data(audiobuf_handle);
|
|
|
|
audio_reset_buffer_noalloc(filebuf);
|
|
}
|
|
|
|
/* Set the buffer margin to begin rebuffering when 'seconds' from empty */
|
|
static void audio_update_filebuf_watermark(int seconds)
|
|
{
|
|
size_t bytes = 0;
|
|
|
|
#ifdef HAVE_DISK_STORAGE
|
|
int spinup = ata_spinup_time();
|
|
|
|
if (seconds == 0)
|
|
{
|
|
/* By current setting */
|
|
seconds = buffer_margin;
|
|
}
|
|
else
|
|
{
|
|
/* New setting */
|
|
buffer_margin = seconds;
|
|
|
|
if (buf_get_watermark() == 0)
|
|
{
|
|
/* Write a watermark only if the audio thread already did so for
|
|
itself or it will fail to set the event and the watermark - if
|
|
it hasn't yet, it will use the new setting when it does */
|
|
return;
|
|
}
|
|
}
|
|
|
|
if (spinup)
|
|
seconds += (spinup / HZ) + 1;
|
|
else
|
|
seconds += 5;
|
|
|
|
seconds += buffer_margin;
|
|
#else
|
|
/* flash storage */
|
|
seconds = 1;
|
|
#endif
|
|
|
|
/* Watermark is a function of the bitrate of the last track in the buffer */
|
|
struct mp3entry *id3 = NULL;
|
|
struct track_info *info = track_list_last(0);
|
|
|
|
if (info)
|
|
id3 = valid_mp3entry(bufgetid3(info->id3_hid));
|
|
|
|
if (id3)
|
|
{
|
|
if (get_audio_base_data_type(id3->codectype) == TYPE_PACKET_AUDIO)
|
|
{
|
|
bytes = id3->bitrate * (1000/8) * seconds;
|
|
}
|
|
else
|
|
{
|
|
/* Bitrate has no meaning to buffering margin for atomic audio -
|
|
rebuffer when it's the only track left unless it's the only
|
|
track that fits, in which case we should avoid constant buffer
|
|
low events */
|
|
if (track_list_count() > 1)
|
|
bytes = info->filesize + 1;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Then set the minimum - this should not occur anyway */
|
|
logf("fwmark: No id3 for last track (s%u/c%u/e%u)",
|
|
track_list.start, track_list.current, track_list.end);
|
|
}
|
|
|
|
/* Actually setting zero disables the notification and we use that
|
|
to detect that it has been reset */
|
|
buf_set_watermark(MAX(bytes, 1));
|
|
logf("fwmark: %lu", (unsigned long)bytes);
|
|
}
|
|
|
|
|
|
/** -- Track change notification -- **/
|
|
|
|
/* Check the pcmbuf track changes and return write the message into the event
|
|
if there are any */
|
|
static inline bool audio_pcmbuf_track_change_scan(void)
|
|
{
|
|
if (track_change.out != track_change.in)
|
|
{
|
|
track_change.out++;
|
|
return true;
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
/* Clear outstanding track change posts */
|
|
static inline void audio_pcmbuf_track_change_clear(void)
|
|
{
|
|
track_change.out = track_change.in;
|
|
}
|
|
|
|
/* Post a track change notification - called by audio ISR */
|
|
static inline void audio_pcmbuf_track_change_post(void)
|
|
{
|
|
track_change.in++;
|
|
}
|
|
|
|
|
|
/** --- Helper functions --- **/
|
|
|
|
/* Removes messages that might end up in the queue before or while processing
|
|
a manual track change. Responding to them would be harmful since they
|
|
belong to a previous track's playback period. Anything that would generate
|
|
the stale messages must first be put into a state where it will not do so.
|
|
*/
|
|
static void audio_clear_track_notifications(void)
|
|
{
|
|
static const long filter_list[][2] =
|
|
{
|
|
/* codec messages */
|
|
{ Q_AUDIO_CODEC_SEEK_COMPLETE, Q_AUDIO_CODEC_COMPLETE },
|
|
/* track change messages */
|
|
{ Q_AUDIO_TRACK_CHANGED, Q_AUDIO_TRACK_CHANGED },
|
|
};
|
|
|
|
const int filter_count = ARRAYLEN(filter_list) - 1;
|
|
|
|
/* Remove any pcmbuf notifications */
|
|
pcmbuf_monitor_track_change(false);
|
|
audio_pcmbuf_track_change_clear();
|
|
|
|
/* Scrub the audio queue of the old mold */
|
|
while (queue_peek_ex(&audio_queue, NULL,
|
|
filter_count | QPEEK_REMOVE_EVENTS,
|
|
filter_list))
|
|
{
|
|
yield(); /* Not strictly needed, per se, ad infinitum, ra, ra */
|
|
}
|
|
}
|
|
|
|
/* Takes actions based upon track load status codes */
|
|
static void audio_handle_track_load_status(int trackstat)
|
|
{
|
|
switch (trackstat)
|
|
{
|
|
case LOAD_TRACK_ERR_NO_MORE:
|
|
if (track_list_count() > 0)
|
|
break;
|
|
|
|
case LOAD_TRACK_ERR_START_CODEC:
|
|
audio_queue_post(Q_AUDIO_CODEC_COMPLETE, CODEC_ERROR);
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* Announce the end of playing the current track */
|
|
static void audio_playlist_track_finish(void)
|
|
{
|
|
struct mp3entry *ply_id3 = id3_get(PLAYING_ID3);
|
|
struct mp3entry *id3 = valid_mp3entry(ply_id3);
|
|
|
|
playlist_update_resume_info(filling == STATE_ENDED ? NULL : id3);
|
|
|
|
if (id3)
|
|
{
|
|
send_event(PLAYBACK_EVENT_TRACK_FINISH, id3);
|
|
prev_track_elapsed = id3->elapsed;
|
|
}
|
|
else
|
|
{
|
|
prev_track_elapsed = 0;
|
|
}
|
|
}
|
|
|
|
/* Announce the beginning of the new track */
|
|
static void audio_playlist_track_change(void)
|
|
{
|
|
struct mp3entry *id3 = valid_mp3entry(id3_get(PLAYING_ID3));
|
|
|
|
if (id3)
|
|
send_event(PLAYBACK_EVENT_TRACK_CHANGE, id3);
|
|
|
|
position_key = pcmbuf_get_position_key();
|
|
|
|
playlist_update_resume_info(id3);
|
|
}
|
|
|
|
/* Change the data for the next track and send the event */
|
|
static void audio_update_and_announce_next_track(const struct mp3entry *id3_next)
|
|
{
|
|
id3_write_locked(NEXTTRACK_ID3, id3_next);
|
|
send_event(PLAYBACK_EVENT_NEXTTRACKID3_AVAILABLE,
|
|
id3_get(NEXTTRACK_ID3));
|
|
}
|
|
|
|
/* Bring the user current mp3entry up to date and set a new offset for the
|
|
buffered metadata */
|
|
static void playing_id3_sync(struct track_info *user_info, off_t offset)
|
|
{
|
|
id3_mutex_lock();
|
|
|
|
struct mp3entry *id3 = bufgetid3(user_info->id3_hid);
|
|
struct mp3entry *playing_id3 = id3_get(PLAYING_ID3);
|
|
|
|
pcm_play_lock();
|
|
|
|
unsigned long e = playing_id3->elapsed;
|
|
unsigned long o = playing_id3->offset;
|
|
|
|
id3_write(PLAYING_ID3, id3);
|
|
|
|
if (offset < 0)
|
|
{
|
|
playing_id3->elapsed = e;
|
|
playing_id3->offset = o;
|
|
offset = 0;
|
|
}
|
|
|
|
pcm_play_unlock();
|
|
|
|
if (id3)
|
|
id3->offset = offset;
|
|
|
|
id3_mutex_unlock();
|
|
}
|
|
|
|
/* Wipe-out track metadata - current is optional */
|
|
static void wipe_track_metadata(bool current)
|
|
{
|
|
id3_mutex_lock();
|
|
|
|
if (current)
|
|
id3_write(PLAYING_ID3, NULL);
|
|
|
|
id3_write(NEXTTRACK_ID3, NULL);
|
|
id3_write(UNBUFFERED_ID3, NULL);
|
|
|
|
id3_mutex_unlock();
|
|
}
|
|
|
|
/* Called when buffering is completed on the last track handle */
|
|
static void filling_is_finished(void)
|
|
{
|
|
logf("last track finished buffering");
|
|
|
|
/* There's no more to load or watch for */
|
|
buf_set_watermark(0);
|
|
filling = STATE_FINISHED;
|
|
}
|
|
|
|
/* Stop the codec decoding or waiting for its data to be ready - returns
|
|
'false' if the codec ended up stopped */
|
|
static bool halt_decoding_track(bool stop)
|
|
{
|
|
/* If it was waiting for us to clear the buffer to make a rebuffer
|
|
happen, it should cease otherwise codec_stop could deadlock waiting
|
|
for the codec to go to its main loop - codec's request will now
|
|
force-fail */
|
|
bool retval = false;
|
|
|
|
buf_signal_handle(ci.audio_hid, true);
|
|
|
|
if (stop)
|
|
codec_stop();
|
|
else
|
|
retval = codec_pause();
|
|
|
|
audio_clear_track_notifications();
|
|
|
|
/* We now know it's idle and not waiting for buffered data */
|
|
buf_signal_handle(ci.audio_hid, false);
|
|
|
|
codec_skip_pending = false;
|
|
codec_seeking = false;
|
|
|
|
return retval;
|
|
}
|
|
|
|
/* Wait for any in-progress fade to complete */
|
|
static void audio_wait_fade_complete(void)
|
|
{
|
|
/* Just loop until it's done */
|
|
while (pcmbuf_fading())
|
|
sleep(0);
|
|
}
|
|
|
|
/* End the ff/rw mode */
|
|
static void audio_ff_rewind_end(void)
|
|
{
|
|
/* A seamless seek (not calling audio_pre_ff_rewind) skips this
|
|
section */
|
|
if (ff_rw_mode)
|
|
{
|
|
ff_rw_mode = false;
|
|
|
|
if (codec_seeking)
|
|
{
|
|
/* Clear the buffer */
|
|
pcmbuf_play_stop();
|
|
audio_pcmbuf_sync_position();
|
|
}
|
|
|
|
if (play_status != PLAY_PAUSED)
|
|
{
|
|
/* Seeking-while-playing, resume PCM playback */
|
|
pcmbuf_pause(false);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Complete the codec seek */
|
|
static void audio_complete_codec_seek(void)
|
|
{
|
|
/* If a seek completed while paused, 'paused' is true.
