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rockbox/apps/playback.c
Nils Wallménius 441fa76a6d Fix flushing of voice during playback, patch in FS#6239 by Stephane Doyon and Daniel Dalton 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@14517 a1c6a512-1295-4272-9138-f99709370657
2007-08-29 15:11:19 +00:00

3870 lines
110 KiB
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
/* TODO: Can use the track changed callback to detect end of track and seek
* in the previous track until this happens */
/* Design: we have prev_ti already, have a conditional for what type of seek
* to do on a seek request, if it is a previous track seek, skip previous,
* and in the request_next_track callback set the offset up the same way that
* starting from an offset works. */
/* TODO: Pause should be handled in here, rather than PCMBUF so that voice can
* play whilst audio is paused */
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <ctype.h>
#include "system.h"
#include "thread.h"
#include "file.h"
#include "panic.h"
#include "memory.h"
#include "lcd.h"
#include "font.h"
#include "button.h"
#include "kernel.h"
#include "tree.h"
#include "debug.h"
#include "sprintf.h"
#include "settings.h"
#include "codecs.h"
#include "audio.h"
#include "logf.h"
#include "mp3_playback.h"
#include "usb.h"
#include "status.h"
#include "ata.h"
#include "screens.h"
#include "playlist.h"
#include "playback.h"
#include "pcmbuf.h"
#include "buffer.h"
#include "dsp.h"
#include "abrepeat.h"
#include "cuesheet.h"
#ifdef HAVE_TAGCACHE
#include "tagcache.h"
#endif
#ifdef HAVE_LCD_BITMAP
#include "icons.h"
#include "peakmeter.h"
#include "action.h"
#endif
#include "lang.h"
#include "bookmark.h"
#include "misc.h"
#include "sound.h"
#include "metadata.h"
#include "splash.h"
#include "talk.h"
#include "ata_idle_notify.h"
#ifdef HAVE_RECORDING
#include "recording.h"
#include "talk.h"
#endif
#ifdef HAVE_WM8758
#include "menus/eq_menu.h"
#endif
#define PLAYBACK_VOICE
/* default point to start buffer refill */
#define AUDIO_DEFAULT_WATERMARK (1024*512)
/* amount of data to read in one read() call */
#define AUDIO_DEFAULT_FILECHUNK (1024*32)
/* point at which the file buffer will fight for CPU time */
#define AUDIO_FILEBUF_CRITICAL (1024*128)
/* amount of guess-space to allow for codecs that must hunt and peck
* for their correct seeek target, 32k seems a good size */
#define AUDIO_REBUFFER_GUESS_SIZE (1024*32)
/* 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
/* Define one constant that includes recording related functionality */
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
#define AUDIO_HAVE_RECORDING
#endif
enum {
Q_AUDIO_PLAY = 1,
Q_AUDIO_STOP,
Q_AUDIO_PAUSE,
Q_AUDIO_SKIP,
Q_AUDIO_PRE_FF_REWIND,
Q_AUDIO_FF_REWIND,
Q_AUDIO_REBUFFER_SEEK,
Q_AUDIO_CHECK_NEW_TRACK,
Q_AUDIO_FLUSH,
Q_AUDIO_TRACK_CHANGED,
Q_AUDIO_DIR_SKIP,
Q_AUDIO_POSTINIT,
Q_AUDIO_FILL_BUFFER,
#if MEM > 8
Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA,
#endif
Q_CODEC_REQUEST_COMPLETE,
Q_CODEC_REQUEST_FAILED,
Q_VOICE_PLAY,
Q_VOICE_STOP,
Q_CODEC_LOAD,
Q_CODEC_LOAD_DISK,
#ifdef AUDIO_HAVE_RECORDING
Q_ENCODER_LOAD_DISK,
Q_ENCODER_RECORD,
#endif
};
/* As defined in plugins/lib/xxx2wav.h */
#if MEM > 1
#define MALLOC_BUFSIZE (512*1024)
#define GUARD_BUFSIZE (32*1024)
#else
#define MALLOC_BUFSIZE (100*1024)
#define GUARD_BUFSIZE (8*1024)
#endif
/* As defined in plugin.lds */
#if defined(CPU_PP)
#define CODEC_IRAM_ORIGIN ((unsigned char *)0x4000c000)
#define CODEC_IRAM_SIZE ((size_t)0xc000)
#elif defined(IAUDIO_X5) || defined(IAUDIO_M5)
#define CODEC_IRAM_ORIGIN ((unsigned char *)0x10010000)
#define CODEC_IRAM_SIZE ((size_t)0x10000)
#else
#define CODEC_IRAM_ORIGIN ((unsigned char *)0x1000c000)
#define CODEC_IRAM_SIZE ((size_t)0xc000)
#endif
#ifndef IBSS_ATTR_VOICE_STACK
#define IBSS_ATTR_VOICE_STACK IBSS_ATTR
#endif
bool audio_is_initialized = false;
/* Variables are commented with the threads that use them: *
* A=audio, C=codec, V=voice. A suffix of - indicates that *
* the variable is read but not updated on that thread. */
/* TBD: Split out "audio" and "playback" (ie. calling) threads */
/* Main state control */
static volatile bool audio_codec_loaded NOCACHEBSS_ATTR = false; /* Codec loaded? (C/A-) */
static volatile bool playing NOCACHEBSS_ATTR = false; /* Is audio playing? (A) */
static volatile bool paused NOCACHEBSS_ATTR = false; /* Is audio paused? (A/C-) */
static volatile bool filling IDATA_ATTR = false; /* Is file buffer refilling? (A/C-) */
/* Ring buffer where compressed audio and codecs are loaded */
static unsigned char *filebuf = NULL; /* Start of buffer (A/C-) */
static unsigned char *malloc_buf = NULL; /* Start of malloc buffer (A/C-) */
/* FIXME: make filebuflen static */
size_t filebuflen = 0; /* Size of buffer (A/C-) */
/* FIXME: make buf_ridx (C/A-) */
static volatile size_t buf_ridx IDATA_ATTR = 0; /* Buffer read position (A/C)*/
static volatile size_t buf_widx IDATA_ATTR = 0; /* Buffer write position (A/C-) */
/* Possible arrangements of the buffer */
#define BUFFER_STATE_TRASHED -1 /* trashed; must be reset */
#define BUFFER_STATE_INITIALIZED 0 /* voice+audio OR audio-only */
#define BUFFER_STATE_VOICED_ONLY 1 /* voice-only */
static int buffer_state = BUFFER_STATE_TRASHED; /* Buffer state */
/* Compressed ring buffer helper macros */
/* Buffer pointer (p) plus value (v), wrapped if necessary */
#define RINGBUF_ADD(p,v) ((p+v)<filebuflen ? p+v : p+v-filebuflen)
/* Buffer pointer (p) minus value (v), wrapped if necessary */
#define RINGBUF_SUB(p,v) ((p>=v) ? p-v : p+filebuflen-v)
/* How far value (v) plus buffer pointer (p1) will cross buffer pointer (p2) */
#define RINGBUF_ADD_CROSS(p1,v,p2) \
((p1<p2)?(int)(p1+v)-(int)p2:(int)(p1+v-p2)-(int)filebuflen)
/* Bytes available in the buffer */
#define FILEBUFUSED RINGBUF_SUB(buf_widx, buf_ridx)
/* Track info structure about songs in the file buffer (A/C-) */
static struct track_info tracks[MAX_TRACK];
static volatile int track_ridx = 0; /* Track being decoded (A/C-) */
static int track_widx = 0; /* Track being buffered (A) */
static struct track_info *prev_ti = NULL; /* Previous track info pointer (A/C-) */
#define CUR_TI (&tracks[track_ridx]) /* Playing track info pointer (A/C-) */
/* Set by the audio thread when the current track information has updated
* and the WPS may need to update its cached information */
static bool track_changed = false;
/* Information used only for filling the buffer */
/* Playlist steps from playing track to next track to be buffered (A) */
static int last_peek_offset = 0;
/* Partially loaded track file handle to continue buffering (A) */
static int current_fd = -1;
/* Scrobbler support */
static unsigned long prev_track_elapsed = 0; /* Previous track elapsed time (C/A-)*/
/* Track change controls */
static bool automatic_skip = false; /* Who initiated in-progress skip? (C/A-) */
static bool playlist_end = false; /* Has the current playlist ended? (A) */
static bool dir_skip = false; /* Is a directory skip pending? (A) */
static bool new_playlist = false; /* Are we starting a new playlist? (A) */
static int wps_offset = 0; /* Pending track change offset, to keep WPS responsive (A) */
static bool skipped_during_pause = false; /* Do we need to clear the PCM buffer when playback resumes (A) */
/* Callbacks which applications or plugins may set */
/* When the playing track has changed from the user's perspective */
void (*track_changed_callback)(struct mp3entry *id3) = NULL;
/* When a track has been buffered */
void (*track_buffer_callback)(struct mp3entry *id3, bool last_track) = NULL;
/* When a track's buffer has been overwritten or cleared */
void (*track_unbuffer_callback)(struct mp3entry *id3, bool last_track) = NULL;
/* Configuration */
static size_t conf_watermark = 0; /* Level to trigger filebuf fill (A/C) FIXME */
static size_t conf_filechunk = 0; /* Largest chunk the codec accepts (A/C) FIXME */
static size_t conf_preseek = 0; /* Codec pre-seek margin (A/C) FIXME */
static size_t buffer_margin = 0; /* Buffer margin aka anti-skip buffer (A/C-) */
static bool v1first = false; /* ID3 data control, true if V1 then V2 (A) */
#if MEM > 8
static size_t high_watermark = 0; /* High watermark for rebuffer (A/V/other) */
#endif
/* Multiple threads */
static void set_current_codec(int codec_idx);
/* Set the watermark to trigger buffer fill (A/C) FIXME */
static void set_filebuf_watermark(int seconds);
/* Audio thread */
static struct event_queue audio_queue;
static struct queue_sender_list audio_queue_sender_list;
static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
static const char audio_thread_name[] = "audio";
static void audio_thread(void);
static void audio_initiate_track_change(long direction);
static bool audio_have_tracks(void);
static void audio_reset_buffer(void);
/* Codec thread */
extern struct codec_api ci;
static struct event_queue codec_queue NOCACHEBSS_ATTR;
static long codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)]
IBSS_ATTR;
static const char codec_thread_name[] = "codec";
struct thread_entry *codec_thread_p; /* For modifying thread priority later. */
static volatile int current_codec IDATA_ATTR; /* Current codec (normal/voice) */
/* Voice thread */
#ifdef PLAYBACK_VOICE
extern struct codec_api ci_voice;
static struct thread_entry *voice_thread_p = NULL;
static struct event_queue voice_queue NOCACHEBSS_ATTR;
static long voice_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)]
IBSS_ATTR_VOICE_STACK;
static const char voice_thread_name[] = "voice codec";
/* Voice codec swapping control */
extern unsigned char codecbuf[]; /* DRAM codec swap buffer */
#ifdef SIMULATOR
/* IRAM codec swap buffer for sim*/
static unsigned char sim_iram[CODEC_IRAM_SIZE];
#undef CODEC_IRAM_ORIGIN
#define CODEC_IRAM_ORIGIN sim_iram
#endif
/* iram_buf and dram_buf are either both NULL or both non-NULL */
/* Pointer to IRAM buffer for codec swapping */
static unsigned char *iram_buf = NULL;
/* Pointer to DRAM buffer for codec swapping */
static unsigned char *dram_buf = NULL;
/* Parity of swap_codec calls - needed because one codec swapping itself in
automatically swaps in the other and the swap when unlocking should not
happen if the parity is even.
*/
static bool swap_codec_parity = false; /* true=odd, false=even */
/* Mutex to control which codec (normal/voice) is running */
static struct mutex mutex_codecthread NOCACHEBSS_ATTR;
/* Voice state */
static volatile bool voice_thread_start = false; /* Triggers voice playback (A/V) */
static volatile bool voice_is_playing NOCACHEBSS_ATTR = false; /* Is voice currently playing? (V) */
static volatile bool voice_codec_loaded NOCACHEBSS_ATTR = false; /* Is voice codec loaded (V/A-) */
static unsigned char *voicebuf = NULL;
static size_t voice_remaining = 0;
#ifdef IRAM_STEAL
/* Voice IRAM has been stolen for other use */
static bool voice_iram_stolen = false;
#endif
static void (*voice_getmore)(unsigned char** start, size_t* size) = NULL;
struct voice_info {
void (*callback)(unsigned char **start, size_t* size);
size_t size;
unsigned char *buf;
};
static void voice_thread(void);
static void voice_stop(void);
#endif /* PLAYBACK_VOICE */
/* --- External interfaces --- */
void mp3_play_data(const unsigned char* start, int size,
void (*get_more)(unsigned char** start, size_t* size))
{
#ifdef PLAYBACK_VOICE
static struct voice_info voice_clip;
voice_clip.callback = get_more;
voice_clip.buf = (unsigned char*)start;
voice_clip.size = size;
LOGFQUEUE("mp3 > voice Q_VOICE_STOP");
queue_post(&voice_queue, Q_VOICE_STOP, 0);
LOGFQUEUE("mp3 > voice Q_VOICE_PLAY");
queue_post(&voice_queue, Q_VOICE_PLAY, (intptr_t)&voice_clip);
voice_thread_start = true;
trigger_cpu_boost();
#else
(void) start;
(void) size;
(void) get_more;
#endif
}
void mp3_play_stop(void)
{
#ifdef PLAYBACK_VOICE
queue_remove_from_head(&voice_queue, Q_VOICE_STOP);
LOGFQUEUE("mp3 > voice Q_VOICE_STOP");
queue_post(&voice_queue, Q_VOICE_STOP, 1);
#endif
}
void mp3_play_pause(bool play)
{
/* a dummy */
(void)play;
}
bool mp3_is_playing(void)
{
return voice_is_playing;
}
void mpeg_id3_options(bool _v1first)
{
v1first = _v1first;
}
/* If voice could be swapped out - wait for it to return
* Used by buffer claming functions.
