60 files changed, 5376 insertions, 3691 deletions
diff --git a/apps/appevents.h b/apps/appevents.h
index fd578b90a2..a303491ae9 100644
--- a/apps/appevents.h
+++ b/apps/appevents.h
@@ -35,6 +35,7 @@ enum {
    PLAYBACK_EVENT_TRACK_BUFFER,
    PLAYBACK_EVENT_TRACK_FINISH,
    PLAYBACK_EVENT_TRACK_CHANGE,
+    PLAYBACK_EVENT_TRACK_SKIP,
    PLAYBACK_EVENT_NEXTTRACKID3_AVAILABLE,
 };
diff --git a/apps/buffering.c b/apps/buffering.c
index 578f0f261a..a130a787ff 100644
--- a/apps/buffering.c
+++ b/apps/buffering.c
@@ -58,7 +58,7 @@
 #define GUARD_BUFSIZE   (32*1024)
 /* Define LOGF_ENABLE to enable logf output in this file */
-/*#define LOGF_ENABLE*/
+/* #define LOGF_ENABLE */
 #include "logf.h"
 /* macros to enable logf for queues
@@ -82,8 +82,6 @@
 #define LOGFQUEUE_SYS_TIMEOUT(...)
 #endif
-/* default point to start buffer refill */
-#define BUFFERING_DEFAULT_WATERMARK      (1024*128)
 /* amount of data to read in one read() call */
 #define BUFFERING_DEFAULT_FILECHUNK      (1024*32)
@@ -94,6 +92,8 @@
 struct memory_handle {
    int id;                    /* A unique ID for the handle */
    enum data_type type;       /* Type of data buffered with this handle */
+    int8_t pinned;             /* Count of references */
+    int8_t signaled;           /* Stop any attempt at waiting to get the data */
    char path[MAX_PATH];       /* Path if data originated in a file */
    int fd;                    /* File descriptor to path (-1 if closed) */
    size_t data;               /* Start index of the handle's data buffer */
@@ -125,9 +125,7 @@ static volatile size_t buf_ridx;  /* current reading position */
 /* Configuration */
 static size_t conf_watermark = 0; /* Level to trigger filebuf fill */
-#if MEMORYSIZE > 8
 static size_t high_watermark = 0; /* High watermark for rebuffer */
-#endif
 /* current memory handle in the linked list. NULL when the list is empty. */
 static struct memory_handle *cur_handle;
@@ -162,7 +160,6 @@ enum
    Q_REBUFFER_HANDLE,   /* Request reset and rebuffering of a handle at a new
                            file starting position. */
    Q_CLOSE_HANDLE,      /* Request closing a handle */
-    Q_BASE_HANDLE,       /* Set the reference handle for buf_useful_data */
    /* Configuration: */
    Q_START_FILL,        /* Request that the buffering thread initiate a buffer
@@ -222,6 +219,9 @@ static inline ssize_t ringbuf_add_cross(uintptr_t p1, size_t v, uintptr_t p2)
 /* Bytes available in the buffer */
 #define BUF_USED ringbuf_sub(buf_widx, buf_ridx)
+/* Real buffer watermark */
+#define BUF_WATERMARK MIN(conf_watermark, high_watermark)
 /*
 LINKED LIST MANAGEMENT
 ======================
@@ -313,6 +313,12 @@ static struct memory_handle *add_handle(size_t data_size, bool can_wrap,
    /* Prevent buffering thread from looking at it */
    new_handle->filerem = 0;
+    /* Handle can be moved by default */
+    new_handle->pinned = 0;
+    /* Handle data can be waited for by default */
+    new_handle->signaled = 0;
    /* only advance the buffer write index of the size of the struct */
    buf_widx = ringbuf_add(buf_widx, sizeof(struct memory_handle));
@@ -364,6 +370,9 @@ static bool rm_handle(const struct memory_handle *h)
                buf_widx = cur_handle->widx;
            }
        } else {
+            /* If we don't find ourselves, this is a seriously incoherent
+               state with a corrupted list and severe action is needed! */
+            panicf("rm_handle fail: %d", h->id);
            return false;
        }
    }
@@ -385,8 +394,7 @@ static struct memory_handle *find_handle(int handle_id)
    /* simple caching because most of the time the requested handle
    will either be the same as the last, or the one after the last */
-    if (cached_handle)
+    if (cached_handle) {
-    {
        if (cached_handle->id == handle_id) {
            return cached_handle;
        } else if (cached_handle->next &&
@@ -618,20 +626,22 @@ static void update_data_counters(struct data_counters *dc)
 static inline bool buffer_is_low(void)
 {
    update_data_counters(NULL);
-    return data_counters.useful < (conf_watermark / 2);
+    return data_counters.useful < BUF_WATERMARK / 2;
 }
 /* Q_BUFFER_HANDLE event and buffer data for the given handle.
   Return whether or not the buffering should continue explicitly.  */
 static bool buffer_handle(int handle_id, size_t to_buffer)
 {
-    logf("buffer_handle(%d)", handle_id);
+    logf("buffer_handle(%d, %lu)", handle_id, (unsigned long)to_buffer);
    struct memory_handle *h = find_handle(handle_id);
    bool stop = false;
    if (!h)
        return true;
+    logf("  type: %d", (int)h->type);
    if (h->filerem == 0) {
        /* nothing left to buffer */
        return true;
@@ -659,13 +669,13 @@ static bool buffer_handle(int handle_id, size_t to_buffer)
        if (!get_metadata((struct mp3entry *)(buffer + h->data),
                          h->fd, h->path)) {
            /* metadata parsing failed: clear the buffer. */
-            memset(buffer + h->data, 0, sizeof(struct mp3entry));
+            wipe_mp3entry((struct mp3entry *)(buffer + h->data));
        }
        close(h->fd);
        h->fd = -1;
        h->filerem = 0;
        h->available = sizeof(struct mp3entry);
-        h->widx += sizeof(struct mp3entry);
+        h->widx = ringbuf_add(h->widx, sizeof(struct mp3entry));
        send_event(BUFFER_EVENT_FINISHED, &handle_id);
        return true;
    }
@@ -698,7 +708,7 @@ static bool buffer_handle(int handle_id, size_t to_buffer)
                break;
            }
-            DEBUGF("File ended %ld bytes early\n", (long)h->filerem);
+            logf("File ended %ld bytes early\n", (long)h->filerem);
            h->filesize -= h->filerem;
            h->filerem = 0;
            break;
@@ -770,22 +780,31 @@ static bool close_handle(int handle_id)
   part of its data buffer or by moving all the data. */
 static void shrink_handle(struct memory_handle *h)
 {
-    size_t delta;
    if (!h)
        return;
-    if (h->type == TYPE_ID3 || h->type == TYPE_CUESHEET ||
+    if (h->type == TYPE_PACKET_AUDIO) {
-        h->type == TYPE_BITMAP || h->type == TYPE_CODEC ||
+        /* only move the handle struct */
-        h->type == TYPE_ATOMIC_AUDIO)
+        /* data is pinned by default - if we start moving packet audio,
-    {
+           the semantics will determine whether or not data is movable
+           but the handle will remain movable in either case */
+        size_t delta = ringbuf_sub(h->ridx, h->data);
+        /* The value of delta might change for alignment reasons */
+        if (!move_handle(&h, &delta, 0, true))
+            return;
+        h->data = ringbuf_add(h->data, delta);
+        h->available -= delta;
+        h->offset += delta;
+    } else {
        /* metadata handle: we can move all of it */
-        if (!h->next || h->filerem != 0)
+        if (h->pinned || !h->next || h->filerem != 0)
-            return; /* Last handle or not finished loading */
+            return; /* Pinned, last handle or not finished loading */
        uintptr_t handle_distance =
            ringbuf_sub(ringbuf_offset(h->next), h->data);
-        delta = handle_distance - h->available;
+        size_t delta = handle_distance - h->available;
        /* The value of delta might change for alignment reasons */
        if (!move_handle(&h, &delta, h->available, h->type==TYPE_CODEC))
@@ -806,15 +825,6 @@ static void shrink_handle(struct memory_handle *h)
            struct bitmap *bmp = (struct bitmap *)&buffer[h->data];
            bmp->data = &buffer[h->data + sizeof(struct bitmap)];
        }
-    } else {
-        /* only move the handle struct */
-        delta = ringbuf_sub(h->ridx, h->data);
-        if (!move_handle(&h, &delta, 0, true))
-            return;
-        h->data = ringbuf_add(h->data, delta);
-        h->available -= delta;
-        h->offset += delta;
    }
 }
@@ -962,6 +972,8 @@ int bufopen(const char *file, size_t offset, enum data_type type,
        mutex_unlock(&llist_mutex);
        return handle_id;
    }
+    else if (type == TYPE_UNKNOWN)
+        return ERR_UNSUPPORTED_TYPE;
 #ifdef APPLICATION
    /* loading code from memory is not supported in application builds */
    else if (type == TYPE_CODEC)
@@ -1083,7 +1095,12 @@ int bufopen(const char *file, size_t offset, enum data_type type,
 */
 int bufalloc(const void *src, size_t size, enum data_type type)
 {
-    int handle_id = ERR_BUFFER_FULL;
+    int handle_id;
+    if (type == TYPE_UNKNOWN)
+        return ERR_UNSUPPORTED_TYPE;
+    handle_id = ERR_BUFFER_FULL;
    mutex_lock(&llist_mutex);
@@ -1124,7 +1141,14 @@ int bufalloc(const void *src, size_t size, enum data_type type)
 bool bufclose(int handle_id)
 {
    logf("bufclose(%d)", handle_id);
+#if 0
+    /* Don't interrupt the buffering thread if the handle is already
+       stale */
+    if (!find_handle(handle_id)) {
+        logf("  handle already closed");
+        return true;
+    }
+#endif
    LOGFQUEUE("buffering >| Q_CLOSE_HANDLE %d", handle_id);
    return queue_send(&buffering_queue, Q_CLOSE_HANDLE, handle_id);
 }
@@ -1236,9 +1260,10 @@ static int seek_handle(struct memory_handle *h, size_t newpos)
 /* Set reading index in handle (relatively to the start of the file).
   Access before the available data will trigger a rebuffer.
-   Return 0 for success and < 0 for failure:
+   Return 0 for success and for failure:
-     -1 if the handle wasn't found
+     ERR_HANDLE_NOT_FOUND if the handle wasn't found
-     -2 if the new requested position was beyond the end of the file
+     ERR_INVALID_VALUE if the new requested position was beyond the end of
+     the file
 */
 int bufseek(int handle_id, size_t newpos)
 {
@@ -1250,7 +1275,11 @@ int bufseek(int handle_id, size_t newpos)
 }
 /* Advance the reading index in a handle (relatively to its current position).
-   Return 0 for success and < 0 for failure */
+   Return 0 for success and for failure:
+     ERR_HANDLE_NOT_FOUND if the handle wasn't found
+     ERR_INVALID_VALUE if the new requested position was beyond the end of
+     the file
+ */
 int bufadvance(int handle_id, off_t offset)
 {
    struct memory_handle *h = find_handle(handle_id);
@@ -1261,6 +1290,18 @@ int bufadvance(int handle_id, off_t offset)
    return seek_handle(h, newpos);
 }
+/* Get the read position from the start of the file
+   Returns the offset from byte 0 of the file and for failure:
+     ERR_HANDLE_NOT_FOUND if the handle wasn't found
+ */
+off_t bufftell(int handle_id)
+{
+    const struct memory_handle *h = find_handle(handle_id);
+    if (!h)
+        return ERR_HANDLE_NOT_FOUND;
+    return h->offset + ringbuf_sub(h->ridx, h->data);
+}
 /* Used by bufread and bufgetdata to prepare the buffer and retrieve the
 * actual amount of data available for reading.  This function explicitly
 * does not check the validity of the input handle.  It does do range checks
@@ -1306,7 +1347,7 @@ static struct memory_handle *prep_bufdata(int handle_id, size_t *size,
            /* it is not safe for a non-buffering thread to sleep while
             * holding a handle */
            h = find_handle(handle_id);
-            if (!h)
+            if (!h || h->signaled != 0)
                return NULL;
            avail = handle_size_available(h);
        }
@@ -1447,9 +1488,14 @@ SECONDARY EXPORTED FUNCTIONS
 buf_handle_offset
 buf_request_buffer_handle
 buf_set_base_handle
+buf_handle_data_type
+buf_is_handle
+buf_pin_handle
+buf_signal_handle
+buf_length
 buf_used
-register_buffering_callback
+buf_set_watermark
-unregister_buffering_callback
+buf_get_watermark
 These functions are exported, to allow interaction with the buffer.
 They take care of the content of the structs, and rely on the linked list
@@ -1472,8 +1518,61 @@ void buf_request_buffer_handle(int handle_id)
 void buf_set_base_handle(int handle_id)
 {
-    LOGFQUEUE("buffering > Q_BASE_HANDLE %d", handle_id);
+    mutex_lock(&llist_mutex);
-    queue_post(&buffering_queue, Q_BASE_HANDLE, handle_id);
+    base_handle_id = handle_id;
+    mutex_unlock(&llist_mutex);
+}
+enum data_type buf_handle_data_type(int handle_id)
+{
+    const struct memory_handle *h = find_handle(handle_id);
+    if (!h)
+        return TYPE_UNKNOWN;
+    return h->type;
+}
+ssize_t buf_handle_remaining(int handle_id)
+{
+    const struct memory_handle *h = find_handle(handle_id);
+    if (!h)
+        return ERR_HANDLE_NOT_FOUND;
+    return h->filerem;
+}
+bool buf_is_handle(int handle_id)
+{
+    return find_handle(handle_id) != NULL;
+}
+bool buf_pin_handle(int handle_id, bool pin)
+{
+    struct memory_handle *h = find_handle(handle_id);
+    if (!h)
+        return false;
+    if (pin) {
+        h->pinned++;
+    } else if (h->pinned > 0) {
+        h->pinned--;
+    }
+    return true; 
+}
+bool buf_signal_handle(int handle_id, bool signal)
+{
+    struct memory_handle *h = find_handle(handle_id);
+    if (!h)
+        return false;
+    h->signaled = signal ? 1 : 0;
+    return true; 
+}
+/* Return the size of the ringbuffer */
+size_t buf_length(void)
+{
+    return buffer_len;
 }
 /* Return the amount of buffer space used */
@@ -1487,6 +1586,21 @@ void buf_set_watermark(size_t bytes)
    conf_watermark = bytes;
 }
+size_t buf_get_watermark(void)
+{
+    return BUF_WATERMARK;
+}
+#ifdef HAVE_IO_PRIORITY
+void buf_back_off_storage(bool back_off)
+{
+    int priority = back_off ?
+        IO_PRIORITY_BACKGROUND : IO_PRIORITY_IMMEDIATE;
+    thread_set_io_priority(buffering_thread_id, priority);
+}
+#endif
+/** -- buffer thread helpers -- **/
 static void shrink_buffer_inner(struct memory_handle *h)
 {
    if (h == NULL)
@@ -1503,7 +1617,7 @@ static void shrink_buffer(void)
    shrink_buffer_inner(first_handle);
 }
-void buffering_thread(void)
+static void NORETURN_ATTR buffering_thread(void)
 {
    bool filling = false;
    struct queue_event ev;
@@ -1511,19 +1625,21 @@ void buffering_thread(void)
    while (true)
    {
-        if (!filling) {
+        if (num_handles > 0) {
+            if (!filling) {
+                cancel_cpu_boost();
+            }
+            queue_wait_w_tmo(&buffering_queue, &ev, filling ? 1 : HZ/2);
+        } else {
+            filling = false;
            cancel_cpu_boost();
+            queue_wait(&buffering_queue, &ev);
        }
-        queue_wait_w_tmo(&buffering_queue, &ev, filling ? 5 : HZ/2);
        switch (ev.id)
        {
            case Q_START_FILL:
                LOGFQUEUE("buffering < Q_START_FILL %d", (int)ev.data);
-                /* Call buffer callbacks here because this is one of two ways
-                 * to begin a full buffer fill */
-                send_event(BUFFER_EVENT_BUFFER_LOW, 0);
                shrink_buffer();
                queue_reply(&buffering_queue, 1);
                filling |= buffer_handle((int)ev.data, 0);
@@ -1553,36 +1669,21 @@ void buffering_thread(void)
                filling = true;
                break;
-            case Q_BASE_HANDLE:
-                LOGFQUEUE("buffering < Q_BASE_HANDLE %d", (int)ev.data);
-                base_handle_id = (int)ev.data;
-                break;
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
-            case SYS_USB_CONNECTED:
-                LOGFQUEUE("buffering < SYS_USB_CONNECTED");
-                usb_acknowledge(SYS_USB_CONNECTED_ACK);
-                usb_wait_for_disconnect(&buffering_queue);
-                break;
-#endif
            case SYS_TIMEOUT:
                LOGFQUEUE_SYS_TIMEOUT("buffering < SYS_TIMEOUT");
                break;
        }
-        update_data_counters(NULL);
+        if (num_handles == 0 || !queue_empty(&buffering_queue))
+            continue;
-        /* If the buffer is low, call the callbacks to get new data */
-        if (num_handles > 0 && data_counters.useful <= conf_watermark)
-            send_event(BUFFER_EVENT_BUFFER_LOW, 0);
+        update_data_counters(NULL);
 #if 0
        /* TODO: This needs to be fixed to use the idle callback, disable it
         * for simplicity until its done right */
 #if MEMORYSIZE > 8
        /* If the disk is spinning, take advantage by filling the buffer */
-        else if (storage_disk_is_active() && queue_empty(&buffering_queue)) {
+        else if (storage_disk_is_active()) {
            if (num_handles > 0 && data_counters.useful <= high_watermark)
                send_event(BUFFER_EVENT_BUFFER_LOW, 0);
@@ -1597,15 +1698,23 @@ void buffering_thread(void)
 #endif
 #endif
-        if (queue_empty(&buffering_queue)) {
+        if (filling) {
-            if (filling) {
+            if (data_counters.remaining > 0 && BUF_USED < buffer_len) {
-                if (data_counters.remaining > 0 && BUF_USED < buffer_len)
+                filling = fill_buffer();
-                    filling = fill_buffer();
+            }
-                else if (data_counters.remaining == 0)
+            else if (data_counters.remaining == 0) {
-                    filling = false;
+                filling = false;
-            } else if (ev.id == SYS_TIMEOUT) {
+            }
-                if (data_counters.remaining > 0 &&
+        } else if (ev.id == SYS_TIMEOUT) {
-                    data_counters.useful <= conf_watermark) {
+            if (data_counters.useful < BUF_WATERMARK) {
+                /* The buffer is low and we're idle, just watching the levels
+                   - call the callbacks to get new data */
+                send_event(BUFFER_EVENT_BUFFER_LOW, NULL);
+                /* Continue anything else we haven't finished - it might
+                   get booted off or stop early because the receiver hasn't
+                   had a chance to clear anything yet */
+                if (data_counters.remaining > 0) {
                    shrink_buffer();
                    filling = fill_buffer();
                }
@@ -1618,9 +1727,14 @@ void buffering_init(void)
 {
    mutex_init(&llist_mutex);
-    conf_watermark = BUFFERING_DEFAULT_WATERMARK;
+    /* Thread should absolutely not respond to USB because if it waits first,
+       then it cannot properly service the handles and leaks will happen -
-    queue_init(&buffering_queue, true);
+       this is a worker thread and shouldn't need to care about any system
+       notifications.
+                                      ***
+       Whoever is using buffering should be responsible enough to clear all
+       the handles at the right time. */
+    queue_init(&buffering_queue, false);
    buffering_thread_id = create_thread( buffering_thread, buffering_stack,
            sizeof(buffering_stack), CREATE_THREAD_FROZEN,
            buffering_thread_name IF_PRIO(, PRIORITY_BUFFERING)
@@ -1636,6 +1750,9 @@ bool buffering_reset(char *buf, size_t buflen)
    /* Wraps of storage-aligned data must also be storage aligned,
       thus buf and buflen must be a aligned to an integer multiple of
       the storage alignment */
+    buflen -= GUARD_BUFSIZE;
    STORAGE_ALIGN_BUFFER(buf, buflen);
    if (!buf || !buflen)
@@ -1654,10 +1771,13 @@ bool buffering_reset(char *buf, size_t buflen)
    num_handles = 0;
    base_handle_id = -1;
-    /* Set the high watermark as 75% full...or 25% empty :) */
+    /* Set the high watermark as 75% full...or 25% empty :)
-#if MEMORYSIZE > 8
+       This is the greatest fullness that will trigger low-buffer events
+       no matter what the setting because high-bitrate files can have
+       ludicrous margins that even exceed the buffer size - most common
+       with a huge anti-skip buffer but even without that setting,
+       staying constantly active in buffering is pointless */
    high_watermark = 3*buflen / 4;
-#endif
    thread_thaw(buffering_thread_id);
@@ -1673,5 +1793,5 @@ void buffering_get_debugdata(struct buffering_debug *dbgdata)
    dbgdata->wasted_space = dc.wasted;
    dbgdata->buffered_data = dc.buffered;
    dbgdata->useful_data = dc.useful;
-    dbgdata->watermark = conf_watermark;
+    dbgdata->watermark = BUF_WATERMARK;
 }
diff --git a/apps/buffering.h b/apps/buffering.h
index 34d6d641c7..2e4cfd3968 100644
--- a/apps/buffering.h
+++ b/apps/buffering.h
@@ -28,14 +28,13 @@
 enum data_type {
+    TYPE_UNKNOWN = 0, /* invalid type indicator */
+    TYPE_ID3,
    TYPE_CODEC,
    TYPE_PACKET_AUDIO,
    TYPE_ATOMIC_AUDIO,
-    TYPE_ID3,
    TYPE_CUESHEET,
    TYPE_BITMAP,
-    TYPE_BUFFER,
-    TYPE_UNKNOWN,
 };
 /* Error return values */
@@ -63,6 +62,7 @@ bool buffering_reset(char *buf, size_t buflen);
 * bufclose  : Close an open handle
 * bufseek   : Set handle reading index, relatively to the start of the file
 * bufadvance: Move handle reading index, relatively to current position
+ * bufftell  : Return the handle's file read position
 * bufread   : Copy data from a handle to a buffer
 * bufgetdata: Obtain a pointer for linear access to a "size" amount of data
 * bufgettail: Out-of-band get the last size bytes of a handle.
@@ -81,28 +81,40 @@ int bufalloc(const void *src, size_t size, enum data_type type);
 bool bufclose(int handle_id);
 int bufseek(int handle_id, size_t newpos);
 int bufadvance(int handle_id, off_t offset);
+off_t bufftell(int handle_id);
 ssize_t bufread(int handle_id, size_t size, void *dest);
 ssize_t bufgetdata(int handle_id, size_t size, void **data);
 ssize_t bufgettail(int handle_id, size_t size, void **data);
 ssize_t bufcuttail(int handle_id, size_t size);
 /***************************************************************************
 * SECONDARY FUNCTIONS
 * ===================
 *
+ * buf_handle_data_type: return the handle's data type
+ * buf_is_handle: is the handle valid?
+ * buf_pin_handle: Disallow/allow handle movement. Handle may still be removed.
 * buf_handle_offset: Get the offset of the first buffered byte from the file
 * buf_request_buffer_handle: Request buffering of a handle
 * buf_set_base_handle: Tell the buffering thread which handle is currently read
+ * buf_length: Total size of ringbuffer
 * buf_used: Total amount of buffer space used (including allocated space)
+ * buf_back_off_storage: tell buffering thread to take it easy
 ****************************************************************************/
+enum data_type buf_handle_data_type(int handle_id);
+ssize_t buf_handle_remaining(int handle_id);
+bool buf_is_handle(int handle_id);
 ssize_t buf_handle_offset(int handle_id);
 void buf_request_buffer_handle(int handle_id);
 void buf_set_base_handle(int handle_id);
+size_t buf_length(void);
 size_t buf_used(void);
+bool buf_pin_handle(int handle_id, bool pin);
+bool buf_signal_handle(int handle_id, bool signal);
+#ifdef HAVE_IO_PRIORITY
+void buf_back_off_storage(bool back_off);
+#endif
 /* Settings */
 enum {
@@ -110,6 +122,7 @@ enum {
    BUFFERING_SET_CHUNKSIZE,
 };
 void buf_set_watermark(size_t bytes);
+size_t buf_get_watermark(void);
 /* Debugging */
 struct buffering_debug {
diff --git a/apps/codec_thread.c b/apps/codec_thread.c
index 65a7ebc7d5..7cf45c3490 100644
--- a/apps/codec_thread.c
+++ b/apps/codec_thread.c
@@ -9,6 +9,7 @@
 *
 * Copyright (C) 2005-2007 Miika Pekkarinen
 * Copyright (C) 2007-2008 Nicolas Pennequin
+ * Copyright (C) 2011      Michael Sevakis
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License
@@ -21,16 +22,14 @@
 ****************************************************************************/
 #include "config.h"
 #include "system.h"
-#include "playback.h"
-#include "codec_thread.h"
 #include "kernel.h"
 #include "codecs.h"
-#include "buffering.h"
+#include "codec_thread.h"
 #include "pcmbuf.h"
+#include "playback.h"
+#include "buffering.h"
 #include "dsp.h"
-#include "abrepeat.h"
 #include "metadata.h"
-#include "splash.h"
 /* Define LOGF_ENABLE to enable logf output in this file */
 /*#define LOGF_ENABLE*/
@@ -57,38 +56,45 @@
 #define LOGFQUEUE_SYS_TIMEOUT(...)
 #endif
 /* Variables are commented with the threads that use them:
- * A=audio, C=codec, V=voice. A suffix of - indicates that
+ * A=audio, C=codec
- * the variable is read but not updated on that thread.
+ * - = reads only
+ *
 * Unless otherwise noted, the extern variables are located
 * in playback.c.
 */
-/* Main state control */
+/* Q_LOAD_CODEC parameter data */
+struct codec_load_info
-/* Type of codec loaded? (C/A) */
+{
-static int current_codectype SHAREDBSS_ATTR = AFMT_UNKNOWN;
+    int hid;    /* audio handle id (specify < 0 to use afmt) */
+    int afmt;   /* codec specification (AFMT_*) */
+};
-extern struct mp3entry *thistrack_id3,  /* the currently playing track */
-                       *othertrack_id3; /* prev track during track-change-transition, or end of playlist,
-                                         * next track otherwise */
-/* Track change controls */
+/** --- Main state control --- **/
-extern struct event_queue audio_queue SHAREDBSS_ATTR;
+static int codec_type = AFMT_UNKNOWN; /* Codec type (C,A-) */
+/* Private interfaces to main playback control */
+extern void audio_codec_update_elapsed(unsigned long value);
+extern void audio_codec_update_offset(size_t value);
+extern void audio_queue_post(long id, intptr_t data);
 extern struct codec_api ci; /* from codecs.c */
 /* Codec thread */
 static unsigned int codec_thread_id; /* For modifying thread priority later */
 static struct event_queue codec_queue SHAREDBSS_ATTR;
 static struct queue_sender_list codec_queue_sender_list SHAREDBSS_ATTR;
-static long codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)]
+static long codec_stack[(DEFAULT_STACK_SIZE + 0x2000)/sizeof(long)] IBSS_ATTR;
-                IBSS_ATTR;
 static const char codec_thread_name[] = "codec";
+static void unload_codec(void);
+/* Messages are only ever sent one at a time to the codec from the audio
+   thread. This is important for correct operation unless playback is
+   stopped. */
 /* static routines */
 static void codec_queue_ack(intptr_t ackme)
 {
@@ -100,52 +106,63 @@ static intptr_t codec_queue_send(long id, intptr_t data)
    return queue_send(&codec_queue, id, data);
 }
-/**************************************/
+/* Poll the state of the codec queue. Returns < 0 if the message is urgent
+   and any state should exit, > 0 if it's a run message (and it was
+   scrubbed), 0 if message was ignored. */
+static int codec_check_queue__have_msg(void)
+{
+    struct queue_event ev;
-/** misc external functions */
+    queue_peek(&codec_queue, &ev);
-/* Used to check whether a new codec must be loaded. See array audio_formats[]
+    /* Seek, pause or stop? Just peek and return if so. Codec
- * in metadata.c */
+       must handle the command after returing. Inserts will not
-int get_codec_base_type(int type)
+       be allowed until it complies. */
-{
+    switch (ev.id)
-    int base_type = type;
+    {
-    switch (type) {
+    case Q_CODEC_SEEK:
-        case AFMT_MPA_L1:
+        LOGFQUEUE("codec - Q_CODEC_SEEK", ev.id);
-        case AFMT_MPA_L2:
+        return -1;
-        case AFMT_MPA_L3:
+    case Q_CODEC_PAUSE:
-            base_type = AFMT_MPA_L3;
+        LOGFQUEUE("codec - Q_CODEC_PAUSE", ev.id);
-            break;
+        return -1;
-        case AFMT_MPC_SV7:
+    case Q_CODEC_STOP:
-        case AFMT_MPC_SV8:
+        LOGFQUEUE("codec - Q_CODEC_STOP", ev.id);
-            base_type = AFMT_MPC_SV7;
+        return -1;
-            break;
-        case AFMT_MP4_AAC:
-        case AFMT_MP4_AAC_HE:
-            base_type = AFMT_MP4_AAC;
-            break;
-        case AFMT_SAP:
-        case AFMT_CMC:
-        case AFMT_CM3:
-        case AFMT_CMR:
-        case AFMT_CMS:
-        case AFMT_DMC:
-        case AFMT_DLT:
-        case AFMT_MPT:
-        case AFMT_MPD:
-        case AFMT_RMT:
-        case AFMT_TMC:
-        case AFMT_TM8:
-        case AFMT_TM2:
-            base_type = AFMT_SAP;
-            break;
-        default:
-            break;
    }
-    return base_type;
+    /* This is in error in this context unless it's "go, go, go!" */
+    queue_wait(&codec_queue, &ev);
+    if (ev.id == Q_CODEC_RUN)
+    {
+        logf("codec < Q_CODEC_RUN: already running!");
+        codec_queue_ack(Q_CODEC_RUN);
+        return 1;
+    }
+    /* Ignore it */
+    logf("codec < bad req %ld (%s)", ev.id, __func__);
+    codec_queue_ack(Q_NULL);
+    return 0;
+}
+/* Does the audio format type equal CODEC_TYPE_ENCODER? */
+static inline bool type_is_encoder(int afmt)
+{
+#ifdef AUDIO_HAVE_RECORDING
+    return (afmt & CODEC_TYPE_MASK) == CODEC_TYPE_ENCODER;
+#else
+    return false;
+    (void)afmt;
+#endif
 }
-const char *get_codec_filename(int cod_spec)
+/**************************************/
+/** --- Miscellaneous external functions --- **/
+const char * get_codec_filename(int cod_spec)
 {
    const char *fname;
@@ -173,7 +190,7 @@ const char *get_codec_filename(int cod_spec)
 #endif /* HAVE_RECORDING */
    return fname;
-} /* get_codec_filename */
+}
 /* Borrow the codec thread and return the ID */
 void codec_thread_do_callback(void (*fn)(void), unsigned int *id)
@@ -189,9 +206,9 @@ void codec_thread_do_callback(void (*fn)(void), unsigned int *id)
 }
-/** codec API callbacks */
+/** --- codec API callbacks --- **/
-static void* codec_get_buffer(size_t *size)
+static void * codec_get_buffer(size_t *size)
 {
    ssize_t s = CODEC_SIZE - codec_size;
    void *buf = &codecbuf[codec_size];
@@ -215,15 +232,19 @@ static void codec_pcmbuf_insert_callback(
        int inp_count;
        char *dest;
-        /* Prevent audio from a previous track from playing */
+        while (1)
-        if (ci.new_track || ci.stop_codec)
-            return;
-        while ((dest = pcmbuf_request_buffer(&out_count)) == NULL)
        {
+            if ((dest = pcmbuf_request_buffer(&out_count)) != NULL)
+                break;
            cancel_cpu_boost();
-            sleep(1);
-            if (ci.seek_time || ci.new_track || ci.stop_codec)
+            /* It will be awhile before space is available but we want
+               "instant" response to any message */
+            queue_wait_w_tmo(&codec_queue, NULL, HZ/20);
+            if (!queue_empty(&codec_queue) &&
+                codec_check_queue__have_msg() < 0)
                return;
        }
@@ -247,62 +268,28 @@ static void codec_pcmbuf_insert_callback(
        count -= inp_count;
    }
-} /* codec_pcmbuf_insert_callback */
+}
-static void codec_set_elapsed_callback(unsigned long value)
+/* helper function, not a callback */
+static bool codec_advance_buffer_counters(size_t amount)
 {
-    if (ci.seek_time)
+    if (bufadvance(ci.audio_hid, amount) < 0)
-        return;
-#ifdef AB_REPEAT_ENABLE
-    ab_position_report(value);
-#endif
-    unsigned long latency = pcmbuf_get_latency();
-    if (value < latency)
-        thistrack_id3->elapsed = 0;
-    else
    {
-        unsigned long elapsed = value - latency;
+        ci.curpos = ci.filesize;
-        if (elapsed > thistrack_id3->elapsed ||
+        return false;
-            elapsed < thistrack_id3->elapsed - 2)
-            {
-                thistrack_id3->elapsed = elapsed;
-            }
    }
-}
-static void codec_set_offset_callback(size_t value)
-{
-    if (ci.seek_time)
-        return;
-    unsigned long latency = pcmbuf_get_latency() * thistrack_id3->bitrate / 8;
-    if (value < latency)
-        thistrack_id3->offset = 0;
-    else
-        thistrack_id3->offset = value - latency;
-}
-/* helper function, not a callback */
-static void codec_advance_buffer_counters(size_t amount)
-{
-    bufadvance(get_audio_hid(), amount);
    ci.curpos += amount;
+    return true;
 }
 /* copy up-to size bytes into ptr and return the actual size copied */
 static size_t codec_filebuf_callback(void *ptr, size_t size)
 {
-    ssize_t copy_n;
+    ssize_t copy_n = bufread(ci.audio_hid, size, ptr);
-    if (ci.stop_codec)
-        return 0;
-    copy_n = bufread(get_audio_hid(), size, ptr);
    /* Nothing requested OR nothing left */
-    if (copy_n == 0)
+    if (copy_n <= 0)
        return 0;
    /* Update read and other position pointers */
@@ -310,15 +297,15 @@ static size_t codec_filebuf_callback(void *ptr, size_t size)
    /* Return the actual amount of data copied to the buffer */
    return copy_n;
-} /* codec_filebuf_callback */
+}
-static void* codec_request_buffer_callback(size_t *realsize, size_t reqsize)
+static void * codec_request_buffer_callback(size_t *realsize, size_t reqsize)
 {
    size_t copy_n = reqsize;
    ssize_t ret;
    void *ptr;
-    ret = bufgetdata(get_audio_hid(), reqsize, &ptr);
+    ret = bufgetdata(ci.audio_hid, reqsize, &ptr);
    if (ret >= 0)
        copy_n = MIN((size_t)ret, reqsize);
    else
@@ -329,101 +316,103 @@ static void* codec_request_buffer_callback(size_t *realsize, size_t reqsize)
    *realsize = copy_n;
    return ptr;
-} /* codec_request_buffer_callback */
+}
 static void codec_advance_buffer_callback(size_t amount)
 {
-    codec_advance_buffer_counters(amount);
+    if (!codec_advance_buffer_counters(amount))
-    codec_set_offset_callback(ci.curpos);
+        return;
+    audio_codec_update_offset(ci.curpos);
 }
 static bool codec_seek_buffer_callback(size_t newpos)
 {
    logf("codec_seek_buffer_callback");
-    int ret = bufseek(get_audio_hid(), newpos);
+    int ret = bufseek(ci.audio_hid, newpos);
-    if (ret == 0) {
+    if (ret == 0)
+    {
        ci.curpos = newpos;
        return true;
    }
-    else {
-        return false;
+    return false;
-    }
 }
 static void codec_seek_complete_callback(void)
 {
-    struct queue_event ev;
    logf("seek_complete");
    /* Clear DSP */
    dsp_configure(ci.dsp, DSP_FLUSH, 0);
    /* Post notification to audio thread */
-    LOGFQUEUE("audio > Q_AUDIO_SEEK_COMPLETE");
+    LOGFQUEUE("audio > Q_AUDIO_CODEC_SEEK_COMPLETE");
-    queue_post(&audio_queue, Q_AUDIO_SEEK_COMPLETE, 0);
+    audio_queue_post(Q_AUDIO_CODEC_SEEK_COMPLETE, 0);
-    /* Wait for ACK */
-    queue_wait(&codec_queue, &ev);
-    /* ACK back in context */
+    /* Wait for urgent or go message */
-    codec_queue_ack(Q_AUDIO_SEEK_COMPLETE);
+    do
+    {
+        queue_wait(&codec_queue, NULL);
+    }
+    while (codec_check_queue__have_msg() == 0);
 }
-static bool codec_request_next_track_callback(void)
+static void codec_configure_callback(int setting, intptr_t value)
 {
-    struct queue_event ev;
+    if (!dsp_configure(ci.dsp, setting, value))
+    {
-    logf("Request new track");
+        logf("Illegal key: %d", setting);
+    }
+}
-    audio_set_prev_elapsed(thistrack_id3->elapsed);
+static enum codec_command_action
+    codec_get_command_callback(intptr_t *param)
+{
+    yield();
-#ifdef AB_REPEAT_ENABLE
+    if (LIKELY(queue_empty(&codec_queue)))
-    ab_end_of_track_report();
+        return CODEC_ACTION_NULL; /* As you were */
-#endif
-    if (ci.stop_codec)
+    /* Process the message - return requested action and data (if any should
+       be expected) */
+    while (1)
    {
-        /* Handle ACK in outer loop */
+        enum codec_command_action action = CODEC_ACTION_NULL;
-        LOGFQUEUE("codec: already stopping");
+        struct queue_event ev;
-        return false;
+        queue_wait(&codec_queue, &ev);
-    }
-    trigger_cpu_boost();
+        switch (ev.id)
+        {
+        case Q_CODEC_RUN:   /* Already running */
+            LOGFQUEUE("codec < Q_CODEC_RUN");
+            break;
-    /* Post request to audio thread */
+        case Q_CODEC_PAUSE: /* Stay here and wait */
-    LOGFQUEUE("codec > audio Q_AUDIO_CHECK_NEW_TRACK");
+            LOGFQUEUE("codec < Q_CODEC_PAUSE");
-    queue_post(&audio_queue, Q_AUDIO_CHECK_NEW_TRACK, 0);
+            codec_queue_ack(Q_CODEC_PAUSE);
+            continue;
-    /* Wait for ACK */
+        case Q_CODEC_SEEK:  /* Audio wants codec to seek */
-    queue_wait(&codec_queue, &ev);
+            LOGFQUEUE("codec < Q_CODEC_SEEK %ld", ev.data);
+            *param = ev.data;
+            action = CODEC_ACTION_SEEK_TIME;
+            break;
-    if (ev.data == Q_CODEC_REQUEST_COMPLETE)
+        case Q_CODEC_STOP:  /* Must only return 0 in main loop */
-    {
+            LOGFQUEUE("codec < Q_CODEC_STOP");
-        /* Seek to the beginning of the new track because if the struct
+            action = CODEC_ACTION_HALT;
-           mp3entry was buffered, "elapsed" might not be zero (if the track has
+            break;
-           been played already but not unbuffered) */
-        codec_seek_buffer_callback(thistrack_id3->first_frame_offset);
-    }
-    /* ACK back in context */
+        default:            /* This is in error in this context. */
-    codec_queue_ack(Q_AUDIO_CHECK_NEW_TRACK);
+            ev.id = Q_NULL;
+            logf("codec bad req %ld (%s)", ev.id, __func__);
+        }
-    if (ev.data != Q_CODEC_REQUEST_COMPLETE || ci.stop_codec)
+        codec_queue_ack(ev.id);
-    {
+        return action;
-        LOGFQUEUE("codec <= request failed (%d)", ev.data);
-        return false;
    }
-    LOGFQUEUE("codec <= Q_CODEC_REQEST_COMPLETE");
-    return true;
-}
-static void codec_configure_callback(int setting, intptr_t value)
-{
-    if (!dsp_configure(ci.dsp, setting, value))
-        { logf("Illegal key:%d", setting); }
 }
 /* Initialize codec API */
@@ -433,119 +422,215 @@ void codec_init_codec_api(void)
                                                                CODEC_IDX_AUDIO);
    ci.codec_get_buffer    = codec_get_buffer;
    ci.pcmbuf_insert       = codec_pcmbuf_insert_callback;
-    ci.set_elapsed         = codec_set_elapsed_callback;
+    ci.set_elapsed         = audio_codec_update_elapsed;
    ci.read_filebuf        = codec_filebuf_callback;
    ci.request_buffer      = codec_request_buffer_callback;
    ci.advance_buffer      = codec_advance_buffer_callback;
    ci.seek_buffer         = codec_seek_buffer_callback;
    ci.seek_complete       = codec_seek_complete_callback;
-    ci.request_next_track  = codec_request_next_track_callback;
+    ci.set_offset          = audio_codec_update_offset;
-    ci.set_offset          = codec_set_offset_callback;
    ci.configure           = codec_configure_callback;
+    ci.get_command         = codec_get_command_callback;
 }
-/* track change */
+/** --- CODEC THREAD --- **/
-/** CODEC THREAD */
+/* Handle Q_CODEC_LOAD */
-static void codec_thread(void)
+static void load_codec(const struct codec_load_info *ev_data)
 {
-    struct queue_event ev;
+    int status = CODEC_ERROR;
+    /* Save a local copy so we can let the audio thread go ASAP */
+    struct codec_load_info data = *ev_data;
+    bool const encoder = type_is_encoder(data.afmt);
+    if (codec_type != AFMT_UNKNOWN)
+    {
+        /* Must have unloaded it first */
+        logf("a codec is already loaded");
+        if (data.hid >= 0)
+            bufclose(data.hid);
+        return;
+    }
-    while (1)   
+    trigger_cpu_boost();
+    if (!encoder)
    {
-        int status = CODEC_OK;
+        /* Do this now because codec may set some things up at load time */
-        void *handle = NULL;
+        dsp_configure(ci.dsp, DSP_RESET, 0);
-        int hid;
+    }
-        const char *codec_fn;
-        
+    if (data.hid >= 0)
-#ifdef HAVE_CROSSFADE
+    {
-        if (!pcmbuf_is_crossfade_active())
+        /* First try buffer load */
-#endif
+        status = codec_load_buf(data.hid, &ci);
+        bufclose(data.hid);
+    }
+    if (status < 0)
+    {
+        /* Either not a valid handle or the buffer method failed */
+        const char *codec_fn = get_codec_filename(data.afmt);
+        if (codec_fn)
        {
-            cancel_cpu_boost();
+#ifdef HAVE_IO_PRIORITY
+            buf_back_off_storage(true);
+#endif
+            status = codec_load_file(codec_fn, &ci);
+#ifdef HAVE_IO_PRIORITY
+            buf_back_off_storage(false);
+#endif
        }
+    }
+    if (status >= 0)
+    {
+        codec_type = data.afmt;
+        codec_queue_ack(Q_CODEC_LOAD);
+        return;
+    }
+    /* Failed - get rid of it */
+    unload_codec();
+}
+/* Handle Q_CODEC_RUN */
+static void run_codec(void)
+{
+    bool const encoder = type_is_encoder(codec_type);
+    int status;
+    if (codec_type == AFMT_UNKNOWN)
+    {
+        logf("no codec to run");
+        return;
+    }
+    codec_queue_ack(Q_CODEC_RUN);
+    trigger_cpu_boost();
+    if (!encoder)
+    {
+        /* This will be either the initial buffered offset or where it left off
+           if it remained buffered and we're skipping back to it and it is best
+           to have ci.curpos in sync with the handle's read position - it's the
+           codec's responsibility to ensure it has the correct positions -
+           playback is sorta dumb and only has a vague idea about what to
+           buffer based upon what metadata has to say */
+        ci.curpos = bufftell(ci.audio_hid);
+        /* Pin the codec's audio data in place */
+        buf_pin_handle(ci.audio_hid, true);
+    }
+    status = codec_run_proc();
+    if (!encoder)
+    {
+        /* Codec is done with it - let it move */
+        buf_pin_handle(ci.audio_hid, false);
+        /* Notify audio that we're done for better or worse - advise of the
+           status */
+        LOGFQUEUE("codec > audio Q_AUDIO_CODEC_COMPLETE: %d", status);
+        audio_queue_post(Q_AUDIO_CODEC_COMPLETE, status);
+    }
+}
+/* Handle Q_CODEC_SEEK */
+static void seek_codec(unsigned long time)
+{
+    if (codec_type == AFMT_UNKNOWN)
+    {
+        logf("no codec to seek");
+        codec_queue_ack(Q_CODEC_SEEK);
+        codec_seek_complete_callback();
+        return;
+    }
+    /* Post it up one level */
+    queue_post(&codec_queue, Q_CODEC_SEEK, time);
+    codec_queue_ack(Q_CODEC_SEEK);
+    /* Have to run it again */
+    run_codec();
+}
+/* Handle Q_CODEC_UNLOAD */
+static void unload_codec(void)
+{
+    /* Tell codec to clean up */
+    codec_type = AFMT_UNKNOWN;
+    codec_close();
+}
+/* Handle Q_CODEC_DO_CALLBACK */
+static void do_callback(void (* callback)(void))
+{
+    codec_queue_ack(Q_CODEC_DO_CALLBACK);
+    if (callback)
+    {
+        cpucache_commit_discard();
+        callback();
+        cpucache_commit();
+    }
+}
+/* Codec thread function */
+static void NORETURN_ATTR codec_thread(void)
+{
+    struct queue_event ev;
+    while (1)   
+    {
+        cancel_cpu_boost();
        queue_wait(&codec_queue, &ev);
        switch (ev.id)
        {
-            case Q_CODEC_LOAD_DISK:
+        case Q_CODEC_LOAD:
-                LOGFQUEUE("codec < Q_CODEC_LOAD_DISK");
+            LOGFQUEUE("codec < Q_CODEC_LOAD");
-                codec_fn = get_codec_filename(ev.data);
+            load_codec((const struct codec_load_info *)ev.data);
-                if (!codec_fn)
+            break;
-                    break;
-#ifdef AUDIO_HAVE_RECORDING
-                if (ev.data & CODEC_TYPE_ENCODER)
-                {
-                    ev.id = Q_ENCODER_LOAD_DISK;
-                    handle = codec_load_file(codec_fn, &ci);
-                    if (handle)
-                        codec_queue_ack(Q_ENCODER_LOAD_DISK);
-                }
-                else
-#endif
-                {
-                    codec_queue_ack(Q_CODEC_LOAD_DISK);
-                    handle = codec_load_file(codec_fn, &ci);
-                }
-                break;
-            case Q_CODEC_LOAD:
+        case Q_CODEC_RUN:
-                LOGFQUEUE("codec < Q_CODEC_LOAD");
+            LOGFQUEUE("codec < Q_CODEC_RUN");
-                codec_queue_ack(Q_CODEC_LOAD);
+            run_codec();
-                hid = (int)ev.data;
+            break;
-                handle = codec_load_buf(hid, &ci);
-                bufclose(hid);
-                break;
-            case Q_CODEC_DO_CALLBACK:
+        case Q_CODEC_PAUSE:
-                LOGFQUEUE("codec < Q_CODEC_DO_CALLBACK");
+            LOGFQUEUE("codec < Q_CODEC_PAUSE");
-                codec_queue_ack(Q_CODEC_DO_CALLBACK);
+            break;
-                if ((void*)ev.data != NULL)
-                {
-                    cpucache_commit_discard();
-                    ((void (*)(void))ev.data)();
-                    cpucache_commit();
-                }
-                break;
-            default:
+        case Q_CODEC_SEEK:
-                LOGFQUEUE("codec < default : %ld", ev.id);
+            LOGFQUEUE("codec < Q_CODEC_SEEK: %lu", (unsigned long)ev.data);
-        }
+            seek_codec(ev.data);
+            break;
-        if (handle)
+        case Q_CODEC_UNLOAD:
-        {
+            LOGFQUEUE("codec < Q_CODEC_UNLOAD");
-            /* Codec loaded - call the entrypoint */
+            unload_codec();
-            yield();
+            break;
-            logf("codec running");
-            status = codec_begin(handle);
-            logf("codec stopped");
-            codec_close(handle);
-            current_codectype = AFMT_UNKNOWN;
-            if (ci.stop_codec)
-                status = CODEC_OK;
-        }
-        switch (ev.id)
+        case Q_CODEC_DO_CALLBACK:
-        {
+            LOGFQUEUE("codec < Q_CODEC_DO_CALLBACK");
-#ifdef AUDIO_HAVE_RECORDING
+            do_callback((void (*)(void))ev.data);
-            case Q_ENCODER_LOAD_DISK:
+            break;
-#endif
-            case Q_CODEC_LOAD_DISK:
+        default:
-            case Q_CODEC_LOAD:
+            LOGFQUEUE("codec < default : %ld", ev.id);
-                /* Notify about the status */
-                if (!handle)
-                    status = CODEC_ERROR;
-                LOGFQUEUE("codec > audio notify status: %d", status);
-                queue_post(&audio_queue, ev.id, status);
-                break;
        }
    }
 }
+/** --- Miscellaneous external interfaces -- **/
+/* Create the codec thread and init kernel objects */
 void make_codec_thread(void)
 {
    queue_init(&codec_queue, false);
@@ -558,78 +643,86 @@ void make_codec_thread(void)
                            codec_thread_id);
 }
+/* Unfreeze the codec thread */
 void codec_thread_resume(void)
 {
    thread_thaw(codec_thread_id);
 }
+/* Is the current thread the codec thread? */
 bool is_codec_thread(void)
 {
    return thread_self() == codec_thread_id;
 }
 #ifdef HAVE_PRIORITY_SCHEDULING
+/* Obtain codec thread's current priority */
 int codec_thread_get_priority(void)
 {
    return thread_get_priority(codec_thread_id);
 }
+/* Set the codec thread's priority and return the old value */
 int codec_thread_set_priority(int priority)
 {
    return thread_set_priority(codec_thread_id, priority);
 }
 #endif /* HAVE_PRIORITY_SCHEDULING */
-/* functions for audio thread use */
-intptr_t codec_ack_msg(intptr_t data, bool stop_codec)
-{
-    intptr_t resp;
-    LOGFQUEUE("codec >| Q_CODEC_ACK: %d", data);
-    if (stop_codec)
-        ci.stop_codec = true;
-    resp = codec_queue_send(Q_CODEC_ACK, data);
-    if (stop_codec)
-        codec_stop();
-    LOGFQUEUE("  ack: %ld", resp);
-    return resp;
-}
+/** --- Functions for audio thread use --- **/
+/* Load a decoder or encoder and set the format type */
 bool codec_load(int hid, int cod_spec)
 {
-    bool retval = false;
+    struct codec_load_info parm = { hid, cod_spec };
-    ci.stop_codec = false;
+    LOGFQUEUE("audio >| codec Q_CODEC_LOAD: %d, %d", hid, cod_spec);
-    current_codectype = cod_spec;
+    return codec_queue_send(Q_CODEC_LOAD, (intptr_t)&parm) != 0;
+}
-    if (hid >= 0)
+/* Begin decoding the current file */
-    {
+void codec_go(void)
-        LOGFQUEUE("audio >| codec Q_CODEC_LOAD: %d", hid);
+{
-        retval = codec_queue_send(Q_CODEC_LOAD, hid) != Q_NULL;
+    LOGFQUEUE("audio >| codec Q_CODEC_RUN");
-    }
+    codec_queue_send(Q_CODEC_RUN, 0);
-    else
+}
-    {
-        LOGFQUEUE("audio >| codec Q_CODEC_LOAD_DISK: %d", cod_spec);
-        retval = codec_queue_send(Q_CODEC_LOAD_DISK, cod_spec) != Q_NULL;
-    }
-    if (!retval)
+/* Instruct the codec to seek to the specified time (should be properly
-    {
+   paused or stopped first to avoid possible buffering deadlock) */
-        ci.stop_codec = true;
+void codec_seek(long time)
-        current_codectype = AFMT_UNKNOWN;
+{
-    }
+    LOGFQUEUE("audio > codec Q_CODEC_SEEK: %ld", time);
+    codec_queue_send(Q_CODEC_SEEK, time);
+}
-    return retval;
+/* Pause the codec and make it wait for further instructions inside the
+   command callback */
+bool codec_pause(void)
+{
+    LOGFQUEUE("audio >| codec Q_CODEC_PAUSE");
+    return codec_queue_send(Q_CODEC_PAUSE, 0) != Q_NULL;
 }
+/* Stop codec if running - codec stays resident if loaded */
 void codec_stop(void)
 {
-    ci.stop_codec = true;
    /* Wait until it's in the main loop */
-    while (codec_ack_msg(0, false) != Q_NULL);
+    LOGFQUEUE("audio >| codec Q_CODEC_STOP");
-    current_codectype = AFMT_UNKNOWN;
+    while (codec_queue_send(Q_CODEC_STOP, 0) != Q_NULL);
+}
+/* Call the codec's exit routine and close all references */
+void codec_unload(void)
+{
+    codec_stop();
+    LOGFQUEUE("audio >| codec Q_CODEC_UNLOAD");
+    codec_queue_send(Q_CODEC_UNLOAD, 0);
 }
+/* Return the afmt type of the loaded codec - sticks until calling
+   codec_unload unless initial load failed */
 int codec_loaded(void)
 {
-    return current_codectype;
+    return codec_type;
 }
diff --git a/apps/codec_thread.h b/apps/codec_thread.h
index 7056e2cdf5..acd7e556e2 100644
--- a/apps/codec_thread.h
+++ b/apps/codec_thread.h
@@ -25,7 +25,6 @@
 #include <stdbool.h>
 /* codec identity */
-int get_codec_base_type(int type);
 const char *get_codec_filename(int cod_spec);
 /* codec thread */
@@ -44,10 +43,14 @@ int codec_thread_set_priority(int priority);
 #endif
 /* codec commands - on audio thread only! */
-intptr_t codec_ack_msg(intptr_t data, bool stop_codec);
 bool codec_load(int hid, int cod_spec);
+void codec_go(void);
+bool codec_pause(void);
+void codec_seek(long time);
 void codec_stop(void);
+void codec_unload(void);
 int codec_loaded(void);
 /* */
 #endif /* _CODEC_THREAD_H */
diff --git a/apps/codecs.c b/apps/codecs.c
index 28e8c2a1c0..25ace4969b 100644
--- a/apps/codecs.c
+++ b/apps/codecs.c
@@ -79,13 +79,10 @@ static int open(const char* pathname, int flags, ...)
 #endif
 struct codec_api ci = {
-    0, /* filesize */
+    0,    /* filesize */
-    0, /* curpos */
+    0,    /* curpos */
    NULL, /* id3 */
-    NULL, /* taginfo_ready */
+    ERR_HANDLE_NOT_FOUND, /* audio_hid */
-    false, /* stop_codec */
-    0, /* new_track */
-    0, /* seek_time */
    NULL, /* struct dsp_config *dsp */
    NULL, /* codec_get_buffer */
    NULL, /* pcmbuf_insert */
@@ -95,9 +92,9 @@ struct codec_api ci = {
    NULL, /* advance_buffer */
    NULL, /* seek_buffer */
    NULL, /* seek_complete */
-    NULL, /* request_next_track */
    NULL, /* set_offset */
    NULL, /* configure */
+    NULL, /* get_command */
    