|
|
* If seeking from seek mode, 'ff_rw_mode' is true. */
|
|
if (codec_seeking)
|
|
{
|
|
audio_ff_rewind_end();
|
|
codec_seeking = false; /* set _after_ the call! */
|
|
}
|
|
/* else it's waiting and we must repond */
|
|
}
|
|
|
|
/* Get the current cuesheet pointer */
|
|
static inline struct cuesheet * get_current_cuesheet(void)
|
|
{
|
|
return audio_scratch_memory->curr_cue;
|
|
}
|
|
|
|
/* Read the cuesheet from the buffer */
|
|
static void buf_read_cuesheet(int handle_id)
|
|
{
|
|
struct cuesheet *cue = get_current_cuesheet();
|
|
|
|
if (!cue || handle_id < 0)
|
|
return;
|
|
|
|
bufread(handle_id, sizeof (struct cuesheet), cue);
|
|
}
|
|
|
|
/* Backend to peek/current/next track metadata interface functions -
|
|
fill in the mp3entry with as much information as we may obtain about
|
|
the track at the specified offset from the user current track -
|
|
returns false if no information exists with us */
|
|
static bool audio_get_track_metadata(int offset, struct mp3entry *id3)
|
|
{
|
|
if (play_status == PLAY_STOPPED)
|
|
return false;
|
|
|
|
if (id3->path[0] != '\0')
|
|
return true; /* Already filled */
|
|
|
|
struct track_info *info = track_list_user_current(offset);
|
|
|
|
if (!info)
|
|
{
|
|
struct mp3entry *ub_id3 = id3_get(UNBUFFERED_ID3);
|
|
|
|
if (offset > 0 && track_list_user_current(offset - 1))
|
|
{
|
|
/* Try the unbuffered id3 since we're moving forward */
|
|
if (ub_id3->path[0] != '\0')
|
|
{
|
|
copy_mp3entry(id3, ub_id3);
|
|
return true;
|
|
}
|
|
}
|
|
}
|
|
else if (bufreadid3(info->id3_hid, id3))
|
|
{
|
|
id3->cuesheet = NULL;
|
|
return true;
|
|
}
|
|
|
|
/* We didn't find the ID3 metadata, so we fill it with the little info we
|
|
have and return that */
|
|
|
|
char path[MAX_PATH+1];
|
|
if (playlist_peek(offset, path, sizeof (path)))
|
|
{
|
|
#if defined(HAVE_TC_RAMCACHE) && defined(HAVE_DIRCACHE)
|
|
/* Try to get it from the database */
|
|
if (!tagcache_fill_tags(id3, path))
|
|
#endif
|
|
{
|
|
/* By now, filename is the only source of info */
|
|
fill_metadata_from_path(id3, path);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
wipe_mp3entry(id3);
|
|
|
|
return false;
|
|
}
|
|
|
|
/* Get a resume rewind adjusted offset from the ID3 */
|
|
unsigned long resume_rewind_adjusted_offset(const struct mp3entry *id3)
|
|
{
|
|
unsigned long offset = id3->offset;
|
|
size_t resume_rewind = global_settings.resume_rewind *
|
|
id3->bitrate * (1000/8);
|
|
|
|
if (offset < resume_rewind)
|
|
offset = 0;
|
|
else
|
|
offset -= resume_rewind;
|
|
|
|
return offset;
|
|
}
|
|
|
|
/* Get the codec into ram and initialize it - keep it if it's ready */
|
|
static bool audio_init_codec(struct track_info *track_info,
|
|
struct mp3entry *track_id3)
|
|
{
|
|
int codt_loaded = get_audio_base_codec_type(codec_loaded());
|
|
int hid = ERR_HANDLE_NOT_FOUND;
|
|
|
|
if (codt_loaded != AFMT_UNKNOWN)
|
|
{
|
|
int codt = get_audio_base_codec_type(track_id3->codectype);
|
|
|
|
if (codt == codt_loaded)
|
|
{
|
|
/* Codec is the same base type */
|
|
logf("Reusing prev. codec: %d", track_id3->codectype);
|
|
#ifdef HAVE_CODEC_BUFFERING
|
|
/* Close any buffered codec (we could have skipped directly to a
|
|
format transistion that is the same format as the current track
|
|
and the buffered one is no longer needed) */
|
|
track_info_close_handle(&track_info->codec_hid);
|
|
#endif
|
|
return true;
|
|
}
|
|
else
|
|
{
|
|
/* New codec - first make sure the old one's gone */
|
|
logf("Removing prev. codec: %d", codt_loaded);
|
|
codec_unload();
|
|
}
|
|
}
|
|
|
|
logf("New codec: %d/%d", track_id3->codectype, codec_loaded());
|
|
|
|
#ifdef HAVE_CODEC_BUFFERING
|
|
/* Codec thread will close the handle even if it fails and will load from
|
|
storage if hid is not valid or the buffer load fails */
|
|
hid = track_info->codec_hid;
|
|
track_info->codec_hid = ERR_HANDLE_NOT_FOUND;
|
|
#endif
|
|
|
|
return codec_load(hid, track_id3->codectype);
|
|
(void)track_info; /* When codec buffering isn't supported */
|
|
}
|
|
|
|
/* Start the codec for the current track scheduled to be decoded */
|
|
static bool audio_start_codec(bool auto_skip)
|
|
{
|
|
struct track_info *info = track_list_current(0);
|
|
struct mp3entry *cur_id3 = valid_mp3entry(bufgetid3(info->id3_hid));
|
|
|
|
if (!cur_id3)
|
|
return false;
|
|
|
|
buf_pin_handle(info->id3_hid, true);
|
|
|
|
if (!audio_init_codec(info, cur_id3))
|
|
{
|
|
buf_pin_handle(info->id3_hid, false);
|
|
return false;
|
|
}
|
|
|
|
#ifdef HAVE_TAGCACHE
|
|
bool autoresume_enable = global_settings.autoresume_enable;
|
|
|
|
if (autoresume_enable && !cur_id3->offset)
|
|
{
|
|
/* Resume all manually selected tracks */
|
|
bool resume = !auto_skip;
|
|
|
|
/* Send the "buffer" event to obtain the resume position for the codec */
|
|
send_event(PLAYBACK_EVENT_TRACK_BUFFER, cur_id3);
|
|
|
|
if (!resume)
|
|
{
|
|
/* Automatic skip - do further tests to see if we should just
|
|
ignore any autoresume position */
|
|
int autoresume_automatic = global_settings.autoresume_automatic;
|
|
|
|
switch (autoresume_automatic)
|
|
{
|
|
case AUTORESUME_NEXTTRACK_ALWAYS:
|
|
/* Just resume unconditionally */
|
|
resume = true;
|
|
break;
|
|
case AUTORESUME_NEXTTRACK_NEVER:
|
|
/* Force-rewind it */
|
|
break;
|
|
default:
|
|
/* Not "never resume" - pass resume filter? */
|
|
resume = autoresumable(cur_id3);
|
|
}
|
|
}
|
|
|
|
if (!resume)
|
|
cur_id3->offset = 0;
|
|
|
|
logf("%s: Set offset for %s to %lX\n", __func__,
|
|
cur_id3->title, cur_id3->offset);
|
|
}
|
|
#endif /* HAVE_TAGCACHE */
|
|
|
|
/* Rewind the required amount - if an autoresume was done, this also rewinds
|
|
that by the setting's amount
|
|
|
|
It would be best to have bookkeeping about whether or not the track
|
|
sounded or not since skipping to it or else skipping to it while paused
|
|
and back again will cause accumulation of silent rewinds - that's not
|
|
our job to track directly nor could it be in any reasonable way
|
|
*/
|
|
cur_id3->offset = resume_rewind_adjusted_offset(cur_id3);
|
|
|
|
/* Update the codec API with the metadata and track info */
|
|
id3_write(CODEC_ID3, cur_id3);
|
|
|
|
ci.audio_hid = info->audio_hid;
|
|
ci.filesize = info->filesize;
|
|
buf_set_base_handle(info->audio_hid);
|
|
|
|
/* All required data is now available for the codec */
|
|
codec_go();
|
|
|
|
#ifdef HAVE_TAGCACHE
|
|
if (!autoresume_enable || cur_id3->offset)
|
|
#endif
|
|
{
|
|
/* Send the "buffer" event now */
|
|
send_event(PLAYBACK_EVENT_TRACK_BUFFER, cur_id3);
|
|
}
|
|
|
|
buf_pin_handle(info->id3_hid, false);
|
|
return true;
|
|
|
|
(void)auto_skip; /* ifndef HAVE_TAGCACHE */
|
|
}
|
|
|
|
|
|
/** --- Audio thread --- **/
|
|
|
|
/* Load and parse a cuesheet for the file - returns false if the buffer
|
|
is full */
|
|
static bool audio_load_cuesheet(struct track_info *info,
|
|
struct mp3entry *track_id3)
|
|
{
|
|
struct cuesheet *cue = get_current_cuesheet();
|
|
track_id3->cuesheet = NULL;
|
|
|
|
if (cue && info->cuesheet_hid == ERR_HANDLE_NOT_FOUND)
|
|
{
|
|
/* If error other than a full buffer, then mark it "unsupported" to
|
|
avoid reloading attempt */
|
|
int hid = ERR_UNSUPPORTED_TYPE;
|
|
char cuepath[MAX_PATH];
|
|
|
|
#ifdef HAVE_IO_PRIORITY
|
|
buf_back_off_storage(true);
|
|
#endif
|
|
if (look_for_cuesheet_file(track_id3->path, cuepath))
|
|
{
|
|
hid = bufalloc(NULL, sizeof (struct cuesheet), TYPE_CUESHEET);
|
|
|
|
if (hid >= 0)
|
|
{
|
|
void *cuesheet = NULL;
|
|
bufgetdata(hid, sizeof (struct cuesheet), &cuesheet);
|
|
|
|
if (parse_cuesheet(cuepath, (struct cuesheet *)cuesheet))
|
|
{
|
|
/* Indicate cuesheet is present (while track remains
|
|
buffered) */
|
|
track_id3->cuesheet = cue;
|
|
}
|
|
else
|
|
{
|
|
bufclose(hid);
|
|
hid = ERR_UNSUPPORTED_TYPE;
|
|
}
|
|
}
|
|
}
|
|
|
|
#ifdef HAVE_IO_PRIORITY
|
|
buf_back_off_storage(false);
|
|
#endif
|
|
if (hid == ERR_BUFFER_FULL)
|
|
{
|
|
logf("buffer is full for now (%s)", __func__);
|
|
return false;
|
|
}
|
|
else
|
|
{
|
|
if (hid < 0)
|
|
logf("Cuesheet loading failed");
|
|
|
|
info->cuesheet_hid = hid;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
#ifdef HAVE_ALBUMART
|
|
/* Load any album art for the file - returns false if the buffer is full */
|
|
static bool audio_load_albumart(struct track_info *info,
|
|
struct mp3entry *track_id3)
|
|
{
|
|
int i;
|
|
FOREACH_ALBUMART(i)
|
|
{
|
|
struct bufopen_bitmap_data user_data;
|
|
int *aa_hid = &info->aa_hid[i];
|
|
int hid = ERR_UNSUPPORTED_TYPE;
|
|
|
|
/* albumart_slots may change during a yield of bufopen,
|
|
* but that's no problem */
|
|
if (*aa_hid >= 0 || *aa_hid == ERR_UNSUPPORTED_TYPE ||
|
|
!albumart_slots[i].used)
|
|
continue;
|
|
|
|
memset(&user_data, 0, sizeof(user_data));
|
|
user_data.dim = &albumart_slots[i].dim;
|
|
|
|
#ifdef HAVE_IO_PRIORITY
|
|
buf_back_off_storage(true);
|
|
#endif
|
|
|
|
/* We can only decode jpeg for embedded AA */
|
|
if (track_id3->embed_albumart && track_id3->albumart.type == AA_TYPE_JPG)
|
|
{
|
|
user_data.embedded_albumart = &track_id3->albumart;
|
|
hid = bufopen(track_id3->path, 0, TYPE_BITMAP, &user_data);
|
|
}
|
|
|
|
if (hid < 0 && hid != ERR_BUFFER_FULL)
|
|
{
|
|
/* No embedded AA or it couldn't be loaded - try other sources */
|
|
char path[MAX_PATH];
|
|
|
|
if (find_albumart(track_id3, path, sizeof(path),
|
|
&albumart_slots[i].dim))
|
|
{
|
|
user_data.embedded_albumart = NULL;
|
|
hid = bufopen(path, 0, TYPE_BITMAP, &user_data);
|
|
}
|
|
}
|
|
|
|
#ifdef HAVE_IO_PRIORITY
|
|
buf_back_off_storage(false);
|
|
#endif
|
|
if (hid == ERR_BUFFER_FULL)
|
|
{
|
|
logf("buffer is full for now (%s)", __func__);
|
|
return false;
|
|
}
|
|
else
|
|
{
|
|
/* If error other than a full buffer, then mark it "unsupported"
|
|
to avoid reloading attempt */
|
|
if (hid < 0)
|
|
{
|
|
logf("Album art loading failed");
|
|
hid = ERR_UNSUPPORTED_TYPE;
|
|
}
|
|
|
|
*aa_hid = hid;
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
#endif /* HAVE_ALBUMART */
|
|
|
|
#ifdef HAVE_CODEC_BUFFERING
|
|
/* Load a codec for the file onto the buffer - assumes we're working from the
|
|
currently loading track - not called for the current track */
|
|
static bool audio_buffer_codec(struct track_info *track_info,
|
|
struct mp3entry *track_id3)
|
|
{
|
|
/* This will not be the current track -> it cannot be the first and the
|
|
current track cannot be ahead of buffering -> there is a previous
|
|
track entry which is either current or ahead of the current */
|
|
struct track_info *prev_info = track_list_last(-1);
|
|
struct mp3entry *prev_id3 = bufgetid3(prev_info->id3_hid);
|
|
|
|
/* If the previous codec is the same as this one, there is no need to
|
|
put another copy of it on the file buffer (in other words, only
|
|
buffer codecs at format transitions) */
|
|
if (prev_id3)
|
|
{
|
|
int codt = get_audio_base_codec_type(track_id3->codectype);
|
|
int prev_codt = get_audio_base_codec_type(prev_id3->codectype);
|
|
|
|
if (codt == prev_codt)
|
|
{
|
|
logf("Reusing prev. codec: %d", prev_id3->codectype);
|
|
return true;
|
|
}
|
|
}
|
|
/* else just load it (harmless) */
|
|
|
|
/* Load the codec onto the buffer if possible */
|
|
const char *codec_fn = get_codec_filename(track_id3->codectype);
|
|
if (!codec_fn)
|
|
return false;
|
|
|
|
char codec_path[MAX_PATH+1]; /* Full path to codec */
|
|
codec_get_full_path(codec_path, codec_fn);
|
|
|
|
track_info->codec_hid = bufopen(codec_path, 0, TYPE_CODEC, NULL);
|
|
|
|
if (track_info->codec_hid >= 0)
|
|
{
|
|
logf("Buffered codec: %d", afmt);
|
|
return true;
|
|
}
|
|
|
|
return false;
|
|
}
|
|
#endif /* HAVE_CODEC_BUFFERING */
|
|
|
|
/* Load metadata for the next track (with bufopen). The rest of the track
|
|
loading will be handled by audio_finish_load_track once the metadata has
|
|
been actually loaded by the buffering thread.
|
|
|
|
Each track is arranged in the buffer as follows:
|
|
<id3|[cuesheet|][album art|][codec|]audio>
|
|
|
|
The next will not be loaded until the previous succeeds if the buffer was
|
|
full at the time. To put any metadata after audio would make those handles
|
|
unmovable.
