apps/playback.c

/***************************************************************************
 *             __________               __   ___.
 *   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"
#define CODEC_SID      "/.rockbox/codecs/sid.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, bool may_yield);
static bool 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;
    
    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;
    }
    
    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:
                if (playing)
                    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: Voice codec");
                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;
    /* Start with the previously playing track's data and our data */
    size_t amount;
    
    buf_back = buf_ridx;
    amount = ci.curpos;
    if (buf_ridx < buf_widx)
        buf_back += filebuflen;
    buf_back -= buf_widx;
    
    /* If we're not just resetting the current track */
    if (new_track_ridx != old_track_ridx)
    {
        /* Need to wind to before the old track's codec and our filesize */
        amount += tracks[old_track_ridx].codecsize;
        amount += tracks[new_track_ridx].filesize;

        /* Rewind the old track to its beginning */
        tracks[old_track_ridx].available =
            tracks[old_track_ridx].filesize - tracks[old_track_ridx].filerem;
    }

    /* If the codec was ever buffered */
    if (tracks[new_track_ridx].codecsize)
    {
        /* Add the codec to the needed size */
        amount += tracks[new_track_ridx].codecsize;
        tracks[new_track_ridx].has_codec = true;
    }

    /* Then collect all data from tracks between new and old */
    amount += 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 */
    /* Currently this can cause some problems (logf in reverse order):
     * Codec load error:-1
     * Codec load disk
     * Codec: Unsupported
     * Codec finished
     * New codec:0/3
     * Clearing tracks:7/7, 1
     * Forcing rebuffer
     * Check new track buffer
     * Request new track
     * Clearing tracks:5/5, 0
     * Starting buffer fill
     * Clearing tracks:5/5, 1
     * Re-buffering song w/seek
     */
    //if (!filling)
    //    queue_post(&codec_callback_queue, Q_CODEC_REQUEST_PENDING, 0);
    
    /* Stop in progress fill, and clear open file descriptor */
    if (current_fd >= 0)
    {
        close(current_fd);
        current_fd = -1;
    }
    filling = false;

    /* Reset buffer and track pointers */
    buf_ridx = buf_widx = 0;
    track_widx = track_ridx;
    cur_ti = &tracks[track_ridx];
    audio_clear_track_entries(true, true, false);
    filebufused = 0;

    /* Fill the buffer */
    last_peek_offset = -1;
    cur_ti->filesize = 0;
    cur_ti->start_pos = 0;
    ci.curpos = 0;

    if (!cur_ti->taginfo_ready)
        memset(&cur_ti->id3, 0, sizeof(struct mp3entry));

    audio_fill_file_buffer(false, true, 0);
}

static void audio_check_new_track(void)
{
    int track_count = audio_track_count();
    int old_track_ridx = track_ridx;
    bool forward;

    if (dir_skip)
    {
        dir_skip = false;
        if (playlist_next_dir(ci.new_track))
        {
            ci.new_track = 0;
            cur_ti->taginfo_ready = false;
            audio_rebuffer();
            goto skip_done;
        }
        else
        {
            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;

    if (playlist_next(ci.new_track) < 0)
    {
        queue_post(&codec_callback_queue, Q_CODEC_REQUEST_FAILED, 0);
        return;
    }

    if (new_playlist)
    {
        ci.new_track = 1;
        new_playlist = false;
    }

    track_ridx += ci.new_track;
    track_ridx &= MAX_TRACK_MASK;

    /* Save the old track */
    prev_ti = cur_ti;
    /* Move to the new track */
    cur_ti = &tracks[track_ridx];

    if (automatic_skip)
        playlist_end = false;

    track_changed = !automatic_skip;

    /* If it is not safe to even skip this many track entries */
    if (ci.new_track >= track_count || ci.new_track <= track_count - MAX_TRACK)
    {
        ci.new_track = 0;
        cur_ti->taginfo_ready = false;
        audio_rebuffer();
        goto skip_done;
    }

    forward = ci.new_track > 0;
    ci.new_track = 0;

    /* If the target track is clearly not in memory */
    if (cur_ti->filesize == 0 || !cur_ti->taginfo_ready)
    {
        audio_rebuffer();
        goto skip_done;
    }