*/
static void wait_for_voice_swap_in(void)
{
#ifdef PLAYBACK_VOICE
if (NULL == iram_buf)
return;
while (current_codec != CODEC_IDX_VOICE)
yield();
#endif /* PLAYBACK_VOICE */
}
/* This sends a stop message and the audio thread will dump all it's
subsequenct messages */
static void audio_hard_stop(void)
{
/* Stop playback */
LOGFQUEUE("audio >| audio Q_AUDIO_STOP: 1");
queue_send(&audio_queue, Q_AUDIO_STOP, 1);
}
unsigned char *audio_get_buffer(bool talk_buf, size_t *buffer_size)
{
unsigned char *buf, *end;
if (audio_is_initialized)
{
audio_hard_stop();
wait_for_voice_swap_in();
voice_stop();
}
/* else buffer_state will be BUFFER_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 = BUFFER_STATE_TRASHED;
return NULL;
}
if (talk_buf || buffer_state == BUFFER_STATE_TRASHED
|| !talk_voice_required())
{
logf("get buffer: talk, audio");
/* Ok to use everything from audiobuf to audiobufend - voice is loaded,
the talk buffer is not needed because voice isn't being used, or
could be BUFFER_STATE_TRASHED already. If state is
BUFFER_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 != BUFFER_STATE_TRASHED)
{
talk_buffer_steal();
buffer_state = BUFFER_STATE_TRASHED;
}
buf = audiobuf;
end = audiobufend;
}
else
{
/* Safe to just return this if already BUFFER_STATE_VOICED_ONLY or
still BUFFER_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 */
logf("get buffer: audio");
buf = audiobuf + talk_get_bufsize();
end = audiobufend - pcmbuf_init(audiobufend);
buffer_state = BUFFER_STATE_VOICED_ONLY;
}
*buffer_size = end - buf;
return buf;
}
#ifdef IRAM_STEAL
void audio_iram_steal(void)
{
/* We need to stop audio playback in order to use codec IRAM */
audio_hard_stop();
#ifdef PLAYBACK_VOICE
if (NULL != iram_buf)
{
/* Can't already be stolen */
if (voice_iram_stolen)
return;
/* Must wait for voice to be current again if it is swapped which
would cause the caller's buffer to get clobbered when voice locks
and runs - we'll wait for it to lock and yield again then make sure
the ride has come to a complete stop */
wait_for_voice_swap_in();
voice_stop();
/* Save voice IRAM but just memcpy - safe to do here since voice
is current and no audio codec is loaded */
memcpy(iram_buf, CODEC_IRAM_ORIGIN, CODEC_IRAM_SIZE);
voice_iram_stolen = true;
}
else
{
/* Nothing much to do if no voice */
voice_iram_stolen = false;
}
#endif
}
#endif /* IRAM_STEAL */
#ifdef HAVE_RECORDING
unsigned char *audio_get_recording_buffer(size_t *buffer_size)
{
/* Don't allow overwrite of voice swap area or we'll trash the
swapped-out voice codec but can use whole thing if none */
unsigned char *end;
/* Stop audio and voice. Wait for voice to swap in and be clear
of pending events to ensure trouble-free operation of encoders */
audio_hard_stop();
wait_for_voice_swap_in();
voice_stop();
talk_buffer_steal();
#ifdef PLAYBACK_VOICE
/* If no dram_buf, swap space not used and recording gets more
memory. Codec swap areas will remain unaffected by the next init
since they're allocated at the end of the buffer and their sizes
don't change between calls */
end = dram_buf;
if (NULL == end)
#endif /* PLAYBACK_VOICE */
end = audiobufend;
buffer_state = BUFFER_STATE_TRASHED;
*buffer_size = end - audiobuf;
return (unsigned char *)audiobuf;
}
bool audio_load_encoder(int afmt)
{
#ifndef SIMULATOR
const char *enc_fn = get_codec_filename(afmt | CODEC_TYPE_ENCODER);
if (!enc_fn)
return false;
audio_remove_encoder();
ci.enc_codec_loaded = 0; /* clear any previous error condition */
LOGFQUEUE("codec > Q_ENCODER_LOAD_DISK");
queue_post(&codec_queue, Q_ENCODER_LOAD_DISK, (intptr_t)enc_fn);
while (ci.enc_codec_loaded == 0)
yield();
logf("codec loaded: %d", ci.enc_codec_loaded);
return ci.enc_codec_loaded > 0;
#else
(void)afmt;
return true;
#endif
} /* audio_load_encoder */
void audio_remove_encoder(void)
{
#ifndef SIMULATOR
/* force encoder codec unload (if currently loaded) */
if (ci.enc_codec_loaded <= 0)
return;
ci.stop_encoder = true;
while (ci.enc_codec_loaded > 0)
yield();
#endif
} /* audio_remove_encoder */
#endif /* HAVE_RECORDING */
struct mp3entry* audio_current_track(void)
{
const char *filename;
const char *p;
static struct mp3entry temp_id3;
int cur_idx;
int offset = ci.new_track + wps_offset;
cur_idx = track_ridx + offset;
cur_idx &= MAX_TRACK_MASK;
if (tracks[cur_idx].taginfo_ready)
return &tracks[cur_idx].id3;
memset(&temp_id3, 0, sizeof(struct mp3entry));
filename = playlist_peek(0);
if (!filename)
filename = "No file!";
#ifdef HAVE_TC_RAMCACHE
if (tagcache_fill_tags(&temp_id3, filename))
return &temp_id3;
#endif
p = strrchr(filename, '/');
if (!p)
p = filename;
else
p++;
strncpy(temp_id3.path, p, sizeof(temp_id3.path)-1);
temp_id3.title = &temp_id3.path[0];
return &temp_id3;
}
struct mp3entry* audio_next_track(void)
{
int next_idx = track_ridx;
if (!audio_have_tracks())
return NULL;
next_idx++;
next_idx &= MAX_TRACK_MASK;
if (!tracks[next_idx].taginfo_ready)
return NULL;
return &tracks[next_idx].id3;
}
bool audio_has_changed_track(void)
{
if (track_changed)
{
track_changed = false;
return true;
}
return false;
}
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 */
do_shutup();
#endif
/* Start playback */
LOGFQUEUE("audio >| audio Q_AUDIO_PLAY: %ld", offset);
/* Don't return until playback has actually started */
queue_send(&audio_queue, Q_AUDIO_PLAY, offset);
}
void audio_stop(void)
{
/* Stop playback */
LOGFQUEUE("audio >| audio Q_AUDIO_STOP");
/* Don't return until playback has actually stopped */
queue_send(&audio_queue, Q_AUDIO_STOP, 0);
}
void audio_pause(void)
{
LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE");
/* Don't return until playback has actually paused */
queue_send(&audio_queue, Q_AUDIO_PAUSE, true);
}
void audio_resume(void)
{
LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE resume");
/* Don't return until playback has actually resumed */
queue_send(&audio_queue, Q_AUDIO_PAUSE, false);
}
void audio_next(void)
{
if (playlist_check(ci.new_track + wps_offset + 1))
{
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
LOGFQUEUE("audio > audio Q_AUDIO_SKIP 1");
queue_post(&audio_queue, Q_AUDIO_SKIP, 1);
/* Update wps while our message travels inside deep playback queues. */
wps_offset++;
track_changed = true;
}
else
{
/* No more tracks. */
if (global_settings.beep)
pcmbuf_beep(1000, 100, 1000*global_settings.beep);
}
}
void audio_prev(void)
{
if (playlist_check(ci.new_track + wps_offset - 1))
{
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
LOGFQUEUE("audio > audio Q_AUDIO_SKIP -1");
queue_post(&audio_queue, Q_AUDIO_SKIP, -1);
/* Update wps while our message travels inside deep playback queues. */
wps_offset--;
track_changed = true;
}
else
{
/* No more tracks. */
if (global_settings.beep)
pcmbuf_beep(1000, 100, 1000*global_settings.beep);
}
}
void audio_next_dir(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP 1");
queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, 1);
}
void audio_prev_dir(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP -1");
queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, -1);
}
void audio_pre_ff_rewind(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_PRE_FF_REWIND");
queue_post(&audio_queue, Q_AUDIO_PRE_FF_REWIND, 0);
}
void audio_ff_rewind(long newpos)
{
LOGFQUEUE("audio > audio Q_AUDIO_FF_REWIND");
queue_post(&audio_queue, Q_AUDIO_FF_REWIND, newpos);
}
void audio_flush_and_reload_tracks(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_FLUSH");
queue_post(&audio_queue, Q_AUDIO_FLUSH, 0);
}
void audio_error_clear(void)
{
#ifdef AUDIO_HAVE_RECORDING
pcm_rec_error_clear();
#endif
}
int audio_status(void)
{
int ret = 0;
if (playing)
ret |= AUDIO_STATUS_PLAY;
if (paused)
ret |= AUDIO_STATUS_PAUSE;
#ifdef HAVE_RECORDING
/* Do this here for constitency with mpeg.c version */
ret |= pcm_rec_status();
#endif
return ret;
}
int audio_get_file_pos(void)
{
return 0;
}
#ifndef HAVE_FLASH_STORAGE
void audio_set_buffer_margin(int setting)
{
static const int lookup[] = {5, 15, 30, 60, 120, 180, 300, 600};
buffer_margin = lookup[setting];
logf("buffer margin: %ld", buffer_margin);
set_filebuf_watermark(buffer_margin);
}
#endif
/* Take nescessary steps to enable or disable the crossfade setting */
void audio_set_crossfade(int enable)
{
size_t offset;
bool was_playing;
size_t size;
/* Tell it the next setting to use */
pcmbuf_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())
{
/* This function is a copout and just syncs some variables -
to be removed at a later date */
pcmbuf_crossfade_enable_finished();
return;
}
offset = 0;
was_playing = playing;
/* Playback has to be stopped before changing the buffer size */
if (was_playing)
{
/* Store the track resume position */
offset = CUR_TI->id3.offset;
gui_syncsplash(0, str(LANG_RESTARTING_PLAYBACK));
}
/* Blast it - audio buffer will have to be setup again next time
something plays */
audio_get_buffer(true, &size);
/* Restart playback if audio was running previously */
if (was_playing)
audio_play(offset);
}
/* --- Routines called from multiple threads --- */
static void set_current_codec(int codec_idx)
{
current_codec = codec_idx;
dsp_configure(DSP_SWITCH_CODEC, codec_idx);
}
#ifdef PLAYBACK_VOICE
static void swap_codec(void)
{
int my_codec;
/* Swap nothing if no swap buffers exist */
if (dram_buf == NULL)
{
logf("swap: no swap buffers");
return;
}
my_codec = current_codec;
logf("swapping out codec: %d", my_codec);
/* Invert this when a codec thread enters and leaves */
swap_codec_parity = !swap_codec_parity;
/* If this is true, an odd number of calls has occurred and there's
no codec thread waiting to swap us out when it locks and runs. This
occurs when playback is stopped or when just starting playback and
the audio thread is loading a codec; parities should always be even
on entry when a thread calls this during playback */
if (swap_codec_parity)
{
/* Save our current IRAM and DRAM */
#ifdef IRAM_STEAL
if (voice_iram_stolen)
{
logf("swap: iram restore");
voice_iram_stolen = false;
/* Don't swap trashed data into buffer as the voice IRAM will
already be swapped out - should _always_ be the case if
voice_iram_stolen is true since the voice has been swapped
in beforehand */
if (my_codec == CODEC_IDX_VOICE)
{
logf("voice iram already swapped");
goto skip_iram_swap;
}
}
#endif
memswap128(iram_buf, CODEC_IRAM_ORIGIN, CODEC_IRAM_SIZE);
#ifdef IRAM_STEAL
skip_iram_swap:
#endif
memswap128(dram_buf, codecbuf, CODEC_SIZE);
/* No cache invalidation needed; it will be done in codec_load_ram
or we won't be here otherwise */
}
/* Release my semaphore */
mutex_unlock(&mutex_codecthread);
logf("unlocked: %d", my_codec);
/* Loop until the other codec has locked and run */
do {
/* Release my semaphore and force a task switch. */
yield();
} while (my_codec == current_codec);
/* Wait for other codec to unlock */
mutex_lock(&mutex_codecthread);
/* Take control */
logf("waiting for lock: %d", my_codec);
set_current_codec(my_codec);
/* Reload our IRAM and DRAM */
memswap128(iram_buf, CODEC_IRAM_ORIGIN, CODEC_IRAM_SIZE);
memswap128(dram_buf, codecbuf, CODEC_SIZE);
invalidate_icache();
/* Flip parity again */
swap_codec_parity = !swap_codec_parity;
logf("resuming codec: %d", my_codec);
}
/* This function is meant to be used by the buffer stealing functions to
ensure the codec is no longer active and so voice will be swapped-in
before it is called */
static void voice_stop(void)
{
#ifdef PLAYBACK_VOICE
/* Must have a voice codec loaded or we'll hang forever here */
if (!voice_codec_loaded)
return;
do_shutup();
/* Loop until voice empties it's queue, stops and picks up on the new
track; the voice thread must be stopped and waiting for messages
outside the codec */
while (voice_is_playing || !queue_empty(&voice_queue) ||
ci_voice.new_track)
yield();
if (!playing)
pcmbuf_play_stop();
#endif
} /* voice_stop */
/* Is voice still speaking */
/* Unfortunately only reliable when music is not also playing. */
static bool is_voice_speaking(void)
{
return is_voice_queued()
|| voice_is_playing
|| (!playing && pcm_is_playing());
}
/* Wait for voice to finish speaking. */
/* Also only reliable when music is not also playing. */
void voice_wait(void)
{
while (is_voice_speaking())
sleep(HZ/10);
}
#endif /* PLAYBACK_VOICE */
static void set_filebuf_watermark(int seconds)
{
size_t bytes;
if (!filebuf)
return; /* Audio buffers not yet set up */
bytes = MAX(CUR_TI->id3.bitrate * seconds * (1000/8), conf_watermark);
bytes = MIN(bytes, filebuflen / 2);
conf_watermark = bytes;
}
const char * get_codec_filename(int cod_spec)
{
const char *fname;
#ifdef HAVE_RECORDING
/* Can choose decoder or encoder if one available */
int type = cod_spec & CODEC_TYPE_MASK;
int afmt = cod_spec & CODEC_AFMT_MASK;
if ((unsigned)afmt >= AFMT_NUM_CODECS)
type = AFMT_UNKNOWN | (type & CODEC_TYPE_MASK);
fname = (type == CODEC_TYPE_ENCODER) ?
