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rockbox/apps/playback.c

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <ctype.h>
#include "system.h"
#include "thread.h"
#include "file.h"
#include "lcd.h"
#include "font.h"
#include "backlight.h"
#include "button.h"
#include "kernel.h"
#include "tree.h"
#include "debug.h"
#include "sprintf.h"
#include "settings.h"
#include "codecs.h"
#include "wps.h"
#include "wps-display.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 "buffer.h"
#include "dsp.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 "talk.h"
static volatile bool audio_codec_loaded;
static volatile bool voice_codec_loaded;
static volatile bool playing;
static volatile bool paused;
#define CODEC_VORBIS "/.rockbox/codecs/vorbis.codec"
#define CODEC_MPA_L3 "/.rockbox/codecs/mpa.codec"
#define CODEC_FLAC "/.rockbox/codecs/flac.codec"
#define CODEC_WAV "/.rockbox/codecs/wav.codec"
#define CODEC_A52 "/.rockbox/codecs/a52.codec"
#define CODEC_MPC "/.rockbox/codecs/mpc.codec"
#define CODEC_WAVPACK "/.rockbox/codecs/wavpack.codec"
#define CODEC_ALAC "/.rockbox/codecs/alac.codec"
#define AUDIO_FILL_CYCLE (1024*256)
#define AUDIO_DEFAULT_WATERMARK (1024*512)
#define AUDIO_DEFAULT_FILECHUNK (1024*32)
#define AUDIO_PLAY 1
#define AUDIO_STOP 2
#define AUDIO_PAUSE 3
#define AUDIO_RESUME 4
#define AUDIO_NEXT 5
#define AUDIO_PREV 6
#define AUDIO_FF_REWIND 7
#define AUDIO_FLUSH_RELOAD 8
#define AUDIO_CODEC_DONE 9
#define AUDIO_FLUSH 10
#define AUDIO_TRACK_CHANGED 11
#define CODEC_LOAD 1
#define CODEC_LOAD_DISK 2
/* As defined in plugins/lib/xxx2wav.h */
#define MALLOC_BUFSIZE (512*1024)
#define GUARD_BUFSIZE (8*1024)
/* As defined in plugin.lds */
#define CODEC_IRAM_ORIGIN 0x1000c000
#define CODEC_IRAM_SIZE 0xc000
extern bool audio_is_initialized;
/* Buffer control thread. */
static struct event_queue audio_queue;
static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
static const char audio_thread_name[] = "audio";
/* Codec thread. */
static struct event_queue codec_queue;
static long codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)] IDATA_ATTR;
static const char codec_thread_name[] = "codec";
/* Voice codec thread. */
static struct event_queue voice_codec_queue;
/* Not enough IRAM for this. */
static long voice_codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)] IDATA_ATTR;
static const char voice_codec_thread_name[] = "voice codec";
static struct mutex mutex_bufferfill;
static struct mutex mutex_codecthread;
static struct mp3entry id3_voice;
#define CODEC_IDX_AUDIO 0
#define CODEC_IDX_VOICE 1
static char *voicebuf;
static int voice_remaining;
static bool voice_is_playing;
static void (*voice_getmore)(unsigned char** start, int* size);
/* Is file buffer currently being refilled? */
static volatile bool filling;
volatile int current_codec;
extern unsigned char codecbuf[];
/* Ring buffer where tracks and codecs are loaded. */
static char *filebuf;
/* Total size of the ring buffer. */
int filebuflen;
/* Bytes available in the buffer. */
int filebufused;
/* Ring buffer read and write indexes. */
static volatile int buf_ridx;
static volatile int buf_widx;
/* Step count to the next unbuffered track. */
static int last_peek_offset;
/* Index of the last buffered track. */
static int last_index;
/* Track information (count in file buffer, read/write indexes for
track ring structure. */
int track_count;
static volatile int track_ridx;
static volatile int track_widx;
static bool track_changed;
/* Partially loaded song's file handle to continue buffering later. */
static int current_fd;
/* Information about how many bytes left on the buffer re-fill run. */
static long 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;
/* Codec API including function callbacks. */
extern struct codec_api ci;
extern struct codec_api ci_voice;
/* When we change a song and buffer is not in filling state, this
variable keeps information about whether to go a next/previous track. */
static int new_track;
/* 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 int conf_bufferlimit;
static int conf_watermark;
static int conf_filechunk;
static int buffer_margin;
static bool v1first = false;
static void mp3_set_elapsed(struct mp3entry* id3);
int mp3_get_file_pos(void);
static void do_swap(int idx_old, int idx_new)
{
#ifndef SIMULATOR
unsigned char *iram_p = (unsigned char *)(CODEC_IRAM_ORIGIN);
unsigned char *iram_buf[2];
#endif
unsigned char *dram_buf[2];
#ifndef SIMULATOR
iram_buf[0] = &filebuf[filebuflen];
iram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE];
memcpy(iram_buf[idx_old], iram_p, CODEC_IRAM_SIZE);
memcpy(iram_p, iram_buf[idx_new], CODEC_IRAM_SIZE);
#endif
dram_buf[0] = &filebuf[filebuflen+CODEC_IRAM_SIZE*2];
dram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE*2+CODEC_SIZE];
memcpy(dram_buf[idx_old], codecbuf, CODEC_SIZE);
memcpy(codecbuf, dram_buf[idx_new], CODEC_SIZE);
}
static void swap_codec(void)
{
int last_codec;
logf("swapping codec:%d", current_codec);
/* We should swap codecs' IRAM contents and code space. */
do_swap(current_codec, !current_codec);
last_codec = current_codec;
current_codec = !current_codec;
/* Release the semaphore and force a task switch. */
mutex_unlock(&mutex_codecthread);
sleep(1);
/* Waiting until we are ready to run again. */
mutex_lock(&mutex_codecthread);
/* Check if codec swap did not happen. */
if (current_codec != last_codec)
{
logf("no codec switch happened!");
do_swap(current_codec, !current_codec);
current_codec = !current_codec;
}
invalidate_icache();
logf("codec resuming:%d", current_codec);
}
#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;
}
}
#else
#define voice_boost_cpu(state) do { } while(0)
#endif
bool codec_pcmbuf_insert_split_callback(void *ch1, void *ch2,
long length)
{
char* src[2];
char *dest;
long input_size;
long output_size;
src[0] = ch1;
src[1] = ch2;
while (paused)
{
if (pcm_is_playing())
pcm_play_pause(false);
sleep(1);
if (ci.stop_codec || ci.reload_codec || ci.seek_time)
return true;
}
if (dsp_stereo_mode() == STEREO_NONINTERLEAVED)
{
length *= 2; /* Length is per channel */
}
while (length > 0) {
while ((dest = pcmbuf_request_buffer(dsp_output_size(length),
&output_size)) == NULL) {
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 > length) {
DEBUGF("Error: dsp_input_size(%ld=dsp_output_size(%ld))=%ld > %ld\n",
output_size, length, input_size, length);
input_size = length;
}
if (input_size <= 0) {
pcmbuf_flush_buffer(0);
DEBUGF("Warning: dsp_input_size(%ld=dsp_output_size(%ld))=%ld <= 0\n",
output_size, length, input_size);
/* should we really continue, or should we break?