    /* The track may be in memory, see if it really is */
    if (forward)
    {
        if (!buffer_wind_forward(track_ridx, old_track_ridx))
            audio_rebuffer();
    }
    else
    {
        int cur_idx = track_ridx;
        bool taginfo_ready = true;
        bool wrap = track_ridx > old_track_ridx;
        
        while (1) 
        {
            cur_idx++;
            cur_idx &= MAX_TRACK_MASK;
            if (!(wrap || cur_idx < old_track_ridx))
                break;

            /* If we hit a track in between without valid tag info, bail */
            if (!tracks[cur_idx].taginfo_ready)
            {
                taginfo_ready = false;
                break;
            }

            tracks[cur_idx].available = tracks[cur_idx].filesize;
            if (tracks[cur_idx].codecsize)
                tracks[cur_idx].has_codec = true;
        }
        if (taginfo_ready)
        {
            if (!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. */
    track_widx = track_ridx;
    filebufused = 0;
    buf_widx = buf_ridx = 0;

    last_peek_offset = 0;
    filling = false;
    initialize_buffer_fill(true);

    if (newpos > AUDIO_REBUFFER_GUESS_SIZE) 
    {
        buf_ridx += AUDIO_REBUFFER_GUESS_SIZE;
        cur_ti->start_pos = newpos - AUDIO_REBUFFER_GUESS_SIZE;
    } 
    else 
    {
        buf_ridx += newpos;
        cur_ti->start_pos = 0;
    }

    cur_ti->filerem = cur_ti->filesize - cur_ti->start_pos;
    cur_ti->available = 0;

    lseek(current_fd, cur_ti->start_pos, SEEK_SET);

    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();

        track_widx++;
        track_widx &= MAX_TRACK_MASK;

        tracks[track_widx].filesize = 0;
    } 
    else 
    {
        logf("Partially buf:%dB",
                tracks[track_widx].filesize - tracks[track_widx].filerem);
    }
}

static void codec_discard_codec_callback(void)
{
    if (cur_ti->has_codec) 
    {
        cur_ti->has_codec = false;
        filebufused -= cur_ti->codecsize;
        buf_ridx += cur_ti->codecsize;
        if (buf_ridx >= filebuflen)
            buf_ridx -= filebuflen;
    }

#if 0
    /* Check if a buffer desync has happened, log it and stop playback. */
    if (buf_ridx != cur_ti->buf_idx)
    {
        int offset = cur_ti->buf_idx - buf_ridx;
        size_t new_used = filebufused - offset;
        
        logf("Buf off :%d=%d-%d", offset, cur_ti->buf_idx, buf_ridx);
        logf("Used off:%d",filebufused - new_used);
        
        /* This is a fatal internal error and it's not safe to 
         * continue playback. */
        ci.stop_codec = true;
        queue_post(&audio_queue, Q_AUDIO_STOP, 0);
    }
#endif
}

static 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;
        case AFMT_SID:
            logf("Codec: SID");
            return CODEC_SID;
        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) & MAX_TRACK_MASK;
            
            /* If the previous codec is the same as this one, there is no need
             * to put another copy of it on the file buffer */
            if (get_codec_base_type(tracks[track_widx].id3.codectype) ==
                    get_codec_base_type(tracks[prev_track].id3.codectype)
                && audio_codec_loaded)
            {
                logf("Reusing prev. codec");
                return true;
            }
        }
    }

    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)
    {
        next_idx++;
        next_idx &= MAX_TRACK_MASK;

        if (tracks[next_idx].taginfo_ready)
            return true;
    }

    trackname = playlist_peek(last_peek_offset + 1);
    if (!trackname)
        return false;

    fd = open(trackname, O_RDONLY);
    if (fd < 0)
        return false;

    status = get_metadata(&tracks[next_idx],fd,trackname,v1first);
    /* Preload the glyphs in the tags */
    if (status) 
    {
        if (tracks[next_idx].id3.title)
            lcd_getstringsize(tracks[next_idx].id3.title, NULL, NULL);
        if (tracks[next_idx].id3.artist)
            lcd_getstringsize(tracks[next_idx].id3.artist, NULL, NULL);
        if (tracks[next_idx].id3.album)
            lcd_getstringsize(tracks[next_idx].id3.album, NULL, NULL);
    }
    close(fd);

    return status;
}

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);