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foxbox/apps/playback.c
Miika Pekkarinen a85044bf9e New scheduler, with priorities for swcodec platforms. Frequent task 
switching should be more efficient and tasks are stored in linked
lists to eliminate unnecessary task switching to improve performance.
Audio should no longer skip on swcodec targets caused by too CPU
hungry UI thread or background threads.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@10958 a1c6a512-1295-4272-9138-f99709370657
2006-09-16 16:18:11 +00:00

3332 lines
89 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: Check for a possibly broken codepath on a rapid skip, stop event */
/* TODO: same in reverse ^^ */
/* TODO: Also play, stop ^^ */
/* 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. */
/* This is also necesary to prevent the problem with buffer overwriting on
* automatic track changes */
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <ctype.h>
#include "system.h"
#include "thread.h"
#include "file.h"
#include "lcd.h"
#include "font.h"
#include "backlight.h"
#include "button.h"
#include "kernel.h"
#include "tree.h"
#include "debug.h"
#include "sprintf.h"
#include "settings.h"
#include "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 "pcm_playback.h"
#include "pcm_record.h"
#include "buffer.h"
#include "dsp.h"
#include "abrepeat.h"
#include "tagcache.h"
#ifdef HAVE_LCD_BITMAP
#include "icons.h"
#include "peakmeter.h"
#include "action.h"
#endif
#include "lang.h"
#include "bookmark.h"
#include "misc.h"
#include "sound.h"
#include "metadata.h"
#include "splash.h"
#include "talk.h"
#ifdef HAVE_RECORDING
#include "recording.h"
#endif
static volatile bool audio_codec_loaded;
static volatile bool voice_codec_loaded;
static volatile bool playing;
static volatile bool paused;
/* 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 */
#ifdef SIMULATOR
#define LOGF_QUEUES /* Define this for logf output of all queuing */
#endif
#ifdef LOGF_QUEUES
#define LOGFQUEUE(s) logf("%s", s)
#else
#define LOGFQUEUE(s)
#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,
Q_CODEC_REQUEST_PENDING,
Q_CODEC_REQUEST_COMPLETE,
Q_CODEC_REQUEST_FAILED,
Q_VOICE_PLAY,
Q_VOICE_STOP,
Q_CODEC_LOAD,
Q_CODEC_LOAD_DISK,
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
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 CONFIG_CPU == PP5020 || CONFIG_CPU == PP5002
#define CODEC_IRAM_ORIGIN 0x4000c000
#elif defined(IAUDIO_X5)
#define CODEC_IRAM_ORIGIN 0x10014000
#else
#define CODEC_IRAM_ORIGIN 0x1000c000
#endif
#define CODEC_IRAM_SIZE 0xc000
#ifndef SIMULATOR
extern bool audio_is_initialized;
#else
static bool audio_is_initialized = false;
#endif
static struct mutex mutex_codecthread;
static struct event_queue codec_callback_queue;
static struct mp3entry id3_voice;
static char *voicebuf;
static size_t voice_remaining;
static bool voice_is_playing;
static void (*voice_getmore)(unsigned char** start, int* size);
static struct thread_entry *voice_thread_p = NULL;
/* Is file buffer currently being refilled? */
static volatile bool filling IDATA_ATTR;
volatile int current_codec IDATA_ATTR;
extern unsigned char codecbuf[];
/* Ring buffer where tracks and codecs are loaded. */
static char *filebuf;
/* Total size of the ring buffer. */
size_t filebuflen;
/* Bytes available in the buffer. */
size_t filebufused;
/* Ring buffer read and write indexes. */
static volatile size_t buf_ridx IDATA_ATTR;
static volatile size_t buf_widx IDATA_ATTR;
#ifndef SIMULATOR
static unsigned char *iram_buf[2];
#endif
static unsigned char *dram_buf[2];
/* Step count to the next unbuffered track. */
static int last_peek_offset;
/* Track information (count in file buffer, read/write indexes for
track ring structure. */
static int track_ridx;
static int track_widx;
static bool track_changed;
/* Partially loaded song's file handle to continue buffering later. */
static int current_fd;
/* Information about how many bytes left on the buffer re-fill run. */
static size_t fill_bytesleft;
/* Track info structure about songs in the file buffer. */
static struct track_info tracks[MAX_TRACK];
/* Pointer to track info structure about current song playing. */
static struct track_info *cur_ti;
static struct track_info *prev_ti;
/* Have we reached end of the current playlist. */