* We should probably continue because calling pcmbuf_flush_buffer(0)
* will wrap the buffer if it was fully filled and so next call to
* pcmbuf_request_buffer should give the requested output_size. */
continue;
}
output_size = dsp_process(dest, src, input_size);
/* Hotswap between audio and voice codecs as necessary. */
switch (current_codec)
{
case CODEC_IDX_AUDIO:
pcmbuf_flush_buffer(output_size);
if (voice_is_playing && pcmbuf_usage() > 30
&& pcmbuf_mix_usage() < 20)
{
voice_boost_cpu(true);
swap_codec();
voice_boost_cpu(false);
}
break ;
case CODEC_IDX_VOICE:
if (audio_codec_loaded) {
pcmbuf_mix(dest, output_size);
if ((pcmbuf_usage() < 10)
|| pcmbuf_mix_usage() > 70)
swap_codec();
} else {
pcmbuf_flush_buffer(output_size);
}
break ;
}
length -= input_size;
}
return true;
}
bool codec_pcmbuf_insert_callback(char *buf, long 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);
}
void* get_codec_memory_callback(long *size)
{
*size = MALLOC_BUFSIZE;
if (voice_codec_loaded)
return &audiobuf[talk_get_bufsize()];
return &audiobuf[0];
}
void codec_set_elapsed_callback(unsigned int value)
{
unsigned int latency;
if (ci.stop_codec || current_codec == CODEC_IDX_VOICE)
return ;
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;
}
}
void codec_set_offset_callback(unsigned int value)
{
unsigned int latency;
if (ci.stop_codec || current_codec == CODEC_IDX_VOICE)
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;
}
}
long codec_filebuf_callback(void *ptr, long size)
{
char *buf = (char *)ptr;
int copy_n;
int part_n;
if (ci.stop_codec || !playing || current_codec == CODEC_IDX_VOICE)
return 0;
copy_n = MIN((off_t)size, (off_t)cur_ti->available + cur_ti->filerem);
while (copy_n > cur_ti->available) {
yield();
if (ci.stop_codec || ci.reload_codec)
return 0;
}
if (copy_n == 0)
return 0;
part_n = MIN(copy_n, filebuflen - buf_ridx);
memcpy(buf, &filebuf[buf_ridx], part_n);
if (part_n < copy_n) {
memcpy(&buf[part_n], &filebuf[0], copy_n - part_n);
}
buf_ridx += copy_n;
if (buf_ridx >= filebuflen)
buf_ridx -= filebuflen;
ci.curpos += copy_n;
cur_ti->available -= copy_n;
filebufused -= copy_n;
return copy_n;
}
void* voice_request_data(long *realsize, long reqsize)
{
while (queue_empty(&voice_codec_queue) && (voice_remaining == 0
|| voicebuf == NULL) && !ci_voice.stop_codec)
{
yield();
if (audio_codec_loaded && (pcmbuf_usage() < 30
|| !voice_is_playing || voicebuf == NULL))
{
swap_codec();
}
else if (!voice_is_playing)
{
voice_boost_cpu(false);
if (!pcm_is_playing())
pcmbuf_boost(false);
sleep(HZ/16);
}
if (voice_remaining)
{
voice_is_playing = true;
break ;
}
if (voice_getmore != NULL)
{
voice_getmore((unsigned char **)&voicebuf, (int *)&voice_remaining);
if (!voice_remaining)
{
voice_is_playing = false;
/* Force pcm playback. */
pcmbuf_play_start();
}
}
}
if (reqsize < 0)
reqsize = 0;
voice_is_playing = true;
*realsize = voice_remaining;
if (*realsize > reqsize)
*realsize = reqsize;
if (*realsize == 0)
return NULL;
return voicebuf;
}
void* codec_request_buffer_callback(long *realsize, long reqsize)
{
long part_n;
/* Voice codec. */
if (current_codec == CODEC_IDX_VOICE) {
return voice_request_data(realsize, reqsize);
}
if (ci.stop_codec || !playing) {
*realsize = 0;
return NULL;
}
*realsize = MIN((off_t)reqsize, (off_t)cur_ti->available + cur_ti->filerem);
if (*realsize == 0) {
return NULL;
}
while ((int)*realsize > cur_ti->available) {
yield();
if (ci.stop_codec || ci.reload_codec) {
*realsize = 0;
return NULL;
}
}
part_n = MIN((int)*realsize, filebuflen - buf_ridx);
if (part_n < *realsize) {
part_n += GUARD_BUFSIZE;
if (part_n < *realsize)
*realsize = part_n;
memcpy(&filebuf[filebuflen], &filebuf[0], *realsize -
(filebuflen - buf_ridx));
}
return (char *)&filebuf[buf_ridx];
}
static bool rebuffer_and_seek(int newpos)
{
int fd;
logf("Re-buffering song");
mutex_lock(&mutex_bufferfill);
/* (Re-)open current track's file handle. */
fd = open(playlist_peek(0), O_RDONLY);
if (fd < 0) {
logf("Open failed!");
mutex_unlock(&mutex_bufferfill);
return false;
}
if (current_fd >= 0)
close(current_fd);
current_fd = fd;
/* Clear codec buffer. */
audio_invalidate_tracks();
filebufused = 0;
buf_ridx = buf_widx = 0;
cur_ti->filerem = cur_ti->filesize - newpos;
cur_ti->filepos = newpos;
cur_ti->start_pos = newpos;
ci.curpos = newpos;
cur_ti->available = 0;
lseek(current_fd, newpos, SEEK_SET);
mutex_unlock(&mutex_bufferfill);
while (cur_ti->available == 0 && cur_ti->filerem > 0) {
yield();
if (ci.stop_codec || ci.reload_codec)
return false;
}
return true;
}
void codec_advance_buffer_callback(long amount)
{
if (current_codec == CODEC_IDX_VOICE) {
//logf("voice ad.buf:%d", amount);
amount = MAX(0, MIN(amount, voice_remaining));
voicebuf += amount;
voice_remaining -= amount;
return ;
}
if (amount > cur_ti->available + cur_ti->filerem)
amount = cur_ti->available + cur_ti->filerem;
if (amount > cur_ti->available) {
if (!rebuffer_and_seek(ci.curpos + amount))
ci.stop_codec = true;
return ;
}
buf_ridx += amount;
if (buf_ridx >= filebuflen)
buf_ridx -= filebuflen;
cur_ti->available -= amount;
filebufused -= amount;
ci.curpos += amount;
codec_set_offset_callback(ci.curpos);
}
void codec_advance_buffer_loc_callback(void *ptr)
{
long amount;
if (current_codec == CODEC_IDX_VOICE)
amount = (int)ptr - (int)voicebuf;
else
amount = (int)ptr - (int)&filebuf[buf_ridx];
codec_advance_buffer_callback(amount);
}
off_t codec_mp3_get_filepos_callback(int newtime)
{
off_t newpos;
cur_ti->id3.elapsed = newtime;
newpos = mp3_get_file_pos();
return newpos;
}
void codec_seek_complete_callback(void)
{
pcmbuf_flush_audio();
}
bool codec_seek_buffer_callback(off_t newpos)
{
int difference;
if (current_codec == CODEC_IDX_VOICE)
return false;
if (newpos < 0)
newpos = 0;
if (newpos >= cur_ti->filesize)
newpos = cur_ti->filesize - 1;
difference = newpos - ci.curpos;
/* Seeking forward */
if (difference >= 0) {
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)
return rebuffer_and_seek(newpos);
/* Seeking inside buffer space. */
logf("seek: -%d", difference);
filebufused += difference;
cur_ti->available += difference;
buf_ridx -= difference;
if (buf_ridx < 0)
buf_ridx = filebuflen + buf_ridx;
ci.curpos -= difference;
return true;
}
static void set_filebuf_watermark(int seconds)
{
long bytes;
if (current_codec == CODEC_IDX_VOICE)
return ;
bytes = MAX((int)cur_ti->id3.bitrate * seconds * (1000/8), conf_watermark);
bytes = MIN(bytes, filebuflen / 2);
conf_watermark = bytes;
}
void codec_configure_callback(int setting, void *value)
{
switch (setting) {
case CODEC_SET_FILEBUF_WATERMARK:
conf_watermark = (unsigned int)value;
set_filebuf_watermark(buffer_margin);
break;
case CODEC_SET_FILEBUF_CHUNKSIZE:
conf_filechunk = (unsigned int)value;
break;
case CODEC_SET_FILEBUF_LIMIT:
conf_bufferlimit = (unsigned int)value;
break;
case CODEC_DSP_ENABLE:
if ((bool)value)
ci.pcmbuf_insert = codec_pcmbuf_insert_callback;
else
ci.pcmbuf_insert = pcmbuf_insert_buffer;
break ;
default:
if (!dsp_configure(setting, value)) {
logf("Illegal key: %d", setting);
}
}
}
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;
}
void codec_track_changed(void)
{
track_changed = true;
queue_post(&audio_queue, AUDIO_TRACK_CHANGED, 0);
}
/* Give codecs or file buffering the right amount of processing time
to prevent pcm audio buffer from going empty. */
void yield_codecs(void)
{
yield();
if (!pcm_is_playing() && !paused)
sleep(5);
while ((pcmbuf_is_crossfade_active() || pcmbuf_is_lowdata())
&& !ci.stop_codec && playing && queue_empty(&audio_queue)
&& filebufused > (128*1024))
yield();
}
/* FIXME: This code should be made more generic and move to metadata.c */
void strip_id3v1_tag(void)
{
int i;
static const unsigned char tag[] = "TAG";
int tagptr;
bool found = true;
if (filebufused >= 128)
{
tagptr = buf_widx - 128;
if (tagptr < 0)
tagptr += filebuflen;
for(i = 0;i < 3;i++)
{
if(tagptr >= filebuflen)
tagptr -= filebuflen;
if(filebuf[tagptr] != tag[i])
{
found = false;
break;
}
tagptr++;
}
if(found)
{
/* Skip id3v1 tag */
logf("Skipping ID3v1 tag\n");
buf_widx -= 128;
tracks[track_widx].available -= 128;
filebufused -= 128;
}
}
}
void audio_fill_file_buffer(void)
{
long i, size;
int rc;
if (current_fd < 0)
return ;
/* Throw away buffered codec. */
if (tracks[track_widx].start_pos != 0)
tracks[track_widx].codecsize = 0;
i = 0;
size = MIN(tracks[track_widx].filerem, AUDIO_FILL_CYCLE);
while (i < size) {
/* Give codecs some processing time. */
yield_codecs();
if (fill_bytesleft == 0)
break ;
rc = MIN(conf_filechunk, filebuflen - buf_widx);
rc = MIN(rc, fill_bytesleft);
rc = read(current_fd, &filebuf[buf_widx], rc);
if (rc <= 0) {
tracks[track_widx].filerem = 0;
strip_id3v1_tag();
break ;
}
buf_widx += rc;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
i += rc;
tracks[track_widx].available += rc;
tracks[track_widx].filerem -= rc;
tracks[track_widx].filepos += rc;
filebufused += rc;
fill_bytesleft -= rc;
}
/*logf("Filled:%d/%d", tracks[track_widx].available,
tracks[track_widx].filerem);*/
}
bool loadcodec(const char *trackname, bool start_play)
{
char msgbuf[80];
off_t size;
unsigned int filetype;
int fd;
int i, rc;
const char *codec_path;
int copy_n;
int prev_track;
filetype = probe_file_format(trackname);
switch (filetype) {
case AFMT_OGG_VORBIS:
logf("Codec: Vorbis");
codec_path = CODEC_VORBIS;
break;
case AFMT_MPA_L1:
case AFMT_MPA_L2:
case AFMT_MPA_L3:
logf("Codec: MPA L1/L2/L3");
codec_path = CODEC_MPA_L3;
break;
case AFMT_PCM_WAV:
logf("Codec: PCM WAV");
codec_path = CODEC_WAV;
break;
case AFMT_FLAC:
logf("Codec: FLAC");
codec_path = CODEC_FLAC;
break;
case AFMT_A52:
logf("Codec: A52");
codec_path = CODEC_A52;
break;
case AFMT_MPC:
logf("Codec: Musepack");
codec_path = CODEC_MPC;
break;
case AFMT_WAVPACK:
logf("Codec: WAVPACK");
codec_path = CODEC_WAVPACK;
break;
case AFMT_ALAC:
logf("Codec: ALAC");
codec_path = CODEC_ALAC;
break;
default:
logf("Codec: Unsupported");
snprintf(msgbuf, sizeof(msgbuf)-1, "No codec for: %s", trackname);
splash(HZ*2, true, msgbuf);
codec_path = NULL;
}
tracks[track_widx].id3.codectype = filetype;
tracks[track_widx].codecsize = 0;
if (codec_path == NULL)
return false;
if (!start_play) {
prev_track = track_widx - 1;
if (prev_track < 0)
prev_track = MAX_TRACK-1;
if (track_count > 0 && filetype == tracks[prev_track].id3.codectype) {
logf("Reusing prev. codec");
return true;
}
} else {
/* Load the codec directly from disk and save some memory. */
cur_ti = &tracks[track_widx];
ci.filesize = cur_ti->filesize;
ci.id3 = (struct mp3entry *)&cur_ti->id3;
ci.taginfo_ready = (bool *)&cur_ti->taginfo_ready;
ci.curpos = 0;
playing = true;
logf("Starting codec");
queue_post(&codec_queue, CODEC_LOAD_DISK, (void *)codec_path);
return true;
}
fd = open(codec_path, O_RDONLY);
if (fd < 0) {
logf("Codec doesn't exist!");
snprintf(msgbuf, sizeof(msgbuf)-1, "Couldn't load codec: %s", codec_path);
splash(HZ*2, true, msgbuf);
return false;
}
size = filesize(fd);
if ((off_t)fill_bytesleft < size + conf_watermark) {
logf("Not enough space");
/* Set codectype back to zero to indicate no codec was loaded. */
tracks[track_widx].id3.codectype = 0;
fill_bytesleft = 0;
close(fd);
return false;
}
i = 0;
while (i < size) {
yield_codecs();
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
rc = read(fd, &filebuf[buf_widx], copy_n);
if (rc < 0)
return false;
buf_widx += rc;
filebufused += rc;
fill_bytesleft -= rc;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
i += rc;
}
close(fd);
logf("Done: %dB", i);
tracks[track_widx].codecsize = size;
return true;
}
bool read_next_metadata(void)
{
int fd;
char *trackname;
int next_track;
int status;
next_track = track_widx;
if (tracks[track_widx].taginfo_ready)
next_track++;
if (next_track >= MAX_TRACK)
next_track -= MAX_TRACK;
if (tracks[next_track].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;
/* Start buffer refilling also because we need to spin-up the disk. */
filling = true;
tracks[next_track].id3.codectype = probe_file_format(trackname);
status = get_metadata(&tracks[next_track],fd,trackname,v1first);
tracks[next_track].id3.codectype = 0;
track_changed = true;
close(fd);
return status;
}
bool audio_load_track(int offset, bool start_play, int peek_offset)
{
char *trackname;
int fd = -1;
off_t size;
int rc, i;
int copy_n;
int playlist_index;
/* 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 (track_count >= MAX_TRACK - 1) {
fill_bytesleft = 0;
return false;
}
/* Don't start loading track if the current write position already
contains a BUFFERED track. The entry may contain the metadata
which is ok. */
if (tracks[track_widx].filesize != 0)
return false;
last_index = playlist_get_display_index();
playlist_index = playlist_get_display_index() - 1
+ playlist_get_first_index(NULL) + peek_offset;
if (playlist_index >= playlist_amount())
playlist_index -= playlist_amount();
peek_again:
/* Get track name from current playlist read position. */
logf("Buffering track:%d/%d", track_widx, track_ridx);
/* Handle broken playlists. */
while ( (trackname = playlist_peek(peek_offset)) != NULL) {
fd = open(trackname, O_RDONLY);
if (fd < 0) {
logf("Open failed");
/* Delete invalid entry from playlist. */
playlist_delete(NULL, playlist_index);
continue ;
}
break ;
}
if (!trackname) {
logf("End-of-playlist");
conf_watermark = 0;
return false;
}
/* Initialize track entry. */
size = filesize(fd);
tracks[track_widx].filerem = size;
tracks[track_widx].filesize = size;
tracks[track_widx].filepos = 0;
tracks[track_widx].available = 0;
//tracks[track_widx].taginfo_ready = false;
tracks[track_widx].playlist_offset = peek_offset;
last_peek_offset = peek_offset;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
/* Set default values */
if (start_play) {
int last_codec = current_codec;
current_codec = CODEC_IDX_AUDIO;
conf_bufferlimit = 0;
conf_watermark = AUDIO_DEFAULT_WATERMARK;
conf_filechunk = AUDIO_DEFAULT_FILECHUNK;
dsp_configure(DSP_RESET, 0);
ci.configure(CODEC_DSP_ENABLE, false);
current_codec = last_codec;
}
/* Load the codec. */
tracks[track_widx].codecbuf = &filebuf[buf_widx];
if (!loadcodec(trackname, start_play)) {
close(fd);
/* Stop buffer filling if codec load failed. */
fill_bytesleft = 0;
/* Set filesize to zero to indicate no file was loaded. */
tracks[track_widx].filesize = 0;
tracks[track_widx].filerem = 0;
/* Try skipping to next track. */
if (fill_bytesleft > 0) {
/* Delete invalid entry from playlist. */
playlist_delete(NULL, playlist_index);
goto peek_again;
}
return false;
}
// tracks[track_widx].filebuf = &filebuf[buf_widx];
tracks[track_widx].start_pos = 0;
/* Get track metadata if we don't already have it. */
if (!tracks[track_widx].taginfo_ready) {
if (!get_metadata(&tracks[track_widx],fd,trackname,v1first)) {
logf("Metadata error!");
tracks[track_widx].filesize = 0;
tracks[track_widx].filerem = 0;
close(fd);
/* Delete invalid entry from playlist. */
playlist_delete(NULL, playlist_index);
goto peek_again;
}
}
set_filebuf_watermark(buffer_margin);
tracks[track_widx].id3.elapsed = 0;
/* Starting playback from an offset is only support in MPA at the moment */
if (offset > 0) {
switch (tracks[track_widx].id3.codectype) {
case AFMT_MPA_L2:
case AFMT_MPA_L3:
lseek(fd, offset, SEEK_SET);
tracks[track_widx].id3.offset = offset;
mp3_set_elapsed(&tracks[track_widx].id3);
tracks[track_widx].filepos = offset;
tracks[track_widx].filerem = tracks[track_widx].filesize - offset;
ci.curpos = offset;
tracks[track_widx].start_pos = offset;
break;
case AFMT_WAVPACK:
lseek(fd, offset, SEEK_SET);
tracks[track_widx].id3.offset = offset;
tracks[track_widx].id3.elapsed = tracks[track_widx].id3.length / 2;
tracks[track_widx].filepos = offset;
tracks[track_widx].filerem = tracks[track_widx].filesize - offset;
ci.curpos = offset;
tracks[track_widx].start_pos = offset;
break;
case AFMT_OGG_VORBIS:
case AFMT_FLAC:
tracks[track_widx].id3.offset = offset;
break;
}
}
if (start_play) {
track_count++;
codec_track_changed();
}
/* Do some initial file buffering. */
i = tracks[track_widx].start_pos;
size = MIN(size, AUDIO_FILL_CYCLE);
while (i < size) {
/* Give codecs some processing time to prevent glitches. */
yield_codecs();
if (fill_bytesleft == 0)
break ;
copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
copy_n = MIN(size - i, copy_n);
copy_n = MIN((int)fill_bytesleft, copy_n);
rc = read(fd, &filebuf[buf_widx], copy_n);
if (rc < copy_n) {
logf("File error!");
tracks[track_widx].filesize = 0;
tracks[track_widx].filerem = 0;
close(fd);
return false;
}
buf_widx += rc;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
i += rc;
tracks[track_widx].available += rc;
tracks[track_widx].filerem -= rc;
filebufused += rc;
fill_bytesleft -= rc;
}
if (!start_play)
track_count++;
tracks[track_widx].filepos = i;
if (current_fd >= 0) {
close(current_fd);
current_fd = -1;
}
/* Leave the file handle open for faster buffer refill. */
if (tracks[track_widx].filerem != 0) {
current_fd = fd;
logf("Partially buf:%d", tracks[track_widx].available);
} else {
logf("Completely buf.");
close(fd);
strip_id3v1_tag();
if (++track_widx >= MAX_TRACK) {
track_widx = 0;
}
tracks[track_widx].filerem = 0;
}
return true;
}
void audio_play_start(int offset)
{
if (current_fd >= 0) {
close(current_fd);
current_fd = -1;
}
memset(&tracks, 0, sizeof(struct track_info) * MAX_TRACK);
sound_set(SOUND_VOLUME, global_settings.volume);
track_count = 0;
track_widx = 0;
track_ridx = 0;
buf_ridx = 0;
buf_widx = 0;
filebufused = 0;
pcmbuf_set_boost_mode(true);
fill_bytesleft = filebuflen;
filling = true;
last_peek_offset = -1;
if (audio_load_track(offset, true, 0)) {
if (track_buffer_callback) {
cur_ti->event_sent = true;
track_buffer_callback(&cur_ti->id3, true);
}
} else {
logf("Failure");
}
pcmbuf_set_boost_mode(false);
}
void audio_clear_track_entries(bool buffered_only)
{
int cur_idx, event_count;
int i;
cur_idx = track_widx;
event_count = 0;
for (i = 0; i < MAX_TRACK - track_count; i++) {
if (++cur_idx >= MAX_TRACK)
cur_idx = 0;
if (tracks[cur_idx].event_sent)
event_count++;
if (!track_unbuffer_callback)
memset(&tracks[cur_idx], 0, sizeof(struct track_info));
}
if (!track_unbuffer_callback)
return ;
cur_idx = track_widx;
for (i = 0; i < MAX_TRACK - track_count; i++) {
if (++cur_idx >= MAX_TRACK)
cur_idx = 0;
/* Send an event to notify that track has finished. */
if (tracks[cur_idx].event_sent) {
event_count--;
track_unbuffer_callback(&tracks[cur_idx].id3, event_count == 0);
}
if (tracks[cur_idx].event_sent || !buffered_only)
memset(&tracks[cur_idx], 0, sizeof(struct track_info));
}
}
/* Send callback events to notify about new tracks. */
static void generate_postbuffer_events(void)
{
int i;
int cur_ridx, event_count;
/* At first determine how many unsent events we have. */
cur_ridx = track_ridx;
event_count = 0;
for (i = 0; i < track_count; i++) {
if (!tracks[cur_ridx].event_sent)
event_count++;
if (++cur_ridx >= MAX_TRACK)
cur_ridx -= MAX_TRACK;
}
/* Now sent these events. */
cur_ridx = track_ridx;
for (i = 0; i < track_count; i++) {
if (!tracks[cur_ridx].event_sent) {
tracks[cur_ridx].event_sent = true;
event_count--;
/* We still want to set event_sent flags even if not using
event callbacks. */
if (track_buffer_callback)
track_buffer_callback(&tracks[cur_ridx].id3, event_count == 0);
}
if (++cur_ridx >= MAX_TRACK)
cur_ridx -= MAX_TRACK;
}
}
void initialize_buffer_fill(void)
{
int cur_idx, i;
fill_bytesleft = filebuflen - filebufused;
cur_ti->start_pos = ci.curpos;
if (filling)
return ;
pcmbuf_set_boost_mode(true);
filling = true;
/* Calculate real track count after throwing away old tracks. */
cur_idx = track_ridx;
for (i = 0; i < track_count; i++) {
if (cur_idx == track_widx)
break ;
if (++cur_idx >= MAX_TRACK)
cur_idx = 0;
}
track_count = i;
if (tracks[track_widx].filesize == 0) {
if (--track_widx < 0)
track_widx = MAX_TRACK - 1;
} else {
track_count++;
}
/* Mark all buffered entries null (not metadata for next track). */
audio_clear_track_entries(true);
}
void audio_check_buffer(void)
{
/* Start buffer filling as necessary. */
if ((filebufused > conf_watermark || !queue_empty(&audio_queue)
|| !playing || ci.stop_codec || ci.reload_codec) && !filling)
return ;
initialize_buffer_fill();
/* Limit buffering size at first run. */
if (conf_bufferlimit && fill_bytesleft > conf_bufferlimit
- filebufused) {
fill_bytesleft = MAX(0, conf_bufferlimit - filebufused);
}
/* Try to load remainings of the file. */
if (tracks[track_widx].filerem > 0)
audio_fill_file_buffer();
/* Increase track write index as necessary. */
if (tracks[track_widx].filerem == 0 && tracks[track_widx].filesize != 0) {
if (++track_widx == MAX_TRACK)
track_widx = 0;
}
/* Load new files to fill the entire buffer. */
if (audio_load_track(0, false, last_peek_offset + 1)) {
} else if (tracks[track_widx].filerem == 0 || fill_bytesleft == 0) {
/* Read next unbuffered track's metadata as necessary. */
read_next_metadata();
generate_postbuffer_events();
filling = false;
conf_bufferlimit = 0;
pcmbuf_set_boost_mode(false);
#ifndef SIMULATOR
if (playing)
ata_sleep();
#endif
}
}
void audio_update_trackinfo(void)
{
if (new_track >= 0) {
buf_ridx += cur_ti->available;
filebufused -= cur_ti->available;
cur_ti = &tracks[track_ridx];
buf_ridx += cur_ti->codecsize;
filebufused -= cur_ti->codecsize;
if (buf_ridx >= filebuflen)
buf_ridx -= filebuflen;
if (!filling)
pcmbuf_set_boost_mode(false);
} else {
buf_ridx -= ci.curpos + cur_ti->codecsize;
filebufused += ci.curpos + cur_ti->codecsize;
cur_ti->available = cur_ti->filesize - cur_ti->filerem;
cur_ti = &tracks[track_ridx];
buf_ridx -= cur_ti->filesize;
filebufused += cur_ti->filesize;
cur_ti->available = cur_ti->filesize;
if (buf_ridx < 0)
buf_ridx = filebuflen + buf_ridx;
}
ci.filesize = cur_ti->filesize;
cur_ti->id3.elapsed = 0;
cur_ti->id3.offset = 0;
ci.id3 = (struct mp3entry *)&cur_ti->id3;
ci.curpos = 0;
cur_ti->start_pos = 0;
ci.taginfo_ready = (bool *)&cur_ti->taginfo_ready;
if (pcmbuf_is_crossfade_enabled() && !pcmbuf_is_crossfade_active()) {
pcmbuf_crossfade_init(new_track ? CROSSFADE_MODE_CROSSFADE
: global_settings.crossfade);
codec_track_changed();
} else {
pcmbuf_add_event(codec_track_changed);
}
last_index = playlist_get_display_index();
}
static void audio_stop_playback(void)
{
paused = false;
playing = false;
filling = false;
ci.stop_codec = true;
if (current_fd >= 0) {
close(current_fd);
current_fd = -1;
}
pcmbuf_play_stop();
while (audio_codec_loaded)
yield();
track_count = 0;
/* Mark all entries null. */
audio_clear_track_entries(false);
}
/* Request the next track with new codec. */
void audio_change_track(void)
{
logf("change track");
/* Wait for new track data. */
while (track_count <= 1 && filling)
yield();
/* If we are not filling, then it must be end-of-playlist. */
if (track_count <= 1) {
logf("No more tracks");
while (pcm_is_playing())
yield();
audio_stop_playback();
return ;
}
if (++track_ridx >= MAX_TRACK)
track_ridx = 0;
audio_update_trackinfo();
queue_post(&codec_queue, CODEC_LOAD, 0);
}
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;
}
bool codec_request_next_track_callback(void)
{
if (current_codec == CODEC_IDX_VOICE) {
voice_remaining = 0;
/* Terminate the codec if they are messages waiting on the queue or
core has been requested the codec to be terminated. */
return !ci_voice.stop_codec && queue_empty(&voice_codec_queue);
}
if (ci.stop_codec || !playing)
return false;
logf("Request new track");
/* Advance to next track. */
if (ci.reload_codec && new_track > 0) {
if (!playlist_check(new_track)) {
ci.reload_codec = false;
return false;
}
last_peek_offset--;
playlist_next(new_track);
if (++track_ridx == MAX_TRACK)
track_ridx = 0;
/* Wait for new track data (codectype 0 is invalid). When a correct
codectype is set, we can assume that the filesize is correct. */
while (tracks[track_ridx].id3.codectype == 0 && filling
&& !ci.stop_codec)
yield();
if (tracks[track_ridx].filesize == 0) {
logf("Loading from disk...");
new_track = 0;
last_index = -1;
queue_post(&audio_queue, AUDIO_PLAY, 0);
return false;
}
}
/* Advance to previous track. */
else if (ci.reload_codec && new_track < 0) {
if (!playlist_check(new_track)) {
ci.reload_codec = false;
return false;
}
last_peek_offset++;
playlist_next(new_track);
if (--track_ridx < 0)
track_ridx = MAX_TRACK-1;
if (tracks[track_ridx].filesize == 0 ||
filebufused+ci.curpos+tracks[track_ridx].filesize
/*+ (off_t)tracks[track_ridx].codecsize*/ > filebuflen) {
logf("Loading from disk...");
new_track = 0;
last_index = -1;
queue_post(&audio_queue, AUDIO_PLAY, 0);
return false;
}
}
/* Codec requested track change (next track). */
else {
if (!playlist_check(1)) {
ci.reload_codec = false;
return false;
}
last_peek_offset--;
playlist_next(1);
if (++track_ridx >= MAX_TRACK)
track_ridx = 0;
/* Wait for new track data (codectype 0 is invalid). When a correct
codectype is set, we can assume that the filesize is correct. */
while (tracks[track_ridx].id3.codectype == 0 && filling
&& !ci.stop_codec)
yield();
if (tracks[track_ridx].filesize == 0) {
logf("No more tracks [2]");
ci.stop_codec = true;
new_track = 0;
last_index = -1;
queue_post(&audio_queue, AUDIO_PLAY, 0);
return false;
}
}
ci.reload_codec = false;
/* Check if the next codec is the same file. */
if (get_codec_base_type(cur_ti->id3.codectype) !=
get_codec_base_type(tracks[track_ridx].id3.codectype)) {
logf("New codec:%d/%d", cur_ti->id3.codectype,
tracks[track_ridx].id3.codectype);
if (--track_ridx < 0)
track_ridx = MAX_TRACK-1;
new_track = 0;
return false;
}
logf("On-the-fly change");
audio_update_trackinfo();
new_track = 0;
return true;
}
/* Invalidates all but currently playing track. */
void audio_invalidate_tracks(void)
{
if (track_count == 0) {
queue_post(&audio_queue, AUDIO_PLAY, 0);
return ;
}
track_count = 1;
last_peek_offset = 0;
track_widx = track_ridx;
/* Mark all other entries null (also buffered wrong metadata). */
audio_clear_track_entries(false);
filebufused = cur_ti->available;
buf_widx = buf_ridx + cur_ti->available;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
read_next_metadata();
}
static void initiate_track_change(int peek_index)
{
if (!playlist_check(peek_index))
return ;
/* Detect if disk is spinning.. */
if (filling) {
playlist_next(peek_index);
queue_post(&audio_queue, AUDIO_PLAY, 0);
} else {
new_track = peek_index;
ci.reload_codec = true;
if (!pcmbuf_is_crossfade_enabled())
pcmbuf_flush_audio();
}
codec_track_changed();
}
void audio_thread(void)
{
struct event ev;
while (1) {
yield_codecs();
mutex_lock(&mutex_bufferfill);
audio_check_buffer();
mutex_unlock(&mutex_bufferfill);
queue_wait_w_tmo(&audio_queue, &ev, 0);
switch (ev.id) {
case AUDIO_PLAY:
/* Refuse to start playback if we are already playing
the requested track. This is needed because when skipping
tracks fast, AUDIO_PLAY commands will get queued with the
the same track and playback will stutter. */
if (last_index == playlist_get_display_index() && playing
&& pcm_is_playing()) {
logf("already playing req. track");
break ;
}
logf("starting...");
playing = true;
ci.stop_codec = true;
ci.reload_codec = false;
ci.seek_time = 0;
pcmbuf_crossfade_init(CROSSFADE_MODE_CROSSFADE);
while (audio_codec_loaded)
yield();
audio_play_start((int)ev.data);
playlist_update_resume_info(audio_current_track());
break ;
case AUDIO_STOP:
if (playing)
playlist_update_resume_info(audio_current_track());
audio_stop_playback();
break ;
case AUDIO_PAUSE:
logf("audio_pause");
/* We will pause the pcm playback in audiobuffer insert function