    /* kernel/ system */
 #if defined(CPU_ARM) && CONFIG_PLATFORM & PLATFORM_NATIVE
@@ -174,10 +171,16 @@ void codec_get_full_path(char *path, const char *codec_root_fn)
             CODECS_DIR, codec_root_fn);
 }
-static void * codec_load_ram(void *handle, struct codec_api *api)
+/** codec loading and call interface **/
+static void *curr_handle = NULL;
+static struct codec_header *c_hdr = NULL;
+static int codec_load_ram(struct codec_api *api)
 {
-    struct codec_header *c_hdr = lc_get_header(handle);
+    struct lc_header *hdr;
-    struct lc_header    *hdr   = c_hdr ? &c_hdr->lc_hdr : NULL;
+    c_hdr = lc_get_header(curr_handle);
+    hdr   = c_hdr ? &c_hdr->lc_hdr : NULL;
    if (hdr == NULL
        || (hdr->magic != CODEC_MAGIC
@@ -193,15 +196,17 @@ static void * codec_load_ram(void *handle, struct codec_api *api)
        )
    {
        logf("codec header error");
-        lc_close(handle);
+        lc_close(curr_handle);
-        return NULL;
+        curr_handle = NULL;
+        return CODEC_ERROR;
    }
    if (hdr->api_version > CODEC_API_VERSION
        || hdr->api_version < CODEC_MIN_API_VERSION) {
        logf("codec api version error");
-        lc_close(handle);
+        lc_close(curr_handle);
-        return NULL;
+        curr_handle = NULL;
+        return CODEC_ERROR;
    }
 #if (CONFIG_PLATFORM & PLATFORM_NATIVE)
@@ -212,63 +217,66 @@ static void * codec_load_ram(void *handle, struct codec_api *api)
    *(c_hdr->api) = api;
-    return handle;
+    logf("Codec: calling entrypoint");
+    return c_hdr->entry_point(CODEC_LOAD);
 }
-void * codec_load_buf(int hid, struct codec_api *api)
+int codec_load_buf(int hid, struct codec_api *api)
 {
-    int rc;
+    int rc = bufread(hid, CODEC_SIZE, codecbuf);
-    void *handle;
-    rc = bufread(hid, CODEC_SIZE, codecbuf);
    if (rc < 0) {
        logf("Codec: cannot read buf handle");
-        return NULL;
+        return CODEC_ERROR;
    }
-    handle = lc_open_from_mem(codecbuf, rc);
+    curr_handle = lc_open_from_mem(codecbuf, rc);
-    if (handle == NULL) {
+    if (curr_handle == NULL) {
-        logf("error loading codec");
+        logf("Codec: load error");
-        return NULL;
+        return CODEC_ERROR;
    }
-    return codec_load_ram(handle, api);
+    return codec_load_ram(api);
 }
-void * codec_load_file(const char *plugin, struct codec_api *api)
+int codec_load_file(const char *plugin, struct codec_api *api)
 {
    char path[MAX_PATH];
-    void *handle;
    codec_get_full_path(path, plugin);
-    handle = lc_open(path, codecbuf, CODEC_SIZE);
+    curr_handle = lc_open(path, codecbuf, CODEC_SIZE);
-    if (handle == NULL) {
+    if (curr_handle == NULL) {
        logf("Codec: cannot read file");
-        return NULL;
+        return CODEC_ERROR;
    }
-    return codec_load_ram(handle, api);
+    return codec_load_ram(api);
 }
-int codec_begin(void *handle)
+int codec_run_proc(void)
 {
-    int status = CODEC_ERROR;
+    if (curr_handle == NULL) {
-    struct codec_header *c_hdr;
+        logf("Codec: no codec to run");
+        return CODEC_ERROR;
-    c_hdr = lc_get_header(handle);
-    if (c_hdr != NULL) {
-        logf("Codec: calling entry_point");
-        status = c_hdr->entry_point();
    }
-    return status;
+    logf("Codec: entering run state");
+    return c_hdr->run_proc();
 }
-void codec_close(void *handle)
+int codec_close(void)
 {
-    if (handle)
+    int status = CODEC_OK;
-        lc_close(handle);
+    if (curr_handle != NULL) {
+        logf("Codec: cleaning up");
+        status = c_hdr->entry_point(CODEC_UNLOAD);
+        lc_close(curr_handle);
+        curr_handle = NULL;
+    }
+    return status;
 }
diff --git a/apps/codecs.h b/apps/codecs.h
index d96b2a7c9a..5c50116038 100644
--- a/apps/codecs.h
+++ b/apps/codecs.h
@@ -75,12 +75,18 @@
 #define CODEC_ENC_MAGIC 0x52454E43 /* RENC */
 /* increase this every time the api struct changes */
-#define CODEC_API_VERSION 41
+#define CODEC_API_VERSION 42
 /* update this to latest version if a change to the api struct breaks
   backwards compatibility (and please take the opportunity to sort in any
   new function which are "waiting" at the end of the function table) */
-#define CODEC_MIN_API_VERSION 41
+#define CODEC_MIN_API_VERSION 42
+/* reasons for calling codec main entrypoint */
+enum codec_entry_call_reason {
+    CODEC_LOAD = 0,
+    CODEC_UNLOAD
+};
 /* codec return codes */
 enum codec_status {
@@ -88,6 +94,13 @@ enum codec_status {
    CODEC_ERROR = -1,
 };
+/* codec command action codes */
+enum codec_command_action {
+    CODEC_ACTION_HALT = -1,
+    CODEC_ACTION_NULL = 0,
+    CODEC_ACTION_SEEK_TIME = 1,
+};
 /* NOTE: To support backwards compatibility, only add new functions at
         the end of the structure.  Every time you add a new function,
         remember to increase CODEC_API_VERSION.  If you make changes to the
@@ -95,24 +108,12 @@ enum codec_status {
         version
 */
 struct codec_api {
    off_t  filesize;          /* Total file length */
    off_t  curpos;            /* Current buffer position */
    
-    /* For gapless mp3 */
    struct mp3entry *id3;     /* TAG metadata pointer */
-    bool *taginfo_ready;      /* Is metadata read */
+    int    audio_hid;         /* Current audio handle */
    
-    /* Codec should periodically check if stop_codec is set to true.
-       In case it is, codec must return immediately */
-    volatile bool stop_codec;
-    /* Codec should periodically check if new_track is non zero.
-       When it is, the codec should request a new track. */
-    volatile int new_track;
-    /* If seek_time != 0, codec should seek to that song position (in ms)
-       if codec supports seeking. */
-    volatile long seek_time;
    /* The dsp instance to be used for audio output */
    struct dsp_config *dsp;
    
@@ -138,14 +139,12 @@ struct codec_api {
    bool (*seek_buffer)(size_t newpos);
    /* Codec should call this function when it has done the seeking. */
    void (*seek_complete)(void);
-    /* Request file change from file buffer. Returns true is next
+    /* Update the current position */
-       track is available and changed. If return value is false,
-       codec should exit immediately with PLUGIN_OK status. */
-    bool (*request_next_track)(void);
-    
    void (*set_offset)(size_t value);
    /* Configure different codec buffer parameters. */
    void (*configure)(int setting, intptr_t value);
+    /* Obtain command action on what to do next */
+    enum codec_command_action (*get_command)(intptr_t *param);
    /* kernel/ system */
 #if defined(CPU_ARM) && CONFIG_PLATFORM & PLATFORM_NATIVE
@@ -231,7 +230,8 @@ struct codec_api {
 /* codec header */
 struct codec_header {
    struct lc_header lc_hdr; /* must be first */
-    enum codec_status(*entry_point)(void);
+    enum codec_status(*entry_point)(enum codec_entry_call_reason reason);
+    enum codec_status(*run_proc)(void);
    struct codec_api **api;
 };
@@ -248,13 +248,15 @@ extern unsigned char plugin_end_addr[];
        const struct codec_header __header \
        __attribute__ ((section (".header")))= { \
        { CODEC_MAGIC, TARGET_ID, CODEC_API_VERSION, \
-        plugin_start_addr, plugin_end_addr }, codec_start, &ci };
+        plugin_start_addr, plugin_end_addr }, codec_start, \
+        codec_run, &ci };
 /* encoders */
 #define CODEC_ENC_HEADER \
        const struct codec_header __header \
        __attribute__ ((section (".header")))= { \
        { CODEC_ENC_MAGIC, TARGET_ID, CODEC_API_VERSION, \
-        plugin_start_addr, plugin_end_addr }, codec_start, &ci };
+        plugin_start_addr, plugin_end_addr }, codec_start, \
+        codec_run, &ci };
 #else /* def SIMULATOR */
 /* decoders */
@@ -262,12 +264,12 @@ extern unsigned char plugin_end_addr[];
        const struct codec_header __header \
        __attribute__((visibility("default"))) = { \
        { CODEC_MAGIC, TARGET_ID, CODEC_API_VERSION, NULL, NULL }, \
-        codec_start, &ci };
+        codec_start, codec_run, &ci };
 /* encoders */
 #define CODEC_ENC_HEADER \
        const struct codec_header __header = { \
        { CODEC_ENC_MAGIC, TARGET_ID, CODEC_API_VERSION, NULL, NULL }, \
-        codec_start, &ci };
+        codec_start, codec_run, &ci };
 #endif /* SIMULATOR */
 #endif /* CODEC */
@@ -276,13 +278,14 @@ extern unsigned char plugin_end_addr[];
 void codec_get_full_path(char *path, const char *codec_root_fn);
 /* defined by the codec loader (codec.c) */
-void * codec_load_buf(int hid, struct codec_api *api);
+int codec_load_buf(int hid, struct codec_api *api);
-void * codec_load_file(const char* codec, struct codec_api *api);
+int codec_load_file(const char* codec, struct codec_api *api);
-int codec_begin(void *handle);
+int codec_run_proc(void);
-void codec_close(void *handle);
+int codec_halt(void);
+int codec_close(void);
 /* defined by the codec */
-enum codec_status codec_start(void);
+enum codec_status codec_start(enum codec_entry_call_reason reason);
-enum codec_status codec_main(void);
+enum codec_status codec_run(void);
-#endif
+#endif /* _CODECS_H_ */
diff --git a/apps/codecs/a52.c b/apps/codecs/a52.c
index 00fdeea309..4cd293e37f 100644
--- a/apps/codecs/a52.c
+++ b/apps/codecs/a52.c
@@ -116,27 +116,31 @@ static void a52_decode_data(uint8_t *start, uint8_t *end)
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+    }
+    else if (reason == CODEC_UNLOAD) {
+        if (state)
+            a52_free(state);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    size_t n;
    unsigned char *filebuf;
    int sample_loc;
-    int retval;
+    intptr_t param;
-    /* Generic codec initialisation */
+    if (codec_init())
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+        return CODEC_ERROR;
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
-next_track:
-    retval = CODEC_OK;
-    if (codec_init()) {
-        retval = CODEC_ERROR;
-        goto exit;
-    }
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
    codec_set_replaygain(ci->id3);
@@ -153,15 +157,18 @@ next_track:
        }
    }
    else {
+        ci->seek_buffer(ci->id3->first_frame_offset);
        samplesdone = 0;
    }
    while (1) {
-        if (ci->stop_codec || ci->new_track)
+        enum codec_command_action action = ci->get_command(&param);
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            sample_loc = (ci->seek_time - 1)/1000 * ci->id3->frequency;
+            sample_loc = param/1000 * ci->id3->frequency;
            if (ci->seek_buffer((sample_loc/A52_SAMPLESPERFRAME)*ci->id3->bytesperframe)) {
                samplesdone = sample_loc;
@@ -179,11 +186,5 @@ next_track:
        ci->advance_buffer(n);
    }
-request_next_track:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    a52_free(state);
-    return retval;
 }
diff --git a/apps/codecs/a52_rm.c b/apps/codecs/a52_rm.c
index f5e4923292..c1930aa7b4 100644
--- a/apps/codecs/a52_rm.c
+++ b/apps/codecs/a52_rm.c
@@ -124,36 +124,44 @@ static void a52_decode_data(uint8_t *start, uint8_t *end)
    }
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+    }
+    else if (reason == CODEC_UNLOAD) {
+        if (state)
+            a52_free(state);        
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    size_t n;
    uint8_t *filebuf;
-    int retval, consumed, packet_offset;
+    int consumed, packet_offset;
    int playback_on = -1;
    size_t resume_offset;
+    intptr_t param;
-    /* Generic codec initialisation */
+    enum codec_command_action action = CODEC_ACTION_NULL;
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
-next_track:
-    retval = CODEC_OK;
    if (codec_init()) {
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    resume_offset = ci->id3->offset;
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
    codec_set_replaygain(ci->id3);
+    ci->seek_buffer(ci->id3->first_frame_offset);
    /* Intializations */
    state = a52_init(0);
    ci->memset(&rmctx,0,sizeof(RMContext)); 
@@ -165,26 +173,34 @@ next_track:
        resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
        /* put number of subpackets to skip in resume_offset */
        resume_offset /= (rmctx.block_align + PACKET_HEADER_SIZE);
-        ci->seek_time =  (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate);                
+        param = (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate);
+        action = CODEC_ACTION_SEEK_TIME;
+    }
+    else {
+        /* Seek to the first packet */
+        ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE );
    }
-    
-    /* Seek to the first packet */
-    ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE );
    /* The main decoding loop */
    while((unsigned)rmctx.audio_pkt_cnt < rmctx.nb_packets) {
-        ci->yield();
+        if (action == CODEC_ACTION_NULL)
-        if (ci->stop_codec || ci->new_track)
+            action = ci->get_command(&param);
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            packet_offset = ci->seek_time / ((rmctx.block_align*8*1000)/rmctx.bit_rate);
+            packet_offset = param / ((rmctx.block_align*8*1000)/rmctx.bit_rate);
-            ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE));
+            ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE +
+                            packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE));
            rmctx.audio_pkt_cnt = packet_offset;
-            samplesdone = (rmctx.sample_rate/1000 * ci->seek_time);
+            samplesdone = (rmctx.sample_rate/1000 * param);
+            ci->set_elapsed(samplesdone/(frequency/1000));
            ci->seek_complete();
        }
+        action = CODEC_ACTION_NULL;
        filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE);
        consumed = rm_get_packet(&filebuf, &rmctx, &pkt);
@@ -195,8 +211,7 @@ next_track:
                return CODEC_ERROR;
            }
            else {
-                retval = CODEC_OK;
+                break;
-                goto exit;
            }
        }
@@ -205,11 +220,5 @@ next_track:
        ci->advance_buffer(pkt.length);
    }
-request_next_track:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    a52_free(state);
-    return retval;
 }
diff --git a/apps/codecs/aac.c b/apps/codecs/aac.c
index 5638dc49a9..90cf0438ce 100644
--- a/apps/codecs/aac.c
+++ b/apps/codecs/aac.c
@@ -33,7 +33,19 @@ CODEC_HEADER
 #define FAAD_BYTE_BUFFER_SIZE (2048-12)
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    /* Note that when dealing with QuickTime/MPEG4 files, terminology is
     * a bit confusing. Files with sound are split up in chunks, where
@@ -59,25 +71,15 @@ enum codec_status codec_main(void)
    uint32_t sbr_fac = 1;
    unsigned char c = 0;
    void *ret;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
-next_track:
-    err = CODEC_OK;
    /* Clean and initialize decoder structures */
    memset(&demux_res , 0, sizeof(demux_res));
    if (codec_init()) {
        LOGF("FAAD: Codec init error\n");
-        err = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
-  
    file_offset = ci->id3->offset;
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -85,12 +87,13 @@ next_track:
    stream_create(&input_stream,ci);
+    ci->seek_buffer(ci->id3->first_frame_offset);
    /* if qtmovie_read returns successfully, the stream is up to
     * the movie data, which can be used directly by the decoder */
    if (!qtmovie_read(&input_stream, &demux_res)) {
        LOGF("FAAD: File init error\n");
-        err = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* initialise the sound converter */
@@ -98,8 +101,7 @@ next_track:
    if (!decoder) {
        LOGF("FAAD: Decode open error\n");
-        err = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
@@ -109,8 +111,7 @@ next_track:
    err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
    if (err) {
        LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
-        err = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
 #ifdef SBR_DEC
@@ -150,20 +151,19 @@ next_track:
    /* The main decoding loop */
    while (i < demux_res.num_sample_byte_sizes) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
        /* Deal with any pending seek requests */
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
            /* Seek to the desired time position. Important: When seeking in SBR
             * upsampling files the seek_time must be divided by 2 when calling 
             * m4a_seek and the resulting sound_samples_done must be expanded 
             * by a factor 2. This is done via using sbr_fac. */
            if (m4a_seek(&demux_res, &input_stream,
-                          ((ci->seek_time-1)/10/sbr_fac)*(ci->id3->frequency/100),
+                          (param/10/sbr_fac)*(ci->id3->frequency/100),
                          &sound_samples_done, (int*) &i)) {
                sound_samples_done *= sbr_fac;
                elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
@@ -194,8 +194,7 @@ next_track:
        else if (file_offset == 0)
        {
            LOGF("AAC: get_sample_offset error\n");
-            err = CODEC_ERROR;
+            return CODEC_ERROR;
-            goto done;
        }
        