static bool playlist_end = false;
/* Codec API including function callbacks. */
extern struct codec_api ci;
extern struct codec_api ci_voice;
/* Was the skip being executed manual or automatic? */
static bool automatic_skip;
static bool dir_skip = false;
static bool new_playlist = false;
/* Callback function to call when current track has really changed. */
void (*track_changed_callback)(struct mp3entry *id3);
void (*track_buffer_callback)(struct mp3entry *id3, bool last_track);
void (*track_unbuffer_callback)(struct mp3entry *id3, bool last_track);
/* Configuration */
static size_t conf_watermark;
static size_t conf_filechunk;
static size_t conf_preseek;
static size_t buffer_margin;
static bool v1first = false;
/* Multiple threads */
static const char * get_codec_filename(int enc_spec);
/* Audio thread */
static struct event_queue audio_queue;
static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
static const char audio_thread_name[] = "audio";
static void audio_thread(void);
static void set_filebuf_watermark(int seconds);
static void audio_initiate_track_change(long direction);
static bool audio_have_tracks(void);
static void audio_reset_buffer(void);
/* Codec thread */
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";
/* For modifying thread priority later. */
struct thread_entry *codec_thread_p;
/* Voice thread */
static struct event_queue voice_queue;
static long voice_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)]
IBSS_ATTR;
static const char voice_thread_name[] = "voice codec";
struct voice_info {
void (*callback)(unsigned char **start, int *size);
int size;
char *buf;
};
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
static void voice_boost_cpu(bool state);
#else
#define voice_boost_cpu(state) do { } while(0)
#endif
static void voice_thread(void);
/* --- External interfaces --- */
void mp3_play_data(const unsigned char* start, int size,
void (*get_more)(unsigned char** start, int* size))
{
static struct voice_info voice_clip;
voice_clip.callback = get_more;
voice_clip.buf = (char *)start;
voice_clip.size = size;
logf("mp3 > voice Q_VOICE_STOP");
queue_post(&voice_queue, Q_VOICE_STOP, 0);
logf("mp3 > voice Q_VOICE_PLAY");
queue_post(&voice_queue, Q_VOICE_PLAY, &voice_clip);
voice_is_playing = true;
voice_boost_cpu(true);
}
void mp3_play_stop(void)
{
logf("mp3 > voice Q_VOICE_STOP");
queue_post(&voice_queue, Q_VOICE_STOP, 0);
}
bool mp3_pause_done(void)
{
return pcm_is_paused();
}
void mpeg_id3_options(bool _v1first)
{
v1first = _v1first;
}
void audio_load_encoder(int enc_id)
{
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
const char *enc_fn = get_codec_filename(enc_id | CODEC_TYPE_ENCODER);
if (!enc_fn)
return;
audio_remove_encoder();
LOGFQUEUE("audio > codec Q_ENCODER_LOAD_DISK");
queue_post(&codec_queue, Q_ENCODER_LOAD_DISK, (void *)enc_fn);
while (!ci.enc_codec_loaded)
yield();
#endif
return;
(void)enc_id;
} /* audio_load_encoder */
void audio_remove_encoder(void)
{
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
/* force encoder codec unload (if previously loaded) */
if (!ci.enc_codec_loaded)
return;
ci.stop_codec = true;
while (ci.enc_codec_loaded)
yield();
#endif
} /* audio_remove_encoder */
struct mp3entry* audio_current_track(void)
{
const char *filename;
const char *p;
static struct mp3entry temp_id3;
int cur_idx;
cur_idx = track_ridx + ci.new_track;
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(ci.new_track);
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");
if (playing && offset <= 0)
{
LOGFQUEUE("audio > audio Q_AUDIO_NEW_PLAYLIST");
queue_post(&audio_queue, Q_AUDIO_NEW_PLAYLIST, 0);
}
else
{
if (playing)
audio_stop();
playing = true;
LOGFQUEUE("audio > audio Q_AUDIO_PLAY");
queue_post(&audio_queue, Q_AUDIO_PLAY, (void *)offset);
}
}
void audio_stop(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_STOP");
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
while (playing || audio_codec_loaded)
yield();
}
void audio_pause(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_PAUSE");
queue_post(&audio_queue, Q_AUDIO_PAUSE, (void *)true);
}
void audio_resume(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_PAUSE resume");
queue_post(&audio_queue, Q_AUDIO_PAUSE, (void *)false);
}