to prevent a loop inside the pcm buffer. */
// pcm_play_pause(false);
paused = true;
break ;
case AUDIO_RESUME:
logf("audio_resume");
pcm_play_pause(true);
paused = false;
break ;
case AUDIO_NEXT:
logf("audio_next");
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
initiate_track_change(1);
break ;
case AUDIO_PREV:
logf("audio_prev");
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
initiate_track_change(-1);
break;
case AUDIO_FLUSH:
audio_invalidate_tracks();
break ;
case AUDIO_TRACK_CHANGED:
if (track_changed_callback)
track_changed_callback(&cur_ti->id3);
playlist_update_resume_info(audio_current_track());
break ;
case AUDIO_CODEC_DONE:
//if (playing)
// audio_change_track();
break ;
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
logf("USB: Audio core");
audio_stop_playback();
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&audio_queue);
break ;
#endif
case SYS_TIMEOUT:
if (playing)
playlist_update_resume_info(audio_current_track());
break;
}
}
}
void codec_thread(void)
{
struct event ev;
long codecsize;
int status;
int wrap;
while (1) {
status = 0;
queue_wait(&codec_queue, &ev);
switch (ev.id) {
case CODEC_LOAD_DISK:
ci.stop_codec = false;
audio_codec_loaded = true;
mutex_lock(&mutex_codecthread);
current_codec = CODEC_IDX_AUDIO;
status = codec_load_file((char *)ev.data, &ci);
mutex_unlock(&mutex_codecthread);
break ;
case CODEC_LOAD:
logf("Codec start");
codecsize = cur_ti->codecsize;
if (codecsize == 0) {
logf("Codec slot is empty!");
/* Wait for the pcm buffer to go empty */
while (pcm_is_playing())
yield();
audio_stop_playback();
break ;
}
ci.stop_codec = false;
wrap = (int)&filebuf[filebuflen] - (int)cur_ti->codecbuf;
audio_codec_loaded = true;
mutex_lock(&mutex_codecthread);
current_codec = CODEC_IDX_AUDIO;
status = codec_load_ram(cur_ti->codecbuf, codecsize,
&filebuf[0], wrap, &ci);
mutex_unlock(&mutex_codecthread);
break ;
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
while (voice_codec_loaded) {
if (current_codec != CODEC_IDX_VOICE)
swap_codec();
sleep(1);
}
logf("USB: Audio codec");
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&codec_queue);
break ;
#endif
}
audio_codec_loaded = false;
switch (ev.id) {
case CODEC_LOAD_DISK:
case CODEC_LOAD:
if (status != CODEC_OK) {
logf("Codec failure");
audio_stop_playback();
splash(HZ*2, true, "Codec failure");
} else {
logf("Codec finished");
}
if (playing && !ci.stop_codec && !ci.reload_codec) {
audio_change_track();
continue ;
} else if (ci.stop_codec) {
//playing = false;
}
//queue_post(&audio_queue, AUDIO_CODEC_DONE, (void *)status);
}
}
}
static void reset_buffer(void)
{
filebuf = &audiobuf[MALLOC_BUFSIZE];
filebuflen = audiobufend - audiobuf - pcmbuf_get_bufsize()
- PCMBUF_GUARD - MALLOC_BUFSIZE - GUARD_BUFSIZE;
if (talk_get_bufsize() && voice_codec_loaded)
{
filebuf = &filebuf[talk_get_bufsize()];
filebuflen -= 2*CODEC_IRAM_SIZE + 2*CODEC_SIZE + talk_get_bufsize();
}
}
void voice_codec_thread(void)
{
struct event ev;
int status;
current_codec = CODEC_IDX_AUDIO;
voice_codec_loaded = false;
while (1) {
status = 0;
voice_is_playing = false;
queue_wait(&voice_codec_queue, &ev);
switch (ev.id) {
case CODEC_LOAD_DISK:
logf("Loading voice codec");
audio_stop_playback();
mutex_lock(&mutex_codecthread);
current_codec = CODEC_IDX_VOICE;
dsp_configure(DSP_RESET, 0);
ci.configure(CODEC_DSP_ENABLE, (bool *)true);
voice_remaining = 0;
voice_getmore = NULL;
voice_codec_loaded = true;
reset_buffer();
ci_voice.stop_codec = false;
status = codec_load_file((char *)ev.data, &ci_voice);
logf("Voice codec finished");
audio_stop_playback();
mutex_unlock(&mutex_codecthread);
current_codec = CODEC_IDX_AUDIO;
voice_codec_loaded = false;
reset_buffer();
break ;
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
logf("USB: Voice codec");
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&voice_codec_queue);
break ;
#endif
}
}
}
void voice_init(void)
{
while (voice_codec_loaded)
{
logf("Terminating voice codec");
ci_voice.stop_codec = true;
if (current_codec != CODEC_IDX_VOICE)
swap_codec();
sleep(1);
}
if (!talk_get_bufsize())
return ;
logf("Starting voice codec");
queue_post(&voice_codec_queue, CODEC_LOAD_DISK, (void *)CODEC_MPA_L3);
while (!voice_codec_loaded)
sleep(1);
}
struct mp3entry* audio_current_track(void)
{
// logf("audio_current_track");
if (track_count > 0 && cur_ti->taginfo_ready)
return (struct mp3entry *)&cur_ti->id3;
else
return NULL;
}
struct mp3entry* audio_next_track(void)
{
int next_idx = track_ridx + 1;
if (track_count == 0)
return NULL;
if (next_idx >= MAX_TRACK)
next_idx = 0;
if (!tracks[next_idx].taginfo_ready)
return NULL;
//logf("audio_next_track");
return &tracks[next_idx].id3;
}
bool audio_has_changed_track(void)
{
if (track_changed && track_count > 0 && playing) {
if (!cur_ti->taginfo_ready)
return false;
track_changed = false;
return true;
}
return false;
}
void audio_play(int offset)
{
logf("audio_play");
last_index = -1;
paused = false;
pcm_play_pause(true);
queue_post(&audio_queue, AUDIO_PLAY, (void *)offset);
}
void audio_stop(void)
{
logf("audio_stop");
queue_post(&audio_queue, AUDIO_STOP, 0);
while (playing || audio_codec_loaded)
yield();
}
bool mp3_pause_done(void)
{
return paused;
}
void audio_pause(void)
{
queue_post(&audio_queue, AUDIO_PAUSE, 0);
}
void audio_resume(void)
{
queue_post(&audio_queue, AUDIO_RESUME, 0);
}
void audio_next(void)
{
queue_post(&audio_queue, AUDIO_NEXT, 0);
}
void audio_prev(void)
{
queue_post(&audio_queue, AUDIO_PREV, 0);
}
void audio_ff_rewind(int newpos)
{
logf("rewind: %d", newpos);
if (playing) {
pcmbuf_play_stop();
ci.seek_time = newpos+1;
}
}
void audio_flush_and_reload_tracks(void)
{
logf("flush & reload");
queue_post(&audio_queue, 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;
return ret;
}
int audio_get_file_pos(void)
{
return 0;
}
/* Copied from mpeg.c. Should be moved somewhere else. */
static void mp3_set_elapsed(struct mp3entry* id3)
{
if ( id3->vbr ) {
if ( id3->has_toc ) {
/* calculate elapsed time using TOC */
int i;
unsigned int remainder, plen, relpos, nextpos;
/* find wich percent we're at */
for (i=0; i<100; i++ )
{
if ( id3->offset < 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 */
id3->elapsed = id3->offset / (id3->bitrate / 8);
}
/* Copied from mpeg.c. Should be moved somewhere else. */
int mp3_get_file_pos(void)
{
int pos = -1;
struct mp3entry *id3 = audio_current_track();
if (id3->vbr)
{
if (id3->has_toc)
{
/* Use the TOC to find the new position */
unsigned int percent, remainder;
int curtoc, nexttoc, plen;
percent = (id3->elapsed*100)/id3->length;
if (percent > 99)
percent = 99;
curtoc = id3->toc[percent];
if (percent < 99)
nexttoc = id3->toc[percent+1];
else
nexttoc = 256;
pos = (id3->filesize/256)*curtoc;
/* Use the remainder to get a more accurate position */
remainder = (id3->elapsed*100)%id3->length;
remainder = (remainder*100)/id3->length;
plen = (nexttoc - curtoc)*(id3->filesize/256);
pos += (plen/100)*remainder;
}
else
{
/* No TOC exists, estimate the new position */
pos = (id3->filesize / (id3->length / 1000)) *
(id3->elapsed / 1000);
}
}
else if (id3->bitrate)
pos = id3->elapsed * (id3->bitrate / 8);
else
{
return -1;
}
if (pos >= (int)(id3->filesize - id3->id3v1len))
{
/* Don't seek right to the end of the file so that we can
transition properly to the next song */
pos = id3->filesize - id3->id3v1len - 1;
}
else if (pos < (int)id3->first_frame_offset)
{
/* skip past id3v2 tag and other leading garbage */
pos = id3->first_frame_offset;
}
return pos;
}
void mp3_play_data(const unsigned char* start, int size,
void (*get_more)(unsigned char** start, int* size))
{
voice_getmore = get_more;
voicebuf = (unsigned char *)start;
voice_remaining = size;
voice_is_playing = true;
pcmbuf_reset_mixpos();
}
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 type)
{
long size;
bool was_playing = playing;
int offset = 0;
static const int lookup[] = {1, 2, 4, 6, 8, 10, 12, 14};
int seconds = lookup[global_settings.crossfade_duration];
/* Store the track resume position */
if (playing)
offset = cur_ti->id3.offset;
if (type == CROSSFADE_MODE_OFF)
seconds = 1;
/* Buffer has to be at least 2s long. */
seconds += 2;
logf("buf len: %d", seconds);
size = seconds * (NATIVE_FREQUENCY*4);
if (pcmbuf_get_bufsize() == size)
return ;
/* Playback has to be stopped before changing the buffer size. */
audio_stop_playback();
/* Re-initialize audio system. */
if (was_playing)
splash(0, true, str(LANG_RESTARTING_PLAYBACK));
pcmbuf_init(size);
pcmbuf_crossfade_enable(type != CROSSFADE_MODE_OFF);
reset_buffer();
logf("abuf:%dB", pcmbuf_get_bufsize());
logf("fbuf:%dB", filebuflen);
voice_init();
/* Restart playback. */
if (was_playing) {
audio_play(offset);
/* Wait for the playback to start again (and display the splash
screen during that period. */
playing = true;
while (playing && !audio_codec_loaded)
yield();
}
}
void mpeg_id3_options(bool _v1first)
{
v1first = _v1first;
}
void test_buffer_event(struct mp3entry *id3, bool last_track)
{
(void)id3;
(void)last_track;
logf("be:%d%s", last_track, id3->path);
}
void test_unbuffer_event(struct mp3entry *id3, bool last_track)
{
(void)id3;
(void)last_track;
logf("ube:%d%s", last_track, id3->path);
}
void audio_init(void)
{
static bool voicetagtrue = true;
logf("audio api init");
pcm_init();
filebufused = 0;
filling = false;
current_codec = CODEC_IDX_AUDIO;
filebuf = &audiobuf[MALLOC_BUFSIZE];
playing = false;
audio_codec_loaded = false;
voice_is_playing = false;
paused = 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];
audio_set_track_buffer_event(test_buffer_event);
audio_set_track_unbuffer_event(test_unbuffer_event);
/* Initialize codec api. */
ci.read_filebuf = codec_filebuf_callback;
ci.pcmbuf_insert = pcmbuf_insert_buffer;
ci.pcmbuf_insert_split = codec_pcmbuf_insert_split_callback;
ci.get_codec_memory = get_codec_memory_callback;
ci.request_buffer = codec_request_buffer_callback;
ci.advance_buffer = codec_advance_buffer_callback;
ci.advance_buffer_loc = codec_advance_buffer_loc_callback;
ci.request_next_track = codec_request_next_track_callback;
ci.mp3_get_filepos = codec_mp3_get_filepos_callback;
ci.seek_buffer = codec_seek_buffer_callback;
ci.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;
memcpy(&ci_voice, &ci, sizeof(struct codec_api));
memset(&id3_voice, 0, sizeof(struct mp3entry));
ci_voice.taginfo_ready = &voicetagtrue;
ci_voice.id3 = &id3_voice;
ci_voice.pcmbuf_insert = codec_pcmbuf_insert_callback;
id3_voice.frequency = 11200;
id3_voice.length = 1000000L;
mutex_init(&mutex_bufferfill);
mutex_init(&mutex_codecthread);
queue_init(&audio_queue);
queue_init(&codec_queue);
queue_init(&voice_codec_queue);
create_thread(codec_thread, codec_stack, sizeof(codec_stack),
codec_thread_name);
create_thread(voice_codec_thread, voice_codec_stack,
sizeof(voice_codec_stack), voice_codec_thread_name);
create_thread(audio_thread, audio_stack, sizeof(audio_stack),
audio_thread_name);
}
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