|
|
*/
|
|
static int audio_load_track(void)
|
|
{
|
|
if (in_progress_id3_hid >= 0)
|
|
{
|
|
/* There must be an info pointer if the in-progress id3 is even there */
|
|
struct track_info *info = track_list_last(0);
|
|
|
|
if (info->id3_hid == in_progress_id3_hid)
|
|
{
|
|
if (filling == STATE_FILLING)
|
|
{
|
|
/* Haven't finished the metadata but the notification is
|
|
anticipated to come soon */
|
|
logf("%s(): in progress ok: %d". __func__, info->id3_hid);
|
|
return LOAD_TRACK_OK;
|
|
}
|
|
else if (filling == STATE_FULL)
|
|
{
|
|
/* Buffer was full trying to complete the load after the
|
|
metadata finished, so attempt to continue - older handles
|
|
should have been cleared already */
|
|
logf("%s(): finishing load: %d". __func__, info->id3_hid);
|
|
filling = STATE_FILLING;
|
|
buffer_event_finished_callback(&info->id3_hid);
|
|
return LOAD_TRACK_OK;
|
|
}
|
|
}
|
|
|
|
/* Some old, stray buffering message */
|
|
logf("%s(): already in progress: %d". __func__, info->id3_hid);
|
|
return LOAD_TRACK_ERR_BUSY;
|
|
}
|
|
|
|
filling = STATE_FILLING;
|
|
|
|
struct track_info *info = track_list_alloc_track();
|
|
if (info == NULL)
|
|
{
|
|
/* List is full so stop buffering tracks - however, attempt to obtain
|
|
metadata as the unbuffered id3 */
|
|
logf("No free tracks");
|
|
filling = STATE_FULL;
|
|
}
|
|
|
|
playlist_peek_offset++;
|
|
|
|
logf("Buffering track: s%u/c%u/e%u/p%d",
|
|
track_list.start, track_list.current, track_list.end,
|
|
playlist_peek_offset);
|
|
|
|
/* Get track name from current playlist read position */
|
|
int fd = -1;
|
|
char name_buf[MAX_PATH + 1];
|
|
const char *trackname;
|
|
|
|
while (1)
|
|
{
|
|
|
|
trackname = playlist_peek(playlist_peek_offset, name_buf,
|
|
sizeof (name_buf));
|
|
|
|
if (!trackname)
|
|
break;
|
|
|
|
/* Test for broken playlists by probing for the files */
|
|
fd = open(trackname, O_RDONLY);
|
|
if (fd >= 0)
|
|
break;
|
|
|
|
logf("Open failed");
|
|
/* Skip invalid entry from playlist */
|
|
playlist_skip_entry(NULL, playlist_peek_offset);
|
|
|
|
/* Sync the playlist if it isn't finished */
|
|
if (playlist_peek(playlist_peek_offset, NULL, 0))
|
|
playlist_next(0);
|
|
}
|
|
|
|
if (!trackname)
|
|
{
|
|
/* No track - exhausted the playlist entries */
|
|
logf("End-of-playlist");
|
|
id3_write_locked(UNBUFFERED_ID3, NULL);
|
|
|
|
if (filling != STATE_FULL)
|
|
track_list_unalloc_track(); /* Free this entry */
|
|
|
|
playlist_peek_offset--; /* Maintain at last index */
|
|
|
|
/* We can end up here after the real last track signals its completion
|
|
and miss the transition to STATE_FINISHED esp. if dropping the last
|
|
songs of a playlist late in their load (2nd stage) */
|
|
info = track_list_last(0);
|
|
|
|
if (info && buf_handle_remaining(info->audio_hid) == 0)
|
|
filling_is_finished();
|
|
else
|
|
filling = STATE_END_OF_PLAYLIST;
|
|
|
|
return LOAD_TRACK_ERR_NO_MORE;
|
|
}
|
|
|
|
/* Successfully opened the file - get track metadata */
|
|
if (filling == STATE_FULL ||
|
|
(info->id3_hid = bufopen(trackname, 0, TYPE_ID3, NULL)) < 0)
|
|
{
|
|
/* Buffer or track list is full */
|
|
struct mp3entry *ub_id3;
|
|
|
|
playlist_peek_offset--;
|
|
|
|
/* Load the metadata for the first unbuffered track */
|
|
ub_id3 = id3_get(UNBUFFERED_ID3);
|
|
id3_mutex_lock();
|
|
get_metadata(ub_id3, fd, trackname);
|
|
id3_mutex_unlock();
|
|
|
|
if (filling != STATE_FULL)
|
|
{
|
|
track_list_unalloc_track();
|
|
filling = STATE_FULL;
|
|
}
|
|
|
|
logf("%s: buffer is full for now (%u tracks)", __func__,
|
|
track_list_count());
|
|
}
|
|
else
|
|
{
|
|
/* Successful load initiation */
|
|
info->filesize = filesize(fd);
|
|
in_progress_id3_hid = info->id3_hid; /* Remember what's in-progress */
|
|
}
|
|
|
|
close(fd);
|
|
return LOAD_TRACK_OK;
|
|
}
|
|
|
|
/* Second part of the track loading: We now have the metadata available, so we
|
|
can load the codec, the album art and finally the audio data.
|
|
This is called on the audio thread after the buffering thread calls the
|
|
buffering_handle_finished_callback callback. */
|
|
static int audio_finish_load_track(struct track_info *info)
|
|
{
|
|
int trackstat = LOAD_TRACK_OK;
|
|
|
|
if (info->id3_hid != in_progress_id3_hid)
|
|
{
|
|
/* We must not be here if not! */
|
|
logf("%s: wrong track %d/%d", __func__, info->id3_hid,
|
|
in_progress_id3_hid);
|
|
return LOAD_TRACK_ERR_BUSY;
|
|
}
|
|
|
|
/* The current track for decoding (there is always one if the list is
|
|
populated) */
|
|
struct track_info *cur_info = track_list_current(0);
|
|
struct mp3entry *track_id3 = valid_mp3entry(bufgetid3(info->id3_hid));
|
|
|
|
if (!track_id3)
|
|
{
|
|
/* This is an error condition. Track cannot be played without valid
|
|
metadata; skip the track. */
|
|
logf("No metadata");
|
|
trackstat = LOAD_TRACK_ERR_FINISH_FAILED;
|
|
goto audio_finish_load_track_exit;
|
|
}
|
|
|
|
/* Try to load a cuesheet for the track */
|
|
if (!audio_load_cuesheet(info, track_id3))
|
|
{
|
|
/* No space for cuesheet on buffer, not an error */
|
|
filling = STATE_FULL;
|
|
goto audio_finish_load_track_exit;
|
|
}
|
|
|
|
#ifdef HAVE_ALBUMART
|
|
/* Try to load album art for the track */
|
|
if (!audio_load_albumart(info, track_id3))
|
|
{
|
|
/* No space for album art on buffer, not an error */
|
|
filling = STATE_FULL;
|
|
goto audio_finish_load_track_exit;
|
|
}
|
|
#endif
|
|
|
|
/* All handles available to external routines are ready - audio and codec
|
|
information is private */
|
|
|
|
if (info == track_list_user_current(0))
|
|
{
|
|
/* Send only when the track handles could not all be opened ahead of
|
|
time for the user's current track - otherwise everything is ready
|
|
by the time PLAYBACK_EVENT_TRACK_CHANGE is sent */
|
|
send_event(PLAYBACK_EVENT_CUR_TRACK_READY, id3_get(PLAYING_ID3));
|
|
}
|
|
|
|
#ifdef HAVE_CODEC_BUFFERING
|
|
/* Try to buffer a codec for the track */
|
|
if (info != cur_info && !audio_buffer_codec(info, track_id3))
|
|
{
|
|
if (info->codec_hid == ERR_BUFFER_FULL)
|
|
{
|
|
/* No space for codec on buffer, not an error */
|
|
filling = STATE_FULL;
|
|
logf("buffer is full for now (%s)", __func__);
|
|
}
|
|
else
|
|
{
|
|
/* This is an error condition, either no codec was found, or
|
|
reading the codec file failed part way through, either way,
|
|
skip the track */
|
|
logf("No codec for: %s", track_id3->path);
|
|
trackstat = LOAD_TRACK_ERR_FINISH_FAILED;
|
|
}
|
|
|
|
goto audio_finish_load_track_exit;
|
|
}
|
|
#endif /* HAVE_CODEC_BUFFERING */
|
|
|
|
/** Finally, load the audio **/
|
|
size_t file_offset = 0;
|
|
track_id3->elapsed = 0;
|
|
|
|
if (track_id3->offset >= info->filesize)
|
|
track_id3->offset = 0;
|
|
|
|
logf("%s: set offset for %s to %lu\n", __func__,
|
|
id3->title, (unsigned long)offset);
|
|
|
|
/* Adjust for resume rewind so we know what to buffer - starting the codec
|
|
calls it again, so we don't save it (and they shouldn't accumulate) */
|
|
size_t offset = resume_rewind_adjusted_offset(track_id3);
|
|
|
|
enum data_type audiotype = get_audio_base_data_type(track_id3->codectype);
|
|
|
|
if (audiotype == TYPE_ATOMIC_AUDIO)
|
|
logf("Loading atomic %d", track_id3->codectype);
|
|
|
|
if (format_buffers_with_offset(track_id3->codectype))
|
|
{
|
|
/* This format can begin buffering from any point */
|
|
file_offset = offset;
|
|
}
|
|
|
|
logf("load track: %s", track_id3->path);
|
|
|
|
if (file_offset > AUDIO_REBUFFER_GUESS_SIZE)
|
|
{
|
|
/* We can buffer later in the file, adjust the hunt-and-peck margin */
|
|
file_offset -= AUDIO_REBUFFER_GUESS_SIZE;
|
|
}
|
|
else
|
|
{
|
|
/* No offset given or it is very minimal - begin at the first frame
|
|
according to the metadata */
|
|
file_offset = track_id3->first_frame_offset;
|
|
}
|
|
|
|
int hid = bufopen(track_id3->path, file_offset, audiotype, NULL);
|
|
|
|
if (hid >= 0)
|
|
{
|
|
info->audio_hid = hid;
|
|
|
|
if (info == cur_info)
|
|
{
|
|
/* This is the current track to decode - should be started now */
|
|
trackstat = LOAD_TRACK_READY;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Buffer could be full but not properly so if this is the only
|
|
track! */
|
|
if (hid == ERR_BUFFER_FULL && audio_track_count() > 1)
|
|
{
|
|
filling = STATE_FULL;
|
|
logf("Buffer is full for now (%s)", __func__);
|
|
}
|
|
else
|
|
{
|
|
/* Nothing to play if no audio handle - skip this */
|
|
logf("Could not add audio data handle");
|
|
trackstat = LOAD_TRACK_ERR_FINISH_FAILED;
|
|
}
|
|
}
|
|
|
|
audio_finish_load_track_exit:
|
|
if (trackstat < LOAD_TRACK_OK)
|
|
{
|
|
playlist_skip_entry(NULL, playlist_peek_offset);
|
|
track_info_close(info);
|
|
track_list_unalloc_track();
|
|
|
|
if (playlist_peek(playlist_peek_offset, NULL, 0))
|
|
playlist_next(0);
|
|
|
|
playlist_peek_offset--;
|
|
}
|
|
|
|
if (filling != STATE_FULL)
|
|
{
|
|
/* Load next track - error or not */
|
|
in_progress_id3_hid = ERR_HANDLE_NOT_FOUND;
|
|
LOGFQUEUE("audio > audio Q_AUDIO_FILL_BUFFER");
|
|
audio_queue_post(Q_AUDIO_FILL_BUFFER, 0);
|
|
}
|
|
else
|
|
{
|
|
/* Full */
|
|
trackstat = LOAD_TRACK_ERR_FINISH_FULL;
|
|
}
|
|
|
|
return trackstat;
|
|
}
|
|
|
|
/* Start a new track load */
|
|
static int audio_fill_file_buffer(void)
|
|
{
|
|
if (play_status == PLAY_STOPPED)
|
|
return LOAD_TRACK_ERR_FAILED;
|
|
|
|
trigger_cpu_boost();
|
|
|
|
/* Must reset the buffer before use if trashed or voice only - voice
|
|
file size shouldn't have changed so we can go straight from
|
|
AUDIOBUF_STATE_VOICED_ONLY to AUDIOBUF_STATE_INITIALIZED */
|
|
if (buffer_state != AUDIOBUF_STATE_INITIALIZED)
|
|
audio_reset_buffer();
|
|
|
|
logf("Starting buffer fill");
|
|
|
|
int trackstat = audio_load_track();
|
|
|
|
if (trackstat >= LOAD_TRACK_OK)
|
|
{
|
|
if (track_list_current(0) == track_list_user_current(0))
|
|
playlist_next(0);
|
|
|
|
if (filling == STATE_FULL && !track_list_user_current(1))
|
|
{
|
|
/* There are no user tracks on the buffer after this therefore
|
|
this is the next track */
|
|
audio_update_and_announce_next_track(id3_get(UNBUFFERED_ID3));
|
|
}
|
|
}
|
|
|
|
return trackstat;
|
|
}
|
|
|
|
/* Discard unwanted tracks and start refill from after the specified playlist
|
|
offset */
|
|
static int audio_reset_and_rebuffer(
|
|
enum track_clear_action action, int peek_offset)
|
|
{
|
|
logf("Forcing rebuffer: 0x%X, %d", flags, peek_offset);
|
|
|
|
id3_write_locked(UNBUFFERED_ID3, NULL);
|
|
|
|
/* Remove unwanted tracks - caller must have ensured codec isn't using
|
|
any */
|
|
track_list_clear(action);
|
|
|
|
/* Refill at specified position (-1 starts at index offset 0) */
|
|
playlist_peek_offset = peek_offset;
|
|
|
|
/* Fill the buffer */
|
|
return audio_fill_file_buffer();
|
|
}
|
|
|
|
/* Handle buffering events
|
|
(Q_AUDIO_BUFFERING) */