        /* Request the required number of bytes from the input buffer */
@@ -207,8 +206,7 @@ next_track:
        /* NeAACDecDecode may sometimes return NULL without setting error. */
        if (ret == NULL || frame_info.error > 0) {
            LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
-            err = CODEC_ERROR;
+            return CODEC_ERROR;
-            goto done;
        }
        /* Advance codec buffer (no need to call set_offset because of this) */
@@ -251,12 +249,6 @@ next_track:
        i++;
    }
-done:
    LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return err;
 }
diff --git a/apps/codecs/adx.c b/apps/codecs/adx.c
index 832b94797b..a1b57fce58 100644
--- a/apps/codecs/adx.c
+++ b/apps/codecs/adx.c
@@ -45,7 +45,19 @@ static const long cutoff = 500;
 static int16_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        /* we only render 16 bits */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    int channels;
    int sampleswritten, i;
@@ -62,12 +74,8 @@ enum codec_status codec_main(void)
    off_t chanstart, bufoff;
    /*long coef1=0x7298L,coef2=-0x3350L;*/
    long coef1, coef2;
+    intptr_t param;
-    /* Generic codec initialisation */
-    /* we only render 16 bits */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
-  
-next_track:
    DEBUGF("ADX: next_track\n");
    if (codec_init()) {
        return CODEC_ERROR;
@@ -77,10 +85,6 @@ next_track:
    /* init history */
    ch1_1=ch1_2=ch2_1=ch2_2=0;
-    /* wait for track info to load */
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    codec_set_replaygain(ci->id3);
        
    /* Get header */
@@ -226,10 +230,10 @@ next_track:
    /* The main decoder loop */
        
    while (!endofstream) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
        
        /* do we need to loop? */
        if (bufoff > end_adr-18*channels && looping) {
@@ -254,17 +258,17 @@ next_track:
        }
        /* do we need to seek? */
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
            uint32_t newpos;
            
-            DEBUGF("ADX: seek to %ldms\n",ci->seek_time);
+            DEBUGF("ADX: seek to %ldms\n", (long)param);
            endofstream = 0;
            loop_count = 0;
            fade_count = -1; /* disable fade */
            fade_frames = 1;
-            newpos = (((uint64_t)avgbytespersec*(ci->seek_time - 1))
+            newpos = (((uint64_t)avgbytespersec*param)
                      / (1000LL*18*channels))*(18*channels);
            bufoff = chanstart + newpos;
            while (bufoff > end_adr-18*channels) {
@@ -385,9 +389,5 @@ next_track:
           1000LL/avgbytespersec);
    }
-request_next_track:
-    if (ci->request_next_track())
-        goto next_track;
    return CODEC_OK;
 }
diff --git a/apps/codecs/aiff.c b/apps/codecs/aiff.c
index d4cf8660dd..3fc137eaeb 100644
--- a/apps/codecs/aiff.c
+++ b/apps/codecs/aiff.c
@@ -61,9 +61,20 @@ static const struct pcm_codec *get_codec(uint32_t formattag)
    return NULL;
 }
-enum codec_status codec_main(void)
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
-    int status;
    struct pcm_format format;
    uint32_t bytesdone, decodedsamples;
    uint32_t num_sample_frames = 0;
@@ -77,38 +88,28 @@ enum codec_status codec_main(void)
    bool is_aifc = false;
    const struct pcm_codec *codec;
    uint32_t size;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
-  
-next_track:
-    status = CODEC_OK;
    if (codec_init()) {
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    codec_set_replaygain(ci->id3);
    
    /* Need to save offset for later use (cleared indirectly by advance_buffer) */
    bytesdone = ci->id3->offset;
    /* assume the AIFF header is less than 1024 bytes */
+    ci->seek_buffer(0);
    buf = ci->request_buffer(&n, 1024);
    if (n < 54) {
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (memcmp(buf, "FORM", 4) != 0)
    {
        DEBUGF("CODEC_ERROR: does not aiff format %4.4s\n", (char*)&buf[0]);
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (memcmp(&buf[8], "AIFF", 4) == 0)
        is_aifc = false;
@@ -117,8 +118,7 @@ next_track:
    else
    {
        DEBUGF("CODEC_ERROR: does not aiff format %4.4s\n", (char*)&buf[8]);
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    buf += 12;
@@ -141,8 +141,7 @@ next_track:
            {
                DEBUGF("CODEC_ERROR: 'COMM' chunk size=%lu < %d\n",
                       (unsigned long)size, (is_aifc)?22:18);
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            /* num_channels */
            format.channels = ((buf[8]<<8)|buf[9]);
@@ -154,8 +153,7 @@ next_track:
            /* sample_rate (don't use last 4 bytes, only integer fs) */
            if (buf[16] != 0x40) {
                DEBUGF("CODEC_ERROR: weird sampling rate (no @)\n");
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            format.samplespersec = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21])+1;
            format.samplespersec >>= (16 + 14 - buf[17]);
@@ -181,8 +179,7 @@ next_track:
        } else if (memcmp(buf, "SSND", 4)==0) {
            if (format.bitspersample == 0) {
                DEBUGF("CODEC_ERROR: unsupported chunk order\n");
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            /* offset2snd */
            offset2snd = (buf[8]<<24)|(buf[9]<<16)|(buf[10]<<8)|buf[11];
@@ -205,21 +202,18 @@ next_track:
        buf += size;
        if (n < size) {
            DEBUGF("CODEC_ERROR: AIFF header size > 1024\n");
-            status = CODEC_ERROR;
+            return CODEC_ERROR;
-            goto done;
        }
        n -= size;
    } /* while 'SSND' */
    if (format.channels == 0) {
        DEBUGF("CODEC_ERROR: 'COMM' chunk not found or 0-channels file\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.numbytes == 0) {
        DEBUGF("CODEC_ERROR: 'SSND' chunk not found or has zero length\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    codec = get_codec(format.formattag);
@@ -227,14 +221,12 @@ next_track:
    {
        DEBUGF("CODEC_ERROR: AIFC does not support compressionType: 0x%x\n", 
            (unsigned int)format.formattag);
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (!codec->set_format(&format))
    {
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -245,21 +237,18 @@ next_track:
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
    } else {
        DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.samplesperblock == 0)
    {
        DEBUGF("CODEC_ERROR: samplesperblock is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.blockalign == 0)
    {
        DEBUGF("CODEC_ERROR: blockalign is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* check chunksize */
@@ -269,8 +258,7 @@ next_track:
    if (format.chunksize == 0)
    {
        DEBUGF("CODEC_ERROR: chunksize is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    firstblockposn = 1024 - n;
@@ -283,13 +271,13 @@ next_track:
                                                     PCM_SEEK_POS, NULL);
        if (newpos->pos > format.numbytes)
-            goto done;
+            return CODEC_OK;
        if (ci->seek_buffer(firstblockposn + newpos->pos))
        {
            bytesdone      = newpos->pos;
            decodedsamples = newpos->samples;
        }
-        ci->seek_complete();
    } else {
        /* already where we need to be */
        bytesdone = 0;
@@ -299,21 +287,29 @@ next_track:
    endofstream = 0;
    while (!endofstream) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
            /* 3rd args(read_buffer) is unnecessary in the format which AIFF supports. */
-            struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, PCM_SEEK_TIME, NULL);
+            struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME, NULL);
            if (newpos->pos > format.numbytes)
+            {
+                ci->set_elapsed(ci->id3->length);
+                ci->seek_complete();
                break;
+            }
            if (ci->seek_buffer(firstblockposn + newpos->pos))
            {
                bytesdone      = newpos->pos;
                decodedsamples = newpos->samples;
            }
+            ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
            ci->seek_complete();
        }
        aifbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
@@ -326,11 +322,10 @@ next_track:
            endofstream = 1;
        }
-        status = codec->decode(aifbuf, n, samples, &bufcount);
+        if (codec->decode(aifbuf, n, samples, &bufcount) == CODEC_ERROR)
-        if (status == CODEC_ERROR)
        {
            DEBUGF("codec error\n");
-            goto done;
+            return CODEC_ERROR;
        }
        ci->pcmbuf_insert(samples, NULL, bufcount);
@@ -343,13 +338,6 @@ next_track:
        ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
    }
-    status = CODEC_OK;
-done:
-    if (ci->request_next_track())
-        goto next_track;
-exit:
+    return CODEC_OK;
-    return status;
 }
diff --git a/apps/codecs/aiff_enc.c b/apps/codecs/aiff_enc.c
index 69496f70ac..fc44196eb0 100644
--- a/apps/codecs/aiff_enc.c
+++ b/apps/codecs/aiff_enc.c
@@ -359,40 +359,42 @@ static bool init_encoder(void)
    return true;
 } /* init_encoder */
-/* main codec entry point */
+/* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
 {
-    if (!init_encoder())
+    if (reason == CODEC_LOAD) {
-        return CODEC_ERROR;
+        if (!init_encoder())
+            return CODEC_ERROR;
+    }
+    else if (reason == CODEC_UNLOAD) {
+        /* reset parameters to initial state */
+        ci->enc_set_parameters(NULL);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
    /* main encoding loop */
-    while(!ci->stop_codec)
+    while (ci->get_command(NULL) != CODEC_ACTION_HALT)
    {
-        uint32_t *src;
+        uint32_t *src = (uint32_t *)ci->enc_get_pcm_data(PCM_CHUNK_SIZE);
+        struct enc_chunk_hdr *chunk;
-        while ((src = (uint32_t *)ci->enc_get_pcm_data(PCM_CHUNK_SIZE)) != NULL)
+        if (src == NULL)
-        {
+            continue;
-            struct enc_chunk_hdr *chunk;
-            if (ci->stop_codec)
-                break;
-            chunk           = ci->enc_get_chunk();
+        chunk           = ci->enc_get_chunk();
-            chunk->enc_size = enc_size;
+        chunk->enc_size = enc_size;
-            chunk->num_pcm  = PCM_SAMP_PER_CHUNK;
+        chunk->num_pcm  = PCM_SAMP_PER_CHUNK;
-            chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
+        chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
-            chunk_to_aiff_format(src, (uint32_t *)chunk->enc_data);
+        chunk_to_aiff_format(src, (uint32_t *)chunk->enc_data);
-            ci->enc_finish_chunk();
-            ci->yield();
-        }
-        ci->yield();
+        ci->enc_finish_chunk();
    }
-    /* reset parameters to initial state */
-    ci->enc_set_parameters(NULL);
- 
    return CODEC_OK;
-} /* codec_start */
+}
diff --git a/apps/codecs/alac.c b/apps/codecs/alac.c
index cd9129a278..82bda264c3 100644
--- a/apps/codecs/alac.c
+++ b/apps/codecs/alac.c
@@ -32,116 +32,114 @@ CODEC_HEADER
 static int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE] IBSS_ATTR;
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
 {
-  size_t n;
+    if (reason == CODEC_LOAD) {
-  demux_res_t demux_res;
+        /* Generic codec initialisation */
-  stream_t input_stream;
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
-  uint32_t samplesdone;
+        ci->configure(DSP_SET_SAMPLE_DEPTH, ALAC_OUTPUT_DEPTH-1);
-  uint32_t elapsedtime;
+    }
-  int samplesdecoded;
-  unsigned int i;
-  unsigned char* buffer;
-  alac_file alac;
-  int retval;
-  /* Generic codec initialisation */
-  ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
-  ci->configure(DSP_SET_SAMPLE_DEPTH, ALAC_OUTPUT_DEPTH-1);
-  next_track:
-    retval = CODEC_OK;
-  /* Clean and initialize decoder structures */
-   memset(&demux_res , 0, sizeof(demux_res));
-  if (codec_init()) {
-    LOGF("ALAC: Error initialising codec\n");
-    retval = CODEC_ERROR;
-    goto exit;
-  }
-  if (codec_wait_taginfo() != 0)
-    goto done;
-  
-  ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
-  codec_set_replaygain(ci->id3);
-  stream_create(&input_stream,ci);
+    return CODEC_OK;
+}
-  /* Read from ci->id3->offset before calling qtmovie_read. */
+/* this is called for each file to process */
-  samplesdone = (uint32_t)(((uint64_t)(ci->id3->offset) * ci->id3->frequency) /  
+enum codec_status codec_run(void)
-                (ci->id3->bitrate*128));
+{
-  
+    size_t n;
-  /* if qtmovie_read returns successfully, the stream is up to
+    demux_res_t demux_res;
-   * the movie data, which can be used directly by the decoder */
+    stream_t input_stream;
-  if (!qtmovie_read(&input_stream, &demux_res)) {
+    uint32_t samplesdone;
-    LOGF("ALAC: Error initialising file\n");
+    uint32_t elapsedtime = 0;
-    retval = CODEC_ERROR;
+    int samplesdecoded;
-    goto done;
+    unsigned int i;
-  }
+    unsigned char* buffer;
+    alac_file alac;
-  /* initialise the sound converter */
+    intptr_t param;
-  create_alac(demux_res.sound_sample_size, demux_res.num_channels,&alac);
-  alac_set_info(&alac, demux_res.codecdata);
+    /* Clean and initialize decoder structures */
-  
+    memset(&demux_res , 0, sizeof(demux_res));
-  /* Set i for first frame, seek to desired sample position for resuming. */
+    if (codec_init()) {
-  i=0;
+        LOGF("ALAC: Error initialising codec\n");
-  if (samplesdone > 0) {
+        return CODEC_ERROR;
-    if (m4a_seek(&demux_res, &input_stream, samplesdone,
-                  &samplesdone, (int*) &i)) {
-        elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
-        ci->set_elapsed(elapsedtime);
-    } else {
-        samplesdone = 0;
    }
-  }
-  /* The main decoding loop */
+    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
-  while (i < demux_res.num_sample_byte_sizes) {
+    codec_set_replaygain(ci->id3);
-    ci->yield();
-    if (ci->stop_codec || ci->new_track) {
+    ci->seek_buffer(0);
-      break;
+    stream_create(&input_stream,ci);
+    /* Read from ci->id3->offset before calling qtmovie_read. */
+    samplesdone = (uint32_t)(((uint64_t)(ci->id3->offset) * ci->id3->frequency) /  
+                  (ci->id3->bitrate*128));
+  
+    /* if qtmovie_read returns successfully, the stream is up to
+     * the movie data, which can be used directly by the decoder */
+    if (!qtmovie_read(&input_stream, &demux_res)) {
+        LOGF("ALAC: Error initialising file\n");
+        return CODEC_ERROR;
    }
-    /* Deal with any pending seek requests */
+    /* initialise the sound converter */
-    if (ci->seek_time) {
+    create_alac(demux_res.sound_sample_size, demux_res.num_channels,&alac);
-      if (m4a_seek(&demux_res, &input_stream,
+    alac_set_info(&alac, demux_res.codecdata);
-                    ((ci->seek_time-1)/10) * (ci->id3->frequency/100),
+  
-                    &samplesdone, (int *)&i)) {
+    /* Set i for first frame, seek to desired sample position for resuming. */
-        elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
+    i=0;
-        ci->set_elapsed(elapsedtime);
+    if (samplesdone > 0) {
-      }
+        if (m4a_seek(&demux_res, &input_stream, samplesdone,
-      ci->seek_complete();
+                      &samplesdone, (int*) &i)) {
+            elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
+            ci->set_elapsed(elapsedtime);
+        } else {
+            samplesdone = 0;
+        }
    }
-    /* Request the required number of bytes from the input buffer */
+    /* The main decoding loop */
-    buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
+    while (i < demux_res.num_sample_byte_sizes) {
+        enum codec_command_action action = ci->get_command(&param);
+        if (action == CODEC_ACTION_HALT)
+            break;
-    /* Decode one block - returned samples will be host-endian */
+        /* Request the required number of bytes from the input buffer */
-    ci->yield();
+        buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
-    samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, ci->yield);
-    /* Advance codec buffer by amount of consumed bytes */
+        /* Deal with any pending seek requests */
-    ci->advance_buffer(alac.bytes_consumed);
+        if (action == CODEC_ACTION_SEEK_TIME) {
+            if (m4a_seek(&demux_res, &input_stream,
+                         (param/10) * (ci->id3->frequency/100),
+                         &samplesdone, (int *)&i)) {
+                elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
+            }
+            ci->set_elapsed(elapsedtime);
+            ci->seek_complete();
+        }
-    /* Output the audio */
+        /* Request the required number of bytes from the input buffer */
-    ci->yield();
+        buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
-    ci->pcmbuf_insert(outputbuffer[0], outputbuffer[1], samplesdecoded);
-    /* Update the elapsed-time indicator */
+        /* Decode one block - returned samples will be host-endian */
-    samplesdone+=samplesdecoded;
+        samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, ci->yield);
-    elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
+        ci->yield();
-    ci->set_elapsed(elapsedtime);
-    i++;
+        /* Advance codec buffer by amount of consumed bytes */
-  }
+        ci->advance_buffer(alac.bytes_consumed);
-done:
+        /* Output the audio */
-  LOGF("ALAC: Decoded %lu samples\n",(unsigned long)samplesdone);
+        ci->pcmbuf_insert(outputbuffer[0], outputbuffer[1], samplesdecoded);
-  if (ci->request_next_track())
+        /* Update the elapsed-time indicator */
-   goto next_track;
+        samplesdone+=samplesdecoded;
+        elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
+        ci->set_elapsed(elapsedtime);
+        i++;
+    }
-exit:
+    LOGF("ALAC: Decoded %lu samples\n",(unsigned long)samplesdone);
-  return retval;
+    return CODEC_OK;
 }
diff --git a/apps/codecs/ape.c b/apps/codecs/ape.c
index 11d973ab26..8f95a01ec7 100644
--- a/apps/codecs/ape.c
+++ b/apps/codecs/ape.c
@@ -127,13 +127,23 @@ static void ape_resume(struct ape_ctx_t* ape_ctx, size_t resume_offset,
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, APE_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    struct ape_ctx_t ape_ctx;
    uint32_t samplesdone;
    uint32_t elapsedtime;
    size_t bytesleft;
-    int retval;
    uint32_t currentframe;
    uint32_t newfilepos;
@@ -145,33 +155,24 @@ enum codec_status codec_main(void)
    int res;
    int firstbyte;
    size_t resume_offset;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, APE_OUTPUT_DEPTH-1);
-next_track:
-    retval = CODEC_OK;
    if (codec_init()) {
        LOGF("APE: Error initialising codec\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    /* Remember the resume position - when the codec is opened, the
       playback engine will reset it. */
    resume_offset = ci->id3->offset;
+    ci->seek_buffer(0);
    inbuffer = ci->request_buffer(&bytesleft, INPUT_CHUNKSIZE);
    /* Read the file headers to populate the ape_ctx struct */
    if (ape_parseheaderbuf(inbuffer,&ape_ctx) < 0) {
        LOGF("APE: Error reading header\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
    /* Initialise the seektable for this file */
@@ -243,16 +244,16 @@ frame_start:
        /* Decode the frame a chunk at a time */
        while (nblocks > 0)
        {
-            ci->yield();
+            enum codec_command_action action = ci->get_command(&param);
-            if (ci->stop_codec || ci->new_track) {
+            if (action == CODEC_ACTION_HALT)
                goto done;
-            }
            /* Deal with any pending seek requests */
-            if (ci->seek_time) 
+            if (action == CODEC_ACTION_SEEK_TIME) 
            {
                if (ape_calc_seekpos(&ape_ctx,
-                    ((ci->seek_time-1)/10) * (ci->id3->frequency/100),
+                    (param/10) * (ci->id3->frequency/100),
                    &currentframe,
                    &newfilepos,
                    &samplestoskip))
@@ -266,9 +267,12 @@ frame_start:
                    ci->seek_buffer(newfilepos);
                    inbuffer = ci->request_buffer(&bytesleft, INPUT_CHUNKSIZE);
+                    elapsedtime = (samplesdone*10)/(ape_ctx.samplerate/100);
+                    ci->set_elapsed(elapsedtime);
                    ci->seek_complete();
                    goto frame_start;  /* Sorry... */
                }
                ci->seek_complete();
            }
@@ -281,8 +285,7 @@ frame_start:
            {
                /* Frame decoding error, abort */
                LOGF("APE: Frame %lu, error %d\n",(unsigned long)currentframe,res);
-                retval = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            ci->yield();
@@ -320,10 +323,5 @@ frame_start:
 done:
    LOGF("APE: Decoded %lu samples\n",(unsigned long)samplesdone);
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return retval;
 }
diff --git a/apps/codecs/asap.c b/apps/codecs/asap.c
index 9447c738d2..f12dc6a0c5 100644
--- a/apps/codecs/asap.c
+++ b/apps/codecs/asap.c
@@ -29,24 +29,21 @@ CODEC_HEADER
 static byte samples[CHUNK_SIZE] IBSS_ATTR;   /* The sample buffer */
 static ASAP_State asap;         /* asap codec state */
-/* this is the codec entry point */
+/* this is called for each file to process */
-enum codec_status codec_main(void)
+enum codec_status codec_run(void)
 {
    int n_bytes;
    int song;
    int duration;
    char* module;
    int bytesPerSample =2;
+    intptr_t param;
    
-next_track:
    if (codec_init()) {
        DEBUGF("codec init failed\n");
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    codec_set_replaygain(ci->id3);
        
    int bytes_done =0;   
@@ -97,19 +94,20 @@ next_track:
    
    /* The main decoder loop */    
    while (1) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            /* New time is ready in ci->seek_time */
+            /* New time is ready in param */
-                      
            /* seek to pos */
-            ASAP_Seek(&asap,ci->seek_time);
+            ASAP_Seek(&asap,param);
-            /* update elapsed */
-            ci->set_elapsed(ci->seek_time);
            /* update bytes_done */
-            bytes_done = ci->seek_time*44.1*2;    
+            bytes_done = param*44.1*2;    
+            /* update elapsed */
+            ci->set_elapsed((bytes_done / 2) / 44.1);
            /* seek ready */    
            ci->seek_complete();            
        }
@@ -129,10 +127,6 @@ next_track:
        if(n_bytes != sizeof(samples))
            break;
    }
-request_next_track:
-    if (ci->request_next_track())
-        goto next_track;
 
    return CODEC_OK;    
 }
diff --git a/apps/codecs/atrac3_oma.c b/apps/codecs/atrac3_oma.c
index 73f3ad29fd..ab24783368 100644
--- a/apps/codecs/atrac3_oma.c
+++ b/apps/codecs/atrac3_oma.c
@@ -33,24 +33,22 @@ CODEC_HEADER
 static ATRAC3Context q IBSS_ATTR;
-/* this is the codec entry point */
+/* this is called for each file to process */
-enum codec_status codec_main(void)
+enum codec_status codec_run(void)
-{ 
+{
    static size_t buff_size;
    int datasize, res, frame_counter, total_frames, seek_frame_offset;
    uint8_t *bit_buffer;
    int elapsed = 0;
    size_t resume_offset;
+    intptr_t param;
+    enum codec_command_action action = CODEC_ACTION_NULL;
-next_track:
    if (codec_init()) {
        DEBUGF("codec init failed\n");
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    resume_offset = ci->id3->offset;
    codec_set_replaygain(ci->id3);
@@ -60,84 +58,88 @@ next_track:
    ci->configure(DSP_SET_SAMPLE_DEPTH, 17); /* Remark: atrac3 uses s15.0 by default, s15.2 was hacked. */
    ci->configure(DSP_SET_STEREO_MODE, ci->id3->channels == 1 ?
        STEREO_MONO : STEREO_NONINTERLEAVED);
-    
-    res =atrac3_decode_init(&q, ci->id3);
+    ci->seek_buffer(0);
+    res = atrac3_decode_init(&q, ci->id3);
    if(res < 0) {
        DEBUGF("failed to initialize OMA atrac decoder\n");
        return CODEC_ERROR;
    }
+    total_frames = (ci->id3->filesize - ci->id3->first_frame_offset) / FRAMESIZE;
+    frame_counter = 0;
+    
    /* check for a mid-track resume and force a seek time accordingly */
    if(resume_offset > ci->id3->first_frame_offset) {
        resume_offset -= ci->id3->first_frame_offset;
        /* calculate resume_offset in frames */
        resume_offset = (int)resume_offset / FRAMESIZE;
-        ci->seek_time =  (int)resume_offset * ((FRAMESIZE * 8)/BITRATE);                
+        param = (int)resume_offset * ((FRAMESIZE * 8)/BITRATE);
+        action = CODEC_ACTION_SEEK_TIME;
+    }
+    else {
+        ci->set_elapsed(0);
+        ci->seek_buffer(ci->id3->first_frame_offset);
    }
-    total_frames = (ci->id3->filesize - ci->id3->first_frame_offset) / FRAMESIZE;
-    frame_counter = 0;
-    
-    ci->set_elapsed(0);
-    ci->seek_buffer(0);
-    ci->advance_buffer(ci->id3->first_frame_offset);
    /* The main decoder loop */  
-seek_start :         
    while(frame_counter < total_frames)
-    {  
+    {
+        if (action == CODEC_ACTION_NULL)
+            action = ci->get_command(&param);
+        if (action == CODEC_ACTION_HALT)
+            break;
        bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, FRAMESIZE);
-        ci->yield();
+        if (action == CODEC_ACTION_SEEK_TIME) {
-        if (ci->stop_codec || ci->new_track)
+            /* Do not allow seeking beyond the file's length */
-            goto done;
+            if ((unsigned) param > ci->id3->length) {
+                ci->set_elapsed(ci->id3->length);
-            if (ci->seek_time) {
+                ci->seek_complete();
-                ci->set_elapsed(ci->seek_time);
+                break;
+            }       
-                /* Do not allow seeking beyond the file's length */
-                if ((unsigned) ci->seek_time > ci->id3->length) {
+            /* Seek to the start of the track */
-                    ci->seek_complete();
+            if (param == 0) {
-                    goto done;
+                elapsed = 0;
-                }       
+                ci->set_elapsed(0);
+                ci->seek_buffer(ci->id3->first_frame_offset);
-                /* Seek to the start of the track */
+                ci->seek_complete();
-                if (ci->seek_time == 1) {
+                action = CODEC_ACTION_NULL;
-                    ci->set_elapsed(0);
+                continue;           
-                    ci->seek_complete();
+            }                                                                
-                    ci->seek_buffer(ci->id3->first_frame_offset);
-                    elapsed = 0;
+            seek_frame_offset = (param * BITRATE) / (8 * FRAMESIZE);
-                    goto seek_start;           
+            frame_counter = seek_frame_offset;
-                }                                                                
+            ci->seek_buffer(ci->id3->first_frame_offset + seek_frame_offset* FRAMESIZE);
-                seek_frame_offset = (ci->seek_time * BITRATE) / (8 * FRAMESIZE);
+            bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, FRAMESIZE);
-                frame_counter = seek_frame_offset;
+            elapsed = param;
-                ci->seek_buffer(ci->id3->first_frame_offset + seek_frame_offset* FRAMESIZE);
-                bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, FRAMESIZE);
-                elapsed = ci->seek_time;
-                ci->set_elapsed(elapsed);
-                ci->seek_complete(); 
-            }
-            res = atrac3_decode_frame(FRAMESIZE, &q, &datasize, bit_buffer, FRAMESIZE);
-            if(res != (int)FRAMESIZE) {
-                DEBUGF("codec error\n");
-                return CODEC_ERROR;
-            }
-            if(datasize)
-                ci->pcmbuf_insert(q.outSamples, q.outSamples + 1024, q.samples_per_frame / ci->id3->channels);
-            elapsed += (FRAMESIZE * 8) / BITRATE;
            ci->set_elapsed(elapsed);
-        
+            ci->seek_complete(); 
-            ci->advance_buffer(FRAMESIZE);
+        }
-            frame_counter++;
-    }
+        action = CODEC_ACTION_NULL;
-    done:
+        res = atrac3_decode_frame(FRAMESIZE, &q, &datasize, bit_buffer, FRAMESIZE);
-    if (ci->request_next_track())
-        goto next_track;
+        if(res != (int)FRAMESIZE) {
+            DEBUGF("codec error\n");
+            return CODEC_ERROR;
+        }
+        if(datasize)
+            ci->pcmbuf_insert(q.outSamples, q.outSamples + 1024,
+                              q.samples_per_frame / ci->id3->channels);
+        elapsed += (FRAMESIZE * 8) / BITRATE;
+        ci->set_elapsed(elapsed);
+        ci->advance_buffer(FRAMESIZE);
+        frame_counter++;
+    }
    return CODEC_OK;    
 }
diff --git a/apps/codecs/atrac3_rm.c b/apps/codecs/atrac3_rm.c
index 6a77d24283..1322e917ed 100644
--- a/apps/codecs/atrac3_rm.c
+++ b/apps/codecs/atrac3_rm.c
@@ -41,9 +41,9 @@ static void init_rm(RMContext *rmctx)
    memcpy(ci->id3->id3v2buf, (char*)rmctx->codec_extradata, rmctx->extradata_size*sizeof(char));
 }
-/* this is the codec entry point */
+/* this is called for each file to process */
-enum codec_status codec_main(void)
+enum codec_status codec_run(void)
-{ 
+{
    static size_t buff_size;
    int datasize, res, consumed, i, time_offset;
    uint8_t *bit_buffer;
@@ -52,16 +52,14 @@ enum codec_status codec_main(void)
    int scrambling_unit_size, num_units, elapsed = 0;
    int playback_on = -1;
    size_t resume_offset;
+    intptr_t param;
+    enum codec_command_action action = CODEC_ACTION_NULL;
-next_track:
    if (codec_init()) {
        DEBUGF("codec init failed\n");
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    resume_offset = ci->id3->offset;
    codec_set_replaygain(ci->id3);
@@ -69,6 +67,7 @@ next_track:
    ci->memset(&pkt,0,sizeof(RMPacket));
    ci->memset(&q,0,sizeof(ATRAC3Context));
+    ci->seek_buffer(0);
    init_rm(&rmctx);
 
    ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
@@ -84,22 +83,25 @@ next_track:
    h = rmctx.sub_packet_h;
    scrambling_unit_size = h*fs;
    