void audio_next(void)
{
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
/* Should be safe to do outside of thread, that way we get
* the instant wps response at least. */
audio_initiate_track_change(1);
// queue_post(&audio_queue, Q_AUDIO_SKIP, (void *)1);
}
void audio_prev(void)
{
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
audio_initiate_track_change(-1);
// queue_post(&audio_queue, Q_AUDIO_SKIP, (void *)-1);
}
void audio_next_dir(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP 1");
queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)1);
}
void audio_prev_dir(void)
{
LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP -1");
queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)-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, (int *)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)
{
}
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;
}
bool audio_query_poweroff(void)
{
return !(playing && paused);
}
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 (pcmbuf_get_bufsize() == 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. */
LOGFQUEUE("audio > audio Q_AUDIO_STOP");
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
while (audio_codec_loaded)
yield();
gui_syncsplash(0, true, (char *)str(LANG_RESTARTING_PLAYBACK));
}
/* Re-initialize audio system. */
pcmbuf_init(size);
pcmbuf_crossfade_enable(enable);
audio_reset_buffer();
logf("abuf:%dB", pcmbuf_get_bufsize());
logf("fbuf:%dB", filebuflen);
voice_init();
/* Restart playback. */
if (was_playing) {
playing = true;
LOGFQUEUE("audio > audio Q_AUDIO_PLAY");
queue_post(&audio_queue, Q_AUDIO_PLAY, (void *)offset);
/* Wait for the playback to start again (and display the splash
screen during that period. */
while (playing && !audio_codec_loaded)
yield();
}
}
void audio_preinit(void)
{
logf("playback system pre-init");
filebufused = 0;
filling = false;
current_codec = CODEC_IDX_AUDIO;
playing = false;
paused = false;
audio_codec_loaded = false;
voice_is_playing = false;
track_changed = false;
current_fd = -1;
track_buffer_callback = NULL;
track_unbuffer_callback = NULL;
track_changed_callback = NULL;
/* Just to prevent cur_ti never be anything random. */
cur_ti = &tracks[0];
mutex_init(&mutex_codecthread);
queue_init(&audio_queue, true);
queue_init(&codec_queue, true);
/* create a private queue */
queue_init(&codec_callback_queue, false);
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)
{
if (!filebuf)
return; /* Audio buffers not yet set up */
if (voice_thread_p)
{
logf("Terminating voice codec");
remove_thread(voice_thread_p);
if (current_codec == CODEC_IDX_VOICE)
mutex_unlock(&mutex_codecthread);
queue_delete(&voice_queue);
voice_thread_p = NULL;
voice_codec_loaded = false;
}
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();
} /* voice_init */
void voice_stop(void)
{
/* Messages should not be posted to voice codec queue unless it is the
current codec or deadlocks happen. This will be addressed globally soon.
-- jhMikeS */
if (current_codec != CODEC_IDX_VOICE)
return;
mp3_play_stop();
while (voice_is_playing && !queue_empty(&voice_queue))
yield();
} /* voice_stop */
/* --- Routines called from multiple threads --- */
static void swap_codec(void)
{
int my_codec = current_codec;
logf("swapping out codec:%d", my_codec);
/* Save our current IRAM and DRAM */
#ifndef SIMULATOR
memcpy(iram_buf[my_codec], (unsigned char *)CODEC_IRAM_ORIGIN,
CODEC_IRAM_SIZE);
#endif
memcpy(dram_buf[my_codec], codecbuf, CODEC_SIZE);
do {
/* Release my semaphore and force a task switch. */
mutex_unlock(&mutex_codecthread);
yield();
mutex_lock(&mutex_codecthread);
/* Loop until the other codec has locked and run */
} while (my_codec == current_codec);
current_codec = my_codec;
/* Reload our IRAM and DRAM */
#ifndef SIMULATOR
memcpy((unsigned char *)CODEC_IRAM_ORIGIN, iram_buf[my_codec],
CODEC_IRAM_SIZE);
#endif
invalidate_icache();
memcpy(codecbuf, dram_buf[my_codec], CODEC_SIZE);
logf("resuming codec:%d", my_codec);
}
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 enc_spec)
{
const char *fname;
int type = enc_spec & CODEC_TYPE_MASK;
int afmt = enc_spec & CODEC_AFMT_MASK;
if ((unsigned)afmt >= AFMT_NUM_CODECS)
type = AFMT_UNKNOWN | (type & CODEC_TYPE_MASK);
fname = (type == CODEC_TYPE_DECODER) ?
audio_formats[afmt].codec_fn : audio_formats[afmt].codec_enc_fn;
logf("%s: %d - %s",
(type == CODEC_TYPE_ENCODER) ? "Encoder" : "Decoder",
afmt, fname ? fname : "<unknown>");
return fname;
} /* get_codec_filename */
/* --- Voice thread --- */
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
static void voice_boost_cpu(bool state)
{
static bool voice_cpu_boosted = false;
if (state != voice_cpu_boosted)
{
cpu_boost(state);
voice_cpu_boosted = state;
}
}
#endif
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)
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)
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)
{
queue_wait_w_tmo(&voice_queue, &ev, 0);
}
else if (playing)
{
queue_wait_w_tmo(&voice_queue, &ev, 0);
if (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)
swap_codec();
break;
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
case Q_ENCODER_RECORD:
LOGFQUEUE("voice < Q_ENCODER_RECORD");
swap_codec();
break;
#endif
case Q_VOICE_STOP:
LOGFQUEUE("voice < Q_VOICE_STOP");
if (voice_is_playing)
{
/* Clear the current buffer */
voice_is_playing = false;
voice_getmore = NULL;
voice_remaining = 0;
voicebuf = NULL;
voice_boost_cpu(false);
ci_voice.new_track = 1;
/* Force the codec to think it's changing tracks */
*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");
{
struct voice_info *voice_data;
voice_is_playing = true;
voice_boost_cpu(true);
voice_data = ev.data;
voice_remaining = voice_data->size;
voicebuf = voice_data->buf;
voice_getmore = voice_data->callback;
}
case SYS_TIMEOUT:
LOGFQUEUE("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)
{
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 */
/* --- 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);
if (voice_is_playing && pcm_is_playing() &&
pcmbuf_usage() > 30 && pcmbuf_mix_free() > 80)
{
swap_codec();
}
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)
{
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 += amount;
if (buf_ridx >= filebuflen)
buf_ridx -= filebuflen;
ci.curpos += amount;
cur_ti->available -= amount;
filebufused -= amount;
/* Start buffer filling as necessary. */
if (!pcmbuf_is_lowdata() && !filling)
{
if (conf_watermark && filebufused <= conf_watermark && playing)
{
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)
{
struct event ev;
LOGFQUEUE("codec > audio Q_AUDIO_REBUFFER_SEEK");
queue_post(&audio_queue,
Q_AUDIO_REBUFFER_SEEK, (void *)(ci.curpos + amount));
queue_wait(&codec_callback_queue, &ev);
switch (ev.id)
{
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;
/* 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;
/* skip past id3v2 tag and other leading garbage */
else if (pos < (int)id3->first_frame_offset)
pos = id3->first_frame_offset;
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();
/* 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)
{
struct event ev;
LOGFQUEUE("codec > audio Q_AUDIO_REBUFFER_SEEK");
queue_post(&audio_queue, Q_AUDIO_REBUFFER_SEEK, (void *)newpos);
queue_wait(&codec_callback_queue, &ev);
switch (ev.id)
{
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);
filebufused += difference;
cur_ti->available += difference;
if (buf_ridx < (unsigned)difference)
buf_ridx += filebuflen;
buf_ridx -= 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;
track_changed = true;
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;
filebufused -= cur_ti->codecsize;
buf_ridx += cur_ti->codecsize;
if (buf_ridx >= filebuflen)
buf_ridx -= filebuflen;
}
#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)
{
/* 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()
&& global_settings.crossfade != 2)
{
pcmbuf_crossfade_init(false);
codec_track_changed();
}
/* Gapless playback. */
else
{
pcmbuf_set_position_callback(codec_pcmbuf_position_callback);
pcmbuf_set_event_handler(codec_pcmbuf_track_changed_callback);
}
}
static bool codec_load_next_track(void)
{
struct event ev;
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;
}
cpu_boost(true);
LOGFQUEUE("codec > audio Q_AUDIO_CHECK_NEW_TRACK");
queue_post(&audio_queue, Q_AUDIO_CHECK_NEW_TRACK, 0);
while (1)
{
queue_wait(&codec_callback_queue, &ev);
if (ev.id == Q_CODEC_REQUEST_PENDING)
{
if (!automatic_skip)
pcmbuf_play_stop();
}
else
break;
}
cpu_boost(false);
switch (ev.id)
{
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;
if (voice_codec_loaded)
{
LOGFQUEUE("codec > voice Q_AUDIO_PLAY");
queue_post(&voice_queue, Q_AUDIO_PLAY, 0);
}
mutex_lock(&mutex_codecthread);
current_codec = CODEC_IDX_AUDIO;
ci.stop_codec = false;
status = codec_load_file((const char *)ev.data, &ci);
mutex_unlock(&mutex_codecthread);
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;
if (voice_codec_loaded)
{
LOGFQUEUE("codec > voice Q_AUDIO_PLAY");
queue_post(&voice_queue, Q_AUDIO_PLAY, 0);
}
mutex_lock(&mutex_codecthread);
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);
mutex_unlock(&mutex_codecthread);
break ;
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
case Q_ENCODER_LOAD_DISK:
LOGFQUEUE("codec < Q_ENCODER_LOAD_DISK");
audio_codec_loaded = false;
if (voice_codec_loaded && current_codec == CODEC_IDX_VOICE)
{
LOGFQUEUE("codec > voice Q_ENCODER_RECORD");
queue_post(&voice_queue, Q_ENCODER_RECORD, NULL);
}
mutex_lock(&mutex_codecthread);
current_codec = CODEC_IDX_AUDIO;
ci.stop_codec = false;
status = codec_load_file((const char *)ev.data, &ci);
mutex_unlock(&mutex_codecthread);
break;
#endif
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
LOGFQUEUE("codec < SYS_USB_CONNECTED");
queue_clear(&codec_queue);
usb_acknowledge(SYS_USB_CONNECTED_ACK);
if (voice_codec_loaded)
swap_codec();
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())
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,
(void *)codec_fn);
}
}
break;
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;
}
/* 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);
if (amount > filebufused)
return false;
logf("bwf:%ldB",amount);
/* Wind the buffer to the beginning of the target track or its codec */
buf_ridx += amount;
filebufused -= amount;
/* Check and handle buffer wrapping */
if (buf_ridx >= filebuflen)
buf_ridx -= filebuflen;
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;
buf_back = buf_ridx;
amount = ci.curpos;
if (buf_ridx < buf_widx)
buf_back += filebuflen;
buf_back -= 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);
/* Check and handle buffer wrapping */
if (amount > buf_ridx)
buf_ridx += filebuflen;
/* Rewind the buffer to the beginning of the target track or its codec */
buf_ridx -= amount;
filebufused += 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())
{
sleep(1);
if (!queue_empty(&audio_queue))
return true;
}
return false;
}
/* Note that this function might yield(). */
static void audio_clear_track_entries(
bool clear_buffered, bool clear_unbuffered,
bool may_yield)
{
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 (clear_buffered)
{
if (last_idx >= 0)
{
/* If there is an unbuffer callback, call it, otherwise,
* just clear the track */
if (track_unbuffer_callback)
{
if (may_yield)
audio_yield_codecs();
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_id3v1_tag(void)
{
int i;
static const unsigned char tag[] = "TAG";
size_t tag_idx;
size_t cur_idx;
tag_idx = buf_widx;
if (tag_idx < 128)
tag_idx += filebuflen;
tag_idx -= 128;
if (filebufused > 128 && tag_idx > buf_ridx)
{
cur_idx = tag_idx;
for(i = 0;i < 3;i++)
{
if(filebuf[cur_idx] != tag[i])
return;
if(++cur_idx >= filebuflen)
cur_idx -= filebuflen;
}
/* Skip id3v1 tag */
logf("Skipping ID3v1 tag");
buf_widx = tag_idx;
tracks[track_widx].available -= 128;
tracks[track_widx].filesize -= 128;
filebufused -= 128;
}
}
static void audio_read_file(bool quick)
{
size_t copy_n;
int rc;
/* If we're called and no file is open, this is an error */
if (current_fd < 0)
{
logf("Bad fd in arf");
/* Stop this buffer cycle immediately */
fill_bytesleft = 0;
/* Give some hope of miraculous recovery by forcing a track reload */
tracks[track_widx].filesize = 0;
return ;
}
while (tracks[track_widx].filerem > 0)
{
int overlap;
if (fill_bytesleft == 0)
break ;
/* copy_n is the largest chunk that is safe to read */
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
copy_n = MIN(copy_n, fill_bytesleft);
/* rc is the actual amount read */
rc = read(current_fd, &filebuf[buf_widx], copy_n);
if (rc <= 0)
{
/* Reached the end of the file */
tracks[track_widx].filerem = 0;
break ;
}
buf_widx += rc;
overlap = buf_widx - tracks[track_ridx].buf_idx;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
if (overlap > 0 && (unsigned) overlap >= filebuflen)
overlap -= filebuflen;
if (overlap > 0 && overlap <= rc && tracks[track_ridx].available != 0) {
tracks[track_ridx].buf_idx = buf_widx;
tracks[track_ridx].start_pos += overlap;
}
tracks[track_widx].available += rc;
tracks[track_widx].filerem -= rc;
filebufused += rc;
if (fill_bytesleft > (unsigned)rc)
fill_bytesleft -= rc;
else
fill_bytesleft = 0;
/* Let the codec process until it is out of the danger zone, or there
* is an event to handle. In the latter case, break this fill cycle
* immediately */
if (quick || audio_yield_codecs())
break;
}
if (tracks[track_widx].filerem == 0)
{
logf("Finished buf:%dB", tracks[track_widx].filesize);
close(current_fd);
current_fd = -1;
audio_strip_id3v1_tag();
track_widx++;
track_widx &= MAX_TRACK_MASK;
tracks[track_widx].filesize = 0;
}
else
{
logf("Partially buf:%dB",
tracks[track_widx].filesize - tracks[track_widx].filerem);
}
}
static bool audio_loadcodec(bool start_play)
{
size_t size;
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;
tracks[track_widx].codecsize = 0;
if (start_play)
{
/* Load the codec directly from disk and save some memory. */
track_ridx = track_widx;
cur_ti = &tracks[track_ridx];
ci.filesize = cur_ti->filesize;
ci.id3 = &cur_ti->id3;
ci.taginfo_ready = &cur_ti->taginfo_ready;
ci.curpos = 0;
playing = true;
LOGFQUEUE("codec > codec Q_CODEC_LOAD_DISK");
queue_post(&codec_queue, Q_CODEC_LOAD_DISK, (void *)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;
}
size = filesize(fd);
/* Never load a partial codec */
if (fill_bytesleft < size)
{
logf("Not enough space");
fill_bytesleft = 0;
close(fd);
return false;
}
while (tracks[track_widx].codecsize < size)
{
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
rc = read(fd, &filebuf[buf_widx], copy_n);
if (rc < 0)
return false;
filebufused += rc;
if (fill_bytesleft > (unsigned)rc)
fill_bytesleft -= rc;
else
fill_bytesleft = 0;
buf_widx += rc;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
tracks[track_widx].codecsize += rc;
audio_yield_codecs();
}
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)
{
if ( id3->vbr ) {
if ( id3->has_toc ) {
/* calculate elapsed time using TOC */
int i;
unsigned int remainder, plen, relpos, nextpos;
/* find wich percent we're at */
for (i=0; i<100; i++ )
if ( id3->offset < id3->toc[i] * (id3->filesize / 256) )
break;
i--;
if (i < 0)
i = 0;
relpos = id3->toc[i];
if (i < 99)
nextpos = id3->toc[i+1];
else
nextpos = 256;
remainder = id3->offset - (relpos * (id3->filesize / 256));
/* set time for this percent (divide before multiply to prevent
overflow on long files. loss of precision is negligible on
short files) */
id3->elapsed = i * (id3->length / 100);
/* calculate remainder time */
plen = (nextpos - relpos) * (id3->filesize / 256);
id3->elapsed += (((remainder * 100) / plen) *
(id3->length / 10000));
}
else {
/* no TOC exists. set a rough estimate using average bitrate */
int tpk = id3->length / (id3->filesize / 1024);
id3->elapsed = id3->offset / 1024 * tpk;
}
}
else
{
/* constant bitrate, use exact calculation */
if (id3->bitrate != 0)
id3->elapsed = id3->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))
{
if (tracks[track_widx].codecsize)
{
/* Must undo the buffer write of the partial codec */
logf("Partial codec loaded");
fill_bytesleft += tracks[track_widx].codecsize;
filebufused -= tracks[track_widx].codecsize;
if (buf_widx < tracks[track_widx].codecsize)
buf_widx += filebuflen;
buf_widx -= tracks[track_widx].codecsize;
tracks[track_widx].codecsize = 0;
}
/* 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;
/* Try skipping to next track if there is space. */
if (fill_bytesleft > 0)
{
/* This is an error condition unless the fill_bytesleft is 0 */
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;
}
return false;
}
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;
audio_read_file(rebuffer);
return true;
}
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;
/* Don't start buffer fill if buffer is already full. */
if (filebufused > conf_watermark && !filling)
return false;
logf("Starting buffer fill");
fill_bytesleft = filebuflen - filebufused;
/* TODO: This doesn't look right, and might explain some problems with
* seeking in large files (to offsets larger than filebuflen).
* And what about buffer wraps?
*
* This really doesn't look right, so don't use it.
*/
// if (buf_ridx > cur_ti->buf_idx)
// cur_ti->start_pos = buf_ridx - cur_ti->buf_idx;
/* 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(true, false, true);
/* 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;
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)
audio_read_file(false);
else if (!audio_load_track(offset, start_play, rebuffer))
fill_bytesleft = 0;
if (!had_next_track && audio_next_track())
track_changed = true;
/* If we're done buffering */
if (fill_bytesleft == 0)
{
audio_read_next_metadata();
audio_generate_postbuffer_events();
filling = false;
#ifndef SIMULATOR
if (playing)
ata_sleep();
#endif
}
}
static void audio_rebuffer(void)
{
logf("Forcing rebuffer");
/* 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);
/* 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 */
tracks[track_ridx].buf_idx = buf_ridx = buf_widx = 0;
track_widx = track_ridx;
cur_ti = &tracks[track_ridx];
audio_clear_track_entries(true, true, false);
filebufused = 0;
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 void 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");
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0);
return;
}
}
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");
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0);
return;
}
/* 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");
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0);
return;
}
}
/* Update the playlist */
last_peek_offset -= ci.new_track;
if (playlist_next(ci.new_track) < 0)
{
LOGFQUEUE("audio > codec Q_CODEC_REQUEST_FAILED");
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0);
return;
}
if (new_playlist)
{
ci.new_track = 1;
new_playlist = false;
}
track_ridx += ci.new_track;
track_ridx &= MAX_TRACK_MASK;
/* Save the old track */
prev_ti = cur_ti;
/* Move to the new track */
cur_ti = &tracks[track_ridx];
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");
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0);
}
static void audio_rebuffer_and_seek(size_t newpos)
{
int fd;
char *trackname;
trackname = playlist_peek(0);
/* (Re-)open current track's file handle. */
fd = open(trackname, O_RDONLY);
if (fd < 0)
{
LOGFQUEUE("audio > codec Q_CODEC_REQUEST_FAILED");
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0);
return;
}
if (current_fd >= 0)
close(current_fd);
current_fd = fd;
playlist_end = false;
ci.curpos = newpos;
/* Clear codec buffer. */
track_widx = track_ridx;
filebufused = 0;
tracks[track_widx].buf_idx = buf_widx = buf_ridx = 0;
last_peek_offset = 0;
filling = false;
audio_initialize_buffer_fill(true);
filling = true;
if (newpos > conf_preseek) {
buf_ridx += conf_preseek;
cur_ti->start_pos = newpos - conf_preseek;
}
else
{
buf_ridx += newpos;
cur_ti->start_pos = 0;
}
cur_ti->filerem = cur_ti->filesize - cur_ti->start_pos;
cur_ti->available = 0;
lseek(current_fd, cur_ti->start_pos, SEEK_SET);
LOGFQUEUE("audio > codec Q_CODEC_REQUEST_COMPLETE");
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0);
}
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;
}
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());
}
while (voice_is_playing && !queue_empty(&voice_queue))
yield();
filebufused = 0;
playing = false;
filling = false;
paused = false;
audio_stop_codec_flush();
if (current_fd >= 0)
{
close(current_fd);
current_fd = -1;
}
/* Mark all entries null. */
audio_clear_track_entries(true, false, false);
memset(tracks, 0, sizeof(struct track_info) * MAX_TRACK);
}
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? */
while (audio_codec_loaded)
audio_stop_codec_flush();
track_changed = true;
playlist_end = false;
playing = true;
ci.new_track = 0;
ci.seek_time = 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;
filebufused = 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. */
void audio_invalidate_tracks(void)
{
if (audio_have_tracks()) {
last_peek_offset = 0;
playlist_end = false;
track_widx = track_ridx;
audio_clear_track_entries(true, true, 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;
/* Mark all other entries null (also buffered wrong metadata). */
filebufused = cur_ti->available;
buf_widx = buf_ridx + cur_ti->available;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
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, true, 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 */
filebufused = cur_ti->available;
buf_widx = buf_ridx + cur_ti->available;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
}
/* 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)
{
if (playlist_check(direction))
{
playlist_end = false;
/* Flag track changed immediately so wps can update instantly.
* No need to wait for disk to spin up or message to travel
* through the deep queues as this info is only for the wps. */
track_changed = true;
ci.new_track += direction;
}
}
static void audio_initiate_dir_change(long direction)
{
playlist_end = false;
dir_skip = true;
ci.new_track = direction;
}
static void audio_reset_buffer(void)
{
size_t offset;
/* Set up file buffer as all space available */
filebuf = (char *)&audiobuf[talk_get_bufsize()+MALLOC_BUFSIZE];
filebuflen = audiobufend - (unsigned char *) filebuf - GUARD_BUFSIZE -
(pcmbuf_get_bufsize() + get_pcmbuf_descsize() + PCMBUF_MIX_CHUNK * 2);
/* Allow for codec(s) at end of file buffer */
if (talk_voice_required())
{
/* Allow 2 codecs at end of file buffer */
filebuflen -= 2 * (CODEC_IRAM_SIZE + CODEC_SIZE);
#ifndef SIMULATOR
iram_buf[0] = &filebuf[filebuflen];
iram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE];
#endif
dram_buf[0] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE*2];
dram_buf[1] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE*2+CODEC_SIZE];
}
else
{
/* Allow for 1 codec at end of file buffer */
filebuflen -= CODEC_IRAM_SIZE + CODEC_SIZE;
#ifndef SIMULATOR
iram_buf[0] = &filebuf[filebuflen];
iram_buf[1] = NULL;
#endif
dram_buf[0] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE];
dram_buf[1] = NULL;
}
/* Ensure that file buffer is aligned */
offset = (-(size_t)filebuf) & 3;
filebuf += offset;
filebuflen -= offset;
filebuflen &= ~3;
}
#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)
{
static bool voicetagtrue = true;
struct event ev;
logf("playback api init");
pcm_init();
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
/* Set the input multiplexer to Line In */
pcm_rec_mux(0);
#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. */
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;
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
}
filebuf = (char *)&audiobuf[MALLOC_BUFSIZE]; /* Will be reset by reset_buffer */
audio_set_crossfade(global_settings.crossfade);
audio_is_initialized = true;
sound_settings_apply();
}
static void audio_thread(void)
{
struct event ev;
/* At first initialize audio system in background. */
audio_playback_init();
while (1)
{
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);
switch (ev.id) {
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(true, false, true);
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");
audio_rebuffer_and_seek((size_t)ev.data);
break;
case Q_AUDIO_CHECK_NEW_TRACK:
LOGFQUEUE("audio < Q_AUDIO_CHECK_NEW_TRACK");
audio_check_new_track();
break;
case Q_AUDIO_DIR_SKIP:
LOGFQUEUE("audio < Q_AUDIO_DIR_SKIP");
playlist_end = false;
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
audio_initiate_dir_change((long)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 ;
#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("audio < SYS_TIMEOUT");
break;
default:
LOGFQUEUE("audio < default");
}
}
}
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