|
|
static void audio_on_buffering(int event)
|
|
{
|
|
enum track_clear_action action;
|
|
int peek_offset;
|
|
|
|
if (track_list_empty())
|
|
return;
|
|
|
|
switch (event)
|
|
{
|
|
case BUFFER_EVENT_BUFFER_LOW:
|
|
if (filling != STATE_FULL && filling != STATE_END_OF_PLAYLIST)
|
|
return; /* Should be nothing left to fill */
|
|
|
|
/* Clear old tracks and continue buffering where it left off */
|
|
action = TRACK_LIST_KEEP_NEW;
|
|
peek_offset = playlist_peek_offset;
|
|
break;
|
|
|
|
case BUFFER_EVENT_REBUFFER:
|
|
/* Remove all but the currently decoding track and redo buffering
|
|
after that */
|
|
action = TRACK_LIST_KEEP_CURRENT;
|
|
peek_offset = (skip_pending == TRACK_SKIP_AUTO) ? 1 : 0;
|
|
break;
|
|
|
|
default:
|
|
return;
|
|
}
|
|
|
|
switch (skip_pending)
|
|
{
|
|
case TRACK_SKIP_NONE:
|
|
case TRACK_SKIP_AUTO:
|
|
case TRACK_SKIP_AUTO_NEW_PLAYLIST:
|
|
audio_reset_and_rebuffer(action, peek_offset);
|
|
break;
|
|
|
|
case TRACK_SKIP_AUTO_END_PLAYLIST:
|
|
/* Already finished */
|
|
break;
|
|
|
|
default:
|
|
/* Invalid */
|
|
logf("Buffering call, inv. state: %d", (int)skip_pending);
|
|
}
|
|
}
|
|
|
|
/* Handle starting the next track load
|
|
(Q_AUDIO_FILL_BUFFER) */
|
|
static void audio_on_fill_buffer(void)
|
|
{
|
|
audio_handle_track_load_status(audio_fill_file_buffer());
|
|
}
|
|
|
|
/* Handle posted load track finish event
|
|
(Q_AUDIO_FINISH_LOAD_TRACK) */
|
|
static void audio_on_finish_load_track(int id3_hid)
|
|
{
|
|
struct track_info *info = track_list_last(0);
|
|
|
|
if (!info || !buf_is_handle(id3_hid))
|
|
return;
|
|
|
|
if (info == track_list_user_current(1))
|
|
{
|
|
/* Just loaded the metadata right after the current position */
|
|
audio_update_and_announce_next_track(bufgetid3(info->id3_hid));
|
|
}
|
|
|
|
if (audio_finish_load_track(info) != LOAD_TRACK_READY)
|
|
return; /* Not current track */
|
|
|
|
bool is_user_current = info == track_list_user_current(0);
|
|
|
|
if (is_user_current)
|
|
{
|
|
/* Copy cuesheet */
|
|
buf_read_cuesheet(info->cuesheet_hid);
|
|
}
|
|
|
|
if (audio_start_codec(automatic_skip))
|
|
{
|
|
if (is_user_current)
|
|
{
|
|
/* Be sure all tagtree info is synchronized; it will be needed for the
|
|
track finish event - the sync will happen when finalizing a track
|
|
change otherwise */
|
|
bool was_valid = valid_mp3entry(id3_get(PLAYING_ID3));
|
|
|
|
playing_id3_sync(info, -1);
|
|
|
|
if (!was_valid)
|
|
{
|
|
/* Playing id3 hadn't been updated yet because no valid track
|
|
was yet available - treat like the first track */
|
|
audio_playlist_track_change();
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
audio_handle_track_load_status(LOAD_TRACK_ERR_START_CODEC);
|
|
}
|
|
}
|
|
|
|
/* Called when handles other than metadata handles have finished buffering
|
|
(Q_AUDIO_HANDLE_FINISHED) */
|
|
static void audio_on_handle_finished(int hid)
|
|
{
|
|
/* Right now, only audio handles should end up calling this */
|
|
if (filling == STATE_END_OF_PLAYLIST)
|
|
{
|
|
struct track_info *info = track_list_last(0);
|
|
|
|
/* Really we don't know which order the handles will actually complete
|
|
to zero bytes remaining since another thread is doing it - be sure
|
|
it's the right one */
|
|
if (info && info->audio_hid == hid)
|
|
{
|
|
/* This was the last track in the playlist and we now have all the
|
|
data we need */
|
|
filling_is_finished();
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Called to make an outstanding track skip the current track and to send the
|
|
transition events */
|
|
static void audio_finalise_track_change(void)
|
|
{
|
|
switch (skip_pending)
|
|
{
|
|
case TRACK_SKIP_NONE: /* Manual skip */
|
|
break;
|
|
|
|
case TRACK_SKIP_AUTO:
|
|
case TRACK_SKIP_AUTO_NEW_PLAYLIST:
|
|
{
|
|
int playlist_delta = skip_pending == TRACK_SKIP_AUTO ? 1 : 0;
|
|
audio_playlist_track_finish();
|
|
|
|
if (!playlist_peek(playlist_delta, NULL, 0))
|
|
{
|
|
/* Track ended up rejected - push things ahead like the codec blew
|
|
it (because it was never started and now we're here where it
|
|
should have been decoding the next track by now) - next, a
|
|
directory change or end of playback will most likely happen */
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
audio_handle_track_load_status(LOAD_TRACK_ERR_START_CODEC);
|
|
return;
|
|
}
|
|
|
|
if (!playlist_delta)
|
|
break;
|
|
|
|
playlist_peek_offset -= playlist_delta;
|
|
if (playlist_next(playlist_delta) >= 0)
|
|
break;
|
|
/* What!? Disappear? Hopeless bleak despair */
|
|
}
|
|
/* Fallthrough */
|
|
case TRACK_SKIP_AUTO_END_PLAYLIST:
|
|
default: /* Invalid */
|
|
filling = STATE_ENDED;
|
|
audio_stop_playback();
|
|
return;
|
|
}
|
|
|
|
struct track_info *info = track_list_current(0);
|
|
struct mp3entry *track_id3 = NULL;
|
|
|
|
id3_mutex_lock();
|
|
|
|
/* Update the current cuesheet if any and enabled */
|
|
if (info)
|
|
{
|
|
buf_read_cuesheet(info->cuesheet_hid);
|
|
track_id3 = bufgetid3(info->id3_hid);
|
|
}
|
|
|
|
id3_write(PLAYING_ID3, track_id3);
|
|
|
|
/* The skip is technically over */
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
|
|
/* Sync the next track information */
|
|
info = track_list_current(1);
|
|
|
|
id3_write(NEXTTRACK_ID3, info ? bufgetid3(info->id3_hid) :
|
|
id3_get(UNBUFFERED_ID3));
|
|
|
|
id3_mutex_unlock();
|
|
|
|
audio_playlist_track_change();
|
|
}
|
|
|
|
/* Actually begin a transition and take care of the codec change - may complete
|
|
it now or ask pcmbuf for notification depending on the type */
|
|
static void audio_begin_track_change(enum pcm_track_change_type type,
|
|
int trackstat)
|
|
{
|
|
/* Even if the new track is bad, the old track must be finished off */
|
|
pcmbuf_start_track_change(type);
|
|
|
|
bool auto_skip = type != TRACK_CHANGE_MANUAL;
|
|
|
|
if (!auto_skip)
|
|
{
|
|
/* Manual track change happens now */
|
|
audio_finalise_track_change();
|
|
pcmbuf_sync_position_update();
|
|
|
|
if (play_status == PLAY_STOPPED)
|
|
return; /* Stopped us */
|
|
}
|
|
|
|
if (trackstat >= LOAD_TRACK_OK)
|
|
{
|
|
struct track_info *info = track_list_current(0);
|
|
|
|
if (info->audio_hid < 0)
|
|
return;
|
|
|
|
/* Everything needed for the codec is ready - start it */
|
|
if (audio_start_codec(auto_skip))
|
|
{
|
|
if (!auto_skip)
|
|
playing_id3_sync(info, -1);
|
|
return;
|
|
}
|
|
|
|
trackstat = LOAD_TRACK_ERR_START_CODEC;
|
|
}
|
|
|
|
audio_handle_track_load_status(trackstat);
|
|
}
|
|
|
|
/* Transition to end-of-playlist state and begin wait for PCM to finish */
|
|
static void audio_monitor_end_of_playlist(void)
|
|
{
|
|
skip_pending = TRACK_SKIP_AUTO_END_PLAYLIST;
|
|
filling = STATE_ENDING;
|
|
pcmbuf_start_track_change(TRACK_CHANGE_END_OF_DATA);
|
|
}
|
|
|
|
/* Codec has completed decoding the track
|
|
(usually Q_AUDIO_CODEC_COMPLETE) */
|
|
static void audio_on_codec_complete(int status)
|
|
{
|
|
logf("%s(%d)", __func__, status);
|
|
|
|
if (play_status == PLAY_STOPPED)
|
|
return;
|
|
|
|
/* If it didn't notify us first, don't expect "seek complete" message
|
|
since the codec can't post it now - do things like it would have
|
|
done */
|
|
audio_complete_codec_seek();
|
|
|
|
if (play_status == PLAY_PAUSED || skip_pending != TRACK_SKIP_NONE)
|
|
{
|
|
/* Old-hay on the ip-skay - codec has completed decoding
|
|
|
|
Paused: We're not sounding it, so just remember that it happened
|
|
and the resume will begin the transition
|
|
|
|
Skipping: There was already a skip in progress, remember it and
|
|
allow no further progress until the PCM from the previous
|
|
song has finished
|
|
*/
|
|
codec_skip_pending = true;
|
|
codec_skip_status = status;
|
|
return;
|
|
}
|
|
|
|
codec_skip_pending = false;
|
|
|
|
int trackstat = LOAD_TRACK_OK;
|
|
|
|
automatic_skip = true;
|
|
skip_pending = TRACK_SKIP_AUTO;
|
|
|
|
/* Does this track have an entry allocated? */
|
|
struct track_info *info = track_list_advance_current(1);
|
|
|
|
if (!info || info->audio_hid < 0)
|
|
{
|
|
bool end_of_playlist = false;
|
|
|
|
if (info)
|
|
{
|
|
/* Track load is not complete - it might have stopped on a
|
|
full buffer without reaching the audio handle or we just
|
|
arrived at it early
|
|
|
|
If this type is atomic and we couldn't get the audio,
|
|
perhaps it would need to wrap to make the allocation and
|
|
handles are in the way - to maximize the liklihood it can
|
|
be allocated, clear all handles to reset the buffer and
|
|
its indexes to 0 - for packet audio, this should not be an
|
|
issue and a pointless full reload of all the track's
|
|
metadata may be avoided */
|
|
|
|
struct mp3entry *track_id3 = bufgetid3(info->id3_hid);
|
|
|
|
if (track_id3 &&
|
|
get_audio_base_data_type(track_id3->codectype)
|
|
== TYPE_PACKET_AUDIO)
|
|
{
|
|
/* Continue filling after this track */
|
|
audio_reset_and_rebuffer(TRACK_LIST_KEEP_CURRENT, 1);
|
|
audio_begin_track_change(TRACK_CHANGE_AUTO, trackstat);
|
|
return;
|
|
}
|
|
/* else rebuffer at this track; status applies to the track we
|
|
want */
|
|
}
|
|
else if (!playlist_peek(1, NULL, 0))
|
|
{
|
|
/* Play sequence is complete - directory change or other playlist
|
|
resequencing - the playlist must now be advanced in order to
|
|
continue since a peek ahead to the next track is not possible */
|
|
skip_pending = TRACK_SKIP_AUTO_NEW_PLAYLIST;
|
|
end_of_playlist = playlist_next(1) < 0;
|
|
}
|
|
|
|
if (!end_of_playlist)
|
|
{
|
|
trackstat = audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL,
|
|
skip_pending == TRACK_SKIP_AUTO ? 0 : -1);
|
|
|
|
if (trackstat == LOAD_TRACK_ERR_NO_MORE)
|
|
{
|
|
/* Failed to find anything after all - do playlist switchover
|
|
instead */
|
|
skip_pending = TRACK_SKIP_AUTO_NEW_PLAYLIST;
|
|
end_of_playlist = playlist_next(1) < 0;
|
|
}
|
|
}
|
|
|
|
if (end_of_playlist)
|
|
{
|
|
audio_monitor_end_of_playlist();
|
|
return;
|
|
}
|
|
}
|
|
|
|
audio_begin_track_change(TRACK_CHANGE_AUTO, trackstat);
|
|
}
|
|
|
|
/* Called when codec completes seek operation
|
|
(usually Q_AUDIO_CODEC_SEEK_COMPLETE) */
|
|
static void audio_on_codec_seek_complete(void)
|
|
{
|
|
logf("%s()", __func__);
|
|
audio_complete_codec_seek();
|
|
codec_go();
|
|
}
|
|
|
|
/* Called when PCM track change has completed
|
|
(Q_AUDIO_TRACK_CHANGED) */
|
|
static void audio_on_track_changed(void)
|
|
{
|
|
/* Finish whatever is pending so that the WPS is in sync */
|
|
audio_finalise_track_change();
|
|
|
|
if (codec_skip_pending)
|
|
{
|
|
/* Codec got ahead completing a short track - complete the
|
|
codec's skip and begin the next */
|
|
codec_skip_pending = false;
|
|
audio_on_codec_complete(codec_skip_status);
|
|
}
|
|
}
|
|
|
|
/* Begin playback from an idle state, transition to a new playlist or
|
|
invalidate the buffer and resume (if playing).
|
|
(usually Q_AUDIO_PLAY, Q_AUDIO_REMAKE_AUDIO_BUFFER) */
|
|
static void audio_start_playback(size_t offset, unsigned int flags)
|
|
{
|
|
enum play_status old_status = play_status;
|
|
|
|
if (flags & AUDIO_START_NEWBUF)
|
|
{
|
|
/* Mark the buffer dirty - if not playing, it will be reset next
|
|
time */
|
|
if (buffer_state == AUDIOBUF_STATE_INITIALIZED)
|
|
buffer_state = AUDIOBUF_STATE_VOICED_ONLY;
|
|
}
|
|
|
|
if (old_status != PLAY_STOPPED)
|
|
{
|
|
logf("%s(%lu): skipping", __func__, (unsigned long)offset);
|
|
|
|
halt_decoding_track(true);
|
|
|
|
automatic_skip = false;
|
|
ff_rw_mode = false;
|
|
|
|
if (flags & AUDIO_START_RESTART)
|
|
{
|
|
/* Clear out some stuff to resume the current track where it
|
|
left off */
|
|
pcmbuf_play_stop();
|
|
offset = id3_get(PLAYING_ID3)->offset;
|
|
track_list_clear(TRACK_LIST_CLEAR_ALL);
|
|
}
|
|
else
|
|
{
|
|
/* This is more-or-less treated as manual track transition */
|
|
/* Save resume information for current track */
|
|
audio_playlist_track_finish();
|
|
track_list_clear(TRACK_LIST_CLEAR_ALL);
|
|
|
|
/* Indicate manual track change */
|
|
pcmbuf_start_track_change(TRACK_CHANGE_MANUAL);
|
|
wipe_track_metadata(true);
|
|
}
|
|
|
|
/* Set after track finish event in case skip was in progress */
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
}
|
|
else
|
|
{
|
|
if (flags & AUDIO_START_RESTART)
|
|
return; /* Must already be playing */
|
|
|
|
/* Cold playback start from a stopped state */
|
|
logf("%s(%lu): starting", __func__, offset);
|
|
|
|
/* Set audio parameters */
|
|
#if INPUT_SRC_CAPS != 0
|
|
audio_set_input_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
|
|
audio_set_output_source(AUDIO_SRC_PLAYBACK);
|
|
#endif
|
|
#ifndef PLATFORM_HAS_VOLUME_CHANGE
|
|
sound_set_volume(global_settings.volume);
|
|
#endif
|
|
/* Be sure channel is audible */
|
|
pcmbuf_fade(false, true);
|
|
|
|
/* Update our state */
|
|
play_status = PLAY_PLAYING;
|
|
}
|
|
|
|
/* Codec's position should be available as soon as it knows it */
|
|
position_key = pcmbuf_get_position_key();
|
|
pcmbuf_sync_position_update();
|
|
|
|
/* Start fill from beginning of playlist */
|
|
playlist_peek_offset = -1;
|
|
buf_set_base_handle(-1);
|
|
|
|
/* Officially playing */
|
|
queue_reply(&audio_queue, 1);
|
|
|
|
/* Add these now - finish event for the first id3 will most likely be sent
|
|
immediately */
|
|
add_event(BUFFER_EVENT_REBUFFER, false, buffer_event_rebuffer_callback);
|
|
add_event(BUFFER_EVENT_FINISHED, false, buffer_event_finished_callback);
|
|
|
|
if (old_status == PLAY_STOPPED)
|
|
{
|
|
/* Send coldstart event */
|
|
send_event(PLAYBACK_EVENT_START_PLAYBACK, NULL);
|
|
}
|
|
|
|
/* Fill the buffer */
|
|
int trackstat = audio_fill_file_buffer();
|
|
|
|
if (trackstat >= LOAD_TRACK_OK)
|
|
{
|
|
/* This is the currently playing track - get metadata, stat */
|
|
playing_id3_sync(track_list_current(0), offset);
|
|
|
|
if (valid_mp3entry(id3_get(PLAYING_ID3)))
|
|
{
|
|
/* Only if actually changing tracks... */
|
|
if (!(flags & AUDIO_START_RESTART))
|
|
audio_playlist_track_change();
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Found nothing playable */
|
|
audio_handle_track_load_status(trackstat);
|
|
}
|
|
}
|
|
|
|
/* Stop playback and enter an idle state
|
|
(usually Q_AUDIO_STOP) */
|
|
static void audio_stop_playback(void)
|
|
{
|
|
logf("%s()", __func__);
|
|
|
|
if (play_status == PLAY_STOPPED)
|
|
return;
|
|
|
|
bool do_fade = global_settings.fade_on_stop && filling != STATE_ENDED;
|
|
|
|
pcmbuf_fade(do_fade, false);
|
|
|
|
/* Wait for fade-out */
|
|
audio_wait_fade_complete();
|
|
|
|
/* Stop the codec and unload it */
|
|
halt_decoding_track(true);
|
|
pcmbuf_play_stop();
|
|
codec_unload();
|
|
|
|
/* Save resume information - "filling" might have been set to
|
|
"STATE_ENDED" by caller in order to facilitate end of playlist */
|
|
audio_playlist_track_finish();
|
|
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
automatic_skip = false;
|
|
|
|
/* Close all tracks and mark them NULL */
|
|
remove_event(BUFFER_EVENT_REBUFFER, buffer_event_rebuffer_callback);
|
|
remove_event(BUFFER_EVENT_FINISHED, buffer_event_finished_callback);
|
|
remove_event(BUFFER_EVENT_BUFFER_LOW, buffer_event_buffer_low_callback);
|
|
|
|
track_list_clear(TRACK_LIST_CLEAR_ALL);
|
|
|
|
/* Update our state */
|
|
ff_rw_mode = false;
|
|
play_status = PLAY_STOPPED;
|
|
|
|
wipe_track_metadata(true);
|
|
|
|
/* Go idle */
|
|
filling = STATE_IDLE;
|
|
cancel_cpu_boost();
|
|
}
|
|
|
|
/* Pause the playback of the current track
|
|
(Q_AUDIO_PAUSE) */
|
|
static void audio_on_pause(bool pause)
|
|
{
|
|
logf("%s(%s)", __func__, pause ? "true" : "false");
|
|
|
|
if (play_status == PLAY_STOPPED || pause == (play_status == PLAY_PAUSED))
|
|
return;
|
|
|
|
play_status = pause ? PLAY_PAUSED : PLAY_PLAYING;
|
|
|
|
if (!pause && codec_skip_pending)
|
|
{
|
|
/* Actually do the skip that is due - resets the status flag */
|
|
audio_on_codec_complete(codec_skip_status);
|
|
}
|
|
|
|
bool do_fade = global_settings.fade_on_stop;
|
|
|
|
pcmbuf_fade(do_fade, !pause);
|
|
|
|
if (!ff_rw_mode && !(do_fade && pause))
|
|
{
|
|
/* Not in ff/rw mode - can actually change the audio state now */
|
|
pcmbuf_pause(pause);
|
|
}
|
|
}
|
|
|
|
/* Skip a certain number of tracks forwards or backwards
|
|
(Q_AUDIO_SKIP) */
|
|
static void audio_on_skip(void)
|
|
{
|
|
id3_mutex_lock();
|
|
|
|
/* Eat the delta to keep it synced, even if not playing */
|
|
int toskip = skip_offset;
|
|
skip_offset = 0;
|
|
|
|
logf("%s(): %d", __func__, toskip);
|
|
|
|
id3_mutex_unlock();
|
|
|
|
if (play_status == PLAY_STOPPED)
|
|
return;
|
|
|
|
/* Force codec to abort this track */
|
|
halt_decoding_track(true);
|
|
|
|
/* Kill the ff/rw halt */
|
|
ff_rw_mode = false;
|
|
|
|
/* Manual skip */
|
|
automatic_skip = false;
|
|
|
|
/* If there was an auto skip in progress, there will be residual
|
|
advancement of the playlist and/or track list so compensation will be
|
|
required in order to end up in the right spot */
|
|
int track_list_delta = toskip;
|
|
int playlist_delta = toskip;
|
|
|
|
if (skip_pending != TRACK_SKIP_NONE)
|
|
{
|
|
if (skip_pending != TRACK_SKIP_AUTO_END_PLAYLIST)
|
|
track_list_delta--;
|
|
|
|
if (skip_pending == TRACK_SKIP_AUTO_NEW_PLAYLIST)
|
|
playlist_delta--;
|
|
}
|
|
|
|
audio_playlist_track_finish();
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
|
|
/* Update the playlist current track now */
|
|
while (playlist_next(playlist_delta) < 0)
|
|
{
|
|
/* Manual skip out of range (because the playlist wasn't updated
|
|
yet by us and so the check in audio_skip returned 'ok') - bring
|
|
back into range */
|
|
int d = toskip < 0 ? 1 : -1;
|
|
|
|
while (!playlist_check(playlist_delta))
|
|
{
|
|
if (playlist_delta == d)
|
|
{
|
|
/* Had to move the opposite direction to correct, which is
|
|
wrong - this is the end */
|
|
filling = STATE_ENDED;
|
|
audio_stop_playback();
|
|
return;
|
|
}
|
|
|
|
playlist_delta += d;
|
|
track_list_delta += d;
|
|
}
|
|
}
|
|
|
|
/* Adjust things by how much the playlist was manually moved */
|
|
playlist_peek_offset -= playlist_delta;
|
|
|
|
struct track_info *info = track_list_advance_current(track_list_delta);
|
|
int trackstat = LOAD_TRACK_OK;
|
|
|
|
if (!info || info->audio_hid < 0)
|
|
{
|
|
/* We don't know the next track thus we know we don't have it */
|
|
trackstat = audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL, -1);
|
|
}
|
|
|
|
audio_begin_track_change(TRACK_CHANGE_MANUAL, trackstat);
|
|
}
|
|
|
|
/* Skip to the next/previous directory
|
|
(Q_AUDIO_DIR_SKIP) */
|
|
static void audio_on_dir_skip(int direction)
|
|
{
|
|
logf("%s(%d)", __func__, direction);
|
|
|
|
id3_mutex_lock();
|
|
skip_offset = 0;
|
|
id3_mutex_unlock();
|
|
|
|
if (play_status == PLAY_STOPPED)
|
|
return;
|
|
|
|
/* Force codec to abort this track */
|
|
halt_decoding_track(true);
|
|
|
|
/* Kill the ff/rw halt */
|
|
ff_rw_mode = false;
|
|
|
|
/* Manual skip */
|
|
automatic_skip = false;
|
|
|
|
audio_playlist_track_finish();
|
|
|
|
/* Unless automatic and gapless, skips do not pend */
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
|
|
/* Regardless of the return value we need to rebuffer. If it fails the old
|
|
playlist will resume, else the next dir will start playing. */
|
|
playlist_next_dir(direction);
|
|
|
|
wipe_track_metadata(false);
|
|
|
|
int trackstat = audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL, -1);
|
|
|
|
if (trackstat == LOAD_TRACK_ERR_NO_MORE)
|
|
{
|
|
/* The day the music died - finish-off whatever is playing and call it
|
|
quits */
|
|
audio_monitor_end_of_playlist();
|
|
return;
|
|
}
|
|
|
|
audio_begin_track_change(TRACK_CHANGE_MANUAL, trackstat);
|
|
}
|
|
|
|
/* Enter seek mode in order to start a seek
|
|
(Q_AUDIO_PRE_FF_REWIND) */
|
|
static void audio_on_pre_ff_rewind(void)
|
|
{
|
|
logf("%s()", __func__);
|
|
|
|
if (play_status == PLAY_STOPPED || ff_rw_mode)
|
|
return;
|
|
|
|
ff_rw_mode = true;
|
|
|
|
audio_wait_fade_complete();
|
|
|
|
if (play_status == PLAY_PAUSED)
|
|
return;
|
|
|
|
pcmbuf_pause(true);
|
|
}
|
|
|
|
/* Seek the playback of the current track to the specified time
|
|
(Q_AUDIO_FF_REWIND) */
|
|
static void audio_on_ff_rewind(long time)
|
|
{
|
|
logf("%s(%ld)", __func__, time);
|
|
|
|
if (play_status == PLAY_STOPPED)
|
|
return;
|
|
|
|
enum track_skip_type pending = skip_pending;
|
|
|
|
switch (pending)
|
|
{
|
|
case TRACK_SKIP_NONE: /* The usual case */
|
|
case TRACK_SKIP_AUTO: /* Have to back it out (fun!) */
|
|
case TRACK_SKIP_AUTO_END_PLAYLIST: /* Still have the last codec used */
|
|
{
|
|
struct mp3entry *id3 = id3_get(PLAYING_ID3);
|
|
struct mp3entry *ci_id3 = id3_get(CODEC_ID3);
|
|
|
|
automatic_skip = false;
|
|
|
|
/* Send event before clobbering the time */
|
|
/* FIXME: Nasty, but the tagtree expects this so that rewinding and
|
|
then skipping back to this track resumes properly. Something else
|
|
should be sent. We're not _really_ finishing the track are we? */
|
|
if (time == 0)
|
|
send_event(PLAYBACK_EVENT_TRACK_FINISH, id3);
|
|
|
|
id3->elapsed = time;
|
|
queue_reply(&audio_queue, 1);
|
|
|
|
bool haltres = halt_decoding_track(pending == TRACK_SKIP_AUTO);
|
|
|
|
/* Need this set in case ff/rw mode + error but _after_ the codec
|
|
halt that will reset it */
|
|
codec_seeking = true;
|
|
|
|
/* If in transition, key will have changed - sync to it */
|
|
position_key = pcmbuf_get_position_key();
|
|
|
|
if (pending == TRACK_SKIP_AUTO)
|
|
{
|
|
if (!track_list_advance_current(-1))
|
|
{
|
|
/* Not in list - must rebuffer at the current playlist index */
|
|
if (audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL, -1)
|
|
< LOAD_TRACK_OK)
|
|
{
|
|
/* Codec is stopped */
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Set after audio_fill_file_buffer to disable playing id3 clobber if
|
|
rebuffer is needed */
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
struct track_info *cur_info = track_list_current(0);
|
|
|
|
/* Track must complete the loading _now_ since a codec and audio
|
|
handle are needed in order to do the seek */
|
|
if (cur_info->audio_hid < 0 &&
|
|
audio_finish_load_track(cur_info) != LOAD_TRACK_READY)
|
|
{
|
|
/* Call above should push any load sequence - no need for
|
|
halt_decoding_track here if no skip was pending here because
|
|
there would not be a codec started if no audio handle was yet
|
|
opened */
|
|
break;
|
|
}
|
|
|
|
if (pending == TRACK_SKIP_AUTO)
|
|
{
|
|
if (!bufreadid3(cur_info->id3_hid, ci_id3) ||
|
|
!audio_init_codec(cur_info, ci_id3))
|
|
{
|
|
/* We should have still been able to get it - skip it and move
|
|
onto the next one - like it or not this track is broken */
|
|
break;
|
|
}
|
|
|
|
/* Set the codec API to the correct metadata and track info */
|
|
ci.audio_hid = cur_info->audio_hid;
|
|
ci.filesize = cur_info->filesize;
|
|
buf_set_base_handle(cur_info->audio_hid);
|
|
}
|
|
|
|
if (!haltres)
|
|
{
|
|
/* If codec must be (re)started, reset the offset */
|
|
ci_id3->offset = 0;
|
|
}
|
|
|
|
codec_seek(time);
|
|
return;
|
|
}
|
|
|
|
case TRACK_SKIP_AUTO_NEW_PLAYLIST:
|
|
{
|
|
/* We cannot do this because the playlist must be reversed by one
|
|
and it doesn't always return the same song when going backwards
|
|
across boundaries as forwards (either because of randomization
|
|
or inconsistency in deciding what the previous track should be),
|
|
therefore the whole operation would often end up as nonsense -
|
|
lock out seeking for a couple seconds */
|
|
|
|
/* Sure as heck cancel seek mode too! */
|
|
audio_ff_rewind_end();
|
|
return;
|
|
}
|
|
|
|
default:
|
|
/* Won't see this */
|
|
return;
|
|
}
|
|
|
|
if (play_status == PLAY_STOPPED)
|
|
{
|
|
/* Playback ended because of an error completing a track load */
|
|
return;
|
|
}
|
|
|
|
/* Always fake it as a codec start error which will handle mode
|
|
cancellations and skip to the next track */
|
|
audio_handle_track_load_status(LOAD_TRACK_ERR_START_CODEC);
|
|
}
|
|
|
|
/* Invalidates all but currently playing track
|
|
(Q_AUDIO_FLUSH) */
|
|
static void audio_on_audio_flush(void)
|
|
{
|
|
logf("%s", __func__);
|
|
|
|
if (track_list_empty())
|
|
return; /* Nothing to flush out */
|
|
|
|
switch (skip_pending)
|
|
{
|
|
case TRACK_SKIP_NONE:
|
|
case TRACK_SKIP_AUTO_END_PLAYLIST:
|
|
/* Remove all but the currently playing track from the list and
|
|
refill after that */
|
|
track_list_clear(TRACK_LIST_KEEP_CURRENT);
|
|
playlist_peek_offset = 0;
|
|
id3_write_locked(UNBUFFERED_ID3, NULL);
|
|
audio_update_and_announce_next_track(NULL);
|
|
|
|
/* Ignore return since it's about the next track, not this one */
|
|
audio_fill_file_buffer();
|
|
|
|
if (skip_pending == TRACK_SKIP_NONE)
|
|
break;
|
|
|
|
/* There's now a track after this one now - convert to auto skip -
|
|
no skip should pend right now because multiple flush messages can
|
|
be fired which would cause a restart in the below cases */
|
|
skip_pending = TRACK_SKIP_NONE;
|
|
audio_clear_track_notifications();
|
|
audio_queue_post(Q_AUDIO_CODEC_COMPLETE, CODEC_OK);
|
|
break;
|
|
|
|
case TRACK_SKIP_AUTO:
|
|
case TRACK_SKIP_AUTO_NEW_PLAYLIST:
|
|
/* Precisely removing what it already decoded for the next track is
|
|
not possible so a restart is required in order to continue the
|
|
currently playing track without the now invalid future track
|
|
playing */
|
|
audio_start_playback(0, AUDIO_START_RESTART);
|
|
break;
|
|
|
|
default: /* Nothing else is a state */
|
|
break;
|
|
}
|
|
}
|
|
|
|
#ifdef AUDIO_HAVE_RECORDING
|
|
/* Load the requested encoder type
|
|
(Q_AUDIO_LOAD_ENCODER) */
|
|
static void audio_on_load_encoder(int afmt)
|
|
{
|
|
bool res = true;
|
|
|
|
if (play_status != PLAY_STOPPED)
|
|
audio_stop_playback(); /* Can't load both types at once */
|
|
else
|
|
codec_unload(); /* Encoder still loaded, stop and unload it */
|
|
|
|
if (afmt != AFMT_UNKNOWN)
|
|
{
|
|
res = codec_load(-1, afmt | CODEC_TYPE_ENCODER);
|
|
if (res)
|
|
codec_go(); /* These are run immediately */
|
|
}
|
|
|
|
queue_reply(&audio_queue, res);
|
|
}
|
|
#endif /* AUDIO_HAVE_RECORDING */
|
|
|
|
static void audio_thread(void)
|
|
{
|
|
struct queue_event ev;
|
|
|
|
pcm_postinit();
|
|
|
|
while (1)
|
|
{
|
|
switch (filling)
|
|
{
|
|
/* Active states */
|
|
case STATE_FULL:
|
|
case STATE_END_OF_PLAYLIST:
|
|
if (buf_get_watermark() == 0)
|
|
{
|
|
/* End of buffering for now, let's calculate the watermark,
|
|
register for a low buffer event and unboost */
|
|
audio_update_filebuf_watermark(0);
|
|
add_event(BUFFER_EVENT_BUFFER_LOW, true,
|
|
buffer_event_buffer_low_callback);
|
|
}
|
|
/* Fall-through */
|
|
case STATE_FINISHED:
|
|
/* All data was buffered */
|
|
cancel_cpu_boost();
|
|
/* Fall-through */
|
|
case STATE_FILLING:
|
|
case STATE_ENDING:
|
|
if (audio_pcmbuf_track_change_scan())
|
|
{
|
|
/* Transfer notification to audio queue event */
|
|
ev.id = Q_AUDIO_TRACK_CHANGED;
|
|
ev.data = 1;
|
|
}
|
|
else
|
|
{
|
|
/* If doing auto skip, poll pcmbuf track notifications a bit
|
|
faster to promply detect the transition */
|
|
queue_wait_w_tmo(&audio_queue, &ev,
|
|
skip_pending == TRACK_SKIP_NONE ?
|
|
HZ/2 : HZ/10);
|
|
}
|
|
break;
|
|
|
|
/* Idle states */
|
|
default:
|
|
queue_wait(&audio_queue, &ev);
|
|
|
|
#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
|
|
switch (ev.id)
|
|
{
|
|
#ifdef AUDIO_HAVE_RECORDING
|
|
/* Must monitor the encoder message for recording so it can remove
|
|
it if we process the insertion before it does. It cannot simply
|
|
be removed from under recording however. */
|
|
case Q_AUDIO_LOAD_ENCODER:
|
|
break;
|
|
#endif
|
|
case SYS_USB_DISCONNECTED:
|
|
filling = STATE_IDLE;
|
|
break;
|
|
|
|
default:
|
|
if (filling == STATE_USB)
|
|
continue;
|
|
}
|
|
#endif /* CONFIG_PLATFORM */
|
|
}
|
|
|
|
switch (ev.id)
|
|
{
|
|
/** Codec and track change messages **/
|
|
case Q_AUDIO_CODEC_COMPLETE:
|
|
/* Codec is done processing track and has gone idle */
|
|
LOGFQUEUE("audio < Q_AUDIO_CODEC_COMPLETE: %ld", (long)ev.data);
|
|
audio_on_codec_complete(ev.data);
|
|
break;
|
|
|
|
case Q_AUDIO_CODEC_SEEK_COMPLETE:
|
|
/* Codec is done seeking */
|
|
LOGFQUEUE("audio < Q_AUDIO_SEEK_COMPLETE");
|
|
audio_on_codec_seek_complete();
|
|
break;
|
|
|
|
case Q_AUDIO_TRACK_CHANGED:
|
|
/* PCM track change done */
|
|
LOGFQUEUE("audio < Q_AUDIO_TRACK_CHANGED");
|
|
audio_on_track_changed();
|
|
break;
|
|
|
|
/** Control messages **/
|
|
case Q_AUDIO_PLAY:
|
|
LOGFQUEUE("audio < Q_AUDIO_PLAY");
|
|
audio_start_playback(ev.data, 0);
|
|
break;
|
|
|
|
case Q_AUDIO_STOP:
|
|
LOGFQUEUE("audio < Q_AUDIO_STOP");
|
|
audio_stop_playback();
|
|
if (ev.data != 0)
|
|
queue_clear(&audio_queue);
|
|
break;
|
|
|
|
case Q_AUDIO_PAUSE:
|
|
LOGFQUEUE("audio < Q_AUDIO_PAUSE");
|
|
audio_on_pause(ev.data);
|
|
break;
|
|
|
|
case Q_AUDIO_SKIP:
|
|
LOGFQUEUE("audio < Q_AUDIO_SKIP");
|
|
audio_on_skip();
|
|
break;
|
|
|
|
case Q_AUDIO_DIR_SKIP:
|
|
LOGFQUEUE("audio < Q_AUDIO_DIR_SKIP");
|
|
audio_on_dir_skip(ev.data);
|
|
break;
|
|
|
|
case Q_AUDIO_PRE_FF_REWIND:
|
|
LOGFQUEUE("audio < Q_AUDIO_PRE_FF_REWIND");
|
|
audio_on_pre_ff_rewind();
|
|
break;
|
|
|
|
case Q_AUDIO_FF_REWIND:
|
|
LOGFQUEUE("audio < Q_AUDIO_FF_REWIND");
|
|
audio_on_ff_rewind(ev.data);
|
|
break;
|
|
|
|
case Q_AUDIO_FLUSH:
|
|
LOGFQUEUE("audio < Q_AUDIO_FLUSH: %d", (int)ev.data);
|
|
audio_on_audio_flush();
|
|
break;
|
|
|
|
/** Buffering messages **/
|
|
case Q_AUDIO_BUFFERING:
|
|
/* some buffering event */
|
|
LOGFQUEUE("audio < Q_AUDIO_BUFFERING: %d", (int)ev.data);
|
|
audio_on_buffering(ev.data);
|
|
break;
|
|
|
|
case Q_AUDIO_FILL_BUFFER:
|
|
/* continue buffering next track */
|
|
LOGFQUEUE("audio < Q_AUDIO_FILL_BUFFER");
|
|
audio_on_fill_buffer();
|
|
break;
|
|
|
|
case Q_AUDIO_FINISH_LOAD_TRACK:
|
|
/* metadata is buffered */
|
|
LOGFQUEUE("audio < Q_AUDIO_FINISH_LOAD_TRACK");
|
|
audio_on_finish_load_track(ev.data);
|
|
break;
|
|
|
|
case Q_AUDIO_HANDLE_FINISHED:
|
|
/* some other type is buffered */
|
|
LOGFQUEUE("audio < Q_AUDIO_HANDLE_FINISHED");
|
|
audio_on_handle_finished(ev.data);
|
|
break;
|
|
|
|
/** Miscellaneous messages **/
|
|
case Q_AUDIO_REMAKE_AUDIO_BUFFER:
|
|
/* buffer needs to be reinitialized */
|
|
LOGFQUEUE("audio < Q_AUDIO_REMAKE_AUDIO_BUFFER");
|
|
audio_start_playback(0, AUDIO_START_RESTART | AUDIO_START_NEWBUF);
|
|
break;
|
|
|
|
#ifdef HAVE_DISK_STORAGE
|
|
case Q_AUDIO_UPDATE_WATERMARK:
|
|
/* buffering watermark needs updating */
|
|
LOGFQUEUE("audio < Q_AUDIO_UPDATE_WATERMARK: %d", (int)ev.data);
|
|
audio_update_filebuf_watermark(ev.data);
|
|
break;
|
|
#endif /* HAVE_DISK_STORAGE */
|
|
|
|
#ifdef AUDIO_HAVE_RECORDING
|
|
case Q_AUDIO_LOAD_ENCODER:
|
|
/* load an encoder for recording */
|
|
LOGFQUEUE("audio < Q_AUDIO_LOAD_ENCODER: %d", (int)ev.data);
|
|
audio_on_load_encoder(ev.data);
|
|
break;
|
|
#endif /* AUDIO_HAVE_RECORDING */
|
|
|
|
#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
|
|
case SYS_USB_CONNECTED:
|
|
LOGFQUEUE("audio < SYS_USB_CONNECTED");
|
|
audio_stop_playback();
|
|
#ifdef PLAYBACK_VOICE
|
|
voice_stop();
|
|
#endif
|
|
filling = STATE_USB;
|
|
usb_acknowledge(SYS_USB_CONNECTED_ACK);
|
|
break;
|
|
#endif /* (CONFIG_PLATFORM & PLATFORM_NATIVE) */
|
|
|
|
case SYS_TIMEOUT:
|
|
LOGFQUEUE_SYS_TIMEOUT("audio < SYS_TIMEOUT");
|
|
break;
|
|
|
|
default:
|
|
/* LOGFQUEUE("audio < default : %08lX", ev.id); */
|
|
break;
|
|
} /* end switch */
|
|
} /* end while */
|
|
}
|
|
|
|
|
|
/* --- Buffering callbacks --- */
|
|
|
|
/* Called when fullness is below the watermark level */
|
|
static void buffer_event_buffer_low_callback(void *data)
|
|
{
|
|
logf("low buffer callback");
|
|
LOGFQUEUE("buffering > audio Q_AUDIO_BUFFERING: buffer low");
|
|
audio_queue_post(Q_AUDIO_BUFFERING, BUFFER_EVENT_BUFFER_LOW);
|
|
(void)data;
|
|
}
|
|
|
|
/* Called when handles must be discarded in order to buffer new data */
|
|
static void buffer_event_rebuffer_callback(void *data)
|
|
{
|
|
logf("rebuffer callback");
|
|
LOGFQUEUE("buffering > audio Q_AUDIO_BUFFERING: rebuffer");
|
|
audio_queue_post(Q_AUDIO_BUFFERING, BUFFER_EVENT_REBUFFER);
|
|
(void)data;
|
|
}
|
|
|
|
/* A handle has completed buffering and all required data is available */
|
|
static void buffer_event_finished_callback(void *data)
|
|
{
|
|
int hid = *(const int *)data;
|
|
const enum data_type htype = buf_handle_data_type(hid);
|
|
|
|
logf("handle %d finished buffering (type:%u)", hid, (unsigned)htype);
|
|
|
|
/* Limit queue traffic */
|
|
switch (htype)
|
|
{
|
|
case TYPE_ID3:
|
|
/* The metadata handle for the last loaded track has been buffered.
|
|
We can ask the audio thread to load the rest of the track's data. */
|
|
LOGFQUEUE("buffering > audio Q_AUDIO_FINISH_LOAD_TRACK: %d", hid);
|
|
audio_queue_post(Q_AUDIO_FINISH_LOAD_TRACK, hid);
|
|
break;
|
|
|
|
case TYPE_PACKET_AUDIO:
|
|
/* Strip any useless trailing tags that are left. */
|
|
strip_tags(hid);
|
|
/* Fall-through */
|
|
case TYPE_ATOMIC_AUDIO:
|
|
LOGFQUEUE("buffering > audio Q_AUDIO_HANDLE_FINISHED: %d", hid);
|
|
audio_queue_post(Q_AUDIO_HANDLE_FINISHED, hid);
|
|
break;
|
|
|
|
default:
|
|
/* Don't care to know about these */
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
/** -- Codec callbacks -- **/
|
|
|
|
/* Update elapsed time for next PCM insert */
|
|
void audio_codec_update_elapsed(unsigned long elapsed)
|
|
{
|
|
#ifdef AB_REPEAT_ENABLE
|
|
ab_position_report(elapsed);
|
|
#endif
|
|
/* Save in codec's id3 where it is used at next pcm insert */
|
|
id3_get(CODEC_ID3)->elapsed = elapsed;
|
|
}
|
|
|
|
/* Update offset for next PCM insert */
|
|
void audio_codec_update_offset(size_t offset)
|
|
{
|
|
/* Save in codec's id3 where it is used at next pcm insert */
|
|
id3_get(CODEC_ID3)->offset = offset;
|
|
}
|
|
|
|
/* Codec has finished running */
|
|
void audio_codec_complete(int status)
|
|
{
|
|
#ifdef AB_REPEAT_ENABLE
|
|
if (status >= CODEC_OK)
|
|
{
|
|
/* Normal automatic skip */
|
|
ab_end_of_track_report();
|
|
}
|
|
#endif
|
|
|
|
LOGFQUEUE("codec > audio Q_AUDIO_CODEC_COMPLETE: %d", status);
|
|
audio_queue_post(Q_AUDIO_CODEC_COMPLETE, status);
|
|
}
|
|
|
|
/* Codec has finished seeking */
|
|
void audio_codec_seek_complete(void)
|
|
{
|
|
LOGFQUEUE("codec > audio Q_AUDIO_CODEC_SEEK_COMPLETE");
|
|
audio_queue_post(Q_AUDIO_CODEC_SEEK_COMPLETE, 0);
|
|
}
|
|
|
|
|
|
/** --- Pcmbuf callbacks --- **/
|
|
|
|
/* Update the elapsed and offset from the information cached during the
|
|
PCM buffer insert */
|
|
void audio_pcmbuf_position_callback(unsigned long elapsed, off_t offset,
|
|
unsigned int key)
|
|
{
|
|
if (key == position_key)
|
|
{
|
|
struct mp3entry *id3 = id3_get(PLAYING_ID3);
|
|
id3->elapsed = elapsed;
|
|
id3->offset = offset;
|
|
}
|
|
}
|
|
|
|
/* Synchronize position info to the codec's */
|
|
void audio_pcmbuf_sync_position(void)
|
|
{
|
|
audio_pcmbuf_position_callback(ci.id3->elapsed, ci.id3->offset,
|
|
pcmbuf_get_position_key());
|
|
}
|
|
|
|
/* Post message from pcmbuf that the end of the previous track has just
|
|
* been played */
|
|
void audio_pcmbuf_track_change(bool pcmbuf)
|
|
{
|
|
if (pcmbuf)
|
|
{
|
|
/* Notify of the change in special-purpose semaphore object */
|
|
LOGFQUEUE("pcmbuf > pcmbuf Q_AUDIO_TRACK_CHANGED");
|
|
audio_pcmbuf_track_change_post();
|
|
}
|
|
else
|
|
{
|
|
/* Safe to post directly to the queue */
|
|
LOGFQUEUE("pcmbuf > audio Q_AUDIO_TRACK_CHANGED");
|
|
audio_queue_post(Q_AUDIO_TRACK_CHANGED, 0);
|
|
}
|
|
}
|
|
|
|
/* May pcmbuf start PCM playback when the buffer is full enough? */
|
|
bool audio_pcmbuf_may_play(void)
|
|
{
|
|
return play_status == PLAY_PLAYING && !ff_rw_mode;
|
|
}
|
|
|
|
|
|
/** -- External interfaces -- **/
|
|
|
|
/* Return the playback and recording status */
|
|
int audio_status(void)
|
|
{
|
|
unsigned int ret = play_status;
|
|
|
|
#ifdef AUDIO_HAVE_RECORDING
|
|
/* Do this here for constitency with mpeg.c version */
|
|
ret |= pcm_rec_status();
|
|
#endif
|
|
|
|
return (int)ret;
|
|
}
|
|
|
|
/* Clear all accumulated audio errors for playback and recording */
|
|
void audio_error_clear(void)
|
|
{
|
|
#ifdef AUDIO_HAVE_RECORDING
|
|
pcm_rec_error_clear();
|
|
#endif
|
|
}
|
|
|
|
/* Get a copy of the id3 data for the for current track + offset + skip delta */
|
|
bool audio_peek_track(struct mp3entry *id3, int offset)
|
|
{
|
|
bool retval = false;
|
|
|
|
id3_mutex_lock();
|
|
|
|
if (play_status != PLAY_STOPPED)
|
|
{
|
|
id3->path[0] = '\0'; /* Null path means it should be filled now */
|
|
retval = audio_get_track_metadata(offset + skip_offset, id3) &&
|
|
id3->path[0] != '\0';
|
|
}
|
|
|
|
id3_mutex_unlock();
|
|
|
|
return retval;
|
|
}
|
|
|
|
/* Return the mp3entry for the currently playing track */
|
|
struct mp3entry * audio_current_track(void)
|
|
{
|
|
struct mp3entry *id3;
|
|
|
|
id3_mutex_lock();
|
|
|
|
#ifdef AUDIO_FAST_SKIP_PREVIEW
|
|
if (skip_offset != 0)
|
|
{
|
|
/* This is a peekahead */
|
|
id3 = id3_get(PLAYING_PEEK_ID3);
|
|
audio_peek_track(id3, 0);
|
|
}
|
|
else
|
|
#endif
|
|
{
|
|
/* Normal case */
|
|
id3 = id3_get(PLAYING_ID3);
|
|
audio_get_track_metadata(0, id3);
|
|
}
|
|
|
|
id3_mutex_unlock();
|
|
|
|
return id3;
|
|
}
|
|
|
|
/* Obtains the mp3entry for the next track from the current */
|
|
struct mp3entry * audio_next_track(void)
|
|
{
|
|
struct mp3entry *id3 = id3_get(NEXTTRACK_ID3);
|
|
|
|
id3_mutex_lock();
|
|
|
|
#ifdef AUDIO_FAST_SKIP_PREVIEW
|
|
if (skip_offset != 0)
|
|
{
|
|
/* This is a peekahead */
|
|
if (!audio_peek_track(id3, 1))
|
|
id3 = NULL;
|
|
}
|
|
else
|
|
#endif
|
|
{
|
|
/* Normal case */
|
|
if (!audio_get_track_metadata(1, id3))
|
|
id3 = NULL;
|
|
}
|
|
|
|
id3_mutex_unlock();
|
|
|
|
return id3;
|
|
}
|
|
|
|
/* Start playback at the specified offset */
|
|
void audio_play(long offset)
|
|
{
|
|
logf("audio_play");
|
|
|
|
#ifdef PLAYBACK_VOICE
|
|
/* Truncate any existing voice output so we don't have spelling
|
|
* etc. over the first part of the played track */
|
|
talk_force_shutup();
|
|
#endif
|
|
|
|
LOGFQUEUE("audio >| audio Q_AUDIO_PLAY: %ld", offset);
|
|
audio_queue_send(Q_AUDIO_PLAY, offset);
|
|
}
|
|
|
|
/* Stop playback if playing */
|
|
void audio_stop(void)
|
|
{
|
|
LOGFQUEUE("audio >| audio Q_AUDIO_STOP");
|
|
audio_queue_send(Q_AUDIO_STOP, 0);
|
|
}
|
|
|
|
/* Pause playback if playing */
|
|
void audio_pause(void)
|
|
{
|
|
LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE");
|
|
audio_queue_send(Q_AUDIO_PAUSE, true);
|
|
}
|
|
|
|
/* This sends a stop message and the audio thread will dump all its
|
|
subsequent messages */
|
|
void audio_hard_stop(void)
|
|
{
|
|
/* Stop playback */
|
|
LOGFQUEUE("audio >| audio Q_AUDIO_STOP: 1");
|
|
audio_queue_send(Q_AUDIO_STOP, 1);
|
|
#ifdef PLAYBACK_VOICE
|
|
voice_stop();
|
|
#endif
|
|
if (audiobuf_handle > 0)
|
|
audiobuf_handle = core_free(audiobuf_handle);
|
|
}
|
|
|
|
/* Resume playback if paused */
|
|
void audio_resume(void)
|
|
{
|
|
LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE resume");
|
|
audio_queue_send(Q_AUDIO_PAUSE, false);
|
|
}
|
|
|
|
/* Skip the specified number of tracks forward or backward from the current */
|
|
void audio_skip(int offset)
|
|
{
|
|
id3_mutex_lock();
|
|
|
|
/* If offset has to be backed-out to stay in range, no skip is done */
|
|
int accum = skip_offset + offset;
|
|
|
|
while (offset != 0 && !playlist_check(accum))
|
|
{
|
|
offset += offset < 0 ? 1 : -1;
|
|
accum = skip_offset + offset;
|
|
}
|
|
|
|
if (offset != 0)
|
|
{
|
|
/* Accumulate net manual skip count since the audio thread last
|
|
processed one */
|
|
skip_offset = accum;
|
|
|
|
system_sound_play(SOUND_TRACK_SKIP);
|
|
|
|
LOGFQUEUE("audio > audio Q_AUDIO_SKIP %d", offset);
|
|
|
|
#ifdef AUDIO_FAST_SKIP_PREVIEW
|
|
/* Do this before posting so that the audio thread can correct us
|
|
when things settle down - additionally, if audio gets a message
|
|
and the delta is zero, the Q_AUDIO_SKIP handler (audio_on_skip)
|
|
handler a skip event with the correct info but doesn't skip */
|
|
send_event(PLAYBACK_EVENT_TRACK_SKIP, NULL);
|
|
#endif /* AUDIO_FAST_SKIP_PREVIEW */
|
|
|
|
/* Playback only needs the final state even if more than one is
|
|
processed because it wasn't removed in time */
|
|
queue_remove_from_head(&audio_queue, Q_AUDIO_SKIP);
|
|
audio_queue_post(Q_AUDIO_SKIP, 0);
|
|
}
|
|
else
|
|
{
|
|
/* No more tracks */
|
|
system_sound_play(SOUND_TRACK_NO_MORE);
|
|
}
|
|
|
|
id3_mutex_unlock();
|
|
}
|
|
|
|
/* Skip one track forward from the current */
|
|
void audio_next(void)
|
|
{
|
|
audio_skip(1);
|
|
}
|
|
|
|
/* Skip one track backward from the current */
|
|
void audio_prev(void)
|
|
{
|
|
audio_skip(-1);
|
|
}
|
|
|
|
/* Move one directory forward */
|
|
void audio_next_dir(void)
|
|
{
|
|
LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP 1");
|
|
audio_queue_post(Q_AUDIO_DIR_SKIP, 1);
|
|
}
|
|
|
|
/* Move one directory backward */
|
|
void audio_prev_dir(void)
|
|
{
|
|
LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP -1");
|
|
audio_queue_post(Q_AUDIO_DIR_SKIP, -1);
|
|
}
|
|
|
|
/* Pause playback in order to start a seek that flushes the old audio */
|
|
void audio_pre_ff_rewind(void)
|
|
{
|
|
LOGFQUEUE("audio > audio Q_AUDIO_PRE_FF_REWIND");
|
|
audio_queue_post(Q_AUDIO_PRE_FF_REWIND, 0);
|
|
}
|
|
|
|
/* Seek to the new time in the current track */
|
|
void audio_ff_rewind(long time)
|
|
{
|
|
LOGFQUEUE("audio > audio Q_AUDIO_FF_REWIND");
|
|
audio_queue_post(Q_AUDIO_FF_REWIND, time);
|
|
}
|
|
|
|
/* Clear all but the currently playing track then rebuffer */
|
|
void audio_flush_and_reload_tracks(void)
|
|
{
|
|
LOGFQUEUE("audio > audio Q_AUDIO_FLUSH");
|
|
audio_queue_post(Q_AUDIO_FLUSH, 0);
|
|
}
|
|
|
|
/* Return the pointer to the main audio buffer, optionally preserving
|
|
voicing */
|
|
unsigned char * audio_get_buffer(bool talk_buf, size_t *buffer_size)
|
|
{
|
|
unsigned char *buf;
|
|
|
|
if (audio_is_initialized)
|
|
{
|
|
audio_hard_stop();
|
|
}
|
|
/* else buffer_state will be AUDIOBUF_STATE_TRASHED at this point */
|
|
|
|
if (buffer_size == NULL)
|
|
{
|
|
/* Special case for talk_init to use since it already knows it's
|
|
trashed */
|
|
buffer_state = AUDIOBUF_STATE_TRASHED;
|
|
return NULL;
|
|
}
|
|
|
|
/* make sure buffer is freed and re-allocated to simplify code below
|
|
* (audio_hard_stop() likely has done that already) */
|
|
if (audiobuf_handle > 0)
|
|
audiobuf_handle = core_free(audiobuf_handle);
|
|
|
|
audiobuf_handle = core_alloc_maximum("audiobuf", &filebuflen, &ops);
|
|
buf = core_get_data(audiobuf_handle);
|
|
|
|
if (talk_buf || buffer_state == AUDIOBUF_STATE_TRASHED
|
|
|| !talk_voice_required())
|
|
{
|
|
logf("get buffer: talk, audio");
|
|
/* Ok to use everything from audiobuf - voice is loaded,
|
|
the talk buffer is not needed because voice isn't being used, or
|
|
could be AUDIOBUF_STATE_TRASHED already. If state is
|
|
AUDIOBUF_STATE_VOICED_ONLY, no problem as long as memory isn't
|
|
written without the caller knowing what's going on. Changing certain
|
|
settings may move it to a worse condition but the memory in use by
|
|
something else will remain undisturbed.
|
|
*/
|
|
if (buffer_state != AUDIOBUF_STATE_TRASHED)
|
|
{
|
|
talk_buffer_steal();
|
|
buffer_state = AUDIOBUF_STATE_TRASHED;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
logf("get buffer: audio");
|
|
/* Safe to just return this if already AUDIOBUF_STATE_VOICED_ONLY or
|
|
still AUDIOBUF_STATE_INITIALIZED */
|
|
/* Skip talk buffer and move pcm buffer to end to maximize available
|
|
contiguous memory - no audio running means voice will not need the
|
|
swap space */
|
|
size_t talkbuf_size;
|
|
buf += talkbuf_size = talkbuf_init(buf);
|
|
filebuflen -= talkbuf_size;
|
|
filebuflen -= voicebuf_init(buf + filebuflen);
|
|
|
|
buffer_state = AUDIOBUF_STATE_VOICED_ONLY;
|
|
}
|
|
|
|
*buffer_size = filebuflen;
|
|
return buf;
|
|
}
|
|
|
|
#ifdef HAVE_RECORDING
|
|
/* Stop audio, voice and obtain all available buffer space */
|
|
unsigned char * audio_get_recording_buffer(size_t *buffer_size)
|
|
{
|
|
audio_hard_stop();
|
|
return audio_get_buffer(true, buffer_size);
|
|
}
|
|
#endif /* HAVE_RECORDING */
|
|
|
|
/* Restore audio buffer to a particular state (one more valid than the current
|
|
state) */
|
|
bool audio_restore_playback(int type)
|
|
{
|
|
switch (type)
|
|
{
|
|
case AUDIO_WANT_PLAYBACK:
|
|
if (buffer_state != AUDIOBUF_STATE_INITIALIZED)
|
|
audio_reset_buffer();
|
|
return true;
|
|
case AUDIO_WANT_VOICE:
|
|
if (buffer_state == AUDIOBUF_STATE_TRASHED)
|
|
audio_reset_buffer();
|
|
return true;
|
|
default:
|
|
return false;
|
|
}
|
|
}
|
|
|
|
/* Has the playback buffer been completely claimed? */
|
|
bool audio_buffer_state_trashed(void)
|
|
{
|
|
return buffer_state == AUDIOBUF_STATE_TRASHED;
|
|
}
|
|
|
|
|
|
/** --- Miscellaneous public interfaces --- **/
|
|
|
|
#ifdef HAVE_ALBUMART
|
|
/* Return which album art handle is current for the user in the given slot */
|
|
int playback_current_aa_hid(int slot)
|
|
{
|
|
if ((unsigned)slot < MAX_MULTIPLE_AA)
|
|
{
|
|
struct track_info *info = track_list_user_current(skip_offset);
|
|
|
|
if (!info && abs(skip_offset) <= 1)
|
|
{
|
|
/* Give the actual position a go */
|
|
info = track_list_user_current(0);
|
|
}
|
|
|
|
if (info)
|
|
return info->aa_hid[slot];
|
|
}
|
|
|
|
return ERR_HANDLE_NOT_FOUND;
|
|
}
|
|
|
|
/* Find an album art slot that doesn't match the dimensions of another that
|
|
is already claimed - increment the use count if it is */
|
|
int playback_claim_aa_slot(struct dim *dim)
|
|
{
|
|
int i;
|
|
|
|
/* First try to find a slot already having the size to reuse it since we
|
|
don't want albumart of the same size buffered multiple times */
|
|
FOREACH_ALBUMART(i)
|
|
{
|
|
struct albumart_slot *slot = &albumart_slots[i];
|
|
|
|
if (slot->dim.width == dim->width &&
|
|
slot->dim.height == dim->height)
|
|
{
|
|
slot->used++;
|
|
return i;
|
|
}
|
|
}
|
|
|
|
/* Size is new, find a free slot */
|
|
FOREACH_ALBUMART(i)
|
|
{
|
|
if (!albumart_slots[i].used)
|
|
{
|
|
albumart_slots[i].used++;
|
|
albumart_slots[i].dim = *dim;
|
|
return i;
|
|
}
|
|
}
|
|
|
|
/* Sorry, no free slot */
|
|
return -1;
|
|
}
|
|
|
|
/* Invalidate the albumart_slot - decrement the use count if > 0 */
|
|
void playback_release_aa_slot(int slot)
|
|
{
|
|
if ((unsigned)slot < MAX_MULTIPLE_AA)
|
|
{
|
|
struct albumart_slot *aa_slot = &albumart_slots[slot];
|
|
|
|
if (aa_slot->used > 0)
|
|
aa_slot->used--;
|
|
}
|
|
}
|
|
#endif /* HAVE_ALBUMART */
|
|
|
|
|
|
#ifdef HAVE_RECORDING
|
|
/* Load an encoder and run it */
|
|
bool audio_load_encoder(int afmt)
|
|
{
|
|
#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
|
|
LOGFQUEUE("audio >| Q_AUDIO_LOAD_ENCODER: %d", afmt);
|
|
return audio_queue_send(Q_AUDIO_LOAD_ENCODER, afmt) != 0;
|
|
#else
|
|
(void)afmt;
|
|
return true;
|
|
#endif
|
|
}
|
|
|
|
/* Stop an encoder and unload it */
|
|
void audio_remove_encoder(void)
|
|
{
|
|
#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
|
|
LOGFQUEUE("audio >| Q_AUDIO_LOAD_ENCODER: NULL");
|
|
audio_queue_send(Q_AUDIO_LOAD_ENCODER, AFMT_UNKNOWN);
|
|
#endif
|
|
}
|
|
#endif /* HAVE_RECORDING */
|
|
|
|
/* Is an automatic skip in progress? If called outside transition callbacks,
|
|
indicates the last skip type at the time it was processed and isn't very
|
|
meaningful. */
|
|
bool audio_automatic_skip(void)
|
|
{
|
|
return automatic_skip;
|
|
}
|
|
|
|
/* Would normally calculate byte offset from an elapsed time but is not
|
|
used on SWCODEC */
|
|
int audio_get_file_pos(void)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
/* Return the elapsed time of the track previous to the current */
|
|
unsigned long audio_prev_elapsed(void)
|
|
{
|
|
return prev_track_elapsed;
|
|
}
|
|
|
|
/* Return total file buffer length after accounting for the talk buf */
|
|
size_t audio_get_filebuflen(void)
|
|
{
|
|
return buf_length();
|
|
}
|
|
|
|
/* How many tracks exist on the buffer - full or partial */
|
|
int audio_track_count(void)
|
|
__attribute__((alias("track_list_count")));
|
|
|
|
/* Return total ringbuffer space occupied - ridx to widx */
|
|
long audio_filebufused(void)
|
|
{
|
|
return buf_used();
|
|
}
|
|
|
|
|
|
/** -- Settings -- **/
|
|
|
|
/* Enable or disable cuesheet support and allocate/don't allocate the
|
|
extra associated resources */
|
|
void audio_set_cuesheet(int enable)
|
|
{
|
|
if (play_status == PLAY_STOPPED || !enable != !get_current_cuesheet())
|
|
{
|
|
LOGFQUEUE("audio >| audio Q_AUDIO_REMAKE_AUDIO_BUFFER");
|
|
audio_queue_send(Q_AUDIO_REMAKE_AUDIO_BUFFER, 0);
|
|
}
|
|
}
|
|
|
|
#ifdef HAVE_DISK_STORAGE
|
|
/* Set the audio antiskip buffer margin by index */
|
|
void audio_set_buffer_margin(int setting)
|
|
{
|
|
static const unsigned short lookup[] =
|
|
{ 5, 15, 30, 60, 120, 180, 300, 600 };
|
|
|
|
if ((unsigned)setting >= ARRAYLEN(lookup))
|
|
setting = 0;
|
|
|
|
logf("buffer margin: %u", (unsigned)lookup[setting]);
|
|
|
|
LOGFQUEUE("audio > audio Q_AUDIO_UPDATE_WATERMARK: %u",
|
|
(unsigned)lookup[setting]);
|
|
audio_queue_post(Q_AUDIO_UPDATE_WATERMARK, lookup[setting]);
|
|
}
|
|
#endif /* HAVE_DISK_STORAGE */
|
|
|
|
#ifdef HAVE_CROSSFADE
|
|
/* Take necessary steps to enable or disable the crossfade setting */
|
|
void audio_set_crossfade(int enable)
|
|
{
|
|
/* Tell it the next setting to use */
|
|
pcmbuf_request_crossfade_enable(enable);
|
|
|
|
/* Return if size hasn't changed or this is too early to determine
|
|
which in the second case there's no way we could be playing
|
|
anything at all */
|
|
if (!pcmbuf_is_same_size())
|
|
{
|
|
LOGFQUEUE("audio >| audio Q_AUDIO_REMAKE_AUDIO_BUFFER");
|
|
audio_queue_send(Q_AUDIO_REMAKE_AUDIO_BUFFER, 0);
|
|
}
|
|
}
|
|
#endif /* HAVE_CROSSFADE */
|
|
|
|
|
|
/** -- Startup -- **/
|
|
|
|
/* Initialize the audio system - called from init() in main.c */
|
|
void audio_init(void)
|
|
{
|
|
/* Can never do this twice */
|
|
if (audio_is_initialized)
|
|
{
|
|
logf("audio: already initialized");
|
|
return;
|
|
}
|
|
|
|
logf("audio: initializing");
|
|
|
|
/* Initialize queues before giving control elsewhere in case it likes
|
|
to send messages. Thread creation will be delayed however so nothing
|
|
starts running until ready if something yields such as talk_init. */
|
|
queue_init(&audio_queue, true);
|
|
|
|
mutex_init(&id3_mutex);
|
|
|
|
pcm_init();
|
|
|
|
codec_thread_init();
|
|
|
|
/* This thread does buffer, so match its priority */
|
|
audio_thread_id = create_thread(audio_thread, audio_stack,
|
|
sizeof(audio_stack), 0, audio_thread_name
|
|
IF_PRIO(, MIN(PRIORITY_BUFFERING, PRIORITY_USER_INTERFACE))
|
|
IF_COP(, CPU));
|
|
|
|
queue_enable_queue_send(&audio_queue, &audio_queue_sender_list,
|
|
audio_thread_id);
|
|
|
|
#ifdef PLAYBACK_VOICE
|
|
voice_thread_init();
|
|
#endif
|
|
|
|
/* audio_reset_buffer must know the size of voice buffer so init
|
|
talk first */
|
|
talk_init();
|
|
|
|
#ifdef HAVE_CROSSFADE
|
|
/* Set crossfade setting for next buffer init which should be about... */
|
|
pcmbuf_request_crossfade_enable(global_settings.crossfade);
|
|
#endif
|
|
|
|
/* Initialize the buffering system */
|
|
track_list_init();
|
|
buffering_init();
|
|
/* ...now! Set up the buffers */
|
|
audio_reset_buffer();
|
|
|
|
/* Probably safe to say */
|
|
audio_is_initialized = true;
|
|
|
|
sound_settings_apply();
|
|
#ifdef HAVE_DISK_STORAGE
|
|
audio_set_buffer_margin(global_settings.buffer_margin);
|
|
#endif
|
|
}
|