-    res =atrac3_decode_init(&q, ci->id3);
+    res = atrac3_decode_init(&q, ci->id3);
    if(res < 0) {
        DEBUGF("failed to initialize RM atrac decoder\n");
        return CODEC_ERROR;
    }
+    
    /* check for a mid-track resume and force a seek time accordingly */
    if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) {
        resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
        num_units = (int)resume_offset / scrambling_unit_size;        
        /* put number of subpackets to skip in resume_offset */
        resume_offset /= (sps + PACKET_HEADER_SIZE);
-        ci->seek_time =  (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);                
+        param = (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);
+        action = CODEC_ACTION_SEEK_TIME;
    }
-    
+    else {
-    ci->set_elapsed(0);
+        ci->set_elapsed(0);
+    }
    ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
    /* The main decoder loop */  
@@ -115,22 +117,23 @@ seek_start :
                return CODEC_ERROR;
            }
            else
-                goto done;
+                return CODEC_OK;
        }
        for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
        { 
-            ci->yield();
+            if (action == CODEC_ACTION_NULL)
-            if (ci->stop_codec || ci->new_track)
+                action = ci->get_command(&param);
-                goto done;
-            if (ci->seek_time) {
+            if (action == CODEC_ACTION_HALT)
-                ci->set_elapsed(ci->seek_time);
+                return CODEC_OK;
+            if (action == CODEC_ACTION_SEEK_TIME) {
                /* Do not allow seeking beyond the file's length */
-                if ((unsigned) ci->seek_time > ci->id3->length) {
+                if ((unsigned) param > ci->id3->length) {
+                    ci->set_elapsed(ci->id3->length);
                    ci->seek_complete();
-                    goto done;
+                    return CODEC_OK;
                }       
                ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
@@ -139,12 +142,13 @@ seek_start :
                rmctx.frame_number = 0;
                /* Seek to the start of the track */
-                if (ci->seek_time == 1) {
+                if (param == 0) {
                    ci->set_elapsed(0);
                    ci->seek_complete();
+                    action = CODEC_ACTION_NULL;
                    goto seek_start;           
                }                                                                
-                num_units = ((ci->seek_time)/(sps*1000*8/rmctx.bit_rate))/(h*(fs/sps));                    
+                num_units = (param/(sps*1000*8/rmctx.bit_rate))/(h*(fs/sps));                    
                ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * num_units);
                bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
                consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
@@ -155,12 +159,12 @@ seek_start :
                        return CODEC_ERROR;
                    }
                    else
-                        goto done;
+                        return CODEC_OK;
                }
                packet_count = rmctx.nb_packets - rmctx.audio_pkt_cnt * num_units;
-                rmctx.frame_number = ((ci->seek_time)/(sps*1000*8/rmctx.bit_rate)); 
+                rmctx.frame_number = (param/(sps*1000*8/rmctx.bit_rate)); 
-                while(rmctx.audiotimestamp > (unsigned) ci->seek_time) {
+                while(rmctx.audiotimestamp > (unsigned) param) {
                    rmctx.audio_pkt_cnt = 0;
                    ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * (num_units-1));
                    bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size); 
@@ -168,16 +172,19 @@ seek_start :
                    packet_count += rmctx.audio_pkt_cnt;
                    num_units--;
                }
-                time_offset = ci->seek_time - rmctx.audiotimestamp;
+                time_offset = param - rmctx.audiotimestamp;
                i = (time_offset/((sps * 8 * 1000)/rmctx.bit_rate));
                elapsed = rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i;
                ci->set_elapsed(elapsed);
                ci->seek_complete(); 
            }
+            action = CODEC_ACTION_NULL;
            if(pkt.length)    
                res = atrac3_decode_frame(rmctx.block_align, &q, &datasize, pkt.frames[i], rmctx.block_align);
            else /* indicates that there are no remaining frames */
-                goto done;
+                return CODEC_OK;
            if(res != rmctx.block_align) {
                DEBUGF("codec error\n");
@@ -196,9 +203,5 @@ seek_start :
        ci->advance_buffer(consumed);
    }
-    done :
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-    return CODEC_OK;    
 }
diff --git a/apps/codecs/au.c b/apps/codecs/au.c
index 3f9436c9e7..e06f931cf9 100644
--- a/apps/codecs/au.c
+++ b/apps/codecs/au.c
@@ -106,9 +106,19 @@ static int convert_au_format(unsigned int encoding, struct pcm_format *fmt)
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
-    int status;
    struct pcm_format format;
    uint32_t bytesdone, decodedsamples;
    size_t n;
@@ -119,22 +129,13 @@ enum codec_status codec_main(void)
    off_t firstblockposn;     /* position of the first block in file */
    const struct pcm_codec *codec;
    int offset = 0;
+    intptr_t param;
-    /* Generic codec initialisation */
+ 
-    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
-  
-next_track:
-    status = CODEC_OK;
    if (codec_init()) {
        DEBUGF("codec_init() error\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    codec_set_replaygain(ci->id3);
    
    /* Need to save offset for later use (cleared indirectly by advance_buffer) */
@@ -145,6 +146,7 @@ next_track:
    format.is_little_endian = false;
    /* set format */
+    ci->seek_buffer(0);
    buf = ci->request_buffer(&n, 24);
    if (n < 24 || (memcmp(buf, ".snd", 4) != 0))
    {
@@ -170,8 +172,7 @@ next_track:
        if (offset < 24)
        {
            DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
-            status = CODEC_ERROR;
+            return CODEC_ERROR;
-            goto done;
        }
        /* data size */
        format.numbytes = get_be32(buf + 8);
@@ -182,8 +183,7 @@ next_track:
        if (format.formattag == AU_FORMAT_UNSUPPORT)
        {
            DEBUGF("CODEC_ERROR: sun audio unsupport format: %d\n", get_be32(buf + 12));
-            status = CODEC_ERROR;
+            return CODEC_ERROR;
-            goto done;
        }
        /* skip sample rate */
        format.channels = get_be32(buf + 20);
@@ -202,20 +202,17 @@ next_track:
    if (!codec)
    {
        DEBUGF("CODEC_ERROR: unsupport sun audio format: %x\n", (int)format.formattag);
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (!codec->set_format(&format))
    {
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.numbytes == 0) {
        DEBUGF("CODEC_ERROR: data size is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* check chunksize */
@@ -225,8 +222,7 @@ next_track:
    if (format.chunksize == 0)
    {
        DEBUGF("CODEC_ERROR: chunksize is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -236,8 +232,7 @@ next_track:
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
    } else {
        DEBUGF("CODEC_ERROR: more than 2 channels\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* make sure we're at the correct offset */
@@ -253,7 +248,6 @@ next_track:
            bytesdone      = newpos->pos;
            decodedsamples = newpos->samples;
        }
-        ci->seek_complete();
    } else {
        /* already where we need to be */
        bytesdone = 0;
@@ -263,22 +257,29 @@ next_track:
    endofstream = 0;
    while (!endofstream) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
            /* 3rd args(read_buffer) is unnecessary in the format which Sun Audio supports.  */
-            struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, PCM_SEEK_TIME, NULL);
+            struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME, NULL);
            if (newpos->pos > format.numbytes)
+            {
+                ci->set_elapsed(ci->id3->length);
+                ci->seek_complete();
                break;
+            }
            if (ci->seek_buffer(firstblockposn + newpos->pos))
            {
                bytesdone      = newpos->pos;
                decodedsamples = newpos->samples;
            }
+            ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
            ci->seek_complete();
        }
@@ -290,11 +291,10 @@ next_track:
            endofstream = 1;
        }
-        status = codec->decode(aubuf, n, samples, &bufcount);
+        if (codec->decode(aubuf, n, samples, &bufcount) == CODEC_ERROR)
-        if (status == CODEC_ERROR)
        {
            DEBUGF("codec error\n");
-            goto done;
+            return CODEC_ERROR;
        }
        ci->pcmbuf_insert(samples, NULL, bufcount);
@@ -308,9 +308,5 @@ next_track:
    }
 done:
-    if (ci->request_next_track())
+    return CODEC_OK;
-        goto next_track;
-exit:
-    return status;
 }
diff --git a/apps/codecs/codec_crt0.c b/apps/codecs/codec_crt0.c
index cf14e460ec..33876272c6 100644
--- a/apps/codecs/codec_crt0.c
+++ b/apps/codecs/codec_crt0.c
@@ -27,39 +27,45 @@ struct codec_api *ci DATA_ATTR;
 extern unsigned char plugin_bss_start[];
 extern unsigned char plugin_end_addr[];
-extern enum codec_status codec_main(void);
+extern enum codec_status codec_main(enum codec_entry_call_reason reason);
 /* stub, the entry point is called via its reference in __header to
 * avoid warning with certain compilers */
 int _start(void) {return 0;}
-enum codec_status codec_start(void)
+enum codec_status codec_start(enum codec_entry_call_reason reason)
 {
 #if (CONFIG_PLATFORM & PLATFORM_NATIVE)
-#ifdef USE_IRAM
+    if (reason == CODEC_LOAD)
-    extern char iramcopy[], iramstart[], iramend[], iedata[], iend[];
-    size_t iram_size = iramend - iramstart;
-    size_t ibss_size = iend - iedata;
-    if (iram_size > 0 || ibss_size > 0)
    {
-        ci->memcpy(iramstart, iramcopy, iram_size);
+#ifdef USE_IRAM
-        ci->memset(iedata, 0, ibss_size);
+        extern char iramcopy[], iramstart[], iramend[], iedata[], iend[];
-        /* make the icache (if it exists) up to date with the new code */
+        size_t iram_size = iramend - iramstart;
+        size_t ibss_size = iend - iedata;
+        if (iram_size > 0 || ibss_size > 0)
+        {
+            ci->memcpy(iramstart, iramcopy, iram_size);
+            ci->memset(iedata, 0, ibss_size);
+            /* make the icache (if it exists) up to date with the new code */
+            ci->cpucache_invalidate();
+            /* barrier to prevent reordering iram copy and BSS clearing,
+             * because the BSS segment alias the IRAM copy.
+             */
+            asm volatile ("" ::: "memory");
+        }
+#endif /* PLUGIN_USE_IRAM */
+        ci->memset(plugin_bss_start, 0, plugin_end_addr - plugin_bss_start);
+        /* Some parts of bss may be used via a no-cache alias (at least
+         * portalplayer has this). If we don't clear the cache, those aliases
+         * may read garbage */
        ci->cpucache_invalidate();
-        /* barrier to prevent reordering iram copy and BSS clearing,
-         * because the BSS segment alias the IRAM copy.
-         */
-        asm volatile ("" ::: "memory");
    }
-#endif /* PLUGIN_USE_IRAM */
+#endif /* CONFIG_PLATFORM */
-    ci->memset(plugin_bss_start, 0, plugin_end_addr - plugin_bss_start);
-    /* Some parts of bss may be used via a no-cache alias (at least
-     * portalplayer has this). If we don't clear the cache, those aliases
-     * may read garbage */
-    ci->cpucache_invalidate();
-#endif
-    return codec_main();
+    /* Note: If for any reason codec_main would not be called with CODEC_LOAD
+     * because the above code failed then it must not be ever be called with
+     * any other value and some strategy to avoid doing so must be conceived */
+    return codec_main(reason);
 }
 #if defined(CPU_ARM) && (CONFIG_PLATFORM & PLATFORM_NATIVE)
diff --git a/apps/codecs/cook.c b/apps/codecs/cook.c
index 015618986c..a6b4a1153e 100644
--- a/apps/codecs/cook.c
+++ b/apps/codecs/cook.c
@@ -38,9 +38,9 @@ static void init_rm(RMContext *rmctx)
    memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
 }
-/* this is the codec entry point */
+/* this is called for each file to process */
-enum codec_status codec_main(void)
+enum codec_status codec_run(void)
-{ 
+{
    static size_t buff_size;
    int datasize, res, consumed, i, time_offset;
    uint8_t *bit_buffer;
@@ -48,16 +48,14 @@ enum codec_status codec_main(void)
    uint32_t packet_count;
    int scrambling_unit_size, num_units;
    size_t resume_offset;
+    intptr_t param = 0;
+    enum codec_command_action action = CODEC_ACTION_NULL;
-next_track:
    if (codec_init()) {
        DEBUGF("codec init failed\n");
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    resume_offset = ci->id3->offset;
    codec_set_replaygain(ci->id3);
@@ -65,6 +63,8 @@ next_track:
    ci->memset(&pkt,0,sizeof(RMPacket));
    ci->memset(&q,0,sizeof(COOKContext));
+    ci->seek_buffer(0);
    init_rm(&rmctx);
 
    ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
@@ -87,20 +87,21 @@ next_track:
        DEBUGF("failed to initialize cook decoder\n");
        return CODEC_ERROR;
    }
-    
    /* check for a mid-track resume and force a seek time accordingly */
    if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) {
        resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
        num_units = (int)resume_offset / scrambling_unit_size;    
        /* put number of subpackets to skip in resume_offset */
        resume_offset /= (sps + PACKET_HEADER_SIZE);
-        ci->seek_time =  (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);                
+        param = (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);
+        action = CODEC_ACTION_SEEK_TIME;
    }
    ci->set_elapsed(0);
    ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
-    /* The main decoder loop */  
+    /* The main decoder loop */
 seek_start :         
    while(packet_count)
    {  
@@ -112,18 +113,19 @@ seek_start :
        }
       
        for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
-        { 
+        {
-            ci->yield();
+            if (action == CODEC_ACTION_NULL)
-            if (ci->stop_codec || ci->new_track)
+                action = ci->get_command(&param);
-                goto done;
-            if (ci->seek_time) {
+            if (action == CODEC_ACTION_HALT)
-                ci->set_elapsed(ci->seek_time);
+                return CODEC_OK;
+            if (action == CODEC_ACTION_SEEK_TIME) {
                /* Do not allow seeking beyond the file's length */
-                if ((unsigned) ci->seek_time > ci->id3->length) {
+                if ((unsigned) param > ci->id3->length) {
+                    ci->set_elapsed(ci->id3->length);
                    ci->seek_complete();
-                    goto done;
+                    return CODEC_OK;
                }       
                ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
@@ -132,22 +134,24 @@ seek_start :
                rmctx.frame_number = 0;
                /* Seek to the start of the track */
-                if (ci->seek_time == 1) {
+                if (param == 0) {
                    ci->set_elapsed(0);
                    ci->seek_complete();
+                    action = CODEC_ACTION_NULL;
                    goto seek_start;           
                }                                                                
-                num_units = ((ci->seek_time)/(sps*1000*8/rmctx.bit_rate))/(h*(fs/sps));                    
+                num_units = (param/(sps*1000*8/rmctx.bit_rate))/(h*(fs/sps));                    
                ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * num_units);
                bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
                consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
                if(consumed < 0) {
                     DEBUGF("rm_get_packet failed\n");
-                     return CODEC_ERROR;
+                    ci->seek_complete();
+                    return CODEC_ERROR;
                } 
                packet_count = rmctx.nb_packets - rmctx.audio_pkt_cnt * num_units;
-                rmctx.frame_number = ((ci->seek_time)/(sps*1000*8/rmctx.bit_rate)); 
+                rmctx.frame_number = (param/(sps*1000*8/rmctx.bit_rate)); 
-                while(rmctx.audiotimestamp > (unsigned) ci->seek_time) {
+                while(rmctx.audiotimestamp > (unsigned) param) {
                    rmctx.audio_pkt_cnt = 0;
                    ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * (num_units-1));
                    bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size); 
@@ -155,11 +159,14 @@ seek_start :
                    packet_count += rmctx.audio_pkt_cnt;
                    num_units--;
                }
-                time_offset = ci->seek_time - rmctx.audiotimestamp;
+                time_offset = param - rmctx.audiotimestamp;
                i = (time_offset/((sps * 8 * 1000)/rmctx.bit_rate));
                ci->set_elapsed(rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i);
                ci->seek_complete(); 
-            }    
+            }
+            action = CODEC_ACTION_NULL;
            res = cook_decode_frame(&rmctx,&q, rm_outbuf, &datasize, pkt.frames[i], rmctx.block_align);
            rmctx.frame_number++;
@@ -181,9 +188,5 @@ seek_start :
        ci->advance_buffer(consumed);
    }
-    done :
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-    return CODEC_OK;    
 }
diff --git a/apps/codecs/flac.c b/apps/codecs/flac.c
index 89d14b98a7..a5521b584f 100644
--- a/apps/codecs/flac.c
+++ b/apps/codecs/flac.c
@@ -418,40 +418,40 @@ static bool flac_seek_offset(FLACContext* fc, uint32_t offset) {
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, FLAC_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    int8_t *buf;
    FLACContext fc;
-    uint32_t samplesdone = 0;
+    uint32_t samplesdone;
    uint32_t elapsedtime;
    size_t bytesleft;
    int consumed;
    int res;
    int frame;
-    int retval;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, FLAC_OUTPUT_DEPTH-1);
-next_track:
-    retval = CODEC_OK;
-        
    if (codec_init()) {
        LOGF("FLAC: Error initialising codec\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    /* Need to save offset for later use (cleared indirectly by flac_init) */
    samplesdone = ci->id3->offset;
    
    if (!flac_init(&fc,ci->id3->first_frame_offset)) {
        LOGF("FLAC: Error initialising codec\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -459,35 +459,34 @@ next_track:
                  STEREO_MONO : STEREO_NONINTERLEAVED);
    codec_set_replaygain(ci->id3);
-    if (samplesdone) {
+    flac_seek_offset(&fc, samplesdone);
-        flac_seek_offset(&fc, samplesdone);
+    samplesdone=0;
-        samplesdone=0;
-    }
    /* The main decoding loop */
    frame=0;
    buf = ci->request_buffer(&bytesleft, MAX_FRAMESIZE);
    while (bytesleft) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
        /* Deal with any pending seek requests */
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            if (flac_seek(&fc,(uint32_t)(((uint64_t)(ci->seek_time-1)
+            if (flac_seek(&fc,(uint32_t)(((uint64_t)param
                *ci->id3->frequency)/1000))) {
                /* Refill the input buffer */
                buf = ci->request_buffer(&bytesleft, MAX_FRAMESIZE);
            }
+            ci->set_elapsed(param);
            ci->seek_complete();
        }
        if((res=flac_decode_frame(&fc,decoded0,decoded1,buf,
                             bytesleft,ci->yield)) < 0) {
             LOGF("FLAC: Frame %d, error %d\n",frame,res);
-             retval = CODEC_ERROR;
+             return CODEC_ERROR;
-             goto done;
        }
        consumed=fc.gb.index/8;
        frame++;
@@ -507,14 +506,7 @@ next_track:
        buf = ci->request_buffer(&bytesleft, MAX_FRAMESIZE);
    }
-    retval = CODEC_OK;
-done:
    LOGF("FLAC: Decoded %lu samples\n",(unsigned long)samplesdone);
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return retval;
 }
diff --git a/apps/codecs/lib/codeclib.c b/apps/codecs/lib/codeclib.c
index af0894c498..443c0bbdcf 100644
--- a/apps/codecs/lib/codeclib.c
+++ b/apps/codecs/lib/codeclib.c
@@ -33,6 +33,16 @@ unsigned char* mp3buf;     // The actual MP3 buffer from Rockbox
 unsigned char* mallocbuf;  // 512K from the start of MP3 buffer
 unsigned char* filebuf;    // The rest of the MP3 buffer
+/* this is the default codec entry point for when nothing needs to be done
+   on load or unload */
+enum codec_status __attribute__((weak))
+codec_main(enum codec_entry_call_reason reason)
+{
+    /* Nothing to do */
+    return CODEC_OK;
+    (void)reason;
+}
 int codec_init(void)
 {
    mem_ptr = 0;
@@ -41,7 +51,7 @@ int codec_init(void)
    return 0;
 }
-void codec_set_replaygain(struct mp3entry* id3)
+void codec_set_replaygain(const struct mp3entry *id3)
 {
    ci->configure(DSP_SET_TRACK_GAIN, id3->track_gain);
    ci->configure(DSP_SET_ALBUM_GAIN, id3->album_gain);
@@ -49,19 +59,6 @@ void codec_set_replaygain(struct mp3entry* id3)
    ci->configure(DSP_SET_ALBUM_PEAK, id3->album_peak);
 }
-/* Note: codec really needs its own private metdata copy for the current
-   track being processed in order to be stable. */
-int codec_wait_taginfo(void)
-{
-    while (!*ci->taginfo_ready && !ci->stop_codec && !ci->new_track)
-        ci->sleep(0);
-    if (ci->stop_codec)
-        return -1;
-    if (ci->new_track)
-        return 1;
-    return 0;
-}
 /* Various "helper functions" common to all the xxx2wav decoder plugins  */
diff --git a/apps/codecs/lib/codeclib.h b/apps/codecs/lib/codeclib.h
index 41b466ed1f..30091c5333 100644
--- a/apps/codecs/lib/codeclib.h
+++ b/apps/codecs/lib/codeclib.h
@@ -156,8 +156,7 @@ static inline unsigned int bs_generic(unsigned int v, int mode)
 /* Various codec helper functions */
 int codec_init(void);
-void codec_set_replaygain(struct mp3entry* id3);
+void codec_set_replaygain(const struct mp3entry *id3);
-int codec_wait_taginfo(void); /* 0 = success */
 #ifdef RB_PROFILE
 void __cyg_profile_func_enter(void *this_fn, void *call_site)
diff --git a/apps/codecs/mod.c b/apps/codecs/mod.c
index 4ace721a1e..3dfaac663f 100644
--- a/apps/codecs/mod.c
+++ b/apps/codecs/mod.c
@@ -1218,47 +1218,37 @@ void synthrender(int32_t *renderbuffer, int samplecount)
    }
 }
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Make use of 44.1khz */
+        ci->configure(DSP_SET_FREQUENCY, 44100);
+        /* Sample depth is 28 bit host endian */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+        /* Stereo output */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
+    }
+    return CODEC_OK;
+}
-enum codec_status codec_main(void)
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    size_t n;
    unsigned char *modfile;
    int old_patterntableposition;
    int bytesdone;
+    intptr_t param;
-next_track:
    if (codec_init()) {
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    codec_set_replaygain(ci->id3);
    /* Load MOD file */
-    /*
-     * This is the save way
-    size_t bytesfree;
-    unsigned int filesize;
-    p = modfile;
-    bytesfree=sizeof(modfile);
-    while ((n = ci->read_filebuf(p, bytesfree)) > 0) {
-        p += n;
-        bytesfree -= n;
-        if (bytesfree == 0)
-           return CODEC_ERROR;
-    }
-    filesize = p-modfile;
-    if (filesize == 0)
-        return CODEC_ERROR;
-    */
-    /* Directly use mod in buffer */
    ci->seek_buffer(0);
    modfile = ci->request_buffer(&n, ci->filesize);
    if (!modfile || n < (size_t)ci->filesize) {
@@ -1268,27 +1258,22 @@ next_track:
    initmodplayer();
    loadmod(modfile);
-    /* Make use of 44.1khz */
-    ci->configure(DSP_SET_FREQUENCY, 44100);
-    /* Sample depth is 28 bit host endian */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
-    /* Stereo output */
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
    /* The main decoder loop */
    ci->set_elapsed(0);
    bytesdone = 0;
    old_patterntableposition = 0;
    while (1) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            /* New time is ready in ci->seek_time */
+            /* New time is ready in param */
-            modplayer.patterntableposition = ci->seek_time/1000;
+            modplayer.patterntableposition = param/1000;
            modplayer.currentline = 0;
+            ci->set_elapsed(modplayer.patterntableposition*1000+500);
            ci->seek_complete();
        }
@@ -1305,9 +1290,5 @@ next_track:
    }
-request_next_track:
-    if (ci->request_next_track())
-        goto next_track;
    return CODEC_OK;
 }
diff --git a/apps/codecs/mp3_enc.c b/apps/codecs/mp3_enc.c
index e7893fd14a..2f5528f74c 100644
--- a/apps/codecs/mp3_enc.c
+++ b/apps/codecs/mp3_enc.c
@@ -2584,45 +2584,46 @@ static bool enc_init(void)
    return true;
 } /* enc_init */
-enum codec_status codec_main(void)
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
 {
-    /* Generic codec initialisation */
+    if (reason == CODEC_LOAD) {
-    if (!enc_init())
+        if (!enc_init())
-        return CODEC_ERROR;
+            return CODEC_ERROR;
+    }
+    else if (reason == CODEC_UNLOAD) {
+        /* reset parameters to initial state */
+        ci->enc_set_parameters(NULL);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
    /* main encoding loop */
-    while (!ci->stop_codec)
+    while(ci->get_command(NULL) != CODEC_ACTION_HALT)
    {
-        char *buffer;
+        char *buffer = buffer = ci->enc_get_pcm_data(pcm_chunk_size);
+        struct enc_chunk_hdr *chunk;
-        while ((buffer = ci->enc_get_pcm_data(pcm_chunk_size)) != NULL)
-        {
-            struct enc_chunk_hdr *chunk;
-            if (ci->stop_codec)
-                break;
-            chunk           = ci->enc_get_chunk();
+        if(buffer == NULL)
-            chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
+            continue;
-            encode_frame(buffer, chunk);
+        chunk           = ci->enc_get_chunk();
+        chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
-            if (chunk->num_pcm < samp_per_frame)
+        encode_frame(buffer, chunk);
-            {
-                ci->enc_unget_pcm_data(pcm_chunk_size - chunk->num_pcm*4);
-                chunk->num_pcm = samp_per_frame;
-            }
-            ci->enc_finish_chunk();
+        if (chunk->num_pcm < samp_per_frame)
+        {
-            ci->yield();
+            ci->enc_unget_pcm_data(pcm_chunk_size - chunk->num_pcm*4);
+            chunk->num_pcm = samp_per_frame;
        }
-        ci->yield();
+        ci->enc_finish_chunk();
    }
-    /* reset parameters to initial state */
-    ci->enc_set_parameters(NULL);
    return CODEC_OK;
-} /* codec_start */
+}
diff --git a/apps/codecs/mpa.c b/apps/codecs/mpa.c
index 4b49775029..c9e2131450 100644
--- a/apps/codecs/mpa.c
+++ b/apps/codecs/mpa.c
@@ -268,8 +268,8 @@ static bool mad_synth_thread_create(void)
 static void mad_synth_thread_quit(void)
 {
-    /*mop up COP thread*/
+    /* mop up COP thread */
-    die=1;
+    die = 1;
    ci->semaphore_release(&synth_pending_sem);
    ci->thread_wait(mad_synth_thread_id);
    ci->cpucache_invalidate();
@@ -299,9 +299,30 @@ static inline void mad_synth_thread_unwait_pcm(void)
 #endif /* MPA_SYNTH_ON_COP */
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Create a decoder instance */
+        if (codec_init())
+            return CODEC_ERROR;
+        ci->configure(DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);
+        /* does nothing on 1 processor systems except return true */
+        if(!mad_synth_thread_create())
+            return CODEC_ERROR;
+    }
+    else if (reason == CODEC_UNLOAD) {
+        /* mop up COP thread - MT only */
+        mad_synth_thread_quit();
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
-    int status;
    size_t size;
    int file_end;
    int samples_to_skip; /* samples to skip in total for this file (at start) */
@@ -312,27 +333,12 @@ enum codec_status codec_main(void)
    unsigned long current_frequency = 0;
    int framelength;
    int padding = MAD_BUFFER_GUARD; /* to help mad decode the last frame */
+    intptr_t param;
-    if (codec_init())
-        return CODEC_ERROR;
-    /* Create a decoder instance */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);
-    /*does nothing on 1 processor systems except return true*/
-    if(!mad_synth_thread_create())
-        return CODEC_ERROR;
-    
-next_track:
-    status = CODEC_OK;
    /* Reinitializing seems to be necessary to avoid playback quircks when seeking. */
    init_mad();
    file_end = 0;
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
    current_frequency = ci->id3->frequency;
@@ -379,29 +385,35 @@ next_track:
    /* This is the decoding loop. */
    while (1) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
            int newpos;
            /*make sure the synth thread is idle before seeking - MT only*/
            mad_synth_thread_wait_pcm();
            mad_synth_thread_unwait_pcm();
-            samplesdone = ((int64_t)(ci->seek_time-1))*current_frequency/1000;
+            samplesdone = ((int64_t)param)*current_frequency/1000;
-            if (ci->seek_time-1 == 0) {
+            if (param == 0) {
                newpos = ci->id3->first_frame_offset;
                samples_to_skip = start_skip;
            } else {
-                newpos = get_file_pos(ci->seek_time-1);
+                newpos = get_file_pos(param);
                samples_to_skip = 0;
            }
            if (!ci->seek_buffer(newpos))
+            {
+                ci->seek_complete();
                break;
+            }
+            ci->set_elapsed((samplesdone * 1000) / current_frequency);
            ci->seek_complete();
            init_mad();
            framelength = 0;
@@ -435,8 +447,7 @@ next_track:
                continue;
            } else {
                /* Some other unrecoverable error */
-                status = CODEC_ERROR;
+               return CODEC_ERROR;
-                break;
            }
        }
@@ -504,12 +515,5 @@ next_track:
                          framelength - stop_skip);
    }
-request_next_track:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-    /*mop up COP thread - MT only*/
-    mad_synth_thread_quit();
-    return status;
 }
diff --git a/apps/codecs/mpc.c b/apps/codecs/mpc.c
index 187c37e597..bbe2d9943b 100644
--- a/apps/codecs/mpc.c
+++ b/apps/codecs/mpc.c
@@ -52,8 +52,20 @@ static mpc_int32_t get_size_impl(mpc_reader *reader)
    return ci->filesize;
 }
-/* This is the codec entry point. */
+/* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* musepack's sample representation is 18.14
+         * DSP_SET_SAMPLE_DEPTH = 14 (FRACT) + 16 (NATIVE) - 1 (SIGN) = 29 */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    mpc_int64_t samplesdone;
    uint32_t frequency;     /* 0.1 kHz accuracy */
@@ -64,39 +76,27 @@ enum codec_status codec_main(void)
    mpc_streaminfo info;
    mpc_frame_info frame;
    mpc_demux *demux = NULL;
-    int retval;
+    intptr_t param;
    
    frame.buffer = sample_buffer;
-    
-    /* musepack's sample representation is 18.14
-     * DSP_SET_SAMPLE_DEPTH = 14 (FRACT) + 16 (NATIVE) - 1 (SIGN) = 29 */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
-    
    /* Create a decoder instance */
    reader.read     = read_impl;
    reader.seek     = seek_impl;
    reader.tell     = tell_impl;
    reader.get_size = get_size_impl;
-next_track:
-    retval = CODEC_OK;
    if (codec_init()) 
-    {
+        return CODEC_ERROR;
-        retval = CODEC_ERROR;
-        goto exit;
-    }
-    if (codec_wait_taginfo() != 0)
+    /* Prep position */
-        goto done;
+    ci->seek_buffer(0);
    /* Initialize demux/decoder. */
    demux = mpc_demux_init(&reader);
    if (NULL == demux)
-    {
+        return CODEC_ERROR;
-        retval = CODEC_ERROR;
-        goto done;
-    }
    /* Read file's streaminfo data. */
    mpc_demux_get_info(demux, &info);
    
@@ -117,11 +117,8 @@ next_track:
        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
    else if (info.channels == 1)
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
-    else 
+    else
-    {
+        return CODEC_ERROR;
-       retval = CODEC_ERROR;
-       goto done;
-    }
    
    codec_set_replaygain(ci->id3);
@@ -142,21 +139,24 @@ next_track:
    /* This is the decoding loop. */
    do 
    {
+        enum codec_command_action action = ci->get_command(&param);
+        if (action == CODEC_ACTION_HALT)
+            return CODEC_OK;
        /* Complete seek handler. */
-        if (ci->seek_time) 
+        if (action == CODEC_ACTION_SEEK_TIME) 
        {
-            mpc_int64_t new_offset = ((ci->seek_time - 1)/10)*frequency;
+            mpc_int64_t new_offset = (param/10)*frequency;
            if (mpc_demux_seek_sample(demux, new_offset) == MPC_STATUS_OK) 
            {
                samplesdone = new_offset;
-                ci->set_elapsed(ci->seek_time);
            }
+            elapsed_time = (samplesdone*10)/frequency;
+            ci->set_elapsed(elapsed_time);
            ci->seek_complete();
        }
-        
-        /* Stop or skip occured, exit decoding loop. */
-        if (ci->stop_codec || ci->new_track)
-            break;
        /* Decode one frame. */
        status = mpc_demux_decode(demux, &frame);
@@ -164,8 +164,7 @@ next_track:
        if (frame.bits == -1)
        {
            /* Decoding error, exit decoding loop. */
-            retval = (status == MPC_STATUS_OK) ? CODEC_OK : CODEC_ERROR;
+            return (status == MPC_STATUS_OK) ? CODEC_OK : CODEC_ERROR;
-            goto done;
        } 
        else 
        {
@@ -181,11 +180,4 @@ next_track:
            ci->set_offset( (samplesdone * byterate)/(frequency*100) );
        }
    } while (true);
-done:
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return retval;
 }
diff --git a/apps/codecs/nsf.c b/apps/codecs/nsf.c
index 2f37da81d2..72f6974214 100644
--- a/apps/codecs/nsf.c
+++ b/apps/codecs/nsf.c
@@ -4307,46 +4307,44 @@ static void set_codec_track(int t, int d) {
        nSilenceTrackMS=5000;
        SetFadeTime(track,track+fade, fNSFPlaybackSpeed,def);
    }
-    ci->id3->elapsed=d*1000; /* d is track no to display */
+    ci->set_elapsed(d*1000); /* d is track no to display */
 }
+/** Operational info **/
+static int track = 0;
+static char last_path[MAX_PATH];
+static int dontresettrack = 0;
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* we only render 16 bits, 44.1KHz, Stereo */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
+        ci->configure(DSP_SET_FREQUENCY, 44100);
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
+        RebuildOutputTables();
+    }
+    return CODEC_OK;
+}
+    
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    int written;
    uint8_t *buf;
    size_t n;
    int endofstream; /* end of stream flag */
-    int track;
+    int usingplaylist = 0;
-    int dontresettrack;
-    char last_path[MAX_PATH];
-    int usingplaylist;
-    /* we only render 16 bits */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
-    ci->configure(DSP_SET_FREQUENCY, 44100);
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
-    
-    RebuildOutputTables();
-    dontresettrack=0;
-    last_path[0]='\0';
-    track=0;
    
-next_track:
-    usingplaylist=0;
    DEBUGF("NSF: next_track\n");
    if (codec_init()) {
        return CODEC_ERROR;
    }
    DEBUGF("NSF: after init\n");
    
-    /* wait for track info to load */
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
-        
    codec_set_replaygain(ci->id3);
        