audio_formats[afmt].codec_enc_root_fn :
audio_formats[afmt].codec_root_fn;
logf("%s: %d - %s",
(type == CODEC_TYPE_ENCODER) ? "Encoder" : "Decoder",
afmt, fname ? fname : "<unknown>");
#else /* !HAVE_RECORDING */
/* Always decoder */
if ((unsigned)cod_spec >= AFMT_NUM_CODECS)
cod_spec = AFMT_UNKNOWN;
fname = audio_formats[cod_spec].codec_root_fn;
logf("Codec: %d - %s", cod_spec, fname ? fname : "<unknown>");
#endif /* HAVE_RECORDING */
return fname;
} /* get_codec_filename */
/* --- Voice thread --- */
#ifdef PLAYBACK_VOICE
static bool voice_pcmbuf_insert_callback(
const void *ch1, const void *ch2, int count)
{
const char *src[2] = { ch1, ch2 };
while (count > 0)
{
int out_count = dsp_output_count(count);
int inp_count;
char *dest;
while ((dest = pcmbuf_request_voice_buffer(
&out_count, playing)) == NULL)
{
if (playing && audio_codec_loaded)
swap_codec();
else
yield();
}
/* Get the real input_size for output_size bytes, guarding
* against resampling buffer overflows. */
inp_count = dsp_input_count(out_count);
if (inp_count <= 0)
return true;
/* Input size has grown, no error, just don't write more than length */
if (inp_count > count)
inp_count = count;
out_count = dsp_process(dest, src, inp_count);
if (out_count <= 0)
return true;
if (playing)
{
pcmbuf_mix_voice(out_count);
if ((pcmbuf_usage() < 10 || pcmbuf_mix_free() < 30) &&
audio_codec_loaded)
swap_codec();
}
else
pcmbuf_write_complete(out_count);
count -= inp_count;
}
return true;
} /* voice_pcmbuf_insert_callback */
static void* voice_get_memory_callback(size_t *size)
{
/* Voice should have no use for this. If it did, we'd have to
swap the malloc buffer as well. */
*size = 0;
return NULL;
}
static void voice_set_elapsed_callback(unsigned int value)
{
(void)value;
}
static void voice_set_offset_callback(size_t value)
{
(void)value;
}
static void voice_configure_callback(int setting, intptr_t value)
{
if (!dsp_configure(setting, value))
{
logf("Illegal key:%d", setting);
}
}
static size_t voice_filebuf_callback(void *ptr, size_t size)
{
(void)ptr;
(void)size;
return 0;
}
/* Handle Q_VOICE_STOP and part of SYS_USB_CONNECTED */
static bool voice_on_voice_stop(bool aborting, size_t *realsize)
{
if (aborting && !playing && pcm_is_playing())
{
/* Aborting: Slight hack - flush PCM buffer if
only being used for voice */
pcmbuf_play_stop();
}
if (voice_is_playing)
{
/* Clear the current buffer */
voice_is_playing = false;
voice_getmore = NULL;
voice_remaining = 0;
voicebuf = NULL;
/* Cancel any automatic boost if no more clips requested. */
if (!playing || !voice_thread_start)
sleep(0);
/* Force the codec to think it's changing tracks */
ci_voice.new_track = 1;
*realsize = 0;
return true; /* Yes, change tracks */
}
return false;
}
static void* voice_request_buffer_callback(size_t *realsize, size_t reqsize)
{
struct event ev;
if (ci_voice.new_track)
{
*realsize = 0;
return NULL;
}
while (1)
{
if (voice_is_playing || playing)
{
queue_wait_w_tmo(&voice_queue, &ev, 0);
if (!voice_is_playing && ev.id == SYS_TIMEOUT)
ev.id = Q_AUDIO_PLAY;
}
else
{
queue_wait(&voice_queue, &ev);
}
switch (ev.id) {
case Q_AUDIO_PLAY:
LOGFQUEUE("voice < Q_AUDIO_PLAY");
if (playing)
{
if (audio_codec_loaded)
swap_codec();
yield();
}
break;
#ifdef AUDIO_HAVE_RECORDING
case Q_ENCODER_RECORD:
LOGFQUEUE("voice < Q_ENCODER_RECORD");
swap_codec();
break;
#endif
case Q_VOICE_STOP:
LOGFQUEUE("voice < Q_VOICE_STOP");
if (voice_on_voice_stop(ev.data, realsize))
return NULL;
break;
case SYS_USB_CONNECTED:
{
LOGFQUEUE("voice < SYS_USB_CONNECTED");
bool change_tracks = voice_on_voice_stop(ev.data, realsize);
/* Voice is obviously current so let us swap ourselves away if
playing so audio may stop itself - audio_codec_loaded can
only be true in this case if we're here even if the codec
is only about to load */
if (audio_codec_loaded)
swap_codec();
/* Playback should be finished by now - ack and wait */
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&voice_queue);
if (change_tracks)
return NULL;
break;
}
case Q_VOICE_PLAY:
LOGFQUEUE("voice < Q_VOICE_PLAY");
if (!voice_is_playing)
{
/* Set up new voice data */
struct voice_info *voice_data;
#ifdef IRAM_STEAL
if (voice_iram_stolen)
{
/* Voice is the first to run again and is currently
loaded */
logf("voice: iram restore");
memcpy(CODEC_IRAM_ORIGIN, iram_buf, CODEC_IRAM_SIZE);
voice_iram_stolen = false;
}
#endif
/* Must reset the buffer before any playback begins if
needed */
if (buffer_state == BUFFER_STATE_TRASHED)
audio_reset_buffer();
voice_is_playing = true;
trigger_cpu_boost();
voice_data = (struct voice_info *)ev.data;
voice_remaining = voice_data->size;
voicebuf = voice_data->buf;
voice_getmore = voice_data->callback;
}
goto voice_play_clip; /* To exit both switch and while */
case SYS_TIMEOUT:
LOGFQUEUE_SYS_TIMEOUT("voice < SYS_TIMEOUT");
goto voice_play_clip;
default:
LOGFQUEUE("voice < default");
}
}
voice_play_clip:
if (voice_remaining == 0 || voicebuf == NULL)
{
if (voice_getmore)
voice_getmore((unsigned char **)&voicebuf, &voice_remaining);
/* If this clip is done */
if (voice_remaining == 0)
{
LOGFQUEUE("voice > voice Q_VOICE_STOP");
queue_post(&voice_queue, Q_VOICE_STOP, 0);
/* Force pcm playback. */
if (!pcm_is_playing())
pcmbuf_play_start();
}
}
*realsize = MIN(voice_remaining, reqsize);
if (*realsize == 0)
return NULL;
return voicebuf;
} /* voice_request_buffer_callback */
static void voice_advance_buffer_callback(size_t amount)
{
amount = MIN(amount, voice_remaining);
voicebuf += amount;
voice_remaining -= amount;
}
static void voice_advance_buffer_loc_callback(void *ptr)
{
size_t amount = (size_t)ptr - (size_t)voicebuf;
voice_advance_buffer_callback(amount);
}
static off_t voice_mp3_get_filepos_callback(int newtime)
{
(void)newtime;
return 0;
}
static void voice_do_nothing(void)
{
return;
}
static bool voice_seek_buffer_callback(size_t newpos)
{
(void)newpos;
return false;
}
static bool voice_request_next_track_callback(void)
{
ci_voice.new_track = 0;
return true;
}
static void voice_thread(void)
{
logf("Loading voice codec");
voice_codec_loaded = true;
mutex_lock(&mutex_codecthread);
set_current_codec(CODEC_IDX_VOICE);
dsp_configure(DSP_RESET, 0);
voice_remaining = 0;
voice_getmore = NULL;
/* FIXME: If we being starting the voice thread without reboot, the
voice_queue could be full of old stuff and we must flush it. */
codec_load_file(get_codec_filename(AFMT_MPA_L3), &ci_voice);
logf("Voice codec finished");
voice_codec_loaded = false;
mutex_unlock(&mutex_codecthread);
voice_thread_p = NULL;
remove_thread(NULL);
} /* voice_thread */
#endif /* PLAYBACK_VOICE */
/* --- Codec thread --- */
static bool codec_pcmbuf_insert_callback(
const void *ch1, const void *ch2, int count)
{
const char *src[2] = { ch1, ch2 };
while (count > 0)
{
int out_count = dsp_output_count(count);
int inp_count;
char *dest;
/* Prevent audio from a previous track from playing */
if (ci.new_track || ci.stop_codec)
return true;
while ((dest = pcmbuf_request_buffer(&out_count)) == NULL)
{
sleep(1);
if (ci.seek_time || ci.new_track || ci.stop_codec)
return true;
}
/* Get the real input_size for output_size bytes, guarding
* against resampling buffer overflows. */
inp_count = dsp_input_count(out_count);
if (inp_count <= 0)
return true;
/* Input size has grown, no error, just don't write more than length */
if (inp_count > count)
inp_count = count;
out_count = dsp_process(dest, src, inp_count);
if (out_count <= 0)
return true;
pcmbuf_write_complete(out_count);
#ifdef PLAYBACK_VOICE
if ((voice_is_playing || voice_thread_start)
&& pcm_is_playing() && voice_codec_loaded &&
pcmbuf_usage() > 30 && pcmbuf_mix_free() > 80)
{
voice_thread_start = false;
swap_codec();
}
#endif
count -= inp_count;
}
return true;
} /* codec_pcmbuf_insert_callback */
static void* codec_get_memory_callback(size_t *size)
{
*size = MALLOC_BUFSIZE;
return malloc_buf;
}
static void codec_pcmbuf_position_callback(size_t size) ICODE_ATTR;
static void codec_pcmbuf_position_callback(size_t size)
{
/* This is called from an ISR, so be quick */
unsigned int time = size * 1000 / 4 / NATIVE_FREQUENCY +
prev_ti->id3.elapsed;
if (time >= prev_ti->id3.length)
{
pcmbuf_set_position_callback(NULL);
prev_ti->id3.elapsed = prev_ti->id3.length;
}
else
prev_ti->id3.elapsed = time;
}
static void codec_set_elapsed_callback(unsigned int value)
{
unsigned int latency;
if (ci.seek_time)
return;
#ifdef AB_REPEAT_ENABLE
ab_position_report(value);
#endif
latency = pcmbuf_get_latency();
if (value < latency)
CUR_TI->id3.elapsed = 0;
else if (value - latency > CUR_TI->id3.elapsed ||
value - latency < CUR_TI->id3.elapsed - 2)
{
CUR_TI->id3.elapsed = value - latency;
}
}
static void codec_set_offset_callback(size_t value)
{
unsigned int latency;
if (ci.seek_time)
return;
latency = pcmbuf_get_latency() * CUR_TI->id3.bitrate / 8;
if (value < latency)
CUR_TI->id3.offset = 0;
else
CUR_TI->id3.offset = value - latency;
}
static void codec_advance_buffer_counters(size_t amount)
{
buf_ridx = RINGBUF_ADD(buf_ridx, amount);
ci.curpos += amount;
CUR_TI->available -= amount;
/* Start buffer filling as necessary. */
if (!pcmbuf_is_lowdata() && !filling)
{
if (FILEBUFUSED < conf_watermark && playing && !playlist_end)
{
LOGFQUEUE("codec > audio Q_AUDIO_FILL_BUFFER");
queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0);
}
}
}
/* copy up-to size bytes into ptr and return the actual size copied */
static size_t codec_filebuf_callback(void *ptr, size_t size)
{
char *buf = (char *)ptr;
size_t copy_n;
size_t part_n;
if (ci.stop_codec || !playing)
return 0;
/* The ammount to copy is the lesser of the requested amount and the
* amount left of the current track (both on disk and already loaded) */
copy_n = MIN(size, CUR_TI->available + CUR_TI->filerem);
/* Nothing requested OR nothing left */
if (copy_n == 0)
return 0;
/* Let the disk buffer catch fill until enough data is available */
while (copy_n > CUR_TI->available)
{
if (!filling)
{
LOGFQUEUE("codec > audio Q_AUDIO_FILL_BUFFER");
queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0);
}
sleep(1);
if (ci.stop_codec || ci.new_track)
return 0;
}
/* Copy as much as possible without wrapping */
part_n = MIN(copy_n, filebuflen - buf_ridx);
memcpy(buf, &filebuf[buf_ridx], part_n);
/* Copy the rest in the case of a wrap */
if (part_n < copy_n) {
memcpy(&buf[part_n], &filebuf[0], copy_n - part_n);
}
/* Update read and other position pointers */
codec_advance_buffer_counters(copy_n);
/* Return the actual amount of data copied to the buffer */
return copy_n;
} /* codec_filebuf_callback */
static void* codec_request_buffer_callback(size_t *realsize, size_t reqsize)
{
size_t short_n, copy_n, buf_rem;
if (!playing)
{
*realsize = 0;
return NULL;
}
copy_n = MIN(reqsize, CUR_TI->available + CUR_TI->filerem);
if (copy_n == 0)
{
*realsize = 0;
return NULL;
}
while (copy_n > CUR_TI->available)
{
if (!filling)
{
LOGFQUEUE("codec > audio Q_AUDIO_FILL_BUFFER");
queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0);
}
sleep(1);
if (ci.stop_codec || ci.new_track)
{
*realsize = 0;
return NULL;
}
}
/* How much is left at the end of the file buffer before wrap? */
buf_rem = filebuflen - buf_ridx;
/* If we can't satisfy the request without wrapping */
if (buf_rem < copy_n)
{
/* How short are we? */
short_n = copy_n - buf_rem;
/* If we can fudge it with the guardbuf */
if (short_n < GUARD_BUFSIZE)
memcpy(&filebuf[filebuflen], &filebuf[0], short_n);
else
copy_n = buf_rem;
}
*realsize = copy_n;
return (char *)&filebuf[buf_ridx];
} /* codec_request_buffer_callback */
static int get_codec_base_type(int type)
{
switch (type) {
case AFMT_MPA_L1:
case AFMT_MPA_L2:
case AFMT_MPA_L3:
return AFMT_MPA_L3;
}
return type;
}
static void codec_advance_buffer_callback(size_t amount)
{
if (amount > CUR_TI->available + CUR_TI->filerem)
amount = CUR_TI->available + CUR_TI->filerem;
while (amount > CUR_TI->available && filling)
sleep(1);
if (amount > CUR_TI->available)
{
intptr_t result = Q_CODEC_REQUEST_FAILED;
if (!ci.stop_codec)
{
LOGFQUEUE("codec >| audio Q_AUDIO_REBUFFER_SEEK");
result = queue_send(&audio_queue, Q_AUDIO_REBUFFER_SEEK,
ci.curpos + amount);
}
switch (result)
{
case Q_CODEC_REQUEST_FAILED:
LOGFQUEUE("codec |< Q_CODEC_REQUEST_FAILED");
ci.stop_codec = true;
return;
case Q_CODEC_REQUEST_COMPLETE:
LOGFQUEUE("codec |< Q_CODEC_REQUEST_COMPLETE");
return;
default:
LOGFQUEUE("codec |< default");
ci.stop_codec = true;
return;
}
}
codec_advance_buffer_counters(amount);
codec_set_offset_callback(ci.curpos);
}
static void codec_advance_buffer_loc_callback(void *ptr)
{
size_t amount = (size_t)ptr - (size_t)&filebuf[buf_ridx];
codec_advance_buffer_callback(amount);
}
/* Copied from mpeg.c. Should be moved somewhere else. */
static int codec_get_file_pos(void)
{
int pos = -1;
struct mp3entry *id3 = audio_current_track();
if (id3->vbr)
{
if (id3->has_toc)
{
/* Use the TOC to find the new position */
unsigned int percent, remainder;
int curtoc, nexttoc, plen;
percent = (id3->elapsed*100)/id3->length;
if (percent > 99)
percent = 99;
curtoc = id3->toc[percent];
if (percent < 99)
nexttoc = id3->toc[percent+1];
else
nexttoc = 256;
pos = (id3->filesize/256)*curtoc;
/* Use the remainder to get a more accurate position */
remainder = (id3->elapsed*100)%id3->length;
remainder = (remainder*100)/id3->length;
plen = (nexttoc - curtoc)*(id3->filesize/256);
pos += (plen/100)*remainder;
}
else
{
/* No TOC exists, estimate the new position */
pos = (id3->filesize / (id3->length / 1000)) *
(id3->elapsed / 1000);
}
}
else if (id3->bitrate)
pos = id3->elapsed * (id3->bitrate / 8);
else
return -1;
pos += id3->first_frame_offset;
/* Don't seek right to the end of the file so that we can
transition properly to the next song */
if (pos >= (int)(id3->filesize - id3->id3v1len))
pos = id3->filesize - id3->id3v1len - 1;
return pos;
}
static off_t codec_mp3_get_filepos_callback(int newtime)
{
off_t newpos;
CUR_TI->id3.elapsed = newtime;
newpos = codec_get_file_pos();
return newpos;
}
static void codec_seek_complete_callback(void)
{
logf("seek_complete");
if (pcm_is_paused())
{
/* If this is not a seamless seek, clear the buffer */
pcmbuf_play_stop();
dsp_configure(DSP_FLUSH, 0);
/* If playback was not 'deliberately' paused, unpause now */
if (!paused)
pcmbuf_pause(false);
}
ci.seek_time = 0;
}
static bool codec_seek_buffer_callback(size_t newpos)
{
int difference;
logf("codec_seek_buffer_callback");
if (newpos >= CUR_TI->filesize)
newpos = CUR_TI->filesize - 1;
difference = newpos - ci.curpos;
if (difference >= 0)
{
/* Seeking forward */
logf("seek: +%d", difference);
codec_advance_buffer_callback(difference);
return true;
}
/* Seeking backward */
difference = -difference;
if (ci.curpos - difference < 0)
difference = ci.curpos;
/* We need to reload the song. */
if (newpos < CUR_TI->start_pos)
{
intptr_t result = Q_CODEC_REQUEST_FAILED;
if (!ci.stop_codec)
{
LOGFQUEUE("codec >| audio Q_AUDIO_REBUFFER_SEEK");
result = queue_send(&audio_queue, Q_AUDIO_REBUFFER_SEEK,
newpos);
}
switch (result)
{
case Q_CODEC_REQUEST_COMPLETE:
LOGFQUEUE("codec |< Q_CODEC_REQUEST_COMPLETE");
return true;
case Q_CODEC_REQUEST_FAILED:
LOGFQUEUE("codec |< Q_CODEC_REQUEST_FAILED");
ci.stop_codec = true;
return false;
default:
LOGFQUEUE("codec |< default");
return false;
}
}
/* Seeking inside buffer space. */
logf("seek: -%d", difference);
CUR_TI->available += difference;
buf_ridx = RINGBUF_SUB(buf_ridx, (unsigned)difference);
ci.curpos -= difference;
return true;
}
static void codec_configure_callback(int setting, intptr_t value)
{
switch (setting) {
case CODEC_SET_FILEBUF_WATERMARK:
conf_watermark = value;
set_filebuf_watermark(buffer_margin);
break;
case CODEC_SET_FILEBUF_CHUNKSIZE:
conf_filechunk = value;
break;
case CODEC_SET_FILEBUF_PRESEEK:
conf_preseek = value;
break;
default:
if (!dsp_configure(setting, value)) { logf("Illegal key:%d", setting); }
}
}
static void codec_track_changed(void)
{
automatic_skip = false;
LOGFQUEUE("codec > audio Q_AUDIO_TRACK_CHANGED");
queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0);
}
static void codec_pcmbuf_track_changed_callback(void)
{
pcmbuf_set_position_callback(NULL);
codec_track_changed();
}
static void codec_discard_codec_callback(void)
{
if (CUR_TI->has_codec)
{
CUR_TI->has_codec = false;
buf_ridx = RINGBUF_ADD(buf_ridx, CUR_TI->codecsize);
}
#if 0
/* Check if a buffer desync has happened, log it and stop playback. */
if (buf_ridx != CUR_TI->buf_idx)
{
int offset = CUR_TI->buf_idx - buf_ridx;
size_t new_used = FILEBUFUSED - offset;
logf("Buf off :%d=%d-%d", offset, CUR_TI->buf_idx, buf_ridx);
logf("Used off:%d",FILEBUFUSED - new_used);
/* This is a fatal internal error and it's not safe to
* continue playback. */
ci.stop_codec = true;
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
}
#endif
}
static inline void codec_gapless_track_change(void) {
/* callback keeps the progress bar moving while the pcmbuf empties */
pcmbuf_set_position_callback(codec_pcmbuf_position_callback);
/* set the pcmbuf callback for when the track really changes */
pcmbuf_set_event_handler(codec_pcmbuf_track_changed_callback);
}
static inline void codec_crossfade_track_change(void) {
/* Initiate automatic crossfade mode */
pcmbuf_crossfade_init(false);
/* Notify the wps that the track change starts now */
codec_track_changed();
}
static void codec_track_skip_done(bool was_manual)
{
int crossfade_mode = global_settings.crossfade;
/* Manual track change (always crossfade or flush audio). */
if (was_manual)
{
pcmbuf_crossfade_init(true);
LOGFQUEUE("codec > audio Q_AUDIO_TRACK_CHANGED");
queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0);
}
/* Automatic track change w/crossfade, if not in "Track Skip Only" mode. */
else if (pcmbuf_is_crossfade_enabled() && !pcmbuf_is_crossfade_active()
&& crossfade_mode != CROSSFADE_ENABLE_TRACKSKIP)
{
if (crossfade_mode == CROSSFADE_ENABLE_SHUFFLE_AND_TRACKSKIP)
{
if (global_settings.playlist_shuffle)
/* shuffle mode is on, so crossfade: */
codec_crossfade_track_change();
else
/* shuffle mode is off, so do a gapless track change */
codec_gapless_track_change();
}
else
/* normal crossfade: */
codec_crossfade_track_change();
}
else
/* normal gapless playback. */
codec_gapless_track_change();
}
static bool codec_load_next_track(void)
{
intptr_t result = Q_CODEC_REQUEST_FAILED;
prev_track_elapsed = CUR_TI->id3.elapsed;
if (ci.seek_time)
codec_seek_complete_callback();
#ifdef AB_REPEAT_ENABLE
ab_end_of_track_report();
#endif
logf("Request new track");
if (ci.new_track == 0)
{
ci.new_track++;
automatic_skip = true;
}
if (!ci.stop_codec)
{
trigger_cpu_boost();
LOGFQUEUE("codec >| audio Q_AUDIO_CHECK_NEW_TRACK");
result = queue_send(&audio_queue, Q_AUDIO_CHECK_NEW_TRACK, 0);
}
switch (result)
{
case Q_CODEC_REQUEST_COMPLETE:
LOGFQUEUE("codec |< Q_CODEC_REQUEST_COMPLETE");
codec_track_skip_done(!automatic_skip);
return true;
case Q_CODEC_REQUEST_FAILED:
LOGFQUEUE("codec |< Q_CODEC_REQUEST_FAILED");
ci.new_track = 0;
ci.stop_codec = true;
return false;
default:
LOGFQUEUE("codec |< default");
ci.stop_codec = true;
return false;
}
}
static bool codec_request_next_track_callback(void)
{
int prev_codectype;
if (ci.stop_codec || !playing)
return false;
prev_codectype = get_codec_base_type(CUR_TI->id3.codectype);
if (!codec_load_next_track())
return false;
/* Check if the next codec is the same file. */
if (prev_codectype == get_codec_base_type(CUR_TI->id3.codectype))
{
logf("New track loaded");
codec_discard_codec_callback();
return true;
}
else
{
logf("New codec:%d/%d", CUR_TI->id3.codectype, prev_codectype);
return false;
}
}
static void codec_thread(void)
{
struct event ev;
int status;
size_t wrap;
while (1) {
status = 0;
queue_wait(&codec_queue, &ev);
switch (ev.id) {
case Q_CODEC_LOAD_DISK:
LOGFQUEUE("codec < Q_CODEC_LOAD_DISK");
audio_codec_loaded = true;
#ifdef PLAYBACK_VOICE
/* Don't sent messages to voice codec if it's already swapped
out or it will never get this */
if (voice_codec_loaded && current_codec == CODEC_IDX_VOICE)
{
LOGFQUEUE("codec > voice Q_AUDIO_PLAY");
queue_post(&voice_queue, Q_AUDIO_PLAY, 0);
}
mutex_lock(&mutex_codecthread);
#endif
set_current_codec(CODEC_IDX_AUDIO);
ci.stop_codec = false;
status = codec_load_file((const char *)ev.data, &ci);
#ifdef PLAYBACK_VOICE
mutex_unlock(&mutex_codecthread);
#endif
break;
case Q_CODEC_LOAD:
LOGFQUEUE("codec < Q_CODEC_LOAD");
if (!CUR_TI->has_codec) {
logf("Codec slot is empty!");
/* Wait for the pcm buffer to go empty */
while (pcm_is_playing())
yield();
/* This must be set to prevent an infinite loop */
ci.stop_codec = true;
LOGFQUEUE("codec > codec Q_AUDIO_PLAY");
queue_post(&codec_queue, Q_AUDIO_PLAY, 0);
break;
}
audio_codec_loaded = true;
#ifdef PLAYBACK_VOICE
if (voice_codec_loaded && current_codec == CODEC_IDX_VOICE)
{
LOGFQUEUE("codec > voice Q_AUDIO_PLAY");
queue_post(&voice_queue, Q_AUDIO_PLAY, 0);
}
mutex_lock(&mutex_codecthread);
#endif
set_current_codec(CODEC_IDX_AUDIO);
ci.stop_codec = false;
wrap = (size_t)&filebuf[filebuflen] - (size_t)CUR_TI->codecbuf;
status = codec_load_ram(CUR_TI->codecbuf, CUR_TI->codecsize,
&filebuf[0], wrap, &ci);
#ifdef PLAYBACK_VOICE
mutex_unlock(&mutex_codecthread);
#endif
break;
#ifdef AUDIO_HAVE_RECORDING
case Q_ENCODER_LOAD_DISK:
LOGFQUEUE("codec < Q_ENCODER_LOAD_DISK");
audio_codec_loaded = false; /* Not audio codec! */
#ifdef PLAYBACK_VOICE
if (voice_codec_loaded && current_codec == CODEC_IDX_VOICE)
{
LOGFQUEUE("codec > voice Q_ENCODER_RECORD");
queue_post(&voice_queue, Q_ENCODER_RECORD, 0);
}
mutex_lock(&mutex_codecthread);
#endif
logf("loading encoder");
set_current_codec(CODEC_IDX_AUDIO);
ci.stop_encoder = false;
status = codec_load_file((const char *)ev.data, &ci);
#ifdef PLAYBACK_VOICE
mutex_unlock(&mutex_codecthread);
#endif
logf("encoder stopped");
break;
#endif /* AUDIO_HAVE_RECORDING */
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
LOGFQUEUE("codec < SYS_USB_CONNECTED");
queue_clear(&codec_queue);
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&codec_queue);
break;
#endif
default:
LOGFQUEUE("codec < default");
}
if (audio_codec_loaded)
{
if (ci.stop_codec)
{
status = CODEC_OK;
if (!playing)
pcmbuf_play_stop();
}
audio_codec_loaded = false;
}
switch (ev.id) {
case Q_CODEC_LOAD_DISK:
case Q_CODEC_LOAD:
LOGFQUEUE("codec < Q_CODEC_LOAD");
if (playing)
{
if (ci.new_track || status != CODEC_OK)
{
if (!ci.new_track)
{
logf("Codec failure");
gui_syncsplash(HZ*2, "Codec failure");
}
if (!codec_load_next_track())
{
LOGFQUEUE("codec > audio Q_AUDIO_STOP");
/* End of playlist */
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
break;
}
}
else
{
logf("Codec finished");
if (ci.stop_codec)
{
/* Wait for the audio to stop playing before
* triggering the WPS exit */
while(pcm_is_playing())
{
CUR_TI->id3.elapsed =
CUR_TI->id3.length - pcmbuf_get_latency();
sleep(1);
}
LOGFQUEUE("codec > audio Q_AUDIO_STOP");
/* End of playlist */
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
break;
}
}
if (CUR_TI->has_codec)
{
LOGFQUEUE("codec > codec Q_CODEC_LOAD");
queue_post(&codec_queue, Q_CODEC_LOAD, 0);
}
else
{
const char *codec_fn =
get_codec_filename(CUR_TI->id3.codectype);
LOGFQUEUE("codec > codec Q_CODEC_LOAD_DISK");
queue_post(&codec_queue, Q_CODEC_LOAD_DISK,
(intptr_t)codec_fn);
}
}
break;
#ifdef AUDIO_HAVE_RECORDING
case Q_ENCODER_LOAD_DISK:
LOGFQUEUE("codec < Q_ENCODER_LOAD_DISK");
if (status == CODEC_OK)
break;
logf("Encoder failure");
gui_syncsplash(HZ*2, "Encoder failure");
if (ci.enc_codec_loaded < 0)
break;
logf("Encoder failed to load");
ci.enc_codec_loaded = -1;
break;
#endif /* AUDIO_HAVE_RECORDING */
default:
LOGFQUEUE("codec < default");
} /* end switch */
}
}
/* --- Audio thread --- */
static bool audio_filebuf_is_lowdata(void)
{
return FILEBUFUSED < AUDIO_FILEBUF_CRITICAL;
}
static bool audio_have_tracks(void)
{
return track_ridx != track_widx || CUR_TI->filesize;
}
static bool audio_have_free_tracks(void)
{
if (track_widx < track_ridx)
return track_widx + 1 < track_ridx;
else if (track_ridx == 0)
return track_widx < MAX_TRACK - 1;
return true;
}
int audio_track_count(void)
{
if (audio_have_tracks())
{
int relative_track_widx = track_widx;
if (track_ridx > track_widx)
relative_track_widx += MAX_TRACK;
return relative_track_widx - track_ridx + 1;
}
return 0;
}
long audio_filebufused(void)
{
return (long) FILEBUFUSED;
}
/* Count the data BETWEEN the selected tracks */
static size_t audio_buffer_count_tracks(int from_track, int to_track)
{
size_t amount = 0;
bool need_wrap = to_track < from_track;
while (1)
{
if (++from_track >= MAX_TRACK)
{
from_track -= MAX_TRACK;
need_wrap = false;
}
if (from_track >= to_track && !need_wrap)
break;
amount += tracks[from_track].codecsize + tracks[from_track].filesize;
}
return amount;
}
static bool audio_buffer_wind_forward(int new_track_ridx, int old_track_ridx)
{
size_t amount;
/* Start with the remainder of the previously playing track */
amount = tracks[old_track_ridx].filesize - ci.curpos;
/* Then collect all data from tracks in between them */
amount += audio_buffer_count_tracks(old_track_ridx, new_track_ridx);
logf("bwf:%ldB", (long) amount);
if (amount > FILEBUFUSED)
return false;
/* Wind the buffer to the beginning of the target track or its codec */
buf_ridx = RINGBUF_ADD(buf_ridx, amount);
return true;
}
static bool audio_buffer_wind_backward(int new_track_ridx, int old_track_ridx)
{
/* Available buffer data */
size_t buf_back;
/* Start with the previously playing track's data and our data */
size_t amount;
amount = ci.curpos;
buf_back = RINGBUF_SUB(buf_ridx, buf_widx);
/* If we're not just resetting the current track */
if (new_track_ridx != old_track_ridx)
{
/* Need to wind to before the old track's codec and our filesize */
amount += tracks[old_track_ridx].codecsize;
amount += tracks[new_track_ridx].filesize;
/* Rewind the old track to its beginning */
tracks[old_track_ridx].available =
tracks[old_track_ridx].filesize - tracks[old_track_ridx].filerem;
}
/* If the codec was ever buffered */
if (tracks[new_track_ridx].codecsize)
{
/* Add the codec to the needed size */
amount += tracks[new_track_ridx].codecsize;
tracks[new_track_ridx].has_codec = true;
}
/* Then collect all data from tracks between new and old */
amount += audio_buffer_count_tracks(new_track_ridx, old_track_ridx);
/* Do we have space to make this skip? */
if (amount > buf_back)
return false;
logf("bwb:%ldB",amount);
/* Rewind the buffer to the beginning of the target track or its codec */
buf_ridx = RINGBUF_SUB(buf_ridx, amount);
/* Reset to the beginning of the new track */
tracks[new_track_ridx].available = tracks[new_track_ridx].filesize;
return true;
}
static void audio_update_trackinfo(void)
{
ci.filesize = CUR_TI->filesize;
CUR_TI->id3.elapsed = 0;
CUR_TI->id3.offset = 0;
ci.id3 = &CUR_TI->id3;
ci.curpos = 0;
ci.taginfo_ready = &CUR_TI->taginfo_ready;
}
/* Yield to codecs for as long as possible if they are in need of data
* return true if the caller should break to let the audio thread process
* new events */
static bool audio_yield_codecs(void)
{
yield();
if (!queue_empty(&audio_queue))
return true;
while ((pcmbuf_is_crossfade_active() || pcmbuf_is_lowdata())
&& !ci.stop_codec && playing && !audio_filebuf_is_lowdata())
{
if (filling)
yield();
else
sleep(2);
if (!queue_empty(&audio_queue))
return true;
}
return false;
}
static void audio_clear_track_entries(bool clear_unbuffered)
{
int cur_idx = track_widx;
int last_idx = -1;
logf("Clearing tracks:%d/%d, %d", track_ridx, track_widx, clear_unbuffered);
/* Loop over all tracks from write-to-read */
while (1)
{
cur_idx++;
cur_idx &= MAX_TRACK_MASK;
if (cur_idx == track_ridx)
break;
/* If the track is buffered, conditionally clear/notify,
* otherwise clear the track if that option is selected */
if (tracks[cur_idx].event_sent)
{
if (last_idx >= 0)
{
/* If there is an unbuffer callback, call it, otherwise,
* just clear the track */
if (track_unbuffer_callback)
track_unbuffer_callback(&tracks[last_idx].id3, false);
memset(&tracks[last_idx], 0, sizeof(struct track_info));
}
last_idx = cur_idx;
}
else if (clear_unbuffered)
memset(&tracks[cur_idx], 0, sizeof(struct track_info));
}
/* We clear the previous instance of a buffered track throughout
* the above loop to facilitate 'last' detection. Clear/notify
* the last track here */
if (last_idx >= 0)
{
if (track_unbuffer_callback)
track_unbuffer_callback(&tracks[last_idx].id3, true);
memset(&tracks[last_idx], 0, sizeof(struct track_info));
}
}
/* FIXME: This code should be made more generic and move to metadata.c */
static void audio_strip_tags(void)
{
int i;
static const unsigned char tag[] = "TAG";
static const unsigned char apetag[] = "APETAGEX";
size_t tag_idx;
size_t cur_idx;
size_t len, version;
tag_idx = RINGBUF_SUB(buf_widx, 128);
if (FILEBUFUSED > 128 && tag_idx > buf_ridx)
{
cur_idx = tag_idx;
for(i = 0;i < 3;i++)
{
if(filebuf[cur_idx] != tag[i])
goto strip_ape_tag;
cur_idx = RINGBUF_ADD(cur_idx, 1);
}
/* Skip id3v1 tag */
logf("Skipping ID3v1 tag");
buf_widx = tag_idx;
tracks[track_widx].available -= 128;
tracks[track_widx].filesize -= 128;
}
strip_ape_tag:
/* Check for APE tag (look for the APE tag footer) */
tag_idx = RINGBUF_SUB(buf_widx, 32);
if (FILEBUFUSED > 32 && tag_idx > buf_ridx)
{
cur_idx = tag_idx;
for(i = 0;i < 8;i++)
{
if(filebuf[cur_idx] != apetag[i])
return;
cur_idx = RINGBUF_ADD(cur_idx, 1);
}
/* Read the version and length from the footer */
version = filebuf[tag_idx+8] | (filebuf[tag_idx+9] << 8) |
(filebuf[tag_idx+10] << 16) | (filebuf[tag_idx+11] << 24);
len = filebuf[tag_idx+12] | (filebuf[tag_idx+13] << 8) |
(filebuf[tag_idx+14] << 16) | (filebuf[tag_idx+15] << 24);
if (version == 2000)
len += 32; /* APEv2 has a 32 byte header */
/* Skip APE tag */
if (FILEBUFUSED > len)
{
logf("Skipping APE tag (%ldB)", len);
buf_widx = RINGBUF_SUB(buf_widx, len);
tracks[track_widx].available -= len;
tracks[track_widx].filesize -= len;
}
}
}
/* Returns true if a whole file is read, false otherwise */
static bool audio_read_file(size_t minimum)
{
bool ret_val = false;
/* If we're called and no file is open, this is an error */
if (current_fd < 0)
{
logf("Bad fd in arf");
/* Give some hope of miraculous recovery by forcing a track reload */
tracks[track_widx].filesize = 0;
/* Stop this buffering run */
return ret_val;
}
trigger_cpu_boost();
while (tracks[track_widx].filerem > 0)
{
size_t copy_n;
int overlap;
int rc;
/* copy_n is the largest chunk that is safe to read */
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
/* buf_widx == buf_ridx is defined as buffer empty, not buffer full */
if (RINGBUF_ADD_CROSS(buf_widx,copy_n,buf_ridx) >= 0)
break;
/* rc is the actual amount read */
rc = read(current_fd, &filebuf[buf_widx], copy_n);
if (rc < 0)
{
logf("File ended %ldB early", tracks[track_widx].filerem);
tracks[track_widx].filesize -= tracks[track_widx].filerem;
tracks[track_widx].filerem = 0;
break;
}
/* How much of the playing track did we overwrite */
if (buf_widx == CUR_TI->buf_idx)
{
/* Special handling; zero or full overlap? */
if (track_widx == track_ridx && CUR_TI->available == 0)
overlap = 0;
else
overlap = rc;
}
else
overlap = RINGBUF_ADD_CROSS(buf_widx,rc,CUR_TI->buf_idx);
if ((unsigned)rc > tracks[track_widx].filerem)
{
logf("Bad: rc-filerem=%ld, fixing", rc-tracks[track_widx].filerem);
tracks[track_widx].filesize += rc - tracks[track_widx].filerem;
tracks[track_widx].filerem = rc;
}
/* Advance buffer */
buf_widx = RINGBUF_ADD(buf_widx, rc);
tracks[track_widx].available += rc;
tracks[track_widx].filerem -= rc;
/* If we write into the playing track, adjust it's buffer info */
if (overlap > 0)
{
CUR_TI->buf_idx += overlap;
CUR_TI->start_pos += overlap;
}
/* For a rebuffer, fill at least this minimum */
if (minimum > (unsigned)rc)
minimum -= rc;
/* Let the codec process up to the watermark */
/* Break immediately if this is a quick buffer, or there is an event */
else if (minimum || audio_yield_codecs())
{
/* Exit quickly, but don't stop the overall buffering process */
ret_val = true;
break;
}
}
if (tracks[track_widx].filerem == 0)
{
logf("Finished buf:%ldB", tracks[track_widx].filesize);
close(current_fd);
current_fd = -1;
audio_strip_tags();
track_widx++;
track_widx &= MAX_TRACK_MASK;
tracks[track_widx].filesize = 0;
return true;
}
else
{
logf("%s buf:%ldB", ret_val?"Quick":"Partially",
tracks[track_widx].filesize - tracks[track_widx].filerem);
return ret_val;
}
}
static bool audio_loadcodec(bool start_play)
{
size_t size = 0;
int fd;
int rc;
size_t copy_n;
int prev_track;
char codec_path[MAX_PATH]; /* Full path to codec */
const char * codec_fn =
get_codec_filename(tracks[track_widx].id3.codectype);
if (codec_fn == NULL)
return false;
tracks[track_widx].has_codec = false;
if (start_play)
{
/* Load the codec directly from disk and save some memory. */
track_ridx = track_widx;
ci.filesize = CUR_TI->filesize;
ci.id3 = &CUR_TI->id3;
ci.taginfo_ready = &CUR_TI->taginfo_ready;
ci.curpos = 0;
LOGFQUEUE("codec > codec Q_CODEC_LOAD_DISK");
queue_post(&codec_queue, Q_CODEC_LOAD_DISK, (intptr_t)codec_fn);
return true;
}
else
{
/* If we already have another track than this one buffered */
if (track_widx != track_ridx)
{
prev_track = (track_widx - 1) & MAX_TRACK_MASK;
/* If the previous codec is the same as this one, there is no need
* to put another copy of it on the file buffer */
if (get_codec_base_type(tracks[track_widx].id3.codectype) ==
get_codec_base_type(tracks[prev_track].id3.codectype)
&& audio_codec_loaded)
{
logf("Reusing prev. codec");
return true;
}
}
}
codec_get_full_path(codec_path, codec_fn);
fd = open(codec_path, O_RDONLY);
if (fd < 0)
{
logf("Codec doesn't exist!");
return false;
}
tracks[track_widx].codecsize = filesize(fd);
/* Never load a partial codec */
if (RINGBUF_ADD_CROSS(buf_widx,tracks[track_widx].codecsize,buf_ridx) >= 0)
{
logf("Not enough space");
close(fd);
return false;
}
while (size < tracks[track_widx].codecsize)
{
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
rc = read(fd, &filebuf[buf_widx], copy_n);
if (rc < 0)
{
close(fd);
/* This is an error condition, likely the codec file is corrupt */
logf("Partial codec loaded");
/* Must undo the buffer write of the partial codec */
buf_widx = RINGBUF_SUB(buf_widx, size);
tracks[track_widx].codecsize = 0;
return false;
}
buf_widx = RINGBUF_ADD(buf_widx, rc);
size += rc;
}
tracks[track_widx].has_codec = true;
close(fd);
logf("Done: %ldB", size);
return true;
}
/* TODO: Copied from mpeg.c. Should be moved somewhere else. */
static void audio_set_elapsed(struct mp3entry* id3)
{
unsigned long offset = id3->offset > id3->first_frame_offset ?
id3->offset - id3->first_frame_offset : 0;
if ( id3->vbr ) {
if ( id3->has_toc ) {
/* calculate elapsed time using TOC */
int i;
unsigned int remainder, plen, relpos, nextpos;
/* find wich percent we're at */
for (i=0; i<100; i++ )
if ( offset < id3->toc[i] * (id3->filesize / 256) )
break;
i--;
if (i < 0)
i = 0;
relpos = id3->toc[i];
if (i < 99)
nextpos = id3->toc[i+1];
else
nextpos = 256;
remainder = offset - (relpos * (id3->filesize / 256));
/* set time for this percent (divide before multiply to prevent
overflow on long files. loss of precision is negligible on
short files) */
id3->elapsed = i * (id3->length / 100);
/* calculate remainder time */
plen = (nextpos - relpos) * (id3->filesize / 256);
id3->elapsed += (((remainder * 100) / plen) *
(id3->length / 10000));
}
else {
/* no TOC exists. set a rough estimate using average bitrate */
int tpk = id3->length /
((id3->filesize - id3->first_frame_offset - id3->id3v1len) /
1024);
id3->elapsed = offset / 1024 * tpk;
}
}
else
{
/* constant bitrate, use exact calculation */
if (id3->bitrate != 0)
id3->elapsed = offset / (id3->bitrate / 8);
}
}
static bool audio_load_track(int offset, bool start_play, bool rebuffer)
{
char *trackname;
off_t size;
char msgbuf[80];
/* Stop buffer filling if there is no free track entries.
Don't fill up the last track entry (we wan't to store next track
metadata there). */
if (!audio_have_free_tracks())
{
logf("No free tracks");
return false;
}
if (current_fd >= 0)
{
logf("Nonzero fd in alt");
close(current_fd);
current_fd = -1;
}
last_peek_offset++;
peek_again:
logf("Buffering track:%d/%d", track_widx, track_ridx);
/* Get track name from current playlist read position. */
while ((trackname = playlist_peek(last_peek_offset)) != NULL)
{
/* Handle broken playlists. */
current_fd = open(trackname, O_RDONLY);
if (current_fd < 0)
{
logf("Open failed");
/* Skip invalid entry from playlist. */
playlist_skip_entry(NULL, last_peek_offset);
}
else
break;
}
if (!trackname)
{
logf("End-of-playlist");
playlist_end = true;
return false;
}
/* Initialize track entry. */
size = filesize(current_fd);
tracks[track_widx].filerem = size;
tracks[track_widx].filesize = size;
tracks[track_widx].available = 0;
/* Set default values */
if (start_play)
{
int last_codec = current_codec;
set_current_codec(CODEC_IDX_AUDIO);
conf_watermark = AUDIO_DEFAULT_WATERMARK;
conf_filechunk = AUDIO_DEFAULT_FILECHUNK;
conf_preseek = AUDIO_REBUFFER_GUESS_SIZE;
dsp_configure(DSP_RESET, 0);
set_current_codec(last_codec);
}
/* Get track metadata if we don't already have it. */
if (!tracks[track_widx].taginfo_ready)
{
if (get_metadata(&(tracks[track_widx].id3),current_fd,trackname,v1first))
{
tracks[track_widx].taginfo_ready = true;
if (start_play)
{
track_changed = true;
playlist_update_resume_info(audio_current_track());
}
}
else
{
logf("mde:%s!",trackname);
/* Set filesize to zero to indicate no file was loaded. */
tracks[track_widx].filesize = 0;
tracks[track_widx].filerem = 0;
close(current_fd);
current_fd = -1;
/* Skip invalid entry from playlist. */
playlist_skip_entry(NULL, last_peek_offset);
tracks[track_widx].taginfo_ready = false;
goto peek_again;
}
}
if (cuesheet_is_enabled() && tracks[track_widx].id3.cuesheet_type == 1)
{
char cuepath[MAX_PATH];
struct cuesheet *cue = start_play ? curr_cue : temp_cue;
if (look_for_cuesheet_file(trackname, cuepath) &&
parse_cuesheet(cuepath, cue))
{
strcpy((cue)->audio_filename, trackname);
if (start_play)
cue_spoof_id3(curr_cue, &tracks[track_widx].id3);
}
}
/* Load the codec. */
tracks[track_widx].codecbuf = &filebuf[buf_widx];
if (!audio_loadcodec(start_play))
{
/* Set filesize to zero to indicate no file was loaded. */
tracks[track_widx].filesize = 0;
tracks[track_widx].filerem = 0;
close(current_fd);
current_fd = -1;
if (tracks[track_widx].codecsize)
{
/* No space for codec on buffer, not an error */
tracks[track_widx].codecsize = 0;
return false;
}
/* This is an error condition, either no codec was found, or reading
* the codec file failed part way through, either way, skip the track */
snprintf(msgbuf, sizeof(msgbuf)-1, "No codec for: %s", trackname);
/* We should not use gui_syncplash from audio thread! */
gui_syncsplash(HZ*2, msgbuf);
/* Skip invalid entry from playlist. */
playlist_skip_entry(NULL, last_peek_offset);
tracks[track_widx].taginfo_ready = false;
goto peek_again;
}
tracks[track_widx].start_pos = 0;
set_filebuf_watermark(buffer_margin);
tracks[track_widx].id3.elapsed = 0;
if (offset > 0)
{
switch (tracks[track_widx].id3.codectype) {
case AFMT_MPA_L1:
case AFMT_MPA_L2:
case AFMT_MPA_L3:
lseek(current_fd, offset, SEEK_SET);
tracks[track_widx].id3.offset = offset;
audio_set_elapsed(&tracks[track_widx].id3);
tracks[track_widx].filerem = size - offset;
ci.curpos = offset;
tracks[track_widx].start_pos = offset;
break;
case AFMT_WAVPACK:
lseek(current_fd, offset, SEEK_SET);
tracks[track_widx].id3.offset = offset;
tracks[track_widx].id3.elapsed =
tracks[track_widx].id3.length / 2;
tracks[track_widx].filerem = size - offset;
ci.curpos = offset;
tracks[track_widx].start_pos = offset;
break;
case AFMT_OGG_VORBIS:
case AFMT_SPEEX:
case AFMT_FLAC:
case AFMT_PCM_WAV:
case AFMT_A52:
case AFMT_AAC:
case AFMT_MPC:
case AFMT_APE:
tracks[track_widx].id3.offset = offset;
break;
}
}
logf("alt:%s", trackname);
tracks[track_widx].buf_idx = buf_widx;
return audio_read_file(rebuffer);
}
static bool audio_read_next_metadata(void)
{
int fd;
char *trackname;
int next_idx;
int status;
next_idx = track_widx;
if (tracks[next_idx].taginfo_ready)
{
next_idx++;
next_idx &= MAX_TRACK_MASK;
if (tracks[next_idx].taginfo_ready)
return true;
}
trackname = playlist_peek(last_peek_offset + 1);
if (!trackname)
return false;
fd = open(trackname, O_RDONLY);
if (fd < 0)
return false;
status = get_metadata(&(tracks[next_idx].id3),fd,trackname,v1first);
/* Preload the glyphs in the tags */
if (status)
{
tracks[next_idx].taginfo_ready = true;
if (tracks[next_idx].id3.title)
lcd_getstringsize(tracks[next_idx].id3.title, NULL, NULL);
if (tracks[next_idx].id3.artist)
lcd_getstringsize(tracks[next_idx].id3.artist, NULL, NULL);
if (tracks[next_idx].id3.album)
lcd_getstringsize(tracks[next_idx].id3.album, NULL, NULL);
}
close(fd);
return status;
}
/* Send callback events to notify about new tracks. */
static void audio_generate_postbuffer_events(void)
{
int cur_idx;
int last_idx = -1;
logf("Postbuffer:%d/%d",track_ridx,track_widx);
if (audio_have_tracks())
{
cur_idx = track_ridx;
while (1) {
if (!tracks[cur_idx].event_sent)
{
if (last_idx >= 0 && !tracks[last_idx].event_sent)
{
/* Mark the event 'sent' even if we don't really send one */
tracks[last_idx].event_sent = true;
if (track_buffer_callback)
track_buffer_callback(&tracks[last_idx].id3, false);
}
last_idx = cur_idx;
}
if (cur_idx == track_widx)
break;
cur_idx++;
cur_idx &= MAX_TRACK_MASK;
}
if (last_idx >= 0 && !tracks[last_idx].event_sent)
{
tracks[last_idx].event_sent = true;
if (track_buffer_callback)
track_buffer_callback(&tracks[last_idx].id3, true);
}
}
}
static bool audio_initialize_buffer_fill(bool clear_tracks)
{
/* Don't initialize if we're already initialized */
if (filling)
return true;
logf("Starting buffer fill");
/* Set the filling flag true before calling audio_clear_tracks as that
* function can yield and we start looping. */
filling = true;
if (clear_tracks)
audio_clear_track_entries(false);
/* Save the current resume position once. */
playlist_update_resume_info(audio_current_track());
return true;
}
static void audio_fill_file_buffer(
bool start_play, bool rebuffer, size_t offset)
{
bool had_next_track = audio_next_track() != NULL;
bool continue_buffering;
/* Must reset the buffer before use if trashed or voice only - voice
file size shouldn't have changed so we can go straight from
BUFFER_STATE_VOICED_ONLY to BUFFER_STATE_INITIALIZED */
if (buffer_state != BUFFER_STATE_INITIALIZED)
audio_reset_buffer();
if (!audio_initialize_buffer_fill(!start_play))
return ;
/* If we have a partially buffered track, continue loading,
* otherwise load a new track */
if (tracks[track_widx].filesize > 0)
continue_buffering = audio_read_file(rebuffer);
else
continue_buffering = audio_load_track(offset, start_play, rebuffer);
if (!had_next_track && audio_next_track())
track_changed = true;
/* If we're done buffering */
if (!continue_buffering)
{
audio_read_next_metadata();
audio_generate_postbuffer_events();
filling = false;
}
#ifndef SIMULATOR
ata_sleep();
#endif
}
static void audio_rebuffer(void)
{
logf("Forcing rebuffer");
/* Stop in progress fill, and clear open file descriptor */
if (current_fd >= 0)
{
close(current_fd);
current_fd = -1;
}
filling = false;
/* Reset buffer and track pointers */
CUR_TI->buf_idx = buf_ridx = buf_widx = 0;
track_widx = track_ridx;
audio_clear_track_entries(true);
CUR_TI->available = 0;
/* Fill the buffer */
last_peek_offset = -1;
CUR_TI->filesize = 0;
CUR_TI->start_pos = 0;
ci.curpos = 0;
if (!CUR_TI->taginfo_ready)
memset(&CUR_TI->id3, 0, sizeof(struct mp3entry));
audio_fill_file_buffer(false, true, 0);
}
static int audio_check_new_track(void)
{
int track_count = audio_track_count();
int old_track_ridx = track_ridx;
bool forward;
if (dir_skip)
{
dir_skip = false;
if (playlist_next_dir(ci.new_track))
{
ci.new_track = 0;
CUR_TI->taginfo_ready = false;
audio_rebuffer();
goto skip_done;
}
else
{
LOGFQUEUE("audio >|= codec Q_CODEC_REQUEST_FAILED");
return Q_CODEC_REQUEST_FAILED;
}
}
if (new_playlist)
ci.new_track = 0;
/* If the playlist isn't that big */
if (!playlist_check(ci.new_track))
{
if (ci.new_track >= 0)
{
LOGFQUEUE("audio >|= codec Q_CODEC_REQUEST_FAILED");
return Q_CODEC_REQUEST_FAILED;
}
/* Find the beginning backward if the user over-skips it */
while (!playlist_check(++ci.new_track))
if (ci.new_track >= 0)
{
LOGFQUEUE("audio >|= codec Q_CODEC_REQUEST_FAILED");
return Q_CODEC_REQUEST_FAILED;
}
}
/* Update the playlist */
last_peek_offset -= ci.new_track;
if (playlist_next(ci.new_track) < 0)
{
LOGFQUEUE("audio >|= codec Q_CODEC_REQUEST_FAILED");
return Q_CODEC_REQUEST_FAILED;
}
if (new_playlist)
{
ci.new_track = 1;
new_playlist = false;
}
/* Save the old track */
prev_ti = CUR_TI;
/* Move to the new track */
track_ridx += ci.new_track;
track_ridx &= MAX_TRACK_MASK;
if (automatic_skip)
playlist_end = false;
track_changed = !automatic_skip;
/* If it is not safe to even skip this many track entries */
if (ci.new_track >= track_count || ci.new_track <= track_count - MAX_TRACK)
{
ci.new_track = 0;
CUR_TI->taginfo_ready = false;
audio_rebuffer();
goto skip_done;
}
forward = ci.new_track > 0;
ci.new_track = 0;
/* If the target track is clearly not in memory */
if (CUR_TI->filesize == 0 || !CUR_TI->taginfo_ready)
{
audio_rebuffer();
goto skip_done;
}
/* The track may be in memory, see if it really is */
if (forward)
{
if (!audio_buffer_wind_forward(track_ridx, old_track_ridx))
audio_rebuffer();
}
else
{
int cur_idx = track_ridx;
bool taginfo_ready = true;
bool wrap = track_ridx > old_track_ridx;
while (1)
{
cur_idx++;
cur_idx &= MAX_TRACK_MASK;
if (!(wrap || cur_idx < old_track_ridx))
break;
/* If we hit a track in between without valid tag info, bail */
if (!tracks[cur_idx].taginfo_ready)
{
taginfo_ready = false;
break;
}
tracks[cur_idx].available = tracks[cur_idx].filesize;
if (tracks[cur_idx].codecsize)
tracks[cur_idx].has_codec = true;
}
if (taginfo_ready)
{
if (!audio_buffer_wind_backward(track_ridx, old_track_ridx))
audio_rebuffer();
}
else
{
CUR_TI->taginfo_ready = false;
audio_rebuffer();
}
}
skip_done:
audio_update_trackinfo();
LOGFQUEUE("audio >|= codec Q_CODEC_REQUEST_COMPLETE");
return Q_CODEC_REQUEST_COMPLETE;
}
static int audio_rebuffer_and_seek(size_t newpos)
{
size_t real_preseek;
int fd;
char *trackname;
/* (Re-)open current track's file handle. */
trackname = playlist_peek(0);
fd = open(trackname, O_RDONLY);
if (fd < 0)
{
LOGFQUEUE("audio >|= codec Q_CODEC_REQUEST_FAILED");
return Q_CODEC_REQUEST_FAILED;
}
if (current_fd >= 0)
close(current_fd);
current_fd = fd;
playlist_end = false;
ci.curpos = newpos;
/* Clear codec buffer. */
track_widx = track_ridx;
tracks[track_widx].buf_idx = buf_widx = buf_ridx = 0;
last_peek_offset = 0;
filling = false;
audio_initialize_buffer_fill(true);
/* This may have been tweaked by the id3v1 code */
CUR_TI->filesize=filesize(fd);
if (newpos > conf_preseek)
{
CUR_TI->start_pos = newpos - conf_preseek;
lseek(current_fd, CUR_TI->start_pos, SEEK_SET);
CUR_TI->filerem = CUR_TI->filesize - CUR_TI->start_pos;
real_preseek = conf_preseek;
}
else
{
CUR_TI->start_pos = 0;
CUR_TI->filerem = CUR_TI->filesize;
real_preseek = newpos;
}
CUR_TI->available = 0;
audio_read_file(real_preseek);
/* Account for the data we just read that is 'behind' us now */
CUR_TI->available -= real_preseek;
buf_ridx = RINGBUF_ADD(buf_ridx, real_preseek);
LOGFQUEUE("audio >|= codec Q_CODEC_REQUEST_COMPLETE");
return Q_CODEC_REQUEST_COMPLETE;
}
void audio_set_track_buffer_event(void (*handler)(struct mp3entry *id3,
bool last_track))
{
track_buffer_callback = handler;
}
void audio_set_track_unbuffer_event(void (*handler)(struct mp3entry *id3,
bool last_track))
{
track_unbuffer_callback = handler;
}
void audio_set_track_changed_event(void (*handler)(struct mp3entry *id3))
{
track_changed_callback = handler;
}
unsigned long audio_prev_elapsed(void)
{
return prev_track_elapsed;
}
static void audio_stop_codec_flush(void)
{
ci.stop_codec = true;
pcmbuf_pause(true);
while (audio_codec_loaded)
yield();
/* If the audio codec is not loaded any more, and the audio is still
* playing, it is now and _only_ now safe to call this function from the
* audio thread */
if (pcm_is_playing())
pcmbuf_play_stop();
pcmbuf_pause(paused);
}
static void audio_stop_playback(void)
{
/* If we were playing, save resume information */
if (playing)
{
struct mp3entry *id3 = NULL;
if (!playlist_end || !ci.stop_codec)
{
/* Set this early, the outside code yields and may allow the codec
to try to wait for a reply on a buffer wait */
ci.stop_codec = true;
id3 = audio_current_track();
}
/* Save the current playing spot, or NULL if the playlist has ended */
playlist_update_resume_info(id3);
prev_track_elapsed = CUR_TI->id3.elapsed;
/* Increment index so runtime info is saved in audio_clear_track_entries().
* Done here, as audio_stop_playback() may be called more than once.
* Don't update runtime unless playback is stopped because of end of playlist.
* Updating runtime when manually stopping a tracks, can destroy autoscores
* and playcounts.
*/
if (playlist_end)
{
track_ridx++;
track_ridx &= MAX_TRACK_MASK;
}
}
filling = false;
paused = false;
audio_stop_codec_flush();
playing = false;
if (current_fd >= 0)
{
close(current_fd);
current_fd = -1;
}
/* Mark all entries null. */
audio_clear_track_entries(false);
}
static void audio_play_start(size_t offset)
{
#if INPUT_SRC_CAPS != 0
audio_set_input_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
audio_set_output_source(AUDIO_SRC_PLAYBACK);
#endif
/* Wait for any previously playing audio to flush - TODO: Not necessary? */
paused = false;
audio_stop_codec_flush();
track_changed = true;
playlist_end = false;
playing = true;
ci.new_track = 0;
ci.seek_time = 0;
wps_offset = 0;
if (current_fd >= 0)
{
close(current_fd);
current_fd = -1;
}
sound_set_volume(global_settings.volume);
track_widx = track_ridx = 0;
buf_ridx = buf_widx = 0;
/* Mark all entries null. */
memset(tracks, 0, sizeof(struct track_info) * MAX_TRACK);
last_peek_offset = -1;
/* Officially playing */
queue_reply(&audio_queue, 1);
audio_fill_file_buffer(true, false, offset);
LOGFQUEUE("audio > audio Q_AUDIO_TRACK_CHANGED");
queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0);
}
/* Invalidates all but currently playing track. */
static void audio_invalidate_tracks(void)
{
if (audio_have_tracks())
{
last_peek_offset = 0;
playlist_end = false;
track_widx = track_ridx;
/* Mark all other entries null (also buffered wrong metadata). */
audio_clear_track_entries(true);
/* If the current track is fully buffered, advance the write pointer */
if (tracks[track_widx].filerem == 0)
track_widx = (track_widx + 1) & MAX_TRACK_MASK;
buf_widx = RINGBUF_ADD(buf_ridx, CUR_TI->available);
audio_read_next_metadata();
}
}
static void audio_new_playlist(void)
{
/* Prepare to start a new fill from the beginning of the playlist */
last_peek_offset = -1;
if (audio_have_tracks())
{
if (paused)
skipped_during_pause = true;
playlist_end = false;
track_widx = track_ridx;
audio_clear_track_entries(true);
track_widx++;
track_widx &= MAX_TRACK_MASK;
/* Stop reading the current track */
CUR_TI->filerem = 0;
close(current_fd);
current_fd = -1;
/* Mark the current track as invalid to prevent skipping back to it */
CUR_TI->taginfo_ready = false;
/* Invalidate the buffer other than the playing track */
buf_widx = RINGBUF_ADD(buf_ridx, CUR_TI->available);
}
/* Signal the codec to initiate a track change forward */
new_playlist = true;
ci.new_track = 1;
/* Officially playing */
queue_reply(&audio_queue, 1);
audio_fill_file_buffer(false, true, 0);
}
static void audio_initiate_track_change(long direction)
{
playlist_end = false;
ci.new_track += direction;
wps_offset -= direction;
if (paused)
skipped_during_pause = true;
}
static void audio_initiate_dir_change(long direction)
{
playlist_end = false;
dir_skip = true;
ci.new_track = direction;
if (paused)
skipped_during_pause = true;
}
/*
* Layout audio buffer as follows - iram buffer depends on target:
* [|SWAP:iram][|TALK]|MALLOC|FILE|GUARD|PCM|[SWAP:dram[|iram]|]
*/
static void audio_reset_buffer(void)
{
/* see audio_get_recording_buffer if this is modified */
logf("audio_reset_buffer");
/* 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 */
malloc_buf = audiobuf + talk_get_bufsize();
/* Align the malloc buf to line size. Especially important to cf
targets that do line reads/writes. */
malloc_buf = (unsigned char *)(((uintptr_t)malloc_buf + 15) & ~15);
filebuf = malloc_buf + MALLOC_BUFSIZE; /* filebuf line align implied */
filebuflen = audiobufend - filebuf;
/* Allow for codec swap space at end of audio buffer */
if (talk_voice_required())
{
/* Layout of swap buffer:
* #ifdef IRAM_STEAL (dedicated iram_buf):
* |iram_buf|...audiobuf...|dram_buf|audiobufend
* #else:
* audiobuf...|dram_buf|iram_buf|audiobufend
*/
#ifdef PLAYBACK_VOICE
/* Check for an absolutely nasty situation which should never,
ever happen - frankly should just panic */
if (voice_codec_loaded && current_codec != CODEC_IDX_VOICE)
{
logf("buffer reset with voice swapped");
}
/* line align length which line aligns the calculations below since
all sizes are also at least line aligned - needed for memswap128 */
filebuflen &= ~15;
#ifdef IRAM_STEAL
filebuflen -= CODEC_SIZE;
#else
filebuflen -= CODEC_SIZE + CODEC_IRAM_SIZE;
#endif
/* Allocate buffers for swapping voice <=> audio */
/* If using IRAM for plugins voice IRAM swap buffer must be dedicated
and out of the way of buffer usage or else a call to audio_get_buffer
and subsequent buffer use might trash the swap space. A plugin
initializing IRAM after getting the full buffer would present similar
problem. Options include: failing the request if the other buffer
has been obtained already or never allowing use of the voice IRAM
buffer within the audio buffer. Using buffer_alloc basically
implements the second in a more convenient way. */
dram_buf = filebuf + filebuflen;
#ifdef IRAM_STEAL
/* Allocate voice IRAM swap buffer once */
if (iram_buf == NULL)
{
iram_buf = buffer_alloc(CODEC_IRAM_SIZE);
/* buffer_alloc moves audiobuf; this is safe because only the end
* has been touched so far in this function and the address of
* filebuf + filebuflen is not changed */
malloc_buf += CODEC_IRAM_SIZE;
filebuf += CODEC_IRAM_SIZE;
filebuflen -= CODEC_IRAM_SIZE;
}
#else
/* Allocate iram_buf after dram_buf */
iram_buf = dram_buf + CODEC_SIZE;
#endif /* IRAM_STEAL */
#endif /* PLAYBACK_VOICE */
}
else
{
#ifdef PLAYBACK_VOICE
/* No swap buffers needed */
iram_buf = NULL;
dram_buf = NULL;
#endif
}
/* Subtract whatever the pcm buffer says it used plus the guard buffer */
filebuflen -= pcmbuf_init(filebuf + filebuflen) + GUARD_BUFSIZE;
/* Make sure filebuflen is a longword multiple after adjustment - filebuf
will already be line aligned */
filebuflen &= ~3;
/* Set the high watermark as 75% full...or 25% empty :) */
#if MEM > 8
high_watermark = 3*filebuflen / 4;
#endif
/* Clear any references to the file buffer */
buffer_state = BUFFER_STATE_INITIALIZED;
#ifdef ROCKBOX_HAS_LOGF
/* Make sure everything adds up - yes, some info is a bit redundant but
aids viewing and the sumation of certain variables should add up to
the location of others. */
{
size_t pcmbufsize;
unsigned char * pcmbuf = pcmbuf_get_meminfo(&pcmbufsize);
logf("mabuf: %08X", (unsigned)malloc_buf);
logf("mabufe: %08X", (unsigned)(malloc_buf + MALLOC_BUFSIZE));
logf("fbuf: %08X", (unsigned)filebuf);
logf("fbufe: %08X", (unsigned)(filebuf + filebuflen));
logf("gbuf: %08X", (unsigned)(filebuf + filebuflen));
logf("gbufe: %08X", (unsigned)(filebuf + filebuflen + GUARD_BUFSIZE));
logf("pcmb: %08X", (unsigned)pcmbuf);
logf("pcmbe: %08X", (unsigned)(pcmbuf + pcmbufsize));
if (dram_buf)
{
logf("dramb: %08X", (unsigned)dram_buf);
logf("drambe: %08X", (unsigned)(dram_buf + CODEC_SIZE));
}
if (iram_buf)
{
logf("iramb: %08X", (unsigned)iram_buf);
logf("irambe: %08X", (unsigned)(iram_buf + CODEC_IRAM_SIZE));
}
}
#endif
}
#if MEM > 8
/* we dont want this rebuffering on targets with little ram
because the disk may never spin down */
static bool ata_fillbuffer_callback(void)
{
queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA, 0);
return true;
}
#endif
static void audio_thread(void)
{
struct event ev;
pcm_postinit();
#ifdef PLAYBACK_VOICE
/* Unlock mutex that init stage locks before creating this thread */
mutex_unlock(&mutex_codecthread);
/* Buffers must be set up by now - should panic - really */
if (buffer_state != BUFFER_STATE_INITIALIZED)
{
logf("audio_thread start: no buffer");
}
/* Have to wait for voice to load up or else the codec swap will be
invalid when an audio codec is loaded */
wait_for_voice_swap_in();
#endif
while (1)
{
intptr_t result = 0;
if (filling)
{
queue_wait_w_tmo(&audio_queue, &ev, 0);
if (ev.id == SYS_TIMEOUT)
ev.id = Q_AUDIO_FILL_BUFFER;
}
else
{
queue_wait_w_tmo(&audio_queue, &ev, HZ/2);
#if MEM > 8
if (playing && (ev.id == SYS_TIMEOUT) &&
(FILEBUFUSED < high_watermark))
register_ata_idle_func(ata_fillbuffer_callback);
#endif
}
switch (ev.id) {
#if MEM > 8
case Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA:
/* only fill if the disk is still spining */
#ifndef SIMULATOR
if (!ata_disk_is_active())
break;
#endif
#endif /* MEM > 8 */
/* else fall through to Q_AUDIO_FILL_BUFFER */
case Q_AUDIO_FILL_BUFFER:
LOGFQUEUE("audio < Q_AUDIO_FILL_BUFFER");
if (!filling)
if (!playing || playlist_end || ci.stop_codec)
break;
audio_fill_file_buffer(false, false, 0);
break;
case Q_AUDIO_PLAY:
LOGFQUEUE("audio < Q_AUDIO_PLAY");
if (playing && ev.data <= 0)
audio_new_playlist();
else
{
audio_stop_playback();
audio_play_start((size_t)ev.data);
}
break;
case Q_AUDIO_STOP:
LOGFQUEUE("audio < Q_AUDIO_STOP");
if (playing)
audio_stop_playback();
if (ev.data != 0)
queue_clear(&audio_queue);
break;
case Q_AUDIO_PAUSE:
LOGFQUEUE("audio < Q_AUDIO_PAUSE");
if (!(bool) ev.data && skipped_during_pause && !pcmbuf_is_crossfade_active())
pcmbuf_play_stop(); /* Flush old track on resume after skip */
skipped_during_pause = false;
if (!playing)
break;
pcmbuf_pause((bool)ev.data);
paused = (bool)ev.data;
break;
case Q_AUDIO_SKIP:
LOGFQUEUE("audio < Q_AUDIO_SKIP");
audio_initiate_track_change((long)ev.data);
break;
case Q_AUDIO_PRE_FF_REWIND:
LOGFQUEUE("audio < Q_AUDIO_PRE_FF_REWIND");
if (!playing)
break;
pcmbuf_pause(true);
break;
case Q_AUDIO_FF_REWIND:
LOGFQUEUE("audio < Q_AUDIO_FF_REWIND");
if (!playing)
break;
ci.seek_time = (long)ev.data+1;
break;
case Q_AUDIO_REBUFFER_SEEK:
LOGFQUEUE("audio < Q_AUDIO_REBUFFER_SEEK");
result = audio_rebuffer_and_seek(ev.data);
break;
case Q_AUDIO_CHECK_NEW_TRACK:
LOGFQUEUE("audio < Q_AUDIO_CHECK_NEW_TRACK");
result = audio_check_new_track();
break;
case Q_AUDIO_DIR_SKIP:
LOGFQUEUE("audio < Q_AUDIO_DIR_SKIP");
playlist_end = false;
audio_initiate_dir_change(ev.data);
break;
case Q_AUDIO_FLUSH:
LOGFQUEUE("audio < Q_AUDIO_FLUSH");
audio_invalidate_tracks();
break;
case Q_AUDIO_TRACK_CHANGED:
LOGFQUEUE("audio < Q_AUDIO_TRACK_CHANGED");
if (track_changed_callback)
track_changed_callback(&CUR_TI->id3);
track_changed = true;
playlist_update_resume_info(audio_current_track());
break;
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
LOGFQUEUE("audio < SYS_USB_CONNECTED");
if (playing)
audio_stop_playback();
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&audio_queue);
break;
#endif
case SYS_TIMEOUT:
LOGFQUEUE_SYS_TIMEOUT("audio < SYS_TIMEOUT");
break;
default:
LOGFQUEUE("audio < default");
} /* end switch */
queue_reply(&audio_queue, result);
} /* end while */
}
#ifdef ROCKBOX_HAS_LOGF
static void audio_test_track_changed_event(struct mp3entry *id3)
{
(void)id3;
logf("tce:%s", id3->path);
}
#endif
/* Initialize the audio system - called from init() in main.c.
* Last function because of all the references to internal symbols
*/
void audio_init(void)
{
#ifdef PLAYBACK_VOICE
static bool voicetagtrue = true;
static struct mp3entry id3_voice;
#endif
/* 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. */
#ifdef PLAYBACK_VOICE
mutex_init(&mutex_codecthread);
/* Take ownership of lock to prevent playback of anything before audio
hardware is initialized - audio thread unlocks it after final init
stage */
mutex_lock(&mutex_codecthread);
#endif
queue_init(&audio_queue, true);
queue_enable_queue_send(&audio_queue, &audio_queue_sender_list);
queue_init(&codec_queue, true);
pcm_init();
#ifdef ROCKBOX_HAS_LOGF
audio_set_track_changed_event(audio_test_track_changed_event);
#endif
/* Initialize codec api. */
ci.read_filebuf = codec_filebuf_callback;
ci.pcmbuf_insert = codec_pcmbuf_insert_callback;
ci.get_codec_memory = codec_get_memory_callback;
ci.request_buffer = codec_request_buffer_callback;
ci.advance_buffer = codec_advance_buffer_callback;
ci.advance_buffer_loc = codec_advance_buffer_loc_callback;
ci.request_next_track = codec_request_next_track_callback;
ci.mp3_get_filepos = codec_mp3_get_filepos_callback;
ci.seek_buffer = codec_seek_buffer_callback;
ci.seek_complete = codec_seek_complete_callback;
ci.set_elapsed = codec_set_elapsed_callback;
ci.set_offset = codec_set_offset_callback;
ci.configure = codec_configure_callback;
ci.discard_codec = codec_discard_codec_callback;
/* Initialize voice codec api. */
#ifdef PLAYBACK_VOICE
memcpy(&ci_voice, &ci, sizeof(ci_voice));
memset(&id3_voice, 0, sizeof(id3_voice));
ci_voice.read_filebuf = voice_filebuf_callback;
ci_voice.pcmbuf_insert = voice_pcmbuf_insert_callback;
ci_voice.get_codec_memory = voice_get_memory_callback;
ci_voice.request_buffer = voice_request_buffer_callback;
ci_voice.advance_buffer = voice_advance_buffer_callback;
ci_voice.advance_buffer_loc = voice_advance_buffer_loc_callback;
ci_voice.request_next_track = voice_request_next_track_callback;
ci_voice.mp3_get_filepos = voice_mp3_get_filepos_callback;
ci_voice.seek_buffer = voice_seek_buffer_callback;
ci_voice.seek_complete = voice_do_nothing;
ci_voice.set_elapsed = voice_set_elapsed_callback;
ci_voice.set_offset = voice_set_offset_callback;
ci_voice.configure = voice_configure_callback;
ci_voice.discard_codec = voice_do_nothing;
ci_voice.taginfo_ready = &voicetagtrue;
ci_voice.id3 = &id3_voice;
id3_voice.frequency = 11200;
id3_voice.length = 1000000L;
#endif
/* initialize the buffer */
filebuf = audiobuf;
/* audio_reset_buffer must to know the size of voice buffer so init
talk first */
talk_init();
/* Create the threads late now that we shouldn't be yielding again before
returning */
codec_thread_p = create_thread(
codec_thread, codec_stack, sizeof(codec_stack),
codec_thread_name IF_PRIO(, PRIORITY_PLAYBACK)
IF_COP(, CPU, true));
create_thread(audio_thread, audio_stack, sizeof(audio_stack),
audio_thread_name IF_PRIO(, PRIORITY_BUFFERING)
IF_COP(, CPU, false));
#ifdef PLAYBACK_VOICE
/* TODO: Change this around when various speech codecs can be used */
if (talk_voice_required())
{
logf("Starting voice codec");
queue_init(&voice_queue, true);
create_thread(voice_thread, voice_stack,
sizeof(voice_stack), voice_thread_name
IF_PRIO(, PRIORITY_PLAYBACK) IF_COP(, CPU, false));
}
#endif
/* Set crossfade setting for next buffer init which should be about... */
pcmbuf_crossfade_enable(global_settings.crossfade);
/* ...now! Set up the buffers */
audio_reset_buffer();
/* Probably safe to say */
audio_is_initialized = true;
sound_settings_apply();
#ifdef HAVE_WM8758
eq_hw_enable(global_settings.eq_hw_enabled);
#endif
#ifndef HAVE_FLASH_STORAGE
audio_set_buffer_margin(global_settings.buffer_margin);
#endif
} /* audio_init */
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