rendered paste body/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id: playback.c 12125 2007-01-27 16:33:44Z lostlogic $
*
* 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 "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 "main_menu.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"
#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
#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(s) logf("%s", s)
#else
#define LOGFQUEUE(s)
#endif
#ifdef PLAYBACK_LOGQUEUES_SYS_TIMEOUT
#define LOGFQUEUE_SYS_TIMEOUT(s) logf("%s", s)
#else
#define LOGFQUEUE_SYS_TIMEOUT(s)
#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_NEW_PLAYLIST,
Q_AUDIO_POSTINIT,
Q_AUDIO_FILL_BUFFER,
#if MEM > 8
Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA,
#endif
#ifdef AUDIO_HAVE_RECORDING
Q_AUDIO_LOAD_ENCODER,
#endif
#if 0
Q_CODEC_REQUEST_PENDING,
#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 0x4000c000
#define CODEC_IRAM_SIZE 0xc000
#elif defined(IAUDIO_X5)
#define CODEC_IRAM_ORIGIN 0x10010000
#define CODEC_IRAM_SIZE 0x10000
#else
#define CODEC_IRAM_ORIGIN 0x1000c000
#define CODEC_IRAM_SIZE 0xc000
#endif
#ifndef IBSS_ATTR_VOICE_STACK
#define IBSS_ATTR_VOICE_STACK IBSS_ATTR
#endif
#ifndef SIMULATOR
extern bool audio_is_initialized;
#else
static bool audio_is_initialized = false;
#endif
/* 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; /* Is codec loaded? (C/A-) */
static volatile bool playing; /* Is audio playing? (A) */
static volatile bool paused; /* Is audio paused? (A/C-) */
static volatile bool filling IDATA_ATTR; /* Is file buffer refilling? (A/C-) */
/* Ring buffer where compressed audio and codecs are loaded */
static unsigned char *filebuf; /* Start of buffer (A/C-) */
/* FIXME: make filebuflen static */
size_t filebuflen; /* Size of buffer (A/C-) */
/* FIXME: make buf_ridx (C/A-) */
static volatile size_t buf_ridx IDATA_ATTR; /* Buffer read position (A/C)*/
static volatile size_t buf_widx IDATA_ATTR; /* Buffer write position (A/C-) */
/* Possible arrangements of the buffer */
#define BUFFER_STATE_TRASHED -1 /* trashed; must be reset */
#define BUFFER_STATE_NORMAL 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; /* Track being decoded (A/C-) */
static int track_widx; /* Track being buffered (A) */
static struct track_info *prev_ti; /* 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;
/* Information used only for filling the buffer */
/* Playlist steps from playing track to next track to be buffered (A) */
static int last_peek_offset;
/* Partially loaded track file handle to continue buffering (A) */
static int current_fd;
/* Scrobbler support */
static unsigned long prev_track_elapsed; /* 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) */
/* Pending track change offset, to keep WPS responsive (A) */
static int wps_offset = 0;
/* 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);
/* When a track has been buffered */
void (*track_buffer_callback)(struct mp3entry *id3, bool last_track);
/* When a track's buffer has been overwritten or cleared */
void (*track_unbuffer_callback)(struct mp3entry *id3, bool last_track);
/* Configuration */
static size_t conf_watermark; /* Level to trigger filebuf fill (A/C) FIXME */
static size_t conf_filechunk; /* Largest chunk the codec accepts (A/C) FIXME */
static size_t conf_preseek; /* Codec pre-seek margin (A/C) FIXME */
static size_t buffer_margin; /* 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; /* High watermark for rebuffer (A/V/other) */
#endif
/* Multiple threads */
static const char *get_codec_filename(int enc_spec); /* (A-/C-/V-) */
/* 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(size_t pcmbufsize);
/* Codec thread */
extern struct codec_api ci;
static struct event_queue codec_queue;
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. */
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;
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
/* Pointer to IRAM buffers for normal/voice codecs */
static unsigned char *iram_buf[2] = { NULL, NULL };
/* Pointer to DRAM buffers for normal/voice codecs */
static unsigned char *dram_buf[2] = { NULL, NULL };
/* Mutex to control which codec (normal/voice) is running */
static struct mutex mutex_codecthread;
/* Voice state */
static volatile bool voice_thread_start; /* Triggers voice playback (A/V) */
static volatile bool voice_is_playing; /* Is voice currently playing? (V) */
static volatile bool voice_codec_loaded; /* Is voice codec loaded (V/A-) */
static char *voicebuf;
static size_t voice_remaining;
#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, int* size);
struct voice_info {
void (*callback)(unsigned char **start, int *size);
int size;
char *buf;
};
static void voice_thread(void);
#endif /* PLAYBACK_VOICE */
/* --- External interfaces --- */
void mp3_play_data(const unsigned char* start, int size,
void (*get_more)(unsigned char** start, int* size))
{
#ifdef PLAYBACK_VOICE
static struct voice_info voice_clip;
voice_clip.callback = get_more;
voice_clip.buf = (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
}
bool mp3_pause_done(void)
{
return pcm_is_paused();
}
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[CODEC_IDX_VOICE])
return;
while (current_codec != CODEC_IDX_VOICE)
yield();
#endif /* PLAYBACK_VOICE */
}
unsigned char *audio_get_buffer(bool talk_buf, size_t *buffer_size)
{
unsigned char *buf, *end;
if (audio_is_initialized)
{
audio_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 */
buffer_state = BUFFER_STATE_TRASHED;
return NULL;
}
if (talk_buf || buffer_state == BUFFER_STATE_TRASHED
|| !talk_voice_required())
{
logf("get buffer: talk_buf");
/* ok to use everything from audiobuf to audiobufend */
if (buffer_state != BUFFER_STATE_TRASHED)
{
talk_buffer_steal();
buffer_state = BUFFER_STATE_TRASHED;
}
buf = audiobuf;
end = audiobufend;
}
else
{
/* skip talk buffer and move pcm buffer to end */
logf("get buffer: voice");
buf = audiobuf + talk_get_bufsize();
end = audiobufend - pcmbuf_init(pcmbuf_get_bufsize(), 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_stop();
#ifdef PLAYBACK_VOICE
if (NULL != iram_buf[CODEC_IDX_VOICE])
{
/* Can't already be stolen */
if (voice_iram_stolen)
return;
wait_for_voice_swap_in();
voice_stop();
/* Save voice IRAM - safe to do here since state is known */
memcpy(iram_buf[CODEC_IDX_VOICE], (void *)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;
audio_stop();
wait_for_voice_swap_in();
voice_stop();
talk_buffer_steal();
#ifdef PLAYBACK_VOICE
#ifdef IRAM_STEAL
end = dram_buf[CODEC_IDX_VOICE];
#else
end = iram_buf[CODEC_IDX_VOICE];
#endif /* IRAM_STEAL */
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("audio > Q_AUDIO_LOAD_ENCODER");
queue_post(&audio_queue, Q_AUDIO_LOAD_ENCODER, (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_codec = 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(offset);
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 */
LOGFQUEUE("mp3 > voice Q_VOICE_STOP");
queue_post(&voice_queue, Q_VOICE_STOP, 1);
#endif
/* Start playback */
if (playing && offset <= 0)
{
LOGFQUEUE("audio > audio Q_AUDIO_NEW_PLAYLIST");
queue_post(&audio_queue, Q_AUDIO_NEW_PLAYLIST, 0);
}
else
{
LOGFQUEUE("audio > audio Q_AUDIO_STOP");
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
LOGFQUEUE("audio > audio Q_AUDIO_PLAY");
queue_post(&audio_queue, Q_AUDIO_PLAY, offset);
}
/* Don't return until playback has actually started */
while (!playing)
yield();
}
void audio_stop(void)
{
/* Stop playback */
LOGFQUEUE("audio > audio Q_AUDIO_STOP");
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
/* Don't return until playback has actually stopped */
while(playing || !queue_empty(&audio_queue))
yield();
}
void audio_pause(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_PAUSE");
queue_post(&audio_queue, Q_AUDIO_PAUSE, true);
}
void audio_resume(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_PAUSE resume");
queue_post(&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;
}
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: %ds", buffer_margin);
set_filebuf_watermark(buffer_margin);
}
/* Set crossfade & PCM buffer length. */
void audio_set_crossfade(int enable)
{
size_t size;
bool was_playing = (playing && audio_is_initialized);
size_t offset = 0;
#if MEM > 1
int seconds = 1;
#endif
if (!filebuf)
return; /* Audio buffers not yet set up */
#if MEM > 1
if (enable)
seconds = global_settings.crossfade_fade_out_delay
+ global_settings.crossfade_fade_out_duration;
/* Buffer has to be at least 2s long. */
seconds += 2;
logf("buf len: %d", seconds);
size = seconds * (NATIVE_FREQUENCY*4);
#else
enable = 0;
size = NATIVE_FREQUENCY*2;
#endif
if (buffer_state == BUFFER_STATE_NORMAL && pcmbuf_is_same_size(size))
return ;
if (was_playing)
{
/* Store the track resume position */
offset = CUR_TI->id3.offset;
/* Playback has to be stopped before changing the buffer size. */
gui_syncsplash(0, true, (char *)str(LANG_RESTARTING_PLAYBACK));
audio_stop();
}
voice_stop();
/* Re-initialize audio system. */
audio_reset_buffer(size);
pcmbuf_crossfade_enable(enable);
logf("abuf:%dB", pcmbuf_get_bufsize());
logf("fbuf:%dB", filebuflen);
voice_init();
/* Restart playback. */
if (was_playing)
audio_play(offset);
}
void audio_preinit(void)
{
logf("playback system pre-init");
filling = false;
current_codec = CODEC_IDX_AUDIO;
playing = false;
paused = false;
audio_codec_loaded = false;
#ifdef PLAYBACK_VOICE
voice_is_playing = false;
voice_thread_start = false;
voice_codec_loaded = false;
#endif
track_changed = false;
current_fd = -1;
track_buffer_callback = NULL;
track_unbuffer_callback = NULL;
track_changed_callback = NULL;
track_ridx = 0; /* Just to prevent CUR_TI from being anything random. */
prev_ti = &tracks[MAX_TRACK-1]; /* And prevent prev_ti being random too */
#ifdef PLAYBACK_VOICE
mutex_init(&mutex_codecthread);
#endif
queue_init(&audio_queue, true);
queue_enable_queue_send(&audio_queue, &audio_queue_sender_list);
queue_init(&codec_queue, true);
create_thread(audio_thread, audio_stack, sizeof(audio_stack),
audio_thread_name IF_PRIO(, PRIORITY_BUFFERING));
}
void audio_init(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_POSTINIT");
queue_post(&audio_queue, Q_AUDIO_POSTINIT, 0);
}
void voice_init(void)
{
#ifdef PLAYBACK_VOICE
if (!filebuf)
return; /* Audio buffers not yet set up */
if (voice_thread_p)
return;
if (!talk_voice_required())
return;
logf("Starting voice codec");
queue_init(&voice_queue, true);
voice_thread_p = create_thread(voice_thread, voice_stack,
sizeof(voice_stack), voice_thread_name
IF_PRIO(, PRIORITY_PLAYBACK));
while (!voice_codec_loaded)
yield();
#endif
} /* voice_init */
void voice_stop(void)
{
#ifdef PLAYBACK_VOICE
/* Messages should not be posted to voice codec queue unless it is the
current codec or deadlocks happen. */
if (current_codec != CODEC_IDX_VOICE)
return;
LOGFQUEUE("mp3 > voice Q_VOICE_STOP");
queue_post(&voice_queue, Q_VOICE_STOP, 0);
while (voice_is_playing || !queue_empty(&voice_queue))
yield();
if (!playing)
pcmbuf_play_stop();
#endif
} /* voice_stop */
/* --- Routines called from multiple threads --- */
#ifdef PLAYBACK_VOICE
static void swap_codec(void)
{
int my_codec = current_codec;
logf("swapping out codec:%d", my_codec);
/* 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 - _should_ always be the case
if voice_iram_stolen is true since the voice has been swapped in
before hand */
if (my_codec == CODEC_IDX_VOICE)
goto skip_iram_swap;
}
#endif
memcpy(iram_buf[my_codec], (unsigned char *)CODEC_IRAM_ORIGIN,
CODEC_IRAM_SIZE);
#ifdef IRAM_STEAL
skip_iram_swap:
#endif
memcpy(dram_buf[my_codec], codecbuf, CODEC_SIZE);
/* Release my semaphore */
mutex_unlock(&mutex_codecthread);
/* 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 */
current_codec = my_codec;
/* Reload our IRAM and DRAM */
memcpy((unsigned char *)CODEC_IRAM_ORIGIN, iram_buf[my_codec],
CODEC_IRAM_SIZE);
invalidate_icache();
memcpy(codecbuf, dram_buf[my_codec], CODEC_SIZE);
logf("resuming codec:%d", my_codec);
}
#endif
static void set_filebuf_watermark(int seconds)
{
size_t bytes;
if (current_codec == CODEC_IDX_VOICE)
return;
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;
}
static 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_split_callback(
const void *ch1, const void *ch2, size_t length)
{
const char* src[2];
char *dest;
long input_size;
size_t output_size;
src[0] = ch1;
src[1] = ch2;
if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
length *= 2; /* Length is per channel */
while (length)
{
long est_output_size = dsp_output_size(length);
while ((dest = pcmbuf_request_voice_buffer(est_output_size,
&output_size, 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. */
input_size = dsp_input_size(output_size);
if (input_size <= 0)
{
DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld<=0\n",
output_size, length, input_size);
/* If this happens, there are samples of codec data that don't
* become a number of pcm samples, and something is broken */
return false;
}
/* Input size has grown, no error, just don't write more than length */
if ((size_t)input_size > length)
input_size = length;
output_size = dsp_process(dest, src, input_size);
if (playing)
{
pcmbuf_mix_voice(output_size);
if ((pcmbuf_usage() < 10 || pcmbuf_mix_free() < 30) &&
audio_codec_loaded)
swap_codec();
}
else
pcmbuf_write_complete(output_size);
length -= input_size;
}
return true;
} /* voice_pcmbuf_insert_split_callback */
static bool voice_pcmbuf_insert_callback(const char *buf, size_t length)
{
/* TODO: The audiobuffer API should probably be updated, and be based on
* pcmbuf_insert_split(). */
long real_length = length;
if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
length /= 2; /* Length is per channel */
/* Second channel is only used for non-interleaved stereo. */
return voice_pcmbuf_insert_split_callback(buf, buf + (real_length / 2),
length);
}
static void* voice_get_memory_callback(size_t *size)
{
*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 size_t voice_filebuf_callback(void *ptr, size_t size)
{
(void)ptr;
(void)size;
return 0;
}
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);
else
queue_wait(&voice_queue, &ev);
if (!voice_is_playing)
{
if (ev.id == SYS_TIMEOUT)
ev.id = Q_AUDIO_PLAY;
}
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 (ev.data == 1 && !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;
/* Force the codec to think it's changing tracks */
ci_voice.new_track = 1;
*realsize = 0;
return NULL;
}
else
break;
case SYS_USB_CONNECTED:
LOGFQUEUE("voice < SYS_USB_CONNECTED");
usb_acknowledge(SYS_USB_CONNECTED_ACK);
if (audio_codec_loaded)
swap_codec();
usb_wait_for_disconnect(&voice_queue);
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)
{
logf("voice: iram restore");
memcpy((void*)CODEC_IRAM_ORIGIN,
iram_buf[CODEC_IDX_VOICE],
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(pcmbuf_get_bufsize());
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;
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, (int *)&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)
{
while (1)
{
logf("Loading voice codec");
voice_codec_loaded = true;
mutex_lock(&mutex_codecthread);
current_codec = CODEC_IDX_VOICE;
dsp_configure(DSP_RESET, 0);
voice_remaining = 0;
voice_getmore = NULL;
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 */
#endif /* PLAYBACK_VOICE */
/* --- Codec thread --- */
static bool codec_pcmbuf_insert_split_callback(
const void *ch1, const void *ch2, size_t length)
{
const char* src[2];
char *dest;
long input_size;
size_t output_size;
src[0] = ch1;
src[1] = ch2;
if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
length *= 2; /* Length is per channel */
while (length)
{
long est_output_size = dsp_output_size(length);
/* Prevent audio from a previous track from playing */
if (ci.new_track || ci.stop_codec)
return true;
while ((dest = pcmbuf_request_buffer(est_output_size,
&output_size)) == 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. */
input_size = dsp_input_size(output_size);
if (input_size <= 0)
{
DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld<=0\n",
output_size, length, input_size);
/* If this happens, there are samples of codec data that don't
* become a number of pcm samples, and something is broken */
return false;
}
/* Input size has grown, no error, just don't write more than length */
if ((size_t)input_size > length)
input_size = length;
output_size = dsp_process(dest, src, input_size);
pcmbuf_write_complete(output_size);
#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
length -= input_size;
}
return true;
} /* codec_pcmbuf_insert_split_callback */
static bool codec_pcmbuf_insert_callback(const char *buf, size_t length)
{
/* TODO: The audiobuffer API should probably be updated, and be based on
* pcmbuf_insert_split(). */
long real_length = length;
if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
length /= 2; /* Length is per channel */
/* Second channel is only used for non-interleaved stereo. */
return codec_pcmbuf_insert_split_callback(buf, buf + (real_length / 2),
length);
}
static void* codec_get_memory_callback(size_t *size)
{
*size = MALLOC_BUFSIZE;
return &audiobuf[talk_get_bufsize()];
}
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;
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, NULL);
/* 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;
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, void *value)
{
switch (setting) {
case CODEC_SET_FILEBUF_WATERMARK:
conf_watermark = (unsigned long)value;
set_filebuf_watermark(buffer_margin);
break;
case CODEC_SET_FILEBUF_CHUNKSIZE:
conf_filechunk = (unsigned long)value;
break;
case CODEC_SET_FILEBUF_PRESEEK:
conf_preseek = (unsigned long)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 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: */
pcmbuf_crossfade_init(false);
codec_track_changed();
}
else
{
/* shuffle mode is off, so do a gapless track change */
pcmbuf_set_position_callback(codec_pcmbuf_position_callback);
pcmbuf_set_event_handler(codec_pcmbuf_track_changed_callback);
}
}
else
{
/* normal crossfade: */
pcmbuf_crossfade_init(false);
codec_track_changed();
}
}
else
{
/* normal gapless playback. */
pcmbuf_set_position_callback(codec_pcmbuf_position_callback);
pcmbuf_set_event_handler(codec_pcmbuf_track_changed_callback);
}
}
static bool codec_load_next_track(void)
{
intptr_t result;
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;
}
trigger_cpu_boost();
LOGFQUEUE("codec >| audio Q_AUDIO_CHECK_NEW_TRACK");
result = queue_send(&audio_queue, Q_AUDIO_CHECK_NEW_TRACK, 0);
#if 0 /* Q_CODEC_REQUEST_PENDING never posted anyway */
while (1)
{
queue_wait(&codec_callback_queue, &ev);
if (ev.id == Q_CODEC_REQUEST_PENDING)
{
if (!automatic_skip)
pcmbuf_play_stop();
}
else
break;
}
#endif
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 not current */
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
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
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");
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
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, true, "Codec failure");
}
if (!codec_load_next_track())
{
// queue_post(&codec_queue, Q_AUDIO_STOP, 0);
LOGFQUEUE("codec > audio Q_AUDIO_STOP");
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");
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, true, "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 (%dB)", 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 %dB 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=%d, 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:%dB", 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:%dB", 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: %dB", 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;
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);
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],current_fd,trackname,v1first))
{
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;
}
}
/* 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, true, 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_FLAC:
case AFMT_PCM_WAV:
case AFMT_A52:
case AFMT_AAC:
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],fd,trackname,v1first);
/* Preload the glyphs in the tags */
if (status)
{
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);
}
/* Force WPS reload. */
track_changed = 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 */
if (buffer_state != BUFFER_STATE_NORMAL)
audio_reset_buffer(pcmbuf_get_bufsize());
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");
#if 0
/* Notify the codec that this will take a while */
/* Currently this can cause some problems (logf in reverse order):
* Codec load error:-1
* Codec load disk
* Codec: Unsupported
* Codec finished
* New codec:0/3
* Clearing tracks:7/7, 1
* Forcing rebuffer
* Check new track buffer
* Request new track
* Clearing tracks:5/5, 0
* Starting buffer fill
* Clearing tracks:5/5, 1
* Re-buffering song w/seek
*/
if (!filling)
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_PENDING, 0);
#endif
/* 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)
{
/* Save the current playing spot, or NULL if the playlist has ended */
playlist_update_resume_info(
(playlist_end && ci.stop_codec)?NULL:audio_current_track());
}
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 defined(HAVE_RECORDING) || defined(CONFIG_TUNER)
rec_set_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
#endif
/* Wait for any previously playing audio to flush - TODO: Not necessary? */
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;
audio_fill_file_buffer(true, false, offset);
}
/* 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()) {
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;
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;
}
static void audio_initiate_dir_change(long direction)
{
playlist_end = false;
dir_skip = true;
ci.new_track = direction;
}
/*
* Layout audio buffer as follows:
* [|TALK]|MALLOC|FILE|GUARD|PCM|AUDIOCODEC|[VOICECODEC|]
*/
static void audio_reset_buffer(size_t pcmbufsize)
{
/* see audio_get_recording_buffer if this is modified */
size_t offset;
logf("audio_reset_buffer");
logf(" size:%08X", pcmbufsize);
/* Initially set up file buffer as all space available */
filebuf = audiobuf + MALLOC_BUFSIZE + talk_get_bufsize();
filebuflen = audiobufend - filebuf;
/* Allow for codec(s) at end of audio buffer */
if (talk_voice_required())
{
#ifdef PLAYBACK_VOICE
#ifdef IRAM_STEAL
filebuflen -= CODEC_IRAM_SIZE + 2*CODEC_SIZE;
#else
filebuflen -= 2*(CODEC_IRAM_SIZE + CODEC_SIZE);
#endif
/* Allow 2 codecs at end of audio buffer */
/* 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. */
iram_buf[CODEC_IDX_AUDIO] = filebuf + filebuflen;
dram_buf[CODEC_IDX_AUDIO] = iram_buf[CODEC_IDX_AUDIO] + CODEC_IRAM_SIZE;
#ifdef IRAM_STEAL
/* Allocate voice IRAM swap buffer once */
if (iram_buf[CODEC_IDX_VOICE] == NULL)
{
iram_buf[CODEC_IDX_VOICE] = 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 */
filebuf += CODEC_IRAM_SIZE;
filebuflen -= CODEC_IRAM_SIZE;
}
dram_buf[CODEC_IDX_VOICE] = dram_buf[CODEC_IDX_AUDIO] + CODEC_SIZE;
#else
iram_buf[CODEC_IDX_VOICE] = dram_buf[CODEC_IDX_AUDIO] + CODEC_SIZE;
dram_buf[CODEC_IDX_VOICE] = iram_buf[CODEC_IDX_VOICE] + CODEC_IRAM_SIZE;
#endif /* IRAM_STEAL */
#endif /* PLAYBACK_VOICE */
}
else
{
#ifdef PLAYBACK_VOICE
/* Allow for 1 codec at end of audio buffer */
filebuflen -= CODEC_IRAM_SIZE + CODEC_SIZE;
iram_buf[CODEC_IDX_AUDIO] = filebuf + filebuflen;
dram_buf[CODEC_IDX_AUDIO] = iram_buf[CODEC_IDX_AUDIO] + CODEC_IRAM_SIZE;
iram_buf[CODEC_IDX_VOICE] = NULL;
dram_buf[CODEC_IDX_VOICE] = NULL;
#endif
}
filebuflen -= pcmbuf_init(pcmbufsize, filebuf + filebuflen) + GUARD_BUFSIZE;
/* Ensure that file buffer is aligned */
offset = -(size_t)filebuf & 3;
filebuf += offset;
filebuflen -= offset;
filebuflen &= ~3;
#if MEM > 8
high_watermark = (3*filebuflen)/4;
#endif
/* Clear any references to the file buffer */
buffer_state = BUFFER_STATE_NORMAL;
}
#ifdef ROCKBOX_HAS_LOGF
static void audio_test_track_changed_event(struct mp3entry *id3)
{
(void)id3;
logf("tce:%s", id3->path);
}
#endif
static void audio_playback_init(void)
{
#ifdef PLAYBACK_VOICE
static bool voicetagtrue = true;
static struct mp3entry id3_voice;
#endif
struct event ev;
logf("playback api init");
pcm_init();
#ifdef AUDIO_HAVE_RECORDING
rec_set_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
#endif
#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.pcmbuf_insert_split = codec_pcmbuf_insert_split_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(struct codec_api));
memset(&id3_voice, 0, sizeof(struct mp3entry));
ci_voice.read_filebuf = voice_filebuf_callback;
ci_voice.pcmbuf_insert = voice_pcmbuf_insert_callback;
ci_voice.pcmbuf_insert_split = voice_pcmbuf_insert_split_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.discard_codec = voice_do_nothing;
ci_voice.taginfo_ready = &voicetagtrue;
ci_voice.id3 = &id3_voice;
id3_voice.frequency = 11200;
id3_voice.length = 1000000L;
#endif
codec_thread_p = create_thread(
codec_thread, codec_stack, sizeof(codec_stack),
codec_thread_name IF_PRIO(, PRIORITY_PLAYBACK));
while (1)
{
queue_wait(&audio_queue, &ev);
if (ev.id == Q_AUDIO_POSTINIT)
break ;
#ifndef SIMULATOR
if (ev.id == SYS_USB_CONNECTED)
{
logf("USB: Audio preinit");
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&audio_queue);
}
#endif
}
/* initialize the buffer */
filebuf = audiobuf; /* must be non-NULL for audio_set_crossfade */
buffer_state = BUFFER_STATE_TRASHED; /* force it */
audio_set_crossfade(global_settings.crossfade);
audio_is_initialized = true;
sound_settings_apply();
}
#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;
/* At first initialize audio system in background. */
audio_playback_init();
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");
audio_clear_track_entries(false);
audio_play_start((size_t)ev.data);
break ;
case Q_AUDIO_STOP:
LOGFQUEUE("audio < Q_AUDIO_STOP");
audio_stop_playback();
break ;
case Q_AUDIO_PAUSE:
LOGFQUEUE("audio < Q_AUDIO_PAUSE");
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_NEW_PLAYLIST:
LOGFQUEUE("audio < Q_AUDIO_NEW_PLAYLIST");
audio_new_playlist();
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 ;
#ifdef AUDIO_HAVE_RECORDING
case Q_AUDIO_LOAD_ENCODER:
LOGFQUEUE("audio < Q_AUDIO_LOAD_ENCODER");
LOGFQUEUE("audio > codec Q_ENCODER_LOAD_DISK");
queue_post(&codec_queue, Q_ENCODER_LOAD_DISK, ev.data);
break;
#endif
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
LOGFQUEUE("audio < SYS_USB_CONNECTED");
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 */
}