    /* Read the entire file */
@@ -4408,22 +4406,27 @@ init_nsf:
    reset_profile_timers();
    
    while (!endofstream) {
+        intptr_t param;
+        enum codec_command_action action = ci->get_command(&param);
-        ci->yield();
+        if (action == CODEC_ACTION_HALT)
-        if (ci->stop_codec || ci->new_track) {
            break;
-        }
-        if (ci->seek_time >0) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            track=ci->seek_time/1000;
+            if (param > 0) {
-            if (usingplaylist) {
+                track=param/1000;
-                if (track>=nPlaylistSize) break;
+                if (usingplaylist) {
-            } else {
+                    if (track>=nPlaylistSize) break;
-                if (track>=nTrackCount) break;
+                } else {
+                    if (track>=nTrackCount) break;
+                }
+                dontresettrack=1;
+                ci->seek_complete();
+                goto init_nsf;
+            }
+            else {
+                ci->seek_complete();
            }
-            ci->seek_complete();
-            dontresettrack=1;
-            goto init_nsf;
        }
        ENTER_TIMER(total);
@@ -4449,22 +4452,17 @@ init_nsf:
    
    print_timers(last_path,track);
-request_next_track:
-    if (ci->request_next_track()) {
    if (ci->global_settings->repeat_mode==REPEAT_ONE) {
        /* in repeat one mode just advance to the next track */
        track++;
        if (track>=nTrackCount) track=0;
        dontresettrack=1;
        /* at this point we can't tell if another file has been selected */
-        goto next_track;
    } else {
        /* otherwise do a proper load of the next file */
        dontresettrack=0;
        last_path[0]='\0';
    }
-    goto next_track; /* when we fall through here we'll reload the file */
-    }
    
    return CODEC_OK;
 }
diff --git a/apps/codecs/raac.c b/apps/codecs/raac.c
index b322ae7df3..4b73f41462 100644
--- a/apps/codecs/raac.c
+++ b/apps/codecs/raac.c
@@ -35,8 +35,21 @@ static void init_rm(RMContext *rmctx)
 static RMContext rmctx;
 static RMPacket pkt;
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    static NeAACDecFrameInfo frame_info;
    NeAACDecHandle decoder;
@@ -49,26 +62,21 @@ enum codec_status codec_main(void)
    unsigned char c = 0; /* channels */
    int playback_on = -1;
    size_t resume_offset;
+    intptr_t param;
-    /* Generic codec initialisation */
+    enum codec_command_action action = CODEC_ACTION_NULL;
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
-next_track:
-    err = CODEC_OK;
    if (codec_init()) {
        DEBUGF("FAAD: Codec init error\n");
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    resume_offset = ci->id3->offset;
    ci->memset(&rmctx,0,sizeof(RMContext));
    ci->memset(&pkt,0,sizeof(RMPacket));
+    ci->seek_buffer(0);
    init_rm(&rmctx);    
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
    codec_set_replaygain(ci->id3);
@@ -78,9 +86,9 @@ next_track:
    
    if (!decoder) {
        DEBUGF("FAAD: Decode open error\n");
-        err = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
+    }
-    }   
    NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
    conf->outputFormat = FAAD_FMT_16BIT; /* irrelevant, we don't convert */
    NeAACDecSetConfiguration(decoder, conf);    
@@ -91,8 +99,7 @@ next_track:
   
    if (err) {
        DEBUGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
-        err = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    
    /* check for a mid-track resume and force a seek time accordingly */
@@ -100,36 +107,38 @@ next_track:
        resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
        /* put number of subpackets to skip in resume_offset */
        resume_offset /= (rmctx.block_align + PACKET_HEADER_SIZE);
-        ci->seek_time =  (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate);                
+        param = (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate);
+        action = CODEC_ACTION_SEEK_TIME;
    }
-    
-    ci->id3->frequency = s;    
+    ci->id3->frequency = s; /* FIXME: Won't get it to the UI */    
    ci->set_elapsed(0);
    ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
    /* The main decoding loop */
-seek_start:
    while (1) {
-        ci->yield();
+        if (action == CODEC_ACTION_NULL)
-        if (ci->stop_codec || ci->new_track) {
+            action = ci->get_command(&param);
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
-        if (ci->seek_time) {            
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            
            /* Do not allow seeking beyond the file's length */
-            if ((unsigned) ci->seek_time > ci->id3->length) {
+            if ((unsigned) param > ci->id3->length) {
+                ci->set_elapsed(ci->id3->length);
                ci->seek_complete();
-                goto done;
+                break;
            }       
            ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE);            
            /* Seek to the start of the track */
-            if (ci->seek_time == 1) {
+            if (param == 0) {
                ci->set_elapsed(0);
                ci->seek_complete();
-                goto seek_start;           
+                action = CODEC_ACTION_NULL;
+                continue;          
            }
            
            skipped = 0;                                                                                       
@@ -141,21 +150,30 @@ seek_start:
                    if(playback_on == -1) {
                    /* Error only if packet-parsing failed and playback hadn't started */
                        DEBUGF("rm_get_packet failed\n");
+                        ci->seek_complete();
                        return CODEC_ERROR;
                    }
-                    else
+                    else {
-                        goto done;
+                        ci->seek_complete();
+                        return CODEC_OK;
+                    }
                }
                skipped += pkt.length;
-                if(pkt.timestamp > (unsigned)ci->seek_time) break;                
+                if(pkt.timestamp > (unsigned)param)
+                    break;
+                
                ci->advance_buffer(pkt.length);
            }           
            ci->seek_buffer(pkt_offset + rmctx.data_offset + DATA_HEADER_SIZE);
            buffer = ci->request_buffer(&n,rmctx.audio_framesize + 1000);
            NeAACDecPostSeekReset(decoder, decoder->frame);
+            ci->set_elapsed(pkt.timestamp);
            ci->seek_complete();            
        }
+        action = CODEC_ACTION_NULL;
        /* Request the required number of bytes from the input buffer */ 
        buffer=ci->request_buffer(&n,rmctx.audio_framesize + 1000);        
        consumed = rm_get_packet(&buffer, &rmctx, &pkt);
@@ -167,20 +185,20 @@ seek_start:
                return CODEC_ERROR;
            }
            else
-                goto done;
+                break;
        }
        
        playback_on = 1;
        if (pkt.timestamp >= ci->id3->length)
-            goto done;
+            break;
        /* Decode one block - returned samples will be host-endian */                           
        for(i = 0; i < rmctx.sub_packet_cnt; i++) {
            ret = NeAACDecDecode(decoder, &frame_info, buffer, rmctx.sub_packet_lengths[i]);
            buffer += rmctx.sub_packet_lengths[i];                      
            if (frame_info.error > 0) {
                DEBUGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
-                err = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto exit;
            }
            ci->pcmbuf_insert(decoder->time_out[0],
                              decoder->time_out[1],
@@ -191,11 +209,5 @@ seek_start:
        ci->advance_buffer(pkt.length);              
    }
-done:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return err;
 }
- 
diff --git a/apps/codecs/shorten.c b/apps/codecs/shorten.c
index 83a9c34da8..db66991679 100644
--- a/apps/codecs/shorten.c
+++ b/apps/codecs/shorten.c
@@ -37,7 +37,19 @@ static int32_t offset1[MAX_OFFSET_SIZE] IBSS_ATTR;
 static int8_t ibuf[MAX_BUFFER_SIZE] IBSS_ATTR;
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+        ci->configure(DSP_SET_SAMPLE_DEPTH, SHN_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    ShortenContext sc;
    uint32_t samplesdone;
@@ -45,21 +57,14 @@ enum codec_status codec_main(void)
    int8_t *buf;
    int consumed, res, nsamples;
    size_t bytesleft;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
-    ci->configure(DSP_SET_SAMPLE_DEPTH, SHN_OUTPUT_DEPTH-1);
-    
-next_track:
    /* Codec initialization */
    if (codec_init()) {
        LOGF("Shorten: codec_init error\n");
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    codec_set_replaygain(ci->id3);
    /* Shorten decoder initialization */
@@ -103,14 +108,15 @@ seek_start:
    samplesdone = 0;
    buf = ci->request_buffer(&bytesleft, MAX_BUFFER_SIZE);
    while (bytesleft) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
        /* Seek to start of track */
-        if (ci->seek_time == 1) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            if (ci->seek_buffer(sc.header_bits/8 + ci->id3->first_frame_offset)) {
+            if (param == 0 &&
+                ci->seek_buffer(sc.header_bits/8 + ci->id3->first_frame_offset)) {
                sc.bitindex = sc.header_bits - 8*(sc.header_bits/8);
                ci->set_elapsed(0);
                ci->seek_complete();
@@ -128,7 +134,7 @@ seek_start:
        if (res == FN_ERROR) {
            LOGF("Shorten: shorten_decode_frames error (%lu)\n",
                (unsigned long)samplesdone);
-            break;
+            return CODEC_ERROR;
        } else {
            /* Insert decoded samples in pcmbuf */
            if (nsamples) {
@@ -153,9 +159,5 @@ seek_start:
        sc.bitindex = sc.gb.index - 8*consumed;
    }
-request_next_track:
-    if (ci->request_next_track())
-        goto next_track;
    return CODEC_OK;
 }
diff --git a/apps/codecs/sid.c b/apps/codecs/sid.c
index 52c1289fff..0edbabe0b6 100644
--- a/apps/codecs/sid.c
+++ b/apps/codecs/sid.c
@@ -1203,34 +1203,47 @@ unsigned short LoadSIDFromMemory(void *pSidData, unsigned short *load_addr,
    return *load_addr;
 }
+static int nSamplesRendered = 0;
+static int nSamplesPerCall = 882;  /* This is PAL SID single speed (44100/50Hz) */
+static int nSamplesToRender = 0;
-enum codec_status codec_main(void)
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
 {
-    unsigned int filesize;
+    if (reason == CODEC_LOAD) {
+        /* Make use of 44.1khz */
+        ci->configure(DSP_SWITCH_FREQUENCY, 44100);
+        /* Sample depth is 28 bit host endian */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+        /* Mono output */
+        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
+    unsigned int filesize;
    unsigned short load_addr, init_addr, play_addr;
    unsigned char subSongsMax, subSong, song_speed;
+    intptr_t param;
-    int nSamplesRendered = 0;
-    int nSamplesPerCall = 882;  /* This is PAL SID single speed (44100/50Hz) */
-    int nSamplesToRender = 0;
-next_track:
    if (codec_init()) {
        return CODEC_ERROR;
    }
-    if (codec_wait_taginfo() != 0)
-        goto request_next_track;
    codec_set_replaygain(ci->id3);
    
    /* Load SID file the read_filebuf callback will return the full requested
-     * size if at all possible, so there is no need to loop */    
+     * size if at all possible, so there is no need to loop */
+    ci->seek_buffer(0);
    filesize = ci->read_filebuf(sidfile, sizeof(sidfile));
-    if (filesize == 0)
+    if (filesize == 0) {
        return CODEC_ERROR;
+    }
    
    c64Init(44100);
    LoadSIDFromMemory(sidfile, &load_addr, &init_addr, &play_addr,
@@ -1239,37 +1252,30 @@ next_track:
    cpuJSR(init_addr, subSong);     /* Start the song initialize */
    
-    /* Make use of 44.1khz */
-    ci->configure(DSP_SWITCH_FREQUENCY, 44100);
-    /* Sample depth is 28 bit host endian */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
-    /* Mono output */
-    ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
-    
    /* Set the elapsed time to the current subsong (in seconds) */
    ci->set_elapsed(subSong*1000);
    /* The main decoder loop */    
    while (1) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            /* New time is ready in ci->seek_time */
+            /* New time is ready in param */
            /* Start playing from scratch */
            c64Init(44100);
-            LoadSIDFromMemory(sidfile, &load_addr, &init_addr, &play_addr, &subSongsMax, &subSong, &song_speed, filesize);
+            LoadSIDFromMemory(sidfile, &load_addr, &init_addr, &play_addr,
-            sidPoke(24, 15);                /* Turn on full volume */            
+                              &subSongsMax, &subSong, &song_speed, filesize);
-            subSong = ci->seek_time / 1000; /* Now use the current seek time in seconds as subsong */
+            sidPoke(24, 15);            /* Turn on full volume */            
-            cpuJSR(init_addr, subSong);     /* Start the song initialize */
+            subSong = param / 1000;     /* Now use the current seek time in seconds as subsong */
-            nSamplesToRender = 0;           /* Start the rendering from scratch */
+            cpuJSR(init_addr, subSong); /* Start the song initialize */
-            
+            nSamplesToRender = 0;       /* Start the rendering from scratch */
-            ci->seek_complete();
            /* Set the elapsed time to the current subsong (in seconds) */
+            ci->seek_complete();
            ci->set_elapsed(subSong*1000);
        }
        
@@ -1306,9 +1312,5 @@ next_track:
        ci->pcmbuf_insert(samples, NULL, CHUNK_SIZE);
    }
-request_next_track:
-    if (ci->request_next_track())
-        goto next_track;
    return CODEC_OK;
 }
diff --git a/apps/codecs/smaf.c b/apps/codecs/smaf.c
index 3e8a41387d..9211daa9aa 100644
--- a/apps/codecs/smaf.c
+++ b/apps/codecs/smaf.c
@@ -332,9 +332,20 @@ static uint8_t *read_buffer(size_t *realsize)
    return buffer;
 }
-enum codec_status codec_main(void)
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
-    int status;
    uint32_t decodedsamples;
    size_t n;
    int bufcount;
@@ -342,20 +353,10 @@ enum codec_status codec_main(void)
    uint8_t *smafbuf;
    off_t firstblockposn;     /* position of the first block in file */
    const struct pcm_codec *codec;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
  
-next_track:
+    if (codec_init())
-    status = CODEC_OK;
+        return CODEC_ERROR;
-    if (codec_init()) {
-        status = CODEC_ERROR;
-        goto exit;
-    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    codec_set_replaygain(ci->id3);
@@ -365,24 +366,22 @@ next_track:
    decodedsamples = 0;
    codec = 0;
+    ci->seek_buffer(0);
    if (!parse_header(&format, &firstblockposn))
    {
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    codec = get_codec(format.formattag);
    if (codec == 0)
    {
        DEBUGF("CODEC_ERROR: unsupport audio format: 0x%x\n", (int)format.formattag);
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (!codec->set_format(&format))
    {
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* check chunksize */
@@ -392,8 +391,7 @@ next_track:
    if (format.chunksize == 0)
    {
        DEBUGF("CODEC_ERROR: chunksize is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -404,12 +402,10 @@ next_track:
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
    } else {
        DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->seek_buffer(firstblockposn);
-    ci->seek_complete();
    /* make sure we're at the correct offset */
    if (bytesdone > (uint32_t) firstblockposn)
@@ -419,13 +415,13 @@ next_track:
                                                     PCM_SEEK_POS, &read_buffer);
        if (newpos->pos > format.numbytes)
-            goto done;
+            return CODEC_OK;
        if (ci->seek_buffer(firstblockposn + newpos->pos))
        {
            bytesdone      = newpos->pos;
            decodedsamples = newpos->samples;
        }
-        ci->seek_complete();
    }
    else
    {
@@ -437,23 +433,32 @@ next_track:
    endofstream = 0;
    while (!endofstream) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, PCM_SEEK_TIME,
+            struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME,
                                                         &read_buffer);
            if (newpos->pos > format.numbytes)
+            {
+                ci->set_elapsed(ci->id3->length);
+                ci->seek_complete();
                break;
+            }
            if (ci->seek_buffer(firstblockposn + newpos->pos))
            {
                bytesdone      = newpos->pos;
                decodedsamples = newpos->samples;
            }
+            ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
            ci->seek_complete();
        }
        smafbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
        if (n == 0)
@@ -464,11 +469,10 @@ next_track:
            endofstream = 1;
        }
-        status = codec->decode(smafbuf, n, samples, &bufcount);
+        if (codec->decode(smafbuf, n, samples, &bufcount) == CODEC_ERROR)
-        if (status == CODEC_ERROR)
        {
            DEBUGF("codec error\n");
-            goto done;
+            return CODEC_ERROR;
        }
        ci->pcmbuf_insert(samples, NULL, bufcount);
@@ -482,11 +486,5 @@ next_track:
        ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
    }
-done:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return status;
 }
diff --git a/apps/codecs/spc.c b/apps/codecs/spc.c
index 4db2878964..1b49761810 100644
--- a/apps/codecs/spc.c
+++ b/apps/codecs/spc.c
@@ -260,14 +260,6 @@ static inline void samples_release_rdbuf(void)
 static inline int32_t * samples_get_rdbuf(void)
 {
    ci->semaphore_wait(&sample_queue.emu_sem_head, TIMEOUT_BLOCK);
-    if (ci->stop_codec || ci->new_track)
-    {
-        /* Told to stop. Buffer must be released. */
-        samples_release_rdbuf();
-        return NULL;
-    }
    return sample_queue.wav_chunk[sample_queue.head & WAV_CHUNK_MASK].audio;
 }
@@ -390,11 +382,10 @@ static inline void spc_emu_quit(void)
    }
 }
-static inline bool spc_play_get_samples(int32_t **samples)
+static inline int32_t * spc_play_get_samples(void)
 {
    /* obtain filled samples buffer */
-    *samples = samples_get_rdbuf();
+    return samples_get_rdbuf();
-    return *samples != NULL;
 }
 static inline void spc_play_send_samples(int32_t *samples)
@@ -433,15 +424,14 @@ static inline void spc_play_send_samples(int32_t *samples)
 #define spc_emu_quit()
 #define samples_release_rdbuf()
-static inline bool spc_play_get_samples(int32_t **samples)
+static inline int32_t * spc_play_get_samples(void)
 {
    ENTER_TIMER(render);
    /* fill samples buffer */
    if ( SPC_play(&spc_emu,WAV_CHUNK_SIZE*2,wav_chunk) )
        assert( false );
    EXIT_TIMER(render);
-    *samples = wav_chunk;
+    return wav_chunk;
-    return true;
 }
 #endif /* SPC_DUAL_CORE */
@@ -454,6 +444,7 @@ static int play_track( void )
    unsigned long fadeendsample = (ID666.length+ID666.fade)*(long long) SAMPLE_RATE/1000;
    int fadedec = 0;
    int fadevol = 0x7fffffffl;
+    intptr_t param;
    
    if (fadeendsample>fadestartsample)
        fadedec=0x7fffffffl/(fadeendsample-fadestartsample)+1;
@@ -462,25 +453,26 @@ static int play_track( void )
    while ( 1 )
    {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
            int curtime = sampleswritten*1000LL/SAMPLE_RATE;
-            DEBUGF("seek to %ld\ncurrently at %d\n",ci->seek_time,curtime);
+            DEBUGF("seek to %ld\ncurrently at %d\n", (long)param, curtime);
-            if (ci->seek_time < curtime) {
+            if (param < curtime) {
                DEBUGF("seek backwards = reset\n");
+                ci->set_elapsed(0);
                ci->seek_complete();
                return 1;
            }
+            ci->set_elapsed(curtime);
            ci->seek_complete();
        }
-        int32_t *samples;
+        int32_t *samples = spc_play_get_samples();
-        if (!spc_play_get_samples(&samples))
-            break;
        sampleswritten += WAV_CHUNK_SIZE;
@@ -532,67 +524,61 @@ static int play_track( void )
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
 {
-    enum codec_status stat = CODEC_ERROR;
+    if (reason == CODEC_LOAD) {
+        if (!spc_emu_start())
-    if (!spc_emu_start())
+            return CODEC_ERROR;
-        goto codec_quit;
-    do
-    {
-        DEBUGF("SPC: next_track\n");
-        if (codec_init()) {
-            goto codec_quit;
-        }
-        DEBUGF("SPC: after init\n");
        ci->configure(DSP_SET_SAMPLE_DEPTH, 24);
        ci->configure(DSP_SET_FREQUENCY, SAMPLE_RATE);
        ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
+    }
+    else if (reason == CODEC_UNLOAD) {
+        spc_emu_quit();
+    }
-        /* wait for track info to load */
+    return CODEC_OK;
-        if (codec_wait_taginfo() != 0)
+}
-            continue;
-        codec_set_replaygain(ci->id3);
-        /* Read the entire file */
+/* this is called for each file to process */
-        DEBUGF("SPC: request initial buffer\n");
+enum codec_status codec_run(void)
-        ci->seek_buffer(0);
+{
-        size_t buffersize;
+    DEBUGF("SPC: next_track\n");
-        uint8_t* buffer = ci->request_buffer(&buffersize, ci->filesize);
+    if (codec_init())
-        if (!buffer) {
+        return CODEC_ERROR;
-            goto codec_quit;
+    DEBUGF("SPC: after init\n");
+    codec_set_replaygain(ci->id3);
+    /* Read the entire file */
+    DEBUGF("SPC: request initial buffer\n");
+    ci->seek_buffer(0);
+    size_t buffersize;
+    uint8_t* buffer = ci->request_buffer(&buffersize, ci->filesize);
+    if (!buffer)
+        return CODEC_ERROR;
+    DEBUGF("SPC: read size = 0x%lx\n",(unsigned long)buffersize);
+    do
+    {
+        if (load_spc_buffer(buffer, buffersize)) {
+            DEBUGF("SPC load failure\n");
+            return CODEC_ERROR;
        }
-        DEBUGF("SPC: read size = 0x%lx\n",(unsigned long)buffersize);
+        LoadID666(buffer+0x2e);
-        do
-        {
-            if (load_spc_buffer(buffer, buffersize)) {
-                DEBUGF("SPC load failure\n");
-                goto codec_quit;
-            }
-            LoadID666(buffer+0x2e);
-            if (ci->global_settings->repeat_mode!=REPEAT_ONE && ID666.length==0) {
+        if (ci->global_settings->repeat_mode!=REPEAT_ONE && ID666.length==0) {
-                ID666.length=3*60*1000; /* 3 minutes */
+            ID666.length=3*60*1000; /* 3 minutes */
-                ID666.fade=5*1000; /* 5 seconds */
+            ID666.fade=5*1000; /* 5 seconds */
-            }
-            reset_profile_timers();
        }
-        while ( play_track() );
-        print_timers(ci->id3->path);
+        reset_profile_timers();
    }
-    while ( ci->request_next_track() );
+    while ( play_track() );
-    stat = CODEC_OK;
+    print_timers(ci->id3->path);
-codec_quit:
+    return CODEC_OK;
-    spc_emu_quit();
-    
-    return stat;
 }
diff --git a/apps/codecs/speex.c b/apps/codecs/speex.c
index 7a1efa9753..e394efc3d5 100644
--- a/apps/codecs/speex.c
+++ b/apps/codecs/speex.c
@@ -367,11 +367,12 @@ static void *process_header(spx_ogg_packet *op,
    return st;
 }
-/* this is the codec entry point */
+/* this is called for each file to process */
-enum codec_status codec_main(void)
+enum codec_status codec_run(void)
 {
+    int error = CODEC_ERROR;
    SpeexBits bits;
-    int error;
    int eof = 0;
    spx_ogg_sync_state oy;
    spx_ogg_page og;
@@ -383,7 +384,7 @@ enum codec_status codec_main(void)
    int eos = 0;
    SpeexStereoState *stereo;
    int channels = -1;
-    int rate = 0, samplerate = 0;
+    int samplerate = 0;
    int extra_headers = 0;
    int stream_init = 0;
    int page_nb_packets, frame_size, packet_count = 0;
@@ -392,26 +393,22 @@ enum codec_status codec_main(void)
    unsigned long strtoffset = 0;
    void *st = NULL;
    int j = 0;
+    intptr_t param;
    memset(&bits, 0, sizeof(bits));
    memset(&oy, 0, sizeof(oy));
    /* Ogg handling still uses mallocs, so reset the malloc buffer per track */
-next_track:
-    error = CODEC_OK;
    if (codec_init()) {
-        error = CODEC_ERROR;
        goto exit;
    }
+    ci->seek_buffer(0);
    stereo = speex_stereo_state_init();
    spx_ogg_sync_init(&oy);
    spx_ogg_alloc_buffer(&oy,2*CHUNKSIZE);
-    if (codec_wait_taginfo() != 0)
-        goto done;
    strtoffset = ci->id3->offset;
    samplerate = ci->id3->frequency; 
@@ -419,30 +416,32 @@ next_track:
    eof = 0;
    while (!eof) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
        /*seek (seeks to the page before the position) */
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
            if(samplerate!=0&&packet_count>1){
                LOGF("Speex seek page:%lld,%lld,%ld,%lld,%d\n",
-                     ((spx_int64_t)ci->seek_time/1000) *
+                     ((spx_int64_t)param/1000) *
                     (spx_int64_t)samplerate,
-                     page_granule, ci->seek_time,
+                     page_granule, param,
                     (page_granule/samplerate)*1000, samplerate);
-                speex_seek_page_granule(((spx_int64_t)ci->seek_time/1000) *
+                speex_seek_page_granule(((spx_int64_t)param/1000) *
                                        (spx_int64_t)samplerate,
                                        page_granule, &oy, headerssize);
-                ci->seek_complete();
            }
+            ci->set_elapsed(param);
+            ci->seek_complete();
        }
 next_page:
        /*Get the ogg buffer for writing*/
        if(get_more_data(&oy)<1){/*read error*/
-            error=CODEC_ERROR;
            goto done;
        }
@@ -477,8 +476,7 @@ next_page:
                        nframes=1;
                    if (!st){
-                        error=CODEC_ERROR;
+                        goto done;
-                        goto exit;
                    }
                    ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
@@ -557,31 +555,18 @@ next_page:
    }
 done:
-    if (ci->request_next_track()) {
+    /* Clean things up for the next track */
+    speex_bits_destroy(&bits);
-        /* Clean things up for the next track */
+    if (st)
        if (st)
            speex_decoder_destroy(st);
-        if (stream_init == 1)
-            spx_ogg_stream_reset(&os);
-        spx_ogg_sync_reset(&oy);
-        cur_granule = stream_init = rate = samplerate = headerssize 
-            = packet_count = eos = 0;
-        goto next_track;
-    }
-exit:
-    speex_bits_destroy(&bits);
    if (stream_init)
       spx_ogg_stream_destroy(&os);
    spx_ogg_sync_destroy(&oy);
+exit:
    return error;
 }
diff --git a/apps/codecs/tta.c b/apps/codecs/tta.c
index 1d0846ea61..c75f2b0a57 100644
--- a/apps/codecs/tta.c
+++ b/apps/codecs/tta.c
@@ -34,36 +34,36 @@ CODEC_HEADER
 static int32_t samples[PCM_BUFFER_LENGTH * 2] IBSS_ATTR;
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, TTA_OUTPUT_DEPTH - 1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    tta_info info;
-    int status;
    unsigned int decodedsamples;
    int endofstream;
    int new_pos = 0;
    int sample_count;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, TTA_OUTPUT_DEPTH - 1);
  
-next_track:
-    status = CODEC_OK;
    if (codec_init())
    {
        DEBUGF("codec_init() error\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
+    ci->seek_buffer(0);
-        goto done;
    if (set_tta_info(&info) < 0 || player_init(&info) < 0)
-    {
+        return CODEC_ERROR;
-        status = CODEC_ERROR;
-        goto exit;
-    }
    codec_set_replaygain(ci->id3);
@@ -74,8 +74,8 @@ next_track:
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
    } else {
        DEBUGF("CODEC_ERROR: more than 2 channels\n");
-        status = CODEC_ERROR;
+        player_stop();
-        goto done;
+        return CODEC_ERROR;
    }
    /* The main decoder loop */
@@ -88,31 +88,31 @@ next_track:
        new_pos = set_position(ci->id3->offset, TTA_SEEK_POS);
        if (new_pos >= 0)
            decodedsamples = new_pos;
-        ci->seek_complete();
    }
    while (!endofstream)
    {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time)
+        if (action == CODEC_ACTION_SEEK_TIME)
        {
-            new_pos = set_position(ci->seek_time / SEEK_STEP, TTA_SEEK_TIME);
+            new_pos = set_position(param / SEEK_STEP, TTA_SEEK_TIME);
            if (new_pos >= 0)
            {
                decodedsamples = new_pos;
-                ci->seek_complete();
            }
+            ci->set_elapsed((uint64_t)info.LENGTH * 1000 * decodedsamples / info.DATALENGTH);
+            ci->seek_complete();
        }
        sample_count = get_samples(samples);
        if (sample_count < 0)
-        {
-            status = CODEC_ERROR;
            break;
-        }
+ 
        ci->pcmbuf_insert(samples, NULL, sample_count);
        decodedsamples += sample_count;
        if (decodedsamples >= info.DATALENGTH)
@@ -120,11 +120,6 @@ next_track:
        ci->set_elapsed((uint64_t)info.LENGTH * 1000 * decodedsamples / info.DATALENGTH);
    }
-done:
    player_stop();
-    if (ci->request_next_track())
+    return CODEC_OK;
-        goto next_track;
-exit:
-    return status;
 }
diff --git a/apps/codecs/vorbis.c b/apps/codecs/vorbis.c
index 0a36a37c8b..e02d459262 100644
--- a/apps/codecs/vorbis.c
+++ b/apps/codecs/vorbis.c
@@ -104,14 +104,25 @@ static bool vorbis_set_codec_parameters(OggVorbis_File *vf)
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        if (codec_init())
+            return CODEC_ERROR;
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 24);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    ov_callbacks callbacks;
    OggVorbis_File vf;
    ogg_int32_t **pcm;
-    bool initialized = false; /* First init done? */
+    int error = CODEC_ERROR;
-    int error;
    long n;
    int current_section;
    int previous_section;
@@ -120,36 +131,24 @@ enum codec_status codec_main(void)
    ogg_int64_t vf_dataoffsets;
    ogg_uint32_t vf_serialnos;
    ogg_int64_t vf_pcmlengths[2];
+    intptr_t param;
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 24);
-    if (codec_init()) {
-        error = CODEC_ERROR;
-        goto exit;
-    }
 #if defined(CPU_ARM) || defined(CPU_COLDFIRE) || defined(CPU_MIPS)
    if (setjmp(rb_jump_buf) != 0) {
-        /* malloc failed; skip to next track */
+        /* malloc failed; finish with this track */
-        error = CODEC_ERROR;
        goto done;
    }
 #endif
-next_track:
-    error = CODEC_OK;
    ogg_malloc_init();
-    if (codec_wait_taginfo() != 0)
-        goto done;
    /* Create a decoder instance */
    callbacks.read_func = read_handler;
    callbacks.seek_func = initial_seek_handler;
    callbacks.tell_func = tell_handler;
    callbacks.close_func = close_handler;
+    ci->seek_buffer(0);
    /* Open a non-seekable stream */
    error = ov_open_callbacks(ci, &vf, NULL, 0, callbacks);
    
@@ -186,15 +185,13 @@ next_track:
         vf.end = ci->id3->filesize;
         vf.ready_state = OPENED;
         vf.links = 1;
-         initialized = true;
    } else {
         DEBUGF("Vorbis: ov_open failed: %d\n", error);
-         error = CODEC_ERROR;
         goto done;
    }
    if (ci->id3->offset) {
-        ci->advance_buffer(ci->id3->offset);
+        ci->seek_buffer(ci->id3->offset);
        ov_raw_seek(&vf, ci->id3->offset);
        ci->set_elapsed(ov_time_tell(&vf));
        ci->set_offset(ov_raw_tell(&vf));
@@ -203,14 +200,17 @@ next_track:
    previous_section = -1;
    eof = 0;
    while (!eof) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track)
+        if (action == CODEC_ACTION_HALT)
            break;
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            if (ov_time_seek(&vf, ci->seek_time - 1)) {
+            if (ov_time_seek(&vf, param)) {
                //ci->logf("ov_time_seek failed");
            }
+            ci->set_elapsed(ov_time_tell(&vf));
            ci->seek_complete();
        }
@@ -220,7 +220,6 @@ next_track:
        /* Change DSP and buffer settings for this bitstream */
        if (current_section != previous_section) {
            if (!vorbis_set_codec_parameters(&vf)) {
-                error = CODEC_ERROR;
                goto done;
            } else {
                previous_section = current_section;
@@ -238,6 +237,7 @@ next_track:
        }
    }
+    error = CODEC_OK;
 done:
 #if 0 /* defined(SIMULATOR) */
    {
@@ -249,18 +249,12 @@ done:
 #endif
    ogg_malloc_destroy();
-    if (ci->request_next_track()) {
+    /* Clean things up for the next track */
-        if (!initialized)
+    vf.dataoffsets = NULL;
-            goto next_track;
+    vf.offsets = NULL;
-        /* Clean things up for the next track */
+    vf.serialnos = NULL;
-        vf.dataoffsets = NULL;
+    vf.pcmlengths = NULL;
-        vf.offsets = NULL;
+    ov_clear(&vf);
-        vf.serialnos = NULL;
-        vf.pcmlengths = NULL;
-        ov_clear(&vf);
-        goto next_track;
-    }
-        
-exit:
    return error;
 }
diff --git a/apps/codecs/vox.c b/apps/codecs/vox.c
index c7f39342c3..bf274c6917 100644
--- a/apps/codecs/vox.c
+++ b/apps/codecs/vox.c
@@ -44,9 +44,19 @@ static uint8_t *read_buffer(size_t *realsize)
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
-    int status;
    uint32_t decodedsamples;
    size_t n;
    int bufcount;
@@ -54,26 +64,18 @@ enum codec_status codec_main(void)
    uint8_t *voxbuf;
    off_t firstblockposn = 0;     /* position of the first block in file */
    const struct pcm_codec *codec;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
-  
-next_track:
-    status = CODEC_OK;
    if (codec_init()) {
        DEBUGF("codec_init() error\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    codec_set_replaygain(ci->id3);
    /* Need to save offset for later use (cleared indirectly by advance_buffer) */
    bytesdone = ci->id3->offset;
+    ci->seek_buffer(0);
    ci->memset(&format, 0, sizeof(struct pcm_format));
@@ -96,20 +98,16 @@ next_track:
    if (!codec)
    {
        DEBUGF("CODEC_ERROR: dialogic oki adpcm codec does not load.\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
-    if (!codec->set_format(&format))
+    if (!codec->set_format(&format)) {
-    {
+        return CODEC_ERROR;
-        status = CODEC_ERROR;
-        goto done;
    }
    if (format.numbytes == 0) {
        DEBUGF("CODEC_ERROR: data size is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* check chunksize */
@@ -118,8 +116,7 @@ next_track:
    if (format.chunksize == 0)
    {
        DEBUGF("CODEC_ERROR: chunksize is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -131,14 +128,14 @@ next_track:
        struct pcm_pos *newpos = codec->get_seek_pos(bytesdone - firstblockposn,
                                                     PCM_SEEK_POS, &read_buffer);
-        if (newpos->pos > format.numbytes)
+        if (newpos->pos > format.numbytes) {
-            goto done;
+            return CODEC_OK;
+        }
        if (ci->seek_buffer(firstblockposn + newpos->pos))
        {
            bytesdone      = newpos->pos;
            decodedsamples = newpos->samples;
        }
-        ci->seek_complete();
    } else {
        /* already where we need to be */
        bytesdone = 0;
@@ -148,22 +145,29 @@ next_track:
    endofstream = 0;
    while (!endofstream) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, PCM_SEEK_TIME,
+            struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME,
                                                         &read_buffer);
            if (newpos->pos > format.numbytes)
+            {
+                ci->set_elapsed(ci->id3->length);
+                ci->seek_complete();
                break;
+            }
            if (ci->seek_buffer(firstblockposn + newpos->pos))
            {
                bytesdone      = newpos->pos;
                decodedsamples = newpos->samples;
            }
+            ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
            ci->seek_complete();
        }
@@ -175,11 +179,10 @@ next_track:
            endofstream = 1;
        }
-        status = codec->decode(voxbuf, n, samples, &bufcount);
+        if (codec->decode(voxbuf, n, samples, &bufcount) == CODEC_ERROR)
-        if (status == CODEC_ERROR)
        {
            DEBUGF("codec error\n");
-            goto done;
+            return CODEC_ERROR;
        }
        ci->pcmbuf_insert(samples, NULL, bufcount);
@@ -192,10 +195,5 @@ next_track:
        ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
    }
-done:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return status;
 }
diff --git a/apps/codecs/wav.c b/apps/codecs/wav.c
index e179470f27..42bcc7081f 100644
--- a/apps/codecs/wav.c
+++ b/apps/codecs/wav.c
@@ -151,9 +151,19 @@ static uint8_t *read_buffer(size_t *realsize)
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
-    int status;
    uint32_t decodedsamples;
    size_t n;
    int bufcount;
@@ -163,38 +173,28 @@ enum codec_status codec_main(void)
    off_t firstblockposn;     /* position of the first block in file */
    const struct pcm_codec *codec;
    uint32_t size;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
-  
-next_track:
-    status = CODEC_OK;
    if (codec_init()) {
        DEBUGF("codec_init() error\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    codec_set_replaygain(ci->id3);
    
