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rockbox/apps/playback.c
Hardeep Sidhu 6a267a8be7 Don't initiate track change if at end of playlist 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9868 a1c6a512-1295-4272-9138-f99709370657
2006-05-03 16:33:23 +00:00

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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 "talk.h"
#ifdef CONFIG_TUNER
#include "radio.h"
#endif
#include "splash.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 CODEC_AAC "/.rockbox/codecs/aac.codec"
#define CODEC_SHN "/.rockbox/codecs/shorten.codec"
#define CODEC_AIFF "/.rockbox/codecs/aiff.codec"
/* 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)
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,
};
/* As defined in plugins/lib/xxx2wav.h */
#define MALLOC_BUFSIZE (512*1024)
#define GUARD_BUFSIZE (32*1024)
/* As defined in plugin.lds */
#if CONFIG_CPU == PP5020 || CONFIG_CPU == PP5002
#define CODEC_IRAM_ORIGIN 0x4000c000
#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
/* 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)]
IBSS_ATTR;
static const char codec_thread_name[] = "codec";
/* Voice codec thread. */
static struct event_queue voice_codec_queue;
static long voice_codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)]
IBSS_ATTR;
static const char voice_codec_thread_name[] = "voice codec";
struct voice_info {
void (*callback)(unsigned char **start, int *size);
int size;
char *buf;
};
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 int voice_thread_num = -1;
/* 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);
static void playback_init(void);
/* Configuration */
static size_t conf_watermark;
static size_t conf_filechunk;
static size_t buffer_margin;
static bool v1first = false;
static void mp3_set_elapsed(struct mp3entry* id3);
static int mp3_get_file_pos(void);
static void audio_clear_track_entries(
bool clear_buffered, bool clear_unbuffered);
static void initialize_buffer_fill(bool clear_tracks);
static void audio_fill_file_buffer(
bool start_play, bool rebuffer, size_t offset);
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);
}
#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
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;
}
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;
}
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 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* get_voice_memory_callback(size_t *size)
{
*size = 0;
return NULL;
}
static void* get_codec_memory_callback(size_t *size)
{
*size = MALLOC_BUFSIZE;
if (voice_codec_loaded)
return &audiobuf[talk_get_bufsize()];
else
return audiobuf;
}
static void pcmbuf_position_callback(size_t size) ICODE_ATTR;
static void 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 voice_set_elapsed_callback(unsigned int value)
{
(void)value;
}
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 voice_set_offset_callback(size_t value)
{
(void)value;
}
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 bool filebuf_is_lowdata(void)
{
return filebufused < AUDIO_FILEBUF_CRITICAL;
}
static bool have_tracks(void)
{
return track_ridx != track_widx || cur_ti->filesize;
}
static bool 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;
else
return true;
}
int audio_track_count(void)
{
if (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;
} else
return 0;
}
static void 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)
queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0);
}
static size_t voice_filebuf_callback(void *ptr, size_t size)
{
(void)ptr;
(void)size;
return 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)
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 */
advance_buffer_counters(copy_n);
/* Return the actual amount of data copied to the buffer */
return copy_n;
}
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_codec_queue, &ev, 0);
else if (playing)
{
queue_wait_w_tmo(&voice_codec_queue, &ev, 0);
if (ev.id == SYS_TIMEOUT)
ev.id = Q_AUDIO_PLAY;
}
else
queue_wait(&voice_codec_queue, &ev);
switch (ev.id) {
case Q_AUDIO_PLAY:
swap_codec();
break;
case 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:
logf("USB: Audio core");
usb_acknowledge(SYS_USB_CONNECTED_ACK);
if (audio_codec_loaded)
swap_codec();
usb_wait_for_disconnect(&voice_codec_queue);
break;
case 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:
goto voice_play_clip;
}
}
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)
{
queue_post(&voice_codec_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;
}
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)
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];
}
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;
}
/* Count the data BETWEEN the selected tracks */
static size_t 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 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 += 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 buffer_wind_backward(int new_track_ridx, int old_track_ridx) {
/* Available buffer data */
size_t buf_back = buf_ridx;
if (buf_ridx < buf_widx)
buf_back += filebuflen;
buf_back -= buf_widx;
/* Start with the previously playing track's data and our data */
size_t amount = ci.curpos;
/* 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 += 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;
}
static void audio_rebuffer(void)
{
logf("Forcing rebuffer");
/* Notify the codec that this will take a while */
if (!filling)
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_PENDING, 0);
/* Stop in progress fill, and clear open file descriptor */
close(current_fd);
current_fd = -1;
filling = false;
/* Reset buffer and track pointers */
buf_ridx = buf_widx = 0;
track_widx = track_ridx;
audio_clear_track_entries(false, true);
filebufused = 0;
/* Fill the buffer */
last_peek_offset = -1;
cur_ti->filesize = 0;
cur_ti->start_pos = 0;
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
{
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)
{
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)
{
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0);
return;
}
}
/* Update the playlist */
last_peek_offset -= ci.new_track;
playlist_next(ci.new_track);
if (new_playlist)
{
ci.new_track = 1;
new_playlist = false;
}
track_ridx+=ci.new_track;
if (track_ridx >= MAX_TRACK)
track_ridx -= MAX_TRACK;
else if (track_ridx < 0)
track_ridx += MAX_TRACK;
/* Save the old track */
prev_ti = cur_ti;
/* Move to the new track */
cur_ti = &tracks[track_ridx];
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 (!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) {
if (++cur_idx >= MAX_TRACK)
cur_idx -= MAX_TRACK;
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 (!buffer_wind_backward(track_ridx, old_track_ridx))
audio_rebuffer();
}
else
{
cur_ti->taginfo_ready = false;
audio_rebuffer();
}
}
skip_done:
audio_update_trackinfo();
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0);
}
static void 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) {
logf("Open 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. */
filebufused = 0;
buf_widx = cur_ti->buf_idx + newpos;
while (buf_widx >= filebuflen)
buf_widx -= filebuflen;
buf_ridx = buf_widx;
/* Write to the now current track */
track_widx = track_ridx;
last_peek_offset = 0;
initialize_buffer_fill(true);
if (newpos > AUDIO_REBUFFER_GUESS_SIZE)
cur_ti->start_pos = newpos - AUDIO_REBUFFER_GUESS_SIZE;
else
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);
queue_post(&codec_callback_queue, Q_CODEC_REQUEST_COMPLETE, 0);
}
static void voice_advance_buffer_callback(size_t amount)
{
amount = MIN(amount, voice_remaining);
voicebuf += amount;
voice_remaining -= amount;
}
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;
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:
ci.stop_codec = true;
case Q_CODEC_REQUEST_COMPLETE:
return;
default:
logf("Bad event on ccq");
ci.stop_codec = true;
return;
}
}
advance_buffer_counters(amount);
codec_set_offset_callback(ci.curpos);
}
static void voice_advance_buffer_loc_callback(void *ptr)
{
size_t amount = (size_t)ptr - (size_t)voicebuf;
voice_advance_buffer_callback(amount);
}
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);
}
static off_t voice_mp3_get_filepos_callback(int newtime)
{
(void)newtime;
return 0;
}
static off_t codec_mp3_get_filepos_callback(int newtime)
{
off_t newpos;
cur_ti->id3.elapsed = newtime;
newpos = mp3_get_file_pos();
return newpos;
}
static void voice_do_nothing(void)
{
return;
}
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 voice_seek_buffer_callback(size_t newpos)
{
(void)newpos;
return false;
}
static bool codec_seek_buffer_callback(size_t newpos)
{
int difference;
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;
queue_post(&audio_queue, Q_AUDIO_REBUFFER_SEEK, (void *)newpos);
queue_wait(&codec_callback_queue, &ev);
switch (ev.id)
{
case Q_CODEC_REQUEST_COMPLETE:
return true;
case Q_CODEC_REQUEST_FAILED:
ci.stop_codec = true;
return false;
default:
logf("Bad event on ccq");
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 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 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;
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;
}
static void codec_track_changed(void)
{
automatic_skip = false;
track_changed = true;
queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0);
}
static void pcmbuf_track_changed_callback(void)
{
pcmbuf_set_position_callback(NULL);
codec_track_changed();
}
/* 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 yield_codecs(void)
{
yield();
if (!queue_empty(&audio_queue)) return true;
while ((pcmbuf_is_crossfade_active() || pcmbuf_is_lowdata())
&& !ci.stop_codec && playing && !filebuf_is_lowdata())
{
sleep(1);
if (!queue_empty(&audio_queue)) return true;
}
return false;
}
/* FIXME: This code should be made more generic and move to metadata.c */
static void 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) {
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;
if (buf_widx >= filebuflen)
buf_widx -= filebuflen;
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 || yield_codecs())
break;
}
if (tracks[track_widx].filerem == 0) {
logf("Finished buf:%dB", tracks[track_widx].filesize);
close(current_fd);
current_fd = -1;
strip_id3v1_tag();
if (++track_widx >= MAX_TRACK)
track_widx = 0;
tracks[track_widx].filesize = 0;
} else {
logf("Partially buf:%dB",
tracks[track_widx].filesize - tracks[track_widx].filerem);
}
}
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;
}
/* Check if a buffer desync has happened, and log it */
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);
buf_ridx += offset;
/* Reset the buffer used amount based on the read and write pointers */
filebufused = track_widx;
if (track_widx < track_ridx)
filebufused += filebuflen;
filebufused -= track_ridx;
/* If that was not the same amount as the track was off, log it */
if (new_used != filebufused) {
logf("Used off:%d",filebufused - new_used);
}
}
}
static const char *get_codec_path(int codectype) {
switch (codectype) {
case AFMT_OGG_VORBIS:
logf("Codec: Vorbis");
return CODEC_VORBIS;
case AFMT_MPA_L1:
case AFMT_MPA_L2:
case AFMT_MPA_L3:
logf("Codec: MPA L1/L2/L3");
return CODEC_MPA_L3;
case AFMT_PCM_WAV:
logf("Codec: PCM WAV");
return CODEC_WAV;
case AFMT_FLAC:
logf("Codec: FLAC");
return CODEC_FLAC;
case AFMT_A52:
logf("Codec: A52");
return CODEC_A52;
case AFMT_MPC:
logf("Codec: Musepack");
return CODEC_MPC;
case AFMT_WAVPACK:
logf("Codec: WAVPACK");
return CODEC_WAVPACK;
case AFMT_ALAC:
logf("Codec: ALAC");
return CODEC_ALAC;
case AFMT_AAC:
logf("Codec: AAC");
return CODEC_AAC;
case AFMT_SHN:
logf("Codec: SHN");
return CODEC_SHN;
case AFMT_AIFF:
logf("Codec: PCM AIFF");
return CODEC_AIFF;
default:
logf("Codec: Unsupported");
return NULL;
}
}
static bool loadcodec(bool start_play)
{
size_t size;
int fd;
int rc;
size_t copy_n;
int prev_track;
const char *codec_path = get_codec_path(tracks[track_widx].id3.codectype);
if (codec_path == 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;
queue_post(&codec_queue, Q_CODEC_LOAD_DISK, (void *)codec_path);
return true;
}
else
{
/* If we already have another track than this one buffered */
if (track_widx != track_ridx) {
prev_track = track_widx - 1;
if (prev_track < 0)
prev_track += MAX_TRACK;
/* 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))
{
logf("Reusing prev. codec");
return true;
}
}
}
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;
yield_codecs();
}
tracks[track_widx].has_codec = true;
close(fd);
logf("Done: %dB", size);
return true;
}
static bool read_next_metadata(void)
{
int fd;
char *trackname;
int next_idx;
int status;
next_idx = track_widx;
if (tracks[next_idx].taginfo_ready)
{
if (++next_idx >= MAX_TRACK)
next_idx -= MAX_TRACK;
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;
}
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 (!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;
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 (!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_L2:
case AFMT_MPA_L3:
lseek(current_fd, offset, SEEK_SET);
tracks[track_widx].id3.offset = offset;
mp3_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:
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 void audio_clear_track_entries(
bool clear_buffered, bool clear_unbuffered)
{
int cur_idx = track_widx;
int last_idx = -1;
logf("Clearing tracks:%d/%d, %d", track_ridx, track_widx, clear_unbuffered);
/* Loop over all tracks from write-to-read */
while (1) {
if (++cur_idx >= MAX_TRACK)
cur_idx -= MAX_TRACK;
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)
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));
}
}
static void 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());
}
filebufused = 0;
playing = false;
filling = false;
paused = false;
stop_codec_flush();
if (current_fd >= 0) {
close(current_fd);
current_fd = -1;
}
/* Mark all entries null. */
memset(tracks, 0, sizeof(struct track_info) * MAX_TRACK);
}
static void audio_play_start(size_t offset)
{
#ifdef CONFIG_TUNER
/* check if radio is playing */
if (get_radio_status() != FMRADIO_OFF)
radio_stop();
#endif
/* Wait for any previously playing audio to flush - TODO: Not necessary? */
while (audio_codec_loaded)
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);
}
/* Send callback events to notify about new tracks. */
static void generate_postbuffer_events(void)
{
int cur_idx;
int last_idx = -1;
logf("Postbuffer:%d/%d",track_ridx,track_widx);
if (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;
if (++cur_idx >= MAX_TRACK)
cur_idx -= MAX_TRACK;
}
if (last_idx >= 0 && !tracks[last_idx].event_sent)
{
tracks[last_idx].event_sent = true;
if (track_buffer_callback)
track_buffer_callback(&tracks[last_idx].id3, true);
}
}
}
static void initialize_buffer_fill(bool clear_tracks)
{
/* Don't initialize if we're already initialized */
if (filling)
return ;
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?
*/
if (buf_ridx > cur_ti->buf_idx)
cur_ti->start_pos = buf_ridx - cur_ti->buf_idx;
if (clear_tracks)
audio_clear_track_entries(true, false);
/* Save the current resume position once. */
playlist_update_resume_info(audio_current_track());
filling = true;
}
static void audio_fill_file_buffer(
bool start_play, bool rebuffer, size_t offset)
{
bool had_next_track = audio_next_track() != NULL;
initialize_buffer_fill(!start_play);
/* 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)
{
read_next_metadata();
generate_postbuffer_events();
filling = false;
#ifndef SIMULATOR
if (playing)
ata_sleep();
#endif
}
}
static void track_skip_done(bool was_manual)
{
/* Manual track change (always crossfade or flush audio). */
if (was_manual)
{
pcmbuf_crossfade_init(true);
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(pcmbuf_position_callback);
pcmbuf_set_event_handler(pcmbuf_track_changed_callback);
}
}
static bool 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);
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:
track_skip_done(!automatic_skip);
return true;
case Q_CODEC_REQUEST_FAILED:
ci.new_track = 0;
ci.stop_codec = true;
return false;
default:
logf("Bad event on ccq");
ci.stop_codec = true;
return false;
}
}
static bool voice_request_next_track_callback(void)
{
ci_voice.new_track = 0;
return true;
}
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 (!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;
}
}
/* Invalidates all but currently playing track. */
void audio_invalidate_tracks(void)
{
if (have_tracks()) {
last_peek_offset = 0;
playlist_end = false;
track_widx = track_ridx;
audio_clear_track_entries(true, true);
/* If the current track is fully buffered, advance the write pointer */
if (tracks[track_widx].filerem == 0)
if (++track_widx >= MAX_TRACK)
track_widx -= MAX_TRACK;
/* 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;
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 (have_tracks()) {
playlist_end = false;
track_widx = track_ridx;
audio_clear_track_entries(true, true);
if (++track_widx >= MAX_TRACK)
track_widx -= MAX_TRACK;
/* 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 initiate_track_change(long direction)
{
if (playlist_check(direction))
{
playlist_end = false;
ci.new_track += direction;
}
}
static void initiate_dir_change(long direction)
{
playlist_end = false;
dir_skip = true;
ci.new_track = direction;
}
void audio_thread(void)
{
struct event ev;
/* At first initialize audio system in background. */
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);
switch (ev.id) {
case 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:
logf("starting...");
audio_play_start((size_t)ev.data);
break ;
case Q_AUDIO_STOP:
logf("audio_stop");
audio_stop_playback();
break ;
case Q_AUDIO_PAUSE:
logf("audio_%s",ev.data?"pause":"resume");
pcmbuf_pause((bool)ev.data);
paused = (bool)ev.data;
break ;
case Q_AUDIO_SKIP:
logf("audio_skip");
initiate_track_change((long)ev.data);
break;
case Q_AUDIO_PRE_FF_REWIND:
if (!playing)
break;
logf("pre_ff_rewind");
pcmbuf_pause(true);
break;
case Q_AUDIO_FF_REWIND:
if (!playing)
break ;
logf("ff_rewind");
ci.seek_time = (long)ev.data+1;
break ;
case Q_AUDIO_REBUFFER_SEEK:
logf("Re-buffering song w/seek");
rebuffer_and_seek((size_t)ev.data);
break;
case Q_AUDIO_CHECK_NEW_TRACK:
logf("Check new track buffer");
audio_check_new_track();
break;
case Q_AUDIO_DIR_SKIP:
logf("audio_dir_skip");
playlist_end = false;
if (global_settings.beep)
pcmbuf_beep(5000, 100, 2500*global_settings.beep);
initiate_dir_change((long)ev.data);
break;
case Q_AUDIO_NEW_PLAYLIST:
logf("new_playlist");
audio_new_playlist();
break;
case Q_AUDIO_FLUSH:
logf("flush & reload");
audio_invalidate_tracks();
break ;
case 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:
logf("USB: Audio core");
audio_stop_playback();
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&audio_queue);
break ;
#endif
case SYS_TIMEOUT:
break;
}
}
}
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:
logf("Codec load disk");
audio_codec_loaded = true;
if (voice_codec_loaded)
queue_post(&voice_codec_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:
logf("Codec load ram");
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;
queue_post(&codec_queue, Q_AUDIO_PLAY, 0);
break ;
}
audio_codec_loaded = true;
if (voice_codec_loaded)
queue_post(&voice_codec_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 ;
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
queue_clear(&codec_queue);
logf("USB: Audio codec");
usb_acknowledge(SYS_USB_CONNECTED_ACK);
if (voice_codec_loaded)
swap_codec();
usb_wait_for_disconnect(&codec_queue);
break ;
#endif
}
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:
if (playing) {
const char *codec_path;
if (ci.new_track || status != CODEC_OK) {
if (!ci.new_track) {
logf("Codec failure");
gui_syncsplash(HZ*2, true, "Codec failure");
}
if (!load_next_track())
{
queue_post(&codec_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);
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
break;
}
}
if (cur_ti->has_codec)
queue_post(&codec_queue, Q_CODEC_LOAD, 0);
else
{
codec_path = get_codec_path(cur_ti->id3.codectype);
queue_post(&codec_queue,
Q_CODEC_LOAD_DISK, (void *)codec_path);
}
}
}
}
}
static void reset_buffer(void)
{
size_t offset;
filebuf = (char *)&audiobuf[MALLOC_BUFSIZE];
filebuflen = audiobufend - audiobuf - MALLOC_BUFSIZE - GUARD_BUFSIZE -
(pcmbuf_get_bufsize() + get_pcmbuf_descsize() + PCMBUF_MIX_CHUNK * 2);
if (talk_get_bufsize())
{
filebuf = &filebuf[talk_get_bufsize()];
filebuflen -= 2*CODEC_IRAM_SIZE + 2*CODEC_SIZE + talk_get_bufsize();
#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];
}
/* Ensure that everything is aligned */
offset = (-(size_t)filebuf) & 3;
filebuf += offset;
filebuflen -= offset;
filebuflen &= ~3;
}
static void voice_codec_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(CODEC_MPA_L3, &ci_voice);
logf("Voice codec finished");
mutex_unlock(&mutex_codecthread);
voice_codec_loaded = false;
}
}
void voice_init(void)
{
if (!filebuf)
return; /* Audio buffers not yet set up */
if (voice_thread_num >= 0)
{
logf("Terminating voice codec");
remove_thread(voice_thread_num);
if (current_codec == CODEC_IDX_VOICE)
mutex_unlock(&mutex_codecthread);
queue_delete(&voice_codec_queue);
voice_thread_num = -1;
voice_codec_loaded = false;
}
if (!talk_get_bufsize())
return ;
logf("Starting voice codec");
queue_init(&voice_codec_queue);
voice_thread_num = create_thread(voice_codec_thread, voice_codec_stack,
sizeof(voice_codec_stack), voice_codec_thread_name);
while (!voice_codec_loaded)
yield();
}
struct mp3entry* audio_current_track(void)
{
const char *filename;
const char *p;
static struct mp3entry temp_id3;
int cur_idx = track_ridx + ci.new_track;
if (cur_idx >= MAX_TRACK)
cur_idx += MAX_TRACK;
else if (cur_idx < 0)
cur_idx += MAX_TRACK;
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 (!have_tracks())
return NULL;
if (++next_idx >= MAX_TRACK)
next_idx -= MAX_TRACK;
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)
queue_post(&audio_queue, Q_AUDIO_NEW_PLAYLIST, 0);
else
{
if (playing)
audio_stop();
playing = true;
queue_post(&audio_queue, Q_AUDIO_PLAY, (void *)offset);
}
}
void audio_stop(void)
{
queue_post(&audio_queue, Q_AUDIO_STOP, 0);
while (playing || audio_codec_loaded)
yield();
}
bool mp3_pause_done(void)
{
return pcm_is_paused();
}
void audio_pause(void)
{
queue_post(&audio_queue, Q_AUDIO_PAUSE, (void *)true);
}
void audio_resume(void)
{
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);
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);
queue_post(&audio_queue, Q_AUDIO_SKIP, (void *)-1);
}
void audio_next_dir(void)
{
queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)1);
}
void audio_prev_dir(void)
{
queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, (void *)-1);
}
void audio_pre_ff_rewind(void)
{
queue_post(&audio_queue, Q_AUDIO_PRE_FF_REWIND, 0);
}
void audio_ff_rewind(long newpos)
{
queue_post(&audio_queue, Q_AUDIO_FF_REWIND, (int *)newpos);
}
void audio_flush_and_reload_tracks(void)
{
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;
return ret;
}
int audio_get_file_pos(void)
{
return 0;
}
/* TODO: 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 */
if (id3->bitrate != 0)
id3->elapsed = id3->offset / (id3->bitrate / 8);
}
}
/* Copied from mpeg.c. Should be moved somewhere else. */
static 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;
/* 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;
}
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;
queue_post(&voice_codec_queue, Q_VOICE_STOP, 0);
queue_post(&voice_codec_queue, Q_VOICE_PLAY, &voice_clip);
voice_is_playing = true;
voice_boost_cpu(true);
}
void mp3_play_stop(void)
{
queue_post(&voice_codec_queue, Q_VOICE_STOP, 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;
int seconds = 1;
if (!filebuf)
return; /* Audio buffers not yet set up */
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);
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. */
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);
reset_buffer();
logf("abuf:%dB", pcmbuf_get_bufsize());
logf("fbuf:%dB", filebuflen);
voice_init();
/* Restart playback. */
if (was_playing) {
playing = true;
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 mpeg_id3_options(bool _v1first)
{
v1first = _v1first;
}
#if (ROCKBOX_HAS_LOGF == 1)
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 test_track_changed_event(struct mp3entry *id3)
{
(void)id3;
logf("tce:%s", id3->path);
}
#endif
static void 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
#if (ROCKBOX_HAS_LOGF == 1)
audio_set_track_buffer_event(test_buffer_event);
audio_set_track_unbuffer_event(test_unbuffer_event);
audio_set_track_changed_event(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 = 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;
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 = get_voice_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;
create_thread(codec_thread, codec_stack, sizeof(codec_stack),
codec_thread_name);
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];
audio_set_crossfade(global_settings.crossfade);
audio_is_initialized = true;
sound_settings_apply();
}
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);
queue_init(&codec_queue);
/* clear, not init to create a private queue */
queue_clear(&codec_callback_queue);
create_thread(audio_thread, audio_stack, sizeof(audio_stack),
audio_thread_name);
}
void audio_init(void)
{
logf("playback system post-init");
queue_post(&audio_queue, Q_AUDIO_POSTINIT, 0);
}
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