    /* Need to save offset for later use (cleared indirectly by advance_buffer) */
    bytesdone = ci->id3->offset;
    /* get RIFF chunk header */
+    ci->seek_buffer(0);
    buf = ci->request_buffer(&n, 12);
    if (n < 12) {
        DEBUGF("request_buffer error\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if ((memcmp(buf, "RIFF", 4) != 0) || (memcmp(&buf[8], "WAVE", 4) != 0)) {
        DEBUGF("CODEC_ERROR: missing riff header\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* advance to first WAVE chunk */
@@ -215,8 +215,7 @@ next_track:
        if (n < 8) {
            DEBUGF("data chunk request_buffer error\n");
            /* no more chunks, 'data' chunk must not have been found */
-            status = CODEC_ERROR;
+            return CODEC_ERROR;
-            goto done;
        }
        /* chunkSize */
@@ -225,8 +224,7 @@ next_track:
            if (size < 16) {
                DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%lu < 16\n",
                       (unsigned long)size);
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            /* wFormatTag */
            format.formattag=buf[8]|(buf[9]<<8);
@@ -256,8 +254,7 @@ next_track:
                        if (format.size < 22) {
                            DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
                                   "missing extension\n");
-                            status = CODEC_ERROR;
+                            return CODEC_ERROR;
-                            goto done;
                        }
                        /* wValidBitsPerSample */
                        format.bitspersample = buf[26]|(buf[27]<<8);
@@ -273,8 +270,7 @@ next_track:
            {
                if (!set_msadpcm_coeffs(buf))
                {
-                    status = CODEC_ERROR;
+                    return CODEC_ERROR;
-                    goto done;
                }
            }
@@ -284,8 +280,7 @@ next_track:
            {
                DEBUGF("CODEC_ERROR: unsupported wave format 0x%x\n", 
                    (unsigned int) format.formattag);
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            /* riff 8bit linear pcm is unsigned */
@@ -295,8 +290,7 @@ next_track:
            /* set format, parse codec specific tag, check format, and calculate chunk size */
            if (!codec->set_format(&format))
            {
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
        } else if (memcmp(buf, "data", 4) == 0) {
            format.numbytes = size;
@@ -324,31 +318,26 @@ next_track:
    if (!codec)
    {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* common format check */
    if (format.channels == 0) {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.samplesperblock == 0) {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-wSamplesPerBlock file\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.blockalign == 0)
    {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-blockalign file\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.numbytes == 0) {
        DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* check chunksize */
@@ -358,8 +347,7 @@ next_track:
    if (format.chunksize == 0)
    {
        DEBUGF("CODEC_ERROR: chunksize is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -369,8 +357,7 @@ next_track:
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
    } else {
        DEBUGF("CODEC_ERROR: more than 2 channels\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* make sure we're at the correct offset */
@@ -380,13 +367,12 @@ next_track:
                                                     PCM_SEEK_POS, &read_buffer);
        if (newpos->pos > format.numbytes)
-            goto done;
+            return CODEC_OK;
        if (ci->seek_buffer(firstblockposn + newpos->pos))
        {
            bytesdone      = newpos->pos;
            decodedsamples = newpos->samples;
        }
-        ci->seek_complete();
    } else {
        /* already where we need to be */
        bytesdone = 0;
@@ -396,22 +382,28 @@ next_track:
    endofstream = 0;
    while (!endofstream) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, PCM_SEEK_TIME,
+            struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME,
                                                         &read_buffer);
            if (newpos->pos > format.numbytes)
+            {
+                ci->set_elapsed(ci->id3->length);
+                ci->seek_complete();
                break;
+            }
            if (ci->seek_buffer(firstblockposn + newpos->pos))
            {
                bytesdone      = newpos->pos;
                decodedsamples = newpos->samples;
            }
+            ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
            ci->seek_complete();
        }
@@ -423,11 +415,10 @@ next_track:
            endofstream = 1;
        }
-        status = codec->decode(wavbuf, n, samples, &bufcount);
+        if (codec->decode(wavbuf, n, samples, &bufcount) == CODEC_ERROR)
-        if (status == CODEC_ERROR)
        {
            DEBUGF("codec error\n");
-            goto done;
+            return CODEC_ERROR;
        }
        ci->pcmbuf_insert(samples, NULL, bufcount);
@@ -440,10 +431,5 @@ next_track:
        ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
    }
-done:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return status;
 }
diff --git a/apps/codecs/wav64.c b/apps/codecs/wav64.c
index 9dbdab8368..c763e6f7f0 100644
--- a/apps/codecs/wav64.c
+++ b/apps/codecs/wav64.c
@@ -159,9 +159,19 @@ static uint8_t *read_buffer(size_t *realsize)
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
-    int status;
    uint32_t decodedsamples;
    size_t n;
    int bufcount;
@@ -171,39 +181,29 @@ enum codec_status codec_main(void)
    off_t firstblockposn;     /* position of the first block in file */
    const struct pcm_codec *codec;
    uint64_t size;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
-  
-next_track:
-    status = CODEC_OK;
    if (codec_init()) {
        DEBUGF("codec_init() error\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
    codec_set_replaygain(ci->id3);
    
    /* Need to save offset for later use (cleared indirectly by advance_buffer) */
    bytesdone = ci->id3->offset;
    /* get RIFF chunk header */
+    ci->seek_buffer(0);
    buf = ci->request_buffer(&n, 40);
    if (n < 40) {
        DEBUGF("request_buffer error\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if ((memcmp(buf   , WAVE64_GUID_RIFF, 16) != 0) ||
        (memcmp(buf+24, WAVE64_GUID_WAVE, 16) != 0))
    {
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* advance to first WAVE chunk */
@@ -224,8 +224,7 @@ next_track:
        if (n < 8) {
            DEBUGF("data chunk request_buffer error\n");
            /* no more chunks, 'data' chunk must not have been found */
-            status = CODEC_ERROR;
+            return CODEC_ERROR;
-            goto done;
        }
        /* chunkSize */
@@ -233,8 +232,7 @@ next_track:
        if (memcmp(buf, WAVE64_GUID_FMT, 16) == 0) {
            if (size < 16) {
                DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%d < 16\n", (int)size);
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            /* wFormatTag */
            format.formattag=buf[24]|(buf[25]<<8);
@@ -263,8 +261,7 @@ next_track:
                        if (format.size < 22) {
                            DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
                                   "missing extension\n");
-                            status = CODEC_ERROR;
+                            return CODEC_ERROR;
-                            goto done;
                        }
                        /* wValidBitsPerSample */
                        format.bitspersample = buf[42]|(buf[43]<<8);
@@ -279,10 +276,7 @@ next_track:
            if (format.formattag == WAVE_FORMAT_ADPCM)
            {
                if (!set_msadpcm_coeffs(buf))
-                {
+                    return CODEC_ERROR;
-                    status = CODEC_ERROR;
-                    goto done;
-                }
            }
            /* get codec */
@@ -291,8 +285,7 @@ next_track:
            {
                DEBUGF("CODEC_ERROR: unsupported wave format 0x%x\n", 
                    (unsigned int) format.formattag);
-                status = CODEC_ERROR;
+                return CODEC_ERROR;
-                goto done;
            }
            /* riff 8bit linear pcm is unsigned */
@@ -301,10 +294,7 @@ next_track:
            /* check format, and calculate chunk size */
            if (!codec->set_format(&format))
-            {
+                return CODEC_ERROR;
-                status = CODEC_ERROR;
-                goto done;
-            }
        } else if (memcmp(buf, WAVE64_GUID_DATA, 16) == 0) {
            format.numbytes = size;
            /* advance to start of data */
@@ -330,31 +320,26 @@ next_track:
    if (!codec)
    {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* common format check */
    if (format.channels == 0) {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.samplesperblock == 0) {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-wSamplesPerBlock file\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.blockalign == 0)
    {
        DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-blockalign file\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    if (format.numbytes == 0) {
        DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* check chunksize */
@@ -364,8 +349,7 @@ next_track:
    if (format.chunksize == 0)
    {
        DEBUGF("CODEC_ERROR: chunksize is 0\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
@@ -375,8 +359,7 @@ next_track:
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
    } else {
        DEBUGF("CODEC_ERROR: more than 2 channels\n");
-        status = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* make sure we're at the correct offset */
@@ -385,14 +368,14 @@ next_track:
        struct pcm_pos *newpos = codec->get_seek_pos(bytesdone - firstblockposn,
                                                     PCM_SEEK_POS, &read_buffer);
-        if (newpos->pos > format.numbytes)
+        if (newpos->pos > format.numbytes) {
-            goto done;
+            return CODEC_OK;
+        }
        if (ci->seek_buffer(firstblockposn + newpos->pos))
        {
            bytesdone      = newpos->pos;
            decodedsamples = newpos->samples;
        }
-        ci->seek_complete();
    } else {
        /* already where we need to be */
        bytesdone = 0;
@@ -402,22 +385,29 @@ next_track:
    endofstream = 0;
    while (!endofstream) {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
+        if (action == CODEC_ACTION_HALT)
            break;
-        }
-        if (ci->seek_time) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, PCM_SEEK_TIME,
+            struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME,
                                                         &read_buffer);
            if (newpos->pos > format.numbytes)
+            {
+                ci->set_elapsed(ci->id3->length);
+                ci->seek_complete();
                break;
+            }
            if (ci->seek_buffer(firstblockposn + newpos->pos))
            {
                bytesdone      = newpos->pos;
                decodedsamples = newpos->samples;
            }
+            ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
            ci->seek_complete();
        }
@@ -429,11 +419,10 @@ next_track:
            endofstream = 1;
        }
-        status = codec->decode(wavbuf, n, samples, &bufcount);
+        if (codec->decode(wavbuf, n, samples, &bufcount) == CODEC_ERROR)
-        if (status == CODEC_ERROR)
        {
            DEBUGF("codec error\n");
-            goto done;
+            return CODEC_ERROR;
        }
        ci->pcmbuf_insert(samples, NULL, bufcount);
@@ -445,12 +434,6 @@ next_track:
            endofstream = 1;
        ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
    }
-    status = CODEC_OK;
-done:
-    if (ci->request_next_track())
-        goto next_track;
-exit:
+    return CODEC_OK;
-    return status;
 }
diff --git a/apps/codecs/wav_enc.c b/apps/codecs/wav_enc.c
index ef1a88ec23..e4afeaf93c 100644
--- a/apps/codecs/wav_enc.c
+++ b/apps/codecs/wav_enc.c
@@ -345,40 +345,42 @@ static bool init_encoder(void)
    return true;
 } /* init_encoder */
-/* main codec entry point */
+/* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
 {
-    if (!init_encoder())
+    if (reason == CODEC_LOAD) {
-        return CODEC_ERROR;
+        if (!init_encoder())
+            return CODEC_ERROR;
+    }
+    else if (reason == CODEC_UNLOAD) {
+        /* reset parameters to initial state */
+        ci->enc_set_parameters(NULL);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
    /* main encoding loop */
-    while(!ci->stop_codec)
+    while(ci->get_command(NULL) != CODEC_ACTION_HALT)
    {
-        uint32_t *src;
+        uint32_t *src = (uint32_t *)ci->enc_get_pcm_data(PCM_CHUNK_SIZE);
+        struct enc_chunk_hdr *chunk;
-        while ((src = (uint32_t *)ci->enc_get_pcm_data(PCM_CHUNK_SIZE)) != NULL)
+        if(src == NULL)
-        {
+            continue;
-            struct enc_chunk_hdr *chunk;
-            if (ci->stop_codec)
-                break;
-            chunk           = ci->enc_get_chunk();
+        chunk           = ci->enc_get_chunk();
-            chunk->enc_size = enc_size;
+        chunk->enc_size = enc_size;
-            chunk->num_pcm  = PCM_SAMP_PER_CHUNK;
+        chunk->num_pcm  = PCM_SAMP_PER_CHUNK;
-            chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
+        chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
-            chunk_to_wav_format(src, (uint32_t *)chunk->enc_data);
+        chunk_to_wav_format(src, (uint32_t *)chunk->enc_data);
-            ci->enc_finish_chunk();
+        ci->enc_finish_chunk();
-            ci->yield();
-        }
-        ci->yield();
    }
-    /* reset parameters to initial state */
-    ci->enc_set_parameters(NULL);
    return CODEC_OK;
-} /* codec_start */
+}
diff --git a/apps/codecs/wavpack.c b/apps/codecs/wavpack.c
index d27a9fb621..ccb9f41190 100644
--- a/apps/codecs/wavpack.c
+++ b/apps/codecs/wavpack.c
@@ -31,39 +31,39 @@ static int32_t temp_buffer [BUFFER_SIZE] IBSS_ATTR;
 static int32_t read_callback (void *buffer, int32_t bytes)
 {
    int32_t retval = ci->read_filebuf (buffer, bytes);
-    ci->id3->offset = ci->curpos;
+    ci->set_offset(ci->curpos);
    return retval;
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    WavpackContext *wpc;
    char error [80];
    int bps, nchans, sr_100;
-    int retval;
+    intptr_t param;
-    /* Generic codec initialisation */
+    if (codec_init())
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+        return CODEC_ERROR;
-next_track:
+    ci->seek_buffer (ci->id3->offset);
-    retval = CODEC_OK;
-    if (codec_init()) {
-        retval = CODEC_ERROR;
-        goto exit;
-    }
-    if (codec_wait_taginfo() != 0)
-        goto done;
-        
    /* Create a decoder instance */
    wpc = WavpackOpenFileInput (read_callback, error);
-    if (!wpc) {
+    if (!wpc)
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
-    }
    ci->configure(DSP_SWITCH_FREQUENCY, WavpackGetSampleRate (wpc));
    codec_set_replaygain(ci->id3);
@@ -77,56 +77,48 @@ next_track:
    /* The main decoder loop */
    while (1) {
-        int32_t nsamples;  
+        int32_t nsamples;
+        enum codec_command_action action = ci->get_command(&param);
+        if (action == CODEC_ACTION_HALT)
+            break;
-        if (ci->seek_time && ci->taginfo_ready && ci->id3->length) {
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            ci->seek_time--;
            int curpos_ms = WavpackGetSampleIndex (wpc) / sr_100 * 10;
            int n, d, skip;
-            if (ci->seek_time > curpos_ms) {
+            if (param > curpos_ms) {
-                n = ci->seek_time - curpos_ms;
+                n = param - curpos_ms;
                d = ci->id3->length - curpos_ms;
                skip = (int)((int64_t)(ci->filesize - ci->curpos) * n / d);
                ci->seek_buffer (ci->curpos + skip);
            }
-            else {
+            else if (curpos_ms != 0) {
-                n = curpos_ms - ci->seek_time;
+                n = curpos_ms - param;
                d = curpos_ms;
                skip = (int)((int64_t) ci->curpos * n / d);
                ci->seek_buffer (ci->curpos - skip);
            }
            wpc = WavpackOpenFileInput (read_callback, error);
-            ci->seek_complete();
            if (!wpc)
+            {
+                ci->seek_complete();
                break;
+            }
            ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
-            ci->yield ();
+            ci->seek_complete();
        }
        nsamples = WavpackUnpackSamples (wpc, temp_buffer, BUFFER_SIZE / nchans);  
-        if (!nsamples || ci->stop_codec || ci->new_track)
+        if (!nsamples)
-            break;
-        ci->yield ();
-        if (ci->stop_codec || ci->new_track)
            break;
        ci->pcmbuf_insert (temp_buffer, NULL, nsamples);
        ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
-        ci->yield ();
    }
-done:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return retval;
 }
diff --git a/apps/codecs/wavpack_enc.c b/apps/codecs/wavpack_enc.c
index d908e284be..730cf0734b 100644
--- a/apps/codecs/wavpack_enc.c
+++ b/apps/codecs/wavpack_enc.c
@@ -389,77 +389,79 @@ static bool init_encoder(void)
    return true;
 } /* init_encoder */
-enum codec_status codec_main(void)
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
 {
-    /* initialize params and config */
+    if (reason == CODEC_LOAD) {
-    if (!init_encoder())
+        /* initialize params and config */
-        return CODEC_ERROR;
+        if (!init_encoder())
+            return CODEC_ERROR;
+    }
+    else if (reason == CODEC_UNLOAD) {
+        /* reset parameters to initial state */
+        ci->enc_set_parameters(NULL);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
    /* main encoding loop */
-    while(!ci->stop_codec)
+    while(ci->get_command(NULL) != CODEC_ACTION_HALT)
    {
-        uint8_t *src;
+        uint8_t *src = (uint8_t *)ci->enc_get_pcm_data(PCM_CHUNK_SIZE);
+        struct enc_chunk_hdr *chunk;
+        bool abort_chunk;
+        uint8_t *dst;
+        uint8_t *src_end;
-        while ((src = ci->enc_get_pcm_data(PCM_CHUNK_SIZE)) != NULL)
+        if(src == NULL)
-        {
+            continue;
-            struct enc_chunk_hdr *chunk;
-            bool     abort_chunk;
-            uint8_t *dst;
-            uint8_t *src_end;
-            if(ci->stop_codec)
-                break;
-            abort_chunk = true;
-            chunk = ci->enc_get_chunk();
+        chunk = ci->enc_get_chunk();
-            /* reset counts and pointer */
+        /* reset counts and pointer */
-            chunk->enc_size = 0;
+        chunk->enc_size = 0;
-            chunk->num_pcm  = 0;
+        chunk->num_pcm  = 0;
-            chunk->enc_data = NULL;
+        chunk->enc_data = NULL;
-            dst = ENC_CHUNK_SKIP_HDR(dst, chunk);
+        dst = ENC_CHUNK_SKIP_HDR(dst, chunk);
-            WavpackStartBlock(wpc, dst, dst + data_size);
+        WavpackStartBlock(wpc, dst, dst + data_size);
-            chunk_to_int32((uint32_t*)src);
+        chunk_to_int32((uint32_t*)src);
-            src      = input_buffer;
+        src      = input_buffer;
-            src_end  = src + input_size;
+        src_end  = src + input_size;
-            /* encode chunk in four steps yielding between each */
+        /* encode chunk in four steps yielding between each */
-            do
+        do
+        {
+            abort_chunk = true;
+            if (WavpackPackSamples(wpc, (int32_t *)src,
+                                   PCM_SAMP_PER_CHUNK/4))
            {
-                if (WavpackPackSamples(wpc, (int32_t *)src,
+                chunk->num_pcm += PCM_SAMP_PER_CHUNK/4;
-                                       PCM_SAMP_PER_CHUNK/4))
+                ci->yield();
-                {
+                /* could've been stopped in some way */
-                    chunk->num_pcm += PCM_SAMP_PER_CHUNK/4;
+                abort_chunk = chunk->flags & CHUNKF_ABORT;
-                    ci->yield();
-                    /* could've been stopped in some way */
-                    abort_chunk = ci->stop_codec ||
-                                  (chunk->flags & CHUNKF_ABORT);
-                }
-                src += input_step;
            }
-            while (!abort_chunk && src < src_end);
-            if (!abort_chunk)
+            src += input_step;
-            {
-                chunk->enc_data = dst;
-                if (chunk->num_pcm < PCM_SAMP_PER_CHUNK)
-                    ci->enc_unget_pcm_data(PCM_CHUNK_SIZE - chunk->num_pcm*4);
-            /* finish the chunk and store chunk size info */
-                chunk->enc_size = WavpackFinishBlock(wpc);
-                ci->enc_finish_chunk();
-            }
        }
+        while (!abort_chunk && src < src_end);
-        ci->yield();
+        if (!abort_chunk)
+        {
+            chunk->enc_data = dst;
+            if (chunk->num_pcm < PCM_SAMP_PER_CHUNK)
+                ci->enc_unget_pcm_data(PCM_CHUNK_SIZE - chunk->num_pcm*4);
+            /* finish the chunk and store chunk size info */
+            chunk->enc_size = WavpackFinishBlock(wpc);
+            ci->enc_finish_chunk();
+        }
    }
-    /* reset parameters to initial state */
-    ci->enc_set_parameters(NULL);
- 
    return CODEC_OK;
-} /* codec_start */
+}
diff --git a/apps/codecs/wma.c b/apps/codecs/wma.c
index 1b46813444..c327fafb5a 100644
--- a/apps/codecs/wma.c
+++ b/apps/codecs/wma.c
@@ -29,53 +29,52 @@ CODEC_HEADER
 static WMADecodeContext wmadec;
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    uint32_t elapsedtime;
-    int retval;
    asf_waveformatex_t wfx;
    size_t resume_offset;
    int i;
-    int wmares, res;
+    int wmares;
+    int res = 0;
    uint8_t* audiobuf;
    int audiobufsize;
    int packetlength = 0;
    int errcount = 0;
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
-next_track:
-    retval = CODEC_OK;
    /* Proper reset of the decoder context. */
    memset(&wmadec, 0, sizeof(wmadec));
-    /* Wait for the metadata to be read */
-    if (codec_wait_taginfo() != 0)
-        goto done;
    /* Remember the resume position - when the codec is opened, the
       playback engine will reset it. */
    resume_offset = ci->id3->offset;
 restart_track:
-    retval = CODEC_OK;
    if (codec_init()) {
        LOGF("WMA: Error initialising codec\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
    /* Copy the format metadata we've stored in the id3 TOC field.  This
       saves us from parsing it again here. */
    memcpy(&wfx, ci->id3->toc, sizeof(wfx));
+    ci->seek_buffer(ci->id3->first_frame_offset);
    if (wma_decode_init(&wmadec,&wfx) < 0) {
        LOGF("WMA: Unsupported or corrupt file\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto exit;
    }
    if (resume_offset > ci->id3->first_frame_offset)
@@ -101,34 +100,35 @@ restart_track:
    codec_set_replaygain(ci->id3);
    /* The main decoding loop */
-    res = 1;
    while (res >= 0)
    {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
-            goto done;
+        if (action == CODEC_ACTION_HALT)
-        }
+            break;
        /* Deal with any pending seek requests */
-        if (ci->seek_time){
+        if (action == CODEC_ACTION_SEEK_TIME) {
-            if (ci->seek_time == 1) {
+            if (param == 0) {
+                ci->set_elapsed(0);
                ci->seek_complete();
                goto restart_track; /* Pretend you never saw this... */
            }
-            elapsedtime = asf_seek(ci->seek_time, &wfx);
+            elapsedtime = asf_seek(param, &wfx);
            if (elapsedtime < 1){
+                ci->set_elapsed(0);
                ci->seek_complete();
-                goto next_track;
+                break;
            }
            /*DEBUGF("Seek returned %d\n", (int)elapsedtime);*/
-            ci->set_elapsed(elapsedtime);
            /*flush the wma decoder state*/
            wmadec.last_superframe_len = 0;
            wmadec.last_bitoffset = 0;
+            ci->set_elapsed(elapsedtime);
            ci->seek_complete();
        }
        errcount = 0;
@@ -140,10 +140,15 @@ new_packet:
             * times. If we succeed, the error counter will be reset.
             */
+            if (res == ASF_ERROR_EOF) {
+                /* File ended - not an error */
+                break;
+            }
            errcount++;
            DEBUGF("read_packet error %d, errcount %d\n",wmares, errcount);
            if (errcount > 5) {
-                goto done;
+                return CODEC_ERROR;
            } else {
                ci->advance_buffer(packetlength);
                goto new_packet;
@@ -163,7 +168,7 @@ new_packet:
                    errcount++;
                    DEBUGF("WMA decode error %d, errcount %d\n",wmares, errcount);
                    if (errcount > 5) {
-                        goto done;
+                        return CODEC_ERROR;
                    } else {
                        ci->advance_buffer(packetlength);
                        goto new_packet;
@@ -173,18 +178,12 @@ new_packet:
                    elapsedtime += (wmares*10)/(wfx.rate/100);
                    ci->set_elapsed(elapsedtime);
                }
-                ci->yield();
            }
        }
        ci->advance_buffer(packetlength);
    }
-done:
    /*LOGF("WMA: Decoded %ld samples\n",elapsedtime*wfx.rate/1000);*/
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-exit:
-    return retval;
 }
diff --git a/apps/codecs/wmapro.c b/apps/codecs/wmapro.c
index c02dddeeb3..b6a8e47f25 100644
--- a/apps/codecs/wmapro.c
+++ b/apps/codecs/wmapro.c
@@ -27,11 +27,22 @@ CODEC_HEADER
 int32_t *dec[2]; /* pointers to the output buffers in WMAProDecodeCtx in wmaprodec.c */
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, WMAPRO_DSP_SAMPLE_DEPTH);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    uint32_t elapsedtime;
-    int retval;
    asf_waveformatex_t wfx;     /* Holds the stream properties */
    size_t resume_offset;
    int res;                    /* Return values from asf_read_packet() and decode_packet() */
@@ -42,28 +53,15 @@ enum codec_status codec_main(void)
    int pktcnt = 0;             /* Count of the packets played */
    uint8_t *data;              /* Pointer to decoder input buffer */
    int size;                   /* Size of the input frame to the decoder */
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, WMAPRO_DSP_SAMPLE_DEPTH);
-    
-next_track:
-    retval = CODEC_OK;
-    /* Wait for the metadata to be read */
-    if (codec_wait_taginfo() != 0)
-        goto done;
    /* Remember the resume position */
    resume_offset = ci->id3->offset;
 restart_track:
-    retval = CODEC_OK;
    if (codec_init()) {
        LOGF("(WMA PRO) Error: Error initialising codec\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* Copy the format metadata we've stored in the id3 TOC field.  This
@@ -77,8 +75,7 @@ restart_track:
    
    if (decode_init(&wfx) < 0) {
        LOGF("(WMA PRO) Error: Unsupported or corrupt file\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* Now advance the file position to the first frame */
@@ -91,23 +88,24 @@ restart_track:
    while (pktcnt < wfx.numpackets)
    {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
-            goto done;
+        if (action == CODEC_ACTION_HALT)
-        }
+            break;
-        
-        /* Deal with any pending seek requests */
-        if (ci->seek_time){
-            if (ci->seek_time == 1) {
+        /* Deal with any pending seek requests */
+        if (action == CODEC_ACTION_SEEK_TIME) {
+            if (param == 0) {
+                ci->set_elapsed(0);
                ci->seek_complete();
                goto restart_track; /* Pretend you never saw this... */
            }
-            elapsedtime = asf_seek(ci->seek_time, &wfx);
+            elapsedtime = asf_seek(param, &wfx);
            if (elapsedtime < 1){
+                ci->set_elapsed(0);
                ci->seek_complete();
-                goto next_track;
+                break;
            }
            ci->set_elapsed(elapsedtime);
@@ -117,8 +115,8 @@ restart_track:
        res = asf_read_packet(&audiobuf, &audiobufsize, &packetlength, &wfx);
        if (res < 0) {
-           LOGF("(WMA PRO) Warning: asf_read_packet returned %d", res);
+            LOGF("(WMA PRO) Warning: asf_read_packet returned %d", res);
-           goto done;
+            return CODEC_ERROR;
        } else {
            data = audiobuf;
            size = audiobufsize;
@@ -132,7 +130,7 @@ restart_track:
                res = decode_packet(&wfx, dec, &outlen, data, size);
                if(res < 0) {
                    LOGF("(WMA PRO) Error: decode_packet returned %d", res);
-                    goto done;
+                    return CODEC_ERROR;
                }
                data += res;
                size -= res;
@@ -152,10 +150,6 @@ restart_track:
        ci->advance_buffer(packetlength);
    }
-done:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-    
-    return retval;
 }
diff --git a/apps/codecs/wmavoice.c b/apps/codecs/wmavoice.c
index ddf66828f1..64c8cd1692 100644
--- a/apps/codecs/wmavoice.c
+++ b/apps/codecs/wmavoice.c
@@ -52,10 +52,20 @@ static void init_codec_ctx(AVCodecContext *avctx, asf_waveformatex_t *wfx)
 }
 /* this is the codec entry point */
-enum codec_status codec_main(void)
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, 31);
+    }
+    return CODEC_OK;
+}
+/* this is called for each file to process */
+enum codec_status codec_run(void)
 {
    uint32_t elapsedtime;
-    int retval;
    asf_waveformatex_t wfx;     /* Holds the stream properties */
    size_t resume_offset;
    int res;                    /* Return values from asf_read_packet() and decode_packet() */
@@ -64,27 +74,14 @@ enum codec_status codec_main(void)
    int packetlength = 0;       /* Logical packet size (minus the header size) */          
    int outlen = 0;             /* Number of bytes written to the output buffer */
    int pktcnt = 0;             /* Count of the packets played */
+    intptr_t param;
-    /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, 31);
-    
-next_track:
-    retval = CODEC_OK;
-    /* Wait for the metadata to be read */
-    if (codec_wait_taginfo() != 0)
-        goto done;
    /* Remember the resume position */
    resume_offset = ci->id3->offset;
 restart_track:
-    retval = CODEC_OK;
    if (codec_init()) {
        LOGF("(WMA Voice) Error: Error initialising codec\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* Copy the format metadata we've stored in the id3 TOC field.  This
@@ -97,14 +94,15 @@ restart_track:
    ci->configure(DSP_SET_STEREO_MODE, wfx.channels == 1 ?
                  STEREO_MONO : STEREO_INTERLEAVED);
    codec_set_replaygain(ci->id3);
+    ci->seek_buffer(0);
    
    /* Initialise the AVCodecContext */
    init_codec_ctx(&avctx, &wfx);
    if (wmavoice_decode_init(&avctx) < 0) {
        LOGF("(WMA Voice) Error: Unsupported or corrupt file\n");
-        retval = CODEC_ERROR;
+        return CODEC_ERROR;
-        goto done;
    }
    /* Now advance the file position to the first frame */
@@ -117,21 +115,24 @@ restart_track:
    while (pktcnt < wfx.numpackets)
    {
-        ci->yield();
+        enum codec_command_action action = ci->get_command(&param);
-        if (ci->stop_codec || ci->new_track) {
-            goto done;
+        if (action == CODEC_ACTION_HALT)
-        }
+            break;
-        
        /* Deal with any pending seek requests */
-        if (ci->seek_time){
+        if (action == CODEC_ACTION_SEEK_TIME) {
+            ci->set_elapsed(param);
-            if (ci->seek_time == 1) {
+            if (param == 0) {
+                ci->set_elapsed(0);
                ci->seek_complete();
                goto restart_track; /* Pretend you never saw this... */
            }
-            elapsedtime = asf_seek(ci->seek_time, &wfx);
+            elapsedtime = asf_seek(param, &wfx);
            if (elapsedtime < 1){
+                ci->set_elapsed(0);
                ci->seek_complete();
                goto next_track;
            }
@@ -145,7 +146,7 @@ new_packet:
        if (res < 0) {
            LOGF("(WMA Voice) read_packet error %d\n",res);
-            goto done;
+            return CODEC_ERROR;
        } else {
            avpkt.data = audiobuf;
            avpkt.size = audiobufsize;
@@ -165,8 +166,9 @@ new_packet:
                        ci->advance_buffer(packetlength);
                        goto new_packet;    
                    }
-                    else
+                    else {
-                        goto done;
+                        return CODEC_ERROR;
+                    }
                }
                avpkt.data += res;
                avpkt.size -= res;
@@ -186,10 +188,6 @@ new_packet:
        ci->advance_buffer(packetlength);
    }
-done:
+    return CODEC_OK;
-    if (ci->request_next_track())
-        goto next_track;
-    
-    return retval;
 }
diff --git a/apps/gui/wps.c b/apps/gui/wps.c
index c33268e6bd..069df09c36 100644
--- a/apps/gui/wps.c
+++ b/apps/gui/wps.c
@@ -219,10 +219,10 @@ static int skintouch_to_wps(struct wps_data *data)
 #endif
        case ACTION_TOUCH_SCROLLBAR:
            skin_get_global_state()->id3->elapsed = skin_get_global_state()->id3->length*offset/100;
-            if (!skin_get_global_state()->paused)
 #if (CONFIG_CODEC == SWCODEC)
-                audio_pre_ff_rewind();
+            audio_pre_ff_rewind();
 #else
+            if (!skin_get_global_state()->paused)
                audio_pause();
 #endif
            audio_ff_rewind(skin_get_global_state()->id3->elapsed);
@@ -300,10 +300,10 @@ bool ffwd_rew(int button)
                    if ( (audio_status() & AUDIO_STATUS_PLAY) &&
                          skin_get_global_state()->id3 && skin_get_global_state()->id3->length )
                    {
-                        if (!skin_get_global_state()->paused)
 #if (CONFIG_CODEC == SWCODEC)
-                            audio_pre_ff_rewind();
+                        audio_pre_ff_rewind();
 #else
+                        if (!skin_get_global_state()->paused)
                            audio_pause();
 #endif
 #if CONFIG_KEYPAD == PLAYER_PAD
@@ -472,10 +472,10 @@ static void prev_track(unsigned long skip_thresh)
            return;
        }
-        if (!state->paused)
 #if (CONFIG_CODEC == SWCODEC)
-            audio_pre_ff_rewind();
+        audio_pre_ff_rewind();
 #else
+        if (!state->paused)
            audio_pause();
 #endif
@@ -554,16 +554,20 @@ static void play_hop(int direction)
    {
        elapsed += step * direction;
    }
-    if((audio_status() & AUDIO_STATUS_PLAY) && !state->paused)
+    if(audio_status() & AUDIO_STATUS_PLAY)
    {
 #if (CONFIG_CODEC == SWCODEC)
        audio_pre_ff_rewind();
 #else
-        audio_pause();
+        if (!state->paused)
+            audio_pause();
 #endif
    }
+#if (CONFIG_CODEC == SWCODEC)
+    audio_ff_rewind(elapsed);
+#else
    audio_ff_rewind(state->id3->elapsed = elapsed);
-#if (CONFIG_CODEC != SWCODEC)
    if (!state->paused)
        audio_resume();
 #endif
@@ -849,10 +853,10 @@ long gui_wps_show(void)
                {
                    if (state->id3->cuesheet)
                    {
-                        if (!state->paused)
 #if (CONFIG_CODEC == SWCODEC)
-                            audio_pre_ff_rewind();
+                        audio_pre_ff_rewind();
 #else
+                        if (!state->paused)
                            audio_pause();
 #endif
                        audio_ff_rewind(0);
@@ -1146,6 +1150,17 @@ static void nextid3available_callback(void* param)
    skin_request_full_update(WPS);
 }
+#ifdef AUDIO_FAST_SKIP_PREVIEW
+/* this is called on the audio_skip caller thread */
+static void track_skip_callback(void *param)
+{
+    struct wps_state *state = skin_get_global_state();
+    state->id3 = audio_current_track();
+    state->nid3 = audio_next_track();
+    skin_request_full_update(WPS);
+    (void)param;
+}
+#endif /* AUDIO_FAST_SKIP_PREVIEW */
 static void wps_state_init(void)
 {
@@ -1167,6 +1182,9 @@ static void wps_state_init(void)
    /* add the WPS track event callbacks */
    add_event(PLAYBACK_EVENT_TRACK_CHANGE, false, track_changed_callback);
    add_event(PLAYBACK_EVENT_NEXTTRACKID3_AVAILABLE, false, nextid3available_callback);
+#ifdef AUDIO_FAST_SKIP_PREVIEW
+    add_event(PLAYBACK_EVENT_TRACK_SKIP, false, track_skip_callback);
+#endif
 }
diff --git a/apps/menus/playback_menu.c b/apps/menus/playback_menu.c
index 9ed0bc5651..4153c1c776 100644
--- a/apps/menus/playback_menu.c
+++ b/apps/menus/playback_menu.c
@@ -159,9 +159,13 @@ static int cuesheet_callback(int action,const struct menu_item_ex *this_item)
    switch (action)
    {
        case ACTION_EXIT_MENUITEM: /* on exit */
+#if CONFIG_CODEC == SWCODEC
+            audio_set_cuesheet(global_settings.cuesheet);
+#else
            if (global_settings.cuesheet)
                splash(HZ*2, ID2P(LANG_PLEASE_REBOOT));
            break;
+#endif
    }
    return action;
 }
diff --git a/apps/metadata.c b/apps/metadata.c
index 46b9482bc7..e88603721b 100644
--- a/apps/metadata.c
+++ b/apps/metadata.c
@@ -33,7 +33,7 @@
 #if CONFIG_CODEC == SWCODEC
-/* For trailing tag stripping */
+/* For trailing tag stripping and base audio data types */
 #include "buffering.h"
 #include "metadata/metadata_common.h"
@@ -239,6 +239,94 @@ const int afmt_rec_format[AFMT_NUM_CODECS] =
 };
 #endif /* CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING) */
+#if CONFIG_CODEC == SWCODEC
+/* Get the canonical AFMT type */
+int get_audio_base_codec_type(int type)
+{
+    int base_type = type;
+    switch (type) {
+        case AFMT_MPA_L1:
+        case AFMT_MPA_L2:
+        case AFMT_MPA_L3:
+            base_type = AFMT_MPA_L3;
+            break;
+        case AFMT_MPC_SV7:
+        case AFMT_MPC_SV8:
+            base_type = AFMT_MPC_SV7;
+            break;
+        case AFMT_MP4_AAC:
+        case AFMT_MP4_AAC_HE:
+            base_type = AFMT_MP4_AAC;
+            break;
+        case AFMT_SAP:
+        case AFMT_CMC:
+        case AFMT_CM3:
+        case AFMT_CMR:
+        case AFMT_CMS:
+        case AFMT_DMC:
+        case AFMT_DLT:
+        case AFMT_MPT:
+        case AFMT_MPD:
+        case AFMT_RMT:
+        case AFMT_TMC:
+        case AFMT_TM8:
+        case AFMT_TM2:
+            base_type = AFMT_SAP;
+            break;
+        default:
+            break;
+    }
+    return base_type;
+}
+/* Get the basic audio type */
+enum data_type get_audio_base_data_type(int afmt)
+{
+    if ((unsigned)afmt >= AFMT_NUM_CODECS)
+        return TYPE_UNKNOWN;
+    switch (get_audio_base_codec_type(afmt))
+    {
+    case AFMT_NSF:
+    case AFMT_SPC:
+    case AFMT_SID:
+    case AFMT_MOD:
+    case AFMT_SAP:
+        /* Type must be allocated and loaded in its entirety onto
+           the buffer */
+        return TYPE_ATOMIC_AUDIO;
+    default:
+        /* Assume type may be loaded and discarded incrementally */
+        return TYPE_PACKET_AUDIO;
+    case AFMT_UNKNOWN:
+        /* Have no idea at all */
+        return TYPE_UNKNOWN;
+    }
+}
+/* Is the format allowed to buffer starting at some offset other than 0
+   or first frame only for resume purposes? */
+bool format_buffers_with_offset(int afmt)
+{
+    switch (afmt)
+    {
+    case AFMT_MPA_L1:
+    case AFMT_MPA_L2:
+    case AFMT_MPA_L3:
+    case AFMT_WAVPACK:
+        /* Format may be loaded at the first needed frame */
+        return true;
+    default:
+        /* Format must be loaded from the beginning of the file
+           (does not imply 'atomic', while 'atomic' implies 'no offset') */
+        return false;
+    }
+}
+#endif /* CONFIG_CODEC == SWCODEC */
 /* Simple file type probing by looking at the filename extension. */
 unsigned int probe_file_format(const char *filename)
@@ -313,7 +401,7 @@ bool get_metadata(struct mp3entry* id3, int fd, const char* trackname)
    }
    
    /* Clear the mp3entry to avoid having bogus pointers appear */
-    memset(id3, 0, sizeof(struct mp3entry));
+    wipe_mp3entry(id3);
    /* Take our best guess at the codec type based on file extension */
    id3->codectype = probe_file_format(trackname);
@@ -414,6 +502,44 @@ void copy_mp3entry(struct mp3entry *dest, const struct mp3entry *orig)
    adjust_mp3entry(dest, dest, orig);
 }
+/* A shortcut to simplify the common task of clearing the struct */
+void wipe_mp3entry(struct mp3entry *id3)
+{
+    memset(id3, 0, sizeof (struct mp3entry));
+}
+#if CONFIG_CODEC == SWCODEC
+/* Glean what is possible from the filename alone - does not parse metadata */
+void fill_metadata_from_path(struct mp3entry *id3, const char *trackname)
+{
+    char *p;
+    /* Clear the mp3entry to avoid having bogus pointers appear */
+    wipe_mp3entry(id3);
+    /* Find the filename portion of the path */
+    p = strrchr(trackname, '/');
+    strlcpy(id3->id3v2buf, p ? ++p : id3->path, ID3V2_BUF_SIZE);
+    /* Get the format from the extension and trim it off */
+    p = strrchr(id3->id3v2buf, '.');
+    if (p)
+    {
+        /* Might be wrong for container formats - should we bother? */
+        id3->codectype = probe_file_format(p);
+        if (id3->codectype != AFMT_UNKNOWN)
+            *p = '\0';
+    }
+    /* Set the filename as the title */
+    id3->title = id3->id3v2buf;
+    /* Copy the path info */
+    strlcpy(id3->path, trackname, sizeof (id3->path));
+}
+#endif /* CONFIG_CODEC == SWCODEC */
 #ifndef __PCTOOL__
 #ifdef HAVE_TAGCACHE
 #if CONFIG_CODEC == SWCODEC
diff --git a/apps/metadata.h b/apps/metadata.h
index c22c1b3ecf..b268a3d474 100644
--- a/apps/metadata.h
+++ b/apps/metadata.h
@@ -266,9 +266,7 @@ struct mp3entry {
    /* resume related */
    unsigned long offset;  /* bytes played */
-#if CONFIG_CODEC != SWCODEC
    int index;             /* playlist index */
-#endif
 #ifdef HAVE_TAGCACHE
    unsigned char autoresumable; /* caches result of autoresumable() */
@@ -309,9 +307,14 @@ bool get_metadata(struct mp3entry* id3, int fd, const char* trackname);
 bool mp3info(struct mp3entry *entry, const char *filename);
 void adjust_mp3entry(struct mp3entry *entry, void *dest, const void *orig);
 void copy_mp3entry(struct mp3entry *dest, const struct mp3entry *orig);
+void wipe_mp3entry(struct mp3entry *id3);
 #if CONFIG_CODEC == SWCODEC
+void fill_metadata_from_path(struct mp3entry *id3, const char *trackname);
+int get_audio_base_codec_type(int type);
 void strip_tags(int handle_id);
+enum data_type get_audio_base_data_type(int afmt);
+bool format_buffers_with_offset(int afmt);
 #endif
 #ifdef HAVE_TAGCACHE
diff --git a/apps/metadata/nsf.c b/apps/metadata/nsf.c
index abb4e6fd80..9207a14048 100644
--- a/apps/metadata/nsf.c
+++ b/apps/metadata/nsf.c
@@ -40,6 +40,9 @@ bool get_nsf_metadata(int fd, struct mp3entry* id3)
    p = id3->id3v2buf;
+    /* Length */
+    id3->length = buf[6]*1000;
    /* Title */
    memcpy(p, &buf[14], 32);
    id3->title = p;
diff --git a/apps/pcmbuf.c b/apps/pcmbuf.c
index f548b156b3..c7baad08e4 100644
--- a/apps/pcmbuf.c
+++ b/apps/pcmbuf.c
@@ -86,6 +86,8 @@ static size_t pcmbuffer_pos IDATA_ATTR;
 /* Amount pcmbuffer_pos will be increased.*/
 static size_t pcmbuffer_fillpos IDATA_ATTR;
+static struct chunkdesc *first_desc;
 /* Gapless playback */
 static bool track_transition IDATA_ATTR;
@@ -144,6 +146,11 @@ static void write_to_crossfade(size_t length);
 static void pcmbuf_finish_crossfade_enable(void);
 #endif
+/* Callbacks into playback.c */
+extern void audio_pcmbuf_position_callback(unsigned int time);
+extern void audio_pcmbuf_track_change(bool pcmbuf);
+extern bool audio_pcmbuf_may_play(void);
 /**************************************/
@@ -153,9 +160,8 @@ static void pcmbuf_finish_crossfade_enable(void);
 #ifndef SIMULATOR
 #undef DESC_DEBUG
 #endif
 #ifdef DESC_DEBUG
-static struct chunkdesc *first_desc;
-static bool show_desc_in_use = false;
 #define DISPLAY_DESC(caller) while(!show_desc(caller))
 #define DESC_IDX(desc)       (desc ? desc - first_desc : -1)
 #define SHOW_DESC(desc)      if(DESC_IDX(desc)==-1) DEBUGF("--"); \
@@ -231,6 +237,7 @@ static void commit_chunk(bool flush_next_time)
            /* Flush! Discard all data after the currently playing chunk,
               and make the current chunk play next */
            logf("commit_chunk: flush");
+            pcm_play_lock();
            write_end_chunk->link = read_chunk->link;
            read_chunk->link = pcmbuf_current;
            while (write_end_chunk->link)
@@ -238,6 +245,9 @@ static void commit_chunk(bool flush_next_time)
                write_end_chunk = write_end_chunk->link;
                pcmbuf_unplayed_bytes -= write_end_chunk->size;
            }
+            read_chunk->end_of_track = track_transition;
+            pcm_play_unlock();
        }
        /* If there is already a read buffer setup, add to it */
        else
@@ -248,7 +258,7 @@ static void commit_chunk(bool flush_next_time)
        /* Otherwise create the buffer */
        read_chunk = pcmbuf_current;
    }
-    
    /* If flush_next_time is true, then the current chunk will be thrown out
     * and the next chunk to be committed will be the next to be played.
     * This is used to empty the PCM buffer for a track change. */
@@ -354,7 +364,7 @@ static bool prepare_insert(size_t length)
 #endif
        {
            logf("pcm starting");
-            if (!(audio_status() & AUDIO_STATUS_PAUSE))
+            if (audio_pcmbuf_may_play())
                pcmbuf_play_start();
        }
    }
@@ -373,8 +383,12 @@ void *pcmbuf_request_buffer(int *count)
    /* crossfade has begun, put the new track samples in fadebuf */
    if (crossfade_active)
    {
-        *count = MIN(*count, CROSSFADE_BUFSIZE/4);
+        int cnt = MIN(*count, CROSSFADE_BUFSIZE/4);
-        return fadebuf;
+        if (prepare_insert(cnt << 2))
+        {
+            *count = cnt;
+            return fadebuf;
+        }
    }
    else
 #endif
@@ -421,9 +435,7 @@ void pcmbuf_write_complete(int count)
 static inline void init_pcmbuffers(void)
 {
-#ifdef DESC_DEBUG
    first_desc = write_chunk;
-#endif
    struct chunkdesc *next = write_chunk;
    next++;
    write_end_chunk = write_chunk;
@@ -494,19 +506,27 @@ void pcmbuf_monitor_track_change(bool monitor)
           currently playing chunk. If not, cancel notification. */
        track_transition = monitor;
        read_end_chunk->end_of_track = monitor;
+        if (!monitor)
+        {
+            /* Clear all notifications */
+            struct chunkdesc *desc = first_desc;
+            struct chunkdesc *end = desc + pcmbuf_descs();
+            while (desc < end)
+                desc++->end_of_track = false;
+        }
    }
    else
    {
        /* Post now if PCM stopped and last buffer was sent. */
        track_transition = false;
        if (monitor)
-            audio_post_track_change(false);
+            audio_pcmbuf_track_change(false);
    }
    pcm_play_unlock();
 }
-void pcmbuf_start_track_change(bool auto_skip)
+bool pcmbuf_start_track_change(bool auto_skip)
 {
    bool crossfade = false;
 #ifdef HAVE_CROSSFADE
@@ -546,9 +566,6 @@ void pcmbuf_start_track_change(bool auto_skip)
        /* Cancel any pending automatic gapless transition */
        pcmbuf_monitor_track_change(false);
-        
-        /* Notify the wps that the track change starts now */
-        audio_post_track_change(false);
        /* Can't do two crossfades at once and, no fade if pcm is off now */
        if (
@@ -559,7 +576,8 @@ void pcmbuf_start_track_change(bool auto_skip)
        {
            pcmbuf_play_stop();
            pcm_play_unlock();
-            return;
+            /* Notify playback that the track change starts now */
+            return true;
        }
        /* Not enough data, or not crossfading, flush the old data instead */
@@ -584,6 +602,9 @@ void pcmbuf_start_track_change(bool auto_skip)
        /* Keep trigger outside the play lock or HW FIFO underruns can happen
           since frequency scaling is *not* always fast */
        trigger_cpu_boost();
+        /* Notify playback that the track change starts now */
+        return true;
    }
    else    /* automatic and not crossfading, so do gapless track change */
    {
@@ -593,6 +614,7 @@ void pcmbuf_start_track_change(bool auto_skip)
         * as the last one in the track. */
        logf("  gapless track change");
        pcmbuf_monitor_track_change(true);
+        return false;
    }
 }
@@ -623,7 +645,7 @@ static void pcmbuf_pcm_callback(unsigned char** start, size_t* size)
        if (pcmbuf_current->end_of_track)
        {
            track_transition = false;
-            audio_post_track_change(true);
+            audio_pcmbuf_track_change(true);
        }
        /* Put the finished chunk back into circulation */
@@ -955,9 +977,6 @@ static void write_to_crossfade(size_t length)
                return;
        }
-        /* Commit samples to the buffer */
-        while (!prepare_insert(length))
-            sleep(1);
        while (length > 0)
        {
            COMMIT_IF_NEEDED;
diff --git a/apps/pcmbuf.h b/apps/pcmbuf.h
index 618b1babad..b7bf8c2b16 100644
--- a/apps/pcmbuf.h
+++ b/apps/pcmbuf.h
@@ -33,13 +33,21 @@ void pcmbuf_play_start(void);
 void pcmbuf_play_stop(void);
 void pcmbuf_pause(bool pause);
 void pcmbuf_monitor_track_change(bool monitor);
-void pcmbuf_start_track_change(bool manual_skip);
+bool pcmbuf_start_track_change(bool manual_skip);
 /* Crossfade */
 #ifdef HAVE_CROSSFADE
 bool pcmbuf_is_crossfade_active(void);
 void pcmbuf_request_crossfade_enable(bool on_off);
 bool pcmbuf_is_same_size(void);
+#else
+/* Dummy functions with sensible returns */
+static inline bool pcmbuf_is_crossfade_active(void)
+    { return false; }
+static inline void pcmbuf_request_crossfade_enable(bool on_off)
+    { return; (void)on_off; }
+static inline bool pcmbuf_is_same_size(void)
+    { return true; }
 #endif
 /* Voice */
diff --git a/apps/playback.c b/apps/playback.c
index 632fd05d3d..a369d15715 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -9,6 +9,7 @@
 *
 * Copyright (C) 2005-2007 Miika Pekkarinen
 * Copyright (C) 2007-2008 Nicolas Pennequin
+ * Copyright (C) 2011      Michael Sevakis
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License
@@ -19,53 +20,61 @@
 * KIND, either express or implied.
 *
 ****************************************************************************/
-/* TODO: Pause should be handled in here, rather than PCMBUF so that voice can
- * play whilst audio is paused */
 #include "config.h"
 #include "system.h"
-#include <string.h>
-#include "playback.h"
-#include "codec_thread.h"
 #include "kernel.h"
+#include "panic.h"
+#include "buffer.h"
+#include "sound.h"
+#include "ata.h"
+#include "usb.h"
 #include "codecs.h"
-#include "buffering.h"
+#include "codec_thread.h"
 #include "voice_thread.h"
-#include "usb.h"
+#include "metadata.h"
-#include "ata.h"
+#include "cuesheet.h"
+#include "buffering.h"
+#include "talk.h"
 #include "playlist.h"
+#include "abrepeat.h"
 #include "pcmbuf.h"
-#include "buffer.h"
+#include "playback.h"
-#include "cuesheet.h"
 #ifdef HAVE_TAGCACHE
 #include "tagcache.h"
 #endif
+#ifdef AUDIO_HAVE_RECORDING
+#include "pcm_record.h"
+#endif
 #ifdef HAVE_LCD_BITMAP
 #ifdef HAVE_ALBUMART
 #include "albumart.h"
 #endif
 #endif
-#include "sound.h"
-#include "metadata.h"
-#include "splash.h"
-#include "talk.h"
-#include "panic.h"
-#ifdef HAVE_RECORDING
+/* TODO: The audio thread really is doing multitasking of acting like a
-#include "pcm_record.h"
+         consumer and producer of tracks. It may be advantageous to better
-#endif
+         logically separate the two functions. I won't go that far just yet. */
+/* Internal support for voice playback */
 #define PLAYBACK_VOICE
-/* amount of guess-space to allow for codecs that must hunt and peck
+#if CONFIG_PLATFORM & PLATFORM_NATIVE
- * for their correct seeek target, 32k seems a good size */
+/* Application builds don't support direct code loading */
+#define HAVE_CODEC_BUFFERING
+#endif
+/* Amount of guess-space to allow for codecs that must hunt and peck
+ * for their correct seek target, 32k seems a good size */
 #define AUDIO_REBUFFER_GUESS_SIZE    (1024*32)
 /* Define LOGF_ENABLE to enable logf output in this file */
-/*#define LOGF_ENABLE*/
+/* #define LOGF_ENABLE */
 #include "logf.h"
-/* macros to enable logf for queues
+/* Macros to enable logf for queues
   logging on SYS_TIMEOUT can be disabled */
 #ifdef SIMULATOR
 /* Define this for logf output of all queuing except SYS_TIMEOUT */
@@ -86,202 +95,292 @@
 #define LOGFQUEUE_SYS_TIMEOUT(...)
 #endif
+/* Variables are commented with the threads that use them:
+ * A=audio, C=codec, O=other. A suffix of "-" indicates that the variable is
+ * read but not updated on that thread. Audio is the only user unless otherwise
+ * specified.
+ */
-static enum filling_state {
+/** Miscellaneous **/
-    STATE_IDLE,     /* audio is stopped: nothing to do */
+bool audio_is_initialized = false; /* (A,O-) */
-    STATE_FILLING,  /* adding tracks to the buffer */
+extern struct codec_api ci;        /* (A,C) */
-    STATE_FULL,     /* can't add any more tracks */
-    STATE_END_OF_PLAYLIST, /* all remaining tracks have been added */
+/** Possible arrangements of the main buffer **/
-    STATE_FINISHED, /* all remaining tracks are fully buffered */
+static enum audio_buffer_state
-    STATE_ENDING,   /* audio playback is ending */
+{
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
+    AUDIOBUF_STATE_TRASHED     = -1,     /* trashed; must be reset */
-    STATE_USB,      /* USB mode, ignore most messages */
+    AUDIOBUF_STATE_INITIALIZED =  0,     /* voice+audio OR audio-only */
+    AUDIOBUF_STATE_VOICED_ONLY =  1,     /* voice-only */
+} buffer_state = AUDIOBUF_STATE_TRASHED; /* (A,O) */
+/** Main state control **/
+static bool ff_rw_mode SHAREDBSS_ATTR = false; /* Pre-ff-rewind mode (A,O-) */
+enum play_status
+{
+    PLAY_STOPPED = 0,
+    PLAY_PLAYING = AUDIO_STATUS_PLAY,
+    PLAY_PAUSED  = AUDIO_STATUS_PLAY | AUDIO_STATUS_PAUSE,
+} play_status = PLAY_STOPPED;
+/* Sizeable things that only need exist during playback and not when stopped */
+static struct audio_scratch_memory
+{
+    struct mp3entry codec_id3; /* (A,C) */
+    struct mp3entry unbuffered_id3;
+    struct cuesheet *curr_cue; /* Will follow this structure */
+} * audio_scratch_memory = NULL;
+/* These are used to store the current, next and optionally the peek-ahead
+ * mp3entry's - this guarentees that the pointer returned by audio_current/
+ * next_track will be valid for the full duration of the currently playing
+ * track */
+enum audio_id3_types
+{
+    /* These are allocated statically */
+    PLAYING_ID3 = 0,
+    NEXTTRACK_ID3,
+#ifdef AUDIO_FAST_SKIP_PREVIEW
+    /* The real playing metadata must has to be protected since it contains
+       critical info for other features */
+    PLAYING_PEEK_ID3,
 #endif
-} filling;
+    ID3_TYPE_NUM_STATIC,
+    /* These go in the scratch memory */
-/* As defined in plugins/lib/xxx2wav.h */
+    UNBUFFERED_ID3 = ID3_TYPE_NUM_STATIC,
-#define GUARD_BUFSIZE  (32*1024)
+    CODEC_ID3,
+};
-bool audio_is_initialized = false;
+static struct mp3entry static_id3_entries[ID3_TYPE_NUM_STATIC]; /* (A,O) */
-static bool audio_thread_ready SHAREDBSS_ATTR = false;
-/* Variables are commented with the threads that use them: *
+/* Peeking functions can yield and mess us up */
- * A=audio, C=codec, V=voice. A suffix of - indicates that *
+static struct mutex id3_mutex SHAREDBSS_ATTR; /* (A,0)*/
- * the variable is read but not updated on that thread.    */
-/* TBD: Split out "audio" and "playback" (ie. calling) threads */
-/* Main state control */
-static volatile bool playing SHAREDBSS_ATTR = false;/* Is audio playing? (A) */
-static volatile bool paused SHAREDBSS_ATTR = false; /* Is audio paused? (A/C-) */
-/* Ring buffer where compressed audio and codecs are loaded */
+/** For Scrobbler support **/
-static unsigned char *filebuf = NULL;       /* Start of buffer (A/C-) */
-static size_t filebuflen = 0;               /* Size of buffer (A/C-) */
-/* FIXME: make buf_ridx (C/A-) */
-/* Possible arrangements of the buffer */
+/* Previous track elapsed time */
-enum audio_buffer_state
+static unsigned long prev_track_elapsed = 0; /* (A,O-) */
-{
-    AUDIOBUF_STATE_TRASHED = -1,    /* trashed; must be reset */
-    AUDIOBUF_STATE_INITIALIZED = 0, /* voice+audio OR audio-only */
-    AUDIOBUF_STATE_VOICED_ONLY = 1, /* voice-only */
-};
-static int buffer_state = AUDIOBUF_STATE_TRASHED; /* Buffer state */
-/* These are used to store the current and next (or prev if the current is the last)
- * mp3entry's in a round-robin system. This guarentees that the pointer returned
- * by audio_current/next_track will be valid for the full duration of the
- * currently playing track */
-static struct mp3entry mp3entry_buf[2];
-struct mp3entry *thistrack_id3,  /* the currently playing track */
-                *othertrack_id3; /* prev track during track-change-transition, or end of playlist,
-                                  * next track otherwise */
-static struct mp3entry unbuffered_id3; /* the id3 for the first unbuffered track */
-/* for cuesheet support */
-static struct cuesheet *curr_cue = NULL;
+/** For album art support **/
 #define MAX_MULTIPLE_AA SKINNABLE_SCREENS_COUNT
 #ifdef HAVE_ALBUMART
-static struct albumart_slot {
+static struct albumart_slot
-    struct dim dim;     /* holds width, height of the albumart */
+{
-    int used;           /* counter, increments if something uses it */
+    struct dim dim;     /* Holds width, height of the albumart */
-} albumart_slots[MAX_MULTIPLE_AA];
+    int used;           /* Counter; increments if something uses it */
+} albumart_slots[MAX_MULTIPLE_AA]; /* (A,O) */
 #define FOREACH_ALBUMART(i) for(i = 0;i < MAX_MULTIPLE_AA; i++)
-#endif
+#endif /* HAVE_ALBUMART */
-#define MAX_TRACK       128
+/** Information used for tracking buffer fills **/
-#define MAX_TRACK_MASK  (MAX_TRACK-1)
-/* Track info structure about songs in the file buffer (A/C-) */
+/* Buffer and thread state tracking */
-static struct track_info {
+static enum filling_state
-    int audio_hid;             /* The ID for the track's buffer handle */
+{
-    int id3_hid;               /* The ID for the track's metadata handle */
+    STATE_BOOT = 0, /* audio thread is not ready yet */
-    int codec_hid;             /* The ID for the track's codec handle */
+    STATE_IDLE,     /* audio is stopped: nothing to do */
-#ifdef HAVE_ALBUMART
+    STATE_FILLING,  /* adding tracks to the buffer */
-    int aa_hid[MAX_MULTIPLE_AA];/* The ID for the track's album art handle */
+    STATE_FULL,     /* can't add any more tracks */
+    STATE_END_OF_PLAYLIST, /* all remaining tracks have been added */
+    STATE_FINISHED, /* all remaining tracks are fully buffered */
+    STATE_ENDING,   /* audio playback is ending */
+    STATE_ENDED,    /* audio playback is done */
+#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
+    STATE_USB,      /* USB mode, ignore most messages */
 #endif
-    int cuesheet_hid;          /* The ID for the track's parsed cueesheet handle */
+} filling = STATE_BOOT;
-    size_t filesize;           /* File total length */
+/* Track info - holds information about each track in the buffer */
+struct track_info
+{
+    /* In per-track allocated order: */
+    int id3_hid;                /* Metadata handle ID */
+    int cuesheet_hid;           /* Parsed cueesheet handle ID */
+#ifdef HAVE_ALBUMART
+    int aa_hid[MAX_MULTIPLE_AA];/* Album art handle IDs */
+#endif
+#ifdef HAVE_CODEC_BUFFERING
+    int codec_hid;              /* Buffered codec handle ID */
+#endif
+    int audio_hid;              /* Main audio data handle ID */
+    size_t filesize;            /* File total length on disk
+                                   TODO: This should be stored
+                                         in the handle or the
+                                         id3 and would use less
+                                         ram */
+};
-    bool taginfo_ready;        /* Is metadata read */
+/* Track list - holds info about all buffered tracks */
-    
+#if MEMORYSIZE >= 32
-} tracks[MAX_TRACK];
+#define TRACK_LIST_LEN  128 /* Must be 2^int(+n) */
+#elif MEMORYSIZE >= 16
+#define TRACK_LIST_LEN   64
+#elif MEMORYSIZE >= 8
+#define TRACK_LIST_LEN   32
+#else
+#define TRACK_LIST_LEN   16
+#endif
-static volatile int track_ridx = 0;  /* Track being decoded (A/C-) */
+#define TRACK_LIST_MASK (TRACK_LIST_LEN-1)
-static int track_widx = 0;           /* Track being buffered (A) */
-#define CUR_TI (&tracks[track_ridx]) /* Playing track info pointer (A/C-) */
-static struct track_info *prev_ti = NULL;  /* Pointer to the previously played
+static struct
-                                              track */
+{
+    /* read, write and current are maintained unwrapped, limited only by the
+       unsigned int range and wrap-safe comparisons are used */
-/* Information used only for filling the buffer */
+    /* NOTE: there appears to be a bug in arm-elf-eabi-gcc 4.4.4 for ARMv4 where
-/* Playlist steps from playing track to next track to be buffered (A) */
+       if 'end' follows 'start' in this structure, track_list_count performs
-static int last_peek_offset = 0;
+       'start - end' rather than 'end - start', giving negative count values...
+       so leave it this way for now! */
+    unsigned int end;           /* Next open position */
+    unsigned int start;         /* First track in list */
+    unsigned int current;       /* Currently decoding track */
+    struct track_info tracks[TRACK_LIST_LEN]; /* Buffered track information */
+} track_list; /* (A, O-) */
-/* Scrobbler support */
-static unsigned long prev_track_elapsed = 0; /* Previous track elapsed time (C/A-)*/
-/* Track change controls */
+/* Playlist steps from playlist position to next track to be buffered */
-static bool automatic_skip = false; /* Who initiated in-progress skip? (A) */
+static int playlist_peek_offset = 0;
-extern bool track_transition;       /* Are we in a track transition? */
-static bool dir_skip = false;       /* Is a directory skip pending? (A) */
-static bool new_playlist = false;   /* Are we starting a new playlist? (A) */
-static int wps_offset = 0;          /* Pending track change offset, to keep WPS responsive (A) */
-static bool skipped_during_pause = false; /* Do we need to clear the PCM buffer when playback resumes (A) */
-static bool start_play_g = false; /* Used by audio_load_track to notify
+/* Metadata handle of track load in progress (meaning all handles have not
-                                     audio_finish_load_track about start_play */
+   yet been opened for the track, id3 always exists or the track does not)
-/* True when a track load is in progress, i.e. audio_load_track() has returned
+   Tracks are keyed by their metadata handles if track list pointers are
- * but audio_finish_load_track() hasn't been called yet. Used to avoid allowing
+   insufficient to make comparisons */
- * audio_load_track() to get called twice in a row, which would cause problems.
+static int in_progress_id3_hid = ERR_HANDLE_NOT_FOUND;
- */
-static bool track_load_started = false;
 #ifdef HAVE_DISK_STORAGE
-static size_t buffer_margin  = 5; /* Buffer margin aka anti-skip buffer (A/C-) */
+/* Buffer margin A.K.A. anti-skip buffer (in seconds) */
+static size_t buffer_margin = 5;
 #endif
-/* Event queues */
+/* Values returned for track loading */
-struct event_queue audio_queue SHAREDBSS_ATTR;
+enum track_load_status
-static struct event_queue pcmbuf_queue SHAREDBSS_ATTR;
+{
+    LOAD_TRACK_ERR_START_CODEC   = -6,
+    LOAD_TRACK_ERR_FINISH_FAILED = -5,
+    LOAD_TRACK_ERR_FINISH_FULL   = -4,
+    LOAD_TRACK_ERR_BUSY          = -3,
+    LOAD_TRACK_ERR_NO_MORE       = -2,
+    LOAD_TRACK_ERR_FAILED        = -1,
+    LOAD_TRACK_OK                =  0,
+    LOAD_TRACK_READY             =  1,
+};
-extern struct codec_api ci;
+/** Track change controls **/
-extern unsigned int codec_thread_id;
+/* What sort of skip is pending globally? */
+enum track_skip_type
+{
+    /* Relative to what user is intended to see: */
+    /* Codec: +0, Track List: +0, Playlist: +0 */
+    TRACK_SKIP_NONE = 0,          /* no track skip */
+    /* Codec: +1, Track List: +1, Playlist: +0 */
+    TRACK_SKIP_AUTO,              /* codec-initiated skip */
+    /* Codec: +1, Track List: +1, Playlist: +1 */
+    TRACK_SKIP_AUTO_NEW_PLAYLIST, /* codec-initiated skip is new playlist */
+    /* Codec: xx, Track List: +0, Playlist: +0 */
+    TRACK_SKIP_AUTO_END_PLAYLIST, /* codec-initiated end of playlist */
+    /* Manual skip: Never pends */
+    TRACK_SKIP_MANUAL,            /* manual track skip */
+    /* Manual skip: Never pends */
+    TRACK_SKIP_DIR_CHANGE,        /* manual directory skip */
+} skip_pending = TRACK_SKIP_NONE;
+/* Note about TRACK_SKIP_AUTO_NEW_PLAYLIST:
+   Fixing playlist code to be able to peek into the first song of
+   the next playlist would fix any issues and this wouldn't need
+   to be a special case since pre-advancing the playlist would be
+   unneeded - it could be much more like TRACK_SKIP_AUTO and all
+   actions that require reversal during an in-progress transition
+   would work as expected */
+/* Used to indicate status for the events. Must be separate to satisfy all
+   clients so the correct metadata is read when sending the change events
+   and also so that it is read correctly outside the events. */
+static bool automatic_skip = false; /* (A, O-) */
+/* Pending manual track skip offset */
+static int skip_offset = 0; /* (A, O) */
+/* Track change notification */
+static struct
+{
+    unsigned int in;  /* Number of pcmbuf posts (audio isr) */
+    unsigned int out; /* Number of times audio has read the difference */
+} track_change = { 0, 0 };
+/** Codec status **/
+/* Did the codec notify us it finished while we were paused or while still 
+   in an automatic transition?
+   If paused, it is necessary to defer a codec-initiated skip until resuming
+   or else the track will move forward while not playing audio!
+   If in-progress, skips should not build-up ahead of where the WPS is when
+   really short tracks finish decoding.
+   If it is forgotten, it will be missed altogether and playback will just sit
+   there looking stupid and comatose until the user does something */
+static bool codec_skip_pending = false;
+static int  codec_skip_status;
+static bool codec_seeking = false;          /* Codec seeking ack expected? */
-/* Multiple threads */
-/* Set the watermark to trigger buffer fill (A/C) */
+/* Event queues */
-static void set_filebuf_watermark(void);
+static struct event_queue audio_queue SHAREDBSS_ATTR;
 /* Audio thread */
 static struct queue_sender_list audio_queue_sender_list SHAREDBSS_ATTR;
 static long audio_stack[(DEFAULT_STACK_SIZE + 0x1000)/sizeof(long)];
 static const char audio_thread_name[] = "audio";
+static unsigned int audio_thread_id = 0;
+/* Forward declarations */
+enum audio_start_playback_flags
+{
+    AUDIO_START_RESTART = 0x1, /* "Restart" playback (flush _all_ tracks) */
+    AUDIO_START_NEWBUF  = 0x2, /* Mark the audiobuffer as invalid */
+};
-static void audio_thread(void);
+static void audio_start_playback(size_t offset, unsigned int flags);
-static void audio_initiate_track_change(long direction);
-static bool audio_have_tracks(void);
-static void audio_reset_buffer(void);
 static void audio_stop_playback(void);
+static void buffer_event_buffer_low_callback(void *data);
+static void buffer_event_rebuffer_callback(void *data);
+static void buffer_event_finished_callback(void *data);
 /**************************************/
-/** Pcmbuf callbacks */
+/** --- audio_queue helpers --- **/
-/* Between the codec and PCM track change, we need to keep updating the
+/* codec thread needs access */
- * "elapsed" value of the previous (to the codec, but current to the
+void audio_queue_post(long id, intptr_t data)
- * user/PCM/WPS) track, so that the progressbar reaches the end.
- * During that transition, the WPS will display othertrack_id3. */
-void audio_pcmbuf_position_callback(unsigned int time)
 {
-    time += othertrack_id3->elapsed;
+    queue_post(&audio_queue, id, data);
-    othertrack_id3->elapsed = (time >= othertrack_id3->length)
-        ? othertrack_id3->length : time;
 }
-/* Post message from pcmbuf that the end of the previous track
+static intptr_t audio_queue_send(long id, intptr_t data)
- * has just been played. */
-void audio_post_track_change(bool pcmbuf)
 {
-    if (pcmbuf)
+    return queue_send(&audio_queue, id, data);
-    {
-        LOGFQUEUE("pcmbuf > pcmbuf Q_AUDIO_TRACK_CHANGED");
-        queue_post(&pcmbuf_queue, Q_AUDIO_TRACK_CHANGED, 0);
-    }
-    else
-    {
-        LOGFQUEUE("pcmbuf > audio Q_AUDIO_TRACK_CHANGED");
-        queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0);
-    }
 }
-/* Scan the pcmbuf queue and return true if a message pulled */
-static bool pcmbuf_queue_scan(struct queue_event *ev)
-{
-    if (!queue_empty(&pcmbuf_queue))
-    {
-        /* Transfer message to audio queue */
-        pcm_play_lock();
-        /* Pull message - never, ever any blocking call! */
-        queue_wait_w_tmo(&pcmbuf_queue, ev, 0);
-        pcm_play_unlock();
-        return true;
-    }
-    return false;
-}
+/** --- MP3Entry --- **/
-/** Helper functions */
+/* Does the mp3entry have enough info for us to use it? */
+static struct mp3entry * valid_mp3entry(const struct mp3entry *id3)
+{
+    return id3 && (id3->length != 0 || id3->filesize != 0) &&
+           id3->codectype != AFMT_UNKNOWN ? (struct mp3entry *)id3 : NULL;
+}
-static struct mp3entry *bufgetid3(int handle_id)
+/* Return a pointer to an mp3entry on the buffer, as it is */
+static struct mp3entry * bufgetid3(int handle_id)
 {
    if (handle_id < 0)
        return NULL;
@@ -295,6 +394,7 @@ static struct mp3entry *bufgetid3(int handle_id)
    return id3;
 }
+/* Read an mp3entry from the buffer, adjusted */
 static bool bufreadid3(int handle_id, struct mp3entry *id3out)
 {
    struct mp3entry *id3 = bufgetid3(handle_id);
@@ -308,1200 +408,1441 @@ static bool bufreadid3(int handle_id, struct mp3entry *id3out)
    return false;
 }
-static bool clear_track_info(struct track_info *track)
+/* Lock the id3 mutex */
+static void id3_mutex_lock(void)
 {
-    /* bufclose returns true if the handle is not found, or if it is closed
+    mutex_lock(&id3_mutex);
-     * successfully, so these checks are safe on non-existant handles */
-    if (!track)
-        return false;
-    if (track->codec_hid >= 0) {
-        if (bufclose(track->codec_hid))
-            track->codec_hid = -1;
-        else
-            return false;
-    }
-    if (track->id3_hid >= 0) {
-        if (bufclose(track->id3_hid))
-            track->id3_hid = -1;
-        else
-            return false;
-    }
-    if (track->audio_hid >= 0) {
-        if (bufclose(track->audio_hid))
-            track->audio_hid = -1;
-        else
-            return false;
-    }
-#ifdef HAVE_ALBUMART
-    {
-        int i;
-        FOREACH_ALBUMART(i)
-        {
-            if (track->aa_hid[i] >= 0) {
-                if (bufclose(track->aa_hid[i]))
-                    track->aa_hid[i] = -1;
-                else
-                    return false;
-            }   
-        }
-    }
-#endif
-    if (track->cuesheet_hid >= 0) {
-        if (bufclose(track->cuesheet_hid))
-            track->cuesheet_hid = -1;
-        else
-            return false;
-    }
-    track->filesize = 0;
-    track->taginfo_ready = false;
-    return true;
 }
-/* --- External interfaces --- */
+/* Unlock the id3 mutex */
+static void id3_mutex_unlock(void)
-/* This sends a stop message and the audio thread will dump all it's
-   subsequenct messages */
-void audio_hard_stop(void)
 {
-    /* Stop playback */
+    mutex_unlock(&id3_mutex);
-    LOGFQUEUE("audio >| audio Q_AUDIO_STOP: 1");
-    queue_send(&audio_queue, Q_AUDIO_STOP, 1);
-#ifdef PLAYBACK_VOICE
-    voice_stop();
-#endif
 }
-bool audio_restore_playback(int type)
+/* Return one of the collection of mp3entry pointers - collect them all here */
+static inline struct mp3entry * id3_get(enum audio_id3_types id3_num)
 {
-    switch (type)
+    switch (id3_num)
    {
-    case AUDIO_WANT_PLAYBACK:
+    case UNBUFFERED_ID3:
-        if (buffer_state != AUDIOBUF_STATE_INITIALIZED)
+        return &audio_scratch_memory->unbuffered_id3;
-            audio_reset_buffer();
+    case CODEC_ID3:
-        return true;
+        return &audio_scratch_memory->codec_id3;
-    case AUDIO_WANT_VOICE:
-        if (buffer_state == AUDIOBUF_STATE_TRASHED)
-            audio_reset_buffer();
-        return true;
    default:
-        return false;
+        return &static_id3_entries[id3_num];
    }
 }
-unsigned char *audio_get_buffer(bool talk_buf, size_t *buffer_size)
+/* Copy an mp3entry into one of the mp3 entries */
+static void id3_write(enum audio_id3_types id3_num,
+                      const struct mp3entry *id3_src)
 {
-    unsigned char *buf, *end;
+    struct mp3entry *dest_id3 = id3_get(id3_num);
-    if (audio_is_initialized)
-    {
-        audio_hard_stop();
-    }
-    /* else buffer_state will be AUDIOBUF_STATE_TRASHED at this point */
-    /* Reset the buffering thread so that it doesn't try to use the data */
-    buffering_reset(filebuf, filebuflen);
-    if (buffer_size == NULL)
-    {
-        /* Special case for talk_init to use since it already knows it's
-           trashed */
-        buffer_state = AUDIOBUF_STATE_TRASHED;
-        return NULL;
-    }
-    if (talk_buf || buffer_state == AUDIOBUF_STATE_TRASHED
+    if (id3_src)
-           || !talk_voice_required())
+        copy_mp3entry(dest_id3, id3_src);
-    {
-        logf("get buffer: talk, audio");
-        /* Ok to use everything from audiobuf to audiobufend - voice is loaded,
-           the talk buffer is not needed because voice isn't being used, or
-           could be AUDIOBUF_STATE_TRASHED already. If state is
-           AUDIOBUF_STATE_VOICED_ONLY, no problem as long as memory isn't written
-           without the caller knowing what's going on. Changing certain settings
-           may move it to a worse condition but the memory in use by something
-           else will remain undisturbed.
-         */
-        if (buffer_state != AUDIOBUF_STATE_TRASHED)
-        {
-            talk_buffer_steal();
-            buffer_state = AUDIOBUF_STATE_TRASHED;
-        }
-        buf = audiobuf;
-        end = audiobufend;
-    }
    else
-    {
+        wipe_mp3entry(dest_id3);
-        /* Safe to just return this if already AUDIOBUF_STATE_VOICED_ONLY or
-           still AUDIOBUF_STATE_INITIALIZED */
-        /* Skip talk buffer and move pcm buffer to end to maximize available
-           contiguous memory - no audio running means voice will not need the
-           swap space */
-        logf("get buffer: audio");
-        buf = audiobuf + talk_get_bufsize();
-        end = audiobufend - pcmbuf_init(audiobufend);
-        buffer_state = AUDIOBUF_STATE_VOICED_ONLY;
-    }
-    *buffer_size = end - buf;
-    return buf;
 }
-bool audio_buffer_state_trashed(void)
+/* Call id3_write "safely" because peek aheads can yield, even if the fast
+   preview isn't enabled */
+static void id3_write_locked(enum audio_id3_types id3_num,
+                             const struct mp3entry *id3_src)
 {
-    return buffer_state == AUDIOBUF_STATE_TRASHED;
+    id3_mutex_lock();
+    id3_write(id3_num, id3_src);
+    id3_mutex_unlock();
 }
-#ifdef HAVE_RECORDING
-unsigned char *audio_get_recording_buffer(size_t *buffer_size)
+/** --- Track info --- **/
+/* Close a handle and mark it invalid */
+static void track_info_close_handle(int *hid_p)
 {
-    /* Stop audio, voice and obtain all available buffer space */
+    int hid = *hid_p;
-    audio_hard_stop();
-    talk_buffer_steal();
-    unsigned char *end = audiobufend;
+    /* bufclose returns true if the handle is not found, or if it is closed
-    buffer_state = AUDIOBUF_STATE_TRASHED;
+     * successfully, so these checks are safe on non-existant handles */
-    *buffer_size = end - audiobuf;
+    if (hid >= 0)
+        bufclose(hid);
-    return (unsigned char *)audiobuf;
+    /* Always reset to "no handle" in case it was something else */
+    *hid_p = ERR_HANDLE_NOT_FOUND;
 }
-bool audio_load_encoder(int afmt)
+/* Close all handles in a struct track_info and clear it */
+static void track_info_close(struct track_info *info)
 {
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
+    /* Close them in the order they are allocated on the buffer to speed up
-    LOGFQUEUE("audio >| Q_AUDIO_LOAD_ENCODER: %d", afmt);
+       the handle searching */
-    return queue_send(&audio_queue, Q_AUDIO_LOAD_ENCODER, afmt) > 0;
+    track_info_close_handle(&info->id3_hid);
-#else
+    track_info_close_handle(&info->cuesheet_hid);
-    (void)afmt;
+#ifdef HAVE_ALBUMART
-    return true;
+    int i;
+    FOREACH_ALBUMART(i)
+        track_info_close_handle(&info->aa_hid[i]);
 #endif
-} /* audio_load_encoder */
+#ifdef HAVE_CODEC_BUFFERING
+    track_info_close_handle(&info->codec_hid);
+#endif
+    track_info_close_handle(&info->audio_hid);
+    info->filesize = 0;
+}
-void audio_remove_encoder(void)
+/* Invalidate all members to initial values - does not close handles */
+static void track_info_wipe(struct track_info * info)
 {
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
+    info->id3_hid = ERR_HANDLE_NOT_FOUND;
-    LOGFQUEUE("audio >| Q_AUDIO_LOAD_ENCODER: NULL");
+    info->cuesheet_hid = ERR_HANDLE_NOT_FOUND;
-    queue_send(&audio_queue, Q_AUDIO_LOAD_ENCODER, AFMT_UNKNOWN);
+#ifdef HAVE_ALBUMART
+    int i;
+    FOREACH_ALBUMART(i)
+        info->aa_hid[i] = ERR_HANDLE_NOT_FOUND;
+#endif
+#ifdef HAVE_CODEC_BUFFERING
+    info->codec_hid = ERR_HANDLE_NOT_FOUND;
 #endif
-} /* audio_remove_encoder */
+    info->audio_hid = ERR_HANDLE_NOT_FOUND;
+    info->filesize = 0;
+}
-#endif /* HAVE_RECORDING */
+/** --- Track list --- **/
-struct mp3entry* audio_current_track(void)
+/* Initialize the track list */
+static void track_list_init(void)
 {
-    const char *filename;
+    int i;
-    struct playlist_track_info trackinfo;
+    for (i = 0; i < TRACK_LIST_LEN; i++)
-    int cur_idx;
+        track_info_wipe(&track_list.tracks[i]);
-    int offset = ci.new_track + wps_offset;
-    struct mp3entry *write_id3;
-    cur_idx = (track_ridx + offset) & MAX_TRACK_MASK;
-    if (cur_idx == track_ridx && *thistrack_id3->path)
+    track_list.start = track_list.end = track_list.current;
-    {
+}
-        /* The usual case */
-        if (tracks[cur_idx].cuesheet_hid >= 0 && !thistrack_id3->cuesheet)
-        {
-            bufread(tracks[cur_idx].cuesheet_hid, sizeof(struct cuesheet), curr_cue);
-            thistrack_id3->cuesheet = curr_cue;
-        }
-        return thistrack_id3;
-    }
-    else if (automatic_skip && offset == -1 && *othertrack_id3->path)
-    {
-        /* We're in a track transition. The codec has moved on to the next track,
-             but the audio being played is still the same (now previous) track.
-             othertrack_id3.elapsed is being updated in an ISR by
-             codec_pcmbuf_position_callback */
-        if (tracks[cur_idx].cuesheet_hid >= 0 && !thistrack_id3->cuesheet)
-        {
-            bufread(tracks[cur_idx].cuesheet_hid, sizeof(struct cuesheet), curr_cue);
-            othertrack_id3->cuesheet = curr_cue;
-        }
-        return othertrack_id3;
-    }
-    if (offset != 0)
+/* Return number of items allocated in the list */
-    {
+static unsigned int track_list_count(void)
-        /* Codec may be using thistrack_id3, so it must not be overwritten.
+{
-             If this is a manual skip, othertrack_id3 will become 
+    return track_list.end - track_list.start;
-             thistrack_id3 in audio_check_new_track().
+}
-           FIXME: If this is an automatic skip, it probably means multiple 
-             short tracks fit in the PCM buffer.  Overwriting othertrack_id3
-             can lead to an incorrect value later.
-           Note that othertrack_id3 may also be used for next track.
-          */
-        write_id3 = othertrack_id3;
-    }
-    else 
-    {
-        write_id3 = thistrack_id3;
-    }
-    if (tracks[cur_idx].id3_hid >= 0)
+/* Return true if the list is empty */
-    {
+static inline bool track_list_empty(void)
-        /* The current track's info has been buffered but not read yet, so get it */
+{
-        if (bufreadid3(tracks[cur_idx].id3_hid, write_id3))
+    return track_list.end == track_list.start;
-            return write_id3;
+}
-    }
-    /* We didn't find the ID3 metadata, so we fill temp_id3 with the little info
+/* Returns true if the list is holding the maximum number of items */
-       we have and return that. */
+static bool track_list_full(void)
+{
+    return track_list.end - track_list.start >= TRACK_LIST_LEN;
+}
-    memset(write_id3, 0, sizeof(struct mp3entry));
+/* Test if the index is within the allocated range */
+static bool track_list_in_range(int pos)
+{
+    return (int)(pos - track_list.start) >= 0 &&
+           (int)(pos - track_list.end) < 0;
+}
-    playlist_get_track_info(NULL, playlist_next(0)+wps_offset, &trackinfo);
+static struct track_info * track_list_entry(int pos)
-    filename = trackinfo.filename;
+{
-    if (!filename)
+    return &track_list.tracks[pos & TRACK_LIST_MASK];
-        filename = "No file!";
+}
-#if defined(HAVE_TC_RAMCACHE) && defined(HAVE_DIRCACHE)
+/* Return the info of the last allocation plus an offset, NULL if result is
-    if (tagcache_fill_tags(write_id3, filename))
+   out of bounds */
-        return write_id3;
+static struct track_info * track_list_last(int offset)
-#endif
+{
+    /* Last is before the end since the end isn't inclusive */
+    unsigned int pos = track_list.end + offset - 1;
-    strlcpy(write_id3->path, filename, sizeof(write_id3->path));
+    if (!track_list_in_range(pos))
-    write_id3->title = strrchr(write_id3->path, '/');
+        return NULL;
-    if (!write_id3->title)
-        write_id3->title = &write_id3->path[0];
-    else
-        write_id3->title++;
-    return write_id3;
+    return track_list_entry(pos);
 }
-struct mp3entry* audio_next_track(void)
+/* Allocate space at the end for another track if not full */
+static struct track_info * track_list_alloc_track(void)
 {
-    int next_idx;
+    if (track_list_full())
-    int offset = ci.new_track + wps_offset;
-    if (!audio_have_tracks())
        return NULL;
-    if (wps_offset == -1 && *thistrack_id3->path)
+    return track_list_entry(track_list.end++);
-    {
+}
-        /* We're in a track transition. The next track for the WPS is the one
-           currently being decoded. */
-        return thistrack_id3;
-    }
-    next_idx = (track_ridx + offset + 1) & MAX_TRACK_MASK;
+/* Remove the last track entry allocated in order to support backing out
+   of a track load */
+static void track_list_unalloc_track(void)
+{
+    if (track_list_empty())
+        return;
-    if (tracks[next_idx].id3_hid >= 0)
+    track_list.end--;
-    {
-        if (bufreadid3(tracks[next_idx].id3_hid, othertrack_id3))
-            return othertrack_id3;
-        else
-            return NULL;
-    }
-    if (next_idx == track_widx)
+    if (track_list.current == track_list.end &&
+        track_list.current != track_list.start)
    {
-        /* The next track hasn't been buffered yet, so we return the static
+        /* Current _must_ remain within bounds */
-           version of its metadata. */
+        track_list.current--;
-        return &unbuffered_id3;
    }
-    return NULL;
 }
-/* gets a copy of the id3 data */
+/* Return current track plus an offset, NULL if result is out of bounds */
-bool audio_peek_track(struct mp3entry* id3, int offset)
+static struct track_info * track_list_current(int offset)
 {
-    int next_idx;
+    unsigned int pos = track_list.current + offset;
-    int new_offset = ci.new_track + wps_offset + offset;
-    if (!audio_have_tracks())
-        return false;
-    next_idx = (track_ridx + new_offset) & MAX_TRACK_MASK;
-    if (tracks[next_idx].id3_hid >= 0)
+    if (!track_list_in_range(pos))
-        return bufreadid3(tracks[next_idx].id3_hid, id3);
+        return NULL;
-    return false;
-}
-#ifdef HAVE_ALBUMART
-int playback_current_aa_hid(int slot)
-{
-    if (slot < 0)
-        return -1;
-    int cur_idx;
-    int offset = ci.new_track + wps_offset;
-    cur_idx = track_ridx + offset;
+    return track_list_entry(pos);
-    cur_idx &= MAX_TRACK_MASK;
-    return tracks[cur_idx].aa_hid[slot];
 }
-int playback_claim_aa_slot(struct dim *dim)
+/* Return current based upon what's intended that the user sees - not
+   necessarily where decoding is taking place */
+static struct track_info * track_list_user_current(int offset)
 {
-    int i;
+    if (skip_pending == TRACK_SKIP_AUTO ||
+        skip_pending == TRACK_SKIP_AUTO_NEW_PLAYLIST)
-    /* first try to find a slot already having the size to reuse it
-     * since we don't want albumart of the same size buffered multiple times */
-    FOREACH_ALBUMART(i)
-    {
-        struct albumart_slot *slot = &albumart_slots[i];
-        if (slot->dim.width == dim->width
-                && slot->dim.height == dim->height)
-        {
-            slot->used++;
-            return i;
-        }
-    }
-    /* size is new, find a free slot */
-    FOREACH_ALBUMART(i)
    {
-        if (!albumart_slots[i].used)
+        offset--;
-        {
-            albumart_slots[i].used++;
-            albumart_slots[i].dim = *dim;
-            return i;
-        }
    }
-    /* sorry, no free slot */
-    return -1;
-}
-void playback_release_aa_slot(int slot)
+    return track_list_current(offset);
-{
-    /* invalidate the albumart_slot */
-    struct albumart_slot *aa_slot = &albumart_slots[slot];
-    if (aa_slot->used > 0)
-        aa_slot->used--;
 }
-#endif
+/* Advance current track by an offset, return false if result is out of
-void audio_play(long offset)
+   bounds */
+static struct track_info * track_list_advance_current(int offset)
 {
-    logf("audio_play");
+    unsigned int pos = track_list.current + offset;
-#ifdef PLAYBACK_VOICE
+    if (!track_list_in_range(pos))
-    /* Truncate any existing voice output so we don't have spelling
+        return NULL;
-     * etc. over the first part of the played track */
-    talk_force_shutup();
-#endif
-    /* Start playback */
+    track_list.current = pos;
-    LOGFQUEUE("audio >| audio Q_AUDIO_PLAY: %ld", offset);
+    return track_list_entry(pos);
-    /* Don't return until playback has actually started */
-    queue_send(&audio_queue, Q_AUDIO_PLAY, offset);
 }
-void audio_stop(void)
+/* Clear tracks in the list, optionally preserving the current track -
+   returns 'false' if the operation was changed */
+enum track_clear_action
 {
-    /* Stop playback */
+    TRACK_LIST_CLEAR_ALL = 0, /* Clear all tracks */
-    LOGFQUEUE("audio >| audio Q_AUDIO_STOP");
+    TRACK_LIST_KEEP_CURRENT,  /* Keep current only; clear before + after */
-    /* Don't return until playback has actually stopped */
+    TRACK_LIST_KEEP_NEW       /* Keep current and those that follow */
-    queue_send(&audio_queue, Q_AUDIO_STOP, 0);
+};
-}
-void audio_pause(void)
+static void track_list_clear(enum track_clear_action action)
 {
-    LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE");
+    logf("%s(%d)", __func__, (int)action);
-    /* Don't return until playback has actually paused */
-    queue_send(&audio_queue, Q_AUDIO_PAUSE, true);
-}
-void audio_resume(void)
+    /* Don't care now since rebuffering is imminent */
-{
+    buf_set_watermark(0);
-    LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE resume");
-    /* Don't return until playback has actually resumed */
-    queue_send(&audio_queue, Q_AUDIO_PAUSE, false);
-}
-void audio_skip(int direction)
+    if (action != TRACK_LIST_CLEAR_ALL)
-{
-    if (playlist_check(ci.new_track + wps_offset + direction))
    {
-        if (global_settings.beep)
+        struct track_info *cur = track_list_current(0);
-            pcmbuf_beep(2000, 100, 2500*global_settings.beep);
-        LOGFQUEUE("audio > audio Q_AUDIO_SKIP %d", direction);
+        if (!cur || cur->id3_hid < 0)
-        queue_post(&audio_queue, Q_AUDIO_SKIP, direction);
+            action = TRACK_LIST_CLEAR_ALL; /* Nothing worthwhile keeping */
-        /* Update wps while our message travels inside deep playback queues. */
-        wps_offset += direction;
    }
-    else
+    /* Noone should see this progressing */
+    int start = track_list.start;
+    int current = track_list.current;
+    int end = track_list.end;
+    track_list.start = current;
+    switch (action)
    {
-        /* No more tracks. */
+    case TRACK_LIST_CLEAR_ALL:
-        if (global_settings.beep)
+        /* Result: .start = .current, .end = .current */
-            pcmbuf_beep(1000, 100, 1500*global_settings.beep);
+        track_list.end = current;
+        break;
+    case TRACK_LIST_KEEP_CURRENT:
+        /* Result: .start = .current, .end = .current + 1 */
+        track_list.end = current + 1;
+        break;
+    case TRACK_LIST_KEEP_NEW:
+        /* Result: .start = .current, .end = .end */
+        end = current;
+        break;
    }
-}
-void audio_next(void)
+    /* Close all open handles in the range except the for the current track
-{
+       if preserving that */
-    audio_skip(1);
+    while (start != end)
-}
+    {
+        if (action != TRACK_LIST_KEEP_CURRENT || start != current)
+        {
+            struct track_info *info =
+                &track_list.tracks[start & TRACK_LIST_MASK];
-void audio_prev(void)
+            /* If this is the in-progress load, abort it */
-{
+            if (in_progress_id3_hid >= 0 &&
-    audio_skip(-1);
+                info->id3_hid == in_progress_id3_hid)
-}
+            {
+                in_progress_id3_hid = ERR_HANDLE_NOT_FOUND;
+            }
-void audio_next_dir(void)
+            track_info_close(info);
-{
+        }
-    LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP 1");
-    queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, 1);
-}
-void audio_prev_dir(void)
+        start++;
-{
+    }
-    LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP -1");
-    queue_post(&audio_queue, Q_AUDIO_DIR_SKIP, -1);
 }
-void audio_pre_ff_rewind(void)
-{
-    LOGFQUEUE("audio > audio Q_AUDIO_PRE_FF_REWIND");
-    queue_post(&audio_queue, Q_AUDIO_PRE_FF_REWIND, 0);
-}
-void audio_ff_rewind(long newpos)
+/** --- Audio buffer -- **/
-{
-    LOGFQUEUE("audio > audio Q_AUDIO_FF_REWIND");
-    queue_post(&audio_queue, Q_AUDIO_FF_REWIND, newpos);
-}
-void audio_flush_and_reload_tracks(void)
+/* What size is needed for the scratch buffer? */
+static size_t scratch_mem_size(void)
 {
-    LOGFQUEUE("audio > audio Q_AUDIO_FLUSH");
+    size_t size = sizeof (struct audio_scratch_memory);
-    queue_post(&audio_queue, Q_AUDIO_FLUSH, 0);
+    if (global_settings.cuesheet)
+        size += sizeof (struct cuesheet);
+    return size;
 }
-void audio_error_clear(void)
+/* Initialize the memory area where data is stored that is only used when
+   playing audio and anything depending upon it */
+static void scratch_mem_init(void *mem)
 {
-#ifdef AUDIO_HAVE_RECORDING
+    audio_scratch_memory = (struct audio_scratch_memory *)mem;
-    pcm_rec_error_clear();
+    id3_write_locked(UNBUFFERED_ID3, NULL);
-#endif
+    id3_write(CODEC_ID3, NULL);
+    ci.id3 = id3_get(CODEC_ID3);
+    audio_scratch_memory->curr_cue = NULL;
+    if (global_settings.cuesheet)
+    {
+        audio_scratch_memory->curr_cue =
+            SKIPBYTES((struct cuesheet *)audio_scratch_memory,
+                      sizeof (struct audio_scratch_memory));
+    }
 }
-int audio_status(void)
+/* Set up the audio buffer for playback */
+static void audio_reset_buffer(void)
 {
-    int ret = 0;
+    /*
+     * Layout audio buffer as follows:
+     * [[|TALK]|SCRATCH|BUFFERING|PCM|]
+     */
-    if (playing)
+    /* see audio_get_recording_buffer if this is modified */
-        ret |= AUDIO_STATUS_PLAY;
+    logf("%s()", __func__);
-    if (paused)
+    /* If the setup of anything allocated before the file buffer is
-        ret |= AUDIO_STATUS_PAUSE;
+       changed, do check the adjustments after the buffer_alloc call
+       as it will likely be affected and need sliding over */
-#ifdef HAVE_RECORDING
+    /* Initially set up file buffer as all space available */
-    /* Do this here for constitency with mpeg.c version */
+    unsigned char *filebuf = audiobuf + talk_get_bufsize();
-    /* FIXME: pcm_rec_status() is deprecated */
+    size_t filebuflen = audiobufend - filebuf;
-    ret |= pcm_rec_status();
+    size_t allocsize;
-#endif
-    return ret;
+    ALIGN_BUFFER(filebuf, filebuflen, sizeof (intptr_t));
-}
-bool audio_automatic_skip(void)
+    /* Subtract whatever the pcm buffer says it used plus the guard buffer */
-{
+    allocsize = pcmbuf_init(filebuf + filebuflen);
-    return automatic_skip;
-}
-int audio_get_file_pos(void)
+    /* Make sure filebuflen is a pointer sized multiple after adjustment */
-{
+    allocsize = ALIGN_UP(allocsize, sizeof (intptr_t));
-    return 0;
+    if (allocsize > filebuflen)
-}
+        goto bufpanic;
-#ifdef HAVE_DISK_STORAGE
+    filebuflen -= allocsize;
-void audio_set_buffer_margin(int setting)
-{
-    static const unsigned short lookup[] = {5, 15, 30, 60, 120, 180, 300, 600};
-    buffer_margin = lookup[setting];
-    logf("buffer margin: %ld", (long)buffer_margin);
-    set_filebuf_watermark();
-}
-#endif
-#ifdef HAVE_CROSSFADE
+    /* Scratch memory */
-/* Take necessary steps to enable or disable the crossfade setting */
+    allocsize = scratch_mem_size();
-void audio_set_crossfade(int enable)
+    if (allocsize > filebuflen)
-{
+        goto bufpanic;
-    size_t offset;
-    bool was_playing;
-    size_t size;
-    /* Tell it the next setting to use */
+    scratch_mem_init(filebuf);
-    pcmbuf_request_crossfade_enable(enable);
+    filebuf += allocsize;
+    filebuflen -= allocsize;
-    /* Return if size hasn't changed or this is too early to determine
+    buffering_reset(filebuf, filebuflen);
-       which in the second case there's no way we could be playing
-       anything at all */
-    if (pcmbuf_is_same_size()) return;
-    offset = 0;
+    /* Clear any references to the file buffer */
-    was_playing = playing;
+    buffer_state = AUDIOBUF_STATE_INITIALIZED;
-    /* Playback has to be stopped before changing the buffer size */
+#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
-    if (was_playing)
+    /* Make sure everything adds up - yes, some info is a bit redundant but
+       aids viewing and the sumation of certain variables should add up to
+       the location of others. */
    {
-        /* Store the track resume position */
+        size_t pcmbufsize;
-        offset = thistrack_id3->offset;
+        const unsigned char *pcmbuf = pcmbuf_get_meminfo(&pcmbufsize);
+        logf("fbuf:   %08X", (unsigned)filebuf);
+        logf("fbufe:  %08X", (unsigned)(filebuf + filebuflen));
+        logf("sbuf:   %08X", (unsigned)audio_scratch_memory);
+        logf("sbufe:  %08X", (unsigned)(audio_scratch_memory + allocsize));
+        logf("pcmb:   %08X", (unsigned)pcmbuf);
+        logf("pcmbe:  %08X", (unsigned)(pcmbuf + pcmbufsize));
    }
+#endif
-    /* Blast it - audio buffer will have to be setup again next time
+    return;
-       something plays */
-    audio_get_buffer(true, &size);
-    /* Restart playback if audio was running previously */
+bufpanic:
-    if (was_playing)
+    panicf("%s(): EOM (%zu > %zu)", __func__, allocsize, filebuflen);
-        audio_play(offset);
 }
-#endif
-/* --- Routines called from multiple threads --- */
-static void set_filebuf_watermark(void)
+/* Set the buffer margin to begin rebuffering when 'seconds' from empty */
+static void audio_update_filebuf_watermark(int seconds)
 {
-    if (!filebuf)
+    size_t bytes = 0;
-        return;     /* Audio buffers not yet set up */
 #ifdef HAVE_DISK_STORAGE
-    int seconds;
    int spinup = ata_spinup_time();
+    if (seconds == 0)
+    {
+        /* By current setting */
+        seconds = buffer_margin;
+    }
+    else
+    {
+        /* New setting */
+        buffer_margin = seconds;
+        if (buf_get_watermark() == 0)
+        {
+            /* Write a watermark only if the audio thread already did so for
+               itself or it will fail to set the event and the watermark - if
+               it hasn't yet, it will use the new setting when it does */
+            return;
+        }
+    }
    if (spinup)
-        seconds = (spinup / HZ) + 1;
+        seconds += (spinup / HZ) + 1;
    else
-        seconds = 5;
+        seconds += 5;
    seconds += buffer_margin;
 #else
    /* flash storage */
-    int seconds = 1;
+    seconds = 1;
 #endif
-    /* bitrate of last track in buffer dictates watermark */
+    /* Watermark is a function of the bitrate of the last track in the buffer */
-    struct mp3entry* id3 = NULL;
+    struct mp3entry *id3 = NULL;   
-    if (tracks[track_widx].taginfo_ready)
+    struct track_info *info = track_list_last(0);
-        id3 = bufgetid3(tracks[track_widx].id3_hid);
+    if (info)
+        id3 = valid_mp3entry(bufgetid3(info->id3_hid));
+    if (id3)
+    {
+        if (get_audio_base_data_type(id3->codectype) == TYPE_PACKET_AUDIO)
+        {
+            bytes = id3->bitrate * (1000/8) * seconds;
+        }
+        else
+        {
+            /* Bitrate has no meaning to buffering margin for atomic audio -
+               rebuffer when it's the only track left unless it's the only 
+               track that fits, in which case we should avoid constant buffer
+               low events */
+            if (track_list_count() > 1)
+                bytes = info->filesize + 1;
+        }
+    }
    else
-        id3 = bufgetid3(tracks[track_widx-1].id3_hid);
+    {
-    if (!id3) {
+        /* Then set the minimum - this should not occur anyway */
-        logf("fwmark: No id3 for last track (r%d/w%d), aborting!", track_ridx, track_widx);
+        logf("fwmark: No id3 for last track (s%u/c%u/e%u)",
-        return;
+             track_list.start, track_list.current, track_list.end);
    }
-    size_t bytes = id3->bitrate * (1000/8) * seconds;
-    buf_set_watermark(bytes);
+    /* Actually setting zero disables the notification and we use that
-    logf("fwmark: %d", bytes);
+       to detect that it has been reset */
+    buf_set_watermark(MAX(bytes, 1));
+    logf("fwmark: %lu", (unsigned long)bytes);
 }
-/* --- Buffering callbacks --- */
-static void buffering_low_buffer_callback(void *data)
+/** -- Track change notification -- **/
+/* Check the pcmbuf track changes and return write the message into the event
+   if there are any */
+static inline bool audio_pcmbuf_track_change_scan(void)
 {
-    (void)data;
+    if (track_change.out != track_change.in)
-    logf("low buffer callback");
+    {
+        track_change.out++;
+        return true;
+    }
+    return false;
+}
+/* Clear outstanding track change posts */
+static inline void audio_pcmbuf_track_change_clear(void)
+{
+    track_change.out = track_change.in;
+}
+/* Post a track change notification - called by audio ISR */
+static inline void audio_pcmbuf_track_change_post(void)
+{
+    track_change.in++;
+}
-    if (filling == STATE_FULL || filling == STATE_END_OF_PLAYLIST) {
-        /* force a refill */
+/** --- Helper functions --- **/
-        LOGFQUEUE("buffering > audio Q_AUDIO_FILL_BUFFER");
-        queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0);
+/* Removes messages that might end up in the queue before or while processing
+   a manual track change. Responding to them would be harmful since they
+   belong to a previous track's playback period. Anything that would generate
+   the stale messages must first be put into a state where it will not do so.
+ */
+static void audio_clear_track_notifications(void)
+{
+    static const long filter_list[][2] =
+    {
+        /* codec messages */
+        { Q_AUDIO_CODEC_SEEK_COMPLETE, Q_AUDIO_CODEC_COMPLETE },
+        /* track change messages */
+        { Q_AUDIO_TRACK_CHANGED, Q_AUDIO_TRACK_CHANGED },
+    };
+    const int filter_count = ARRAYLEN(filter_list) - 1;
+    /* Remove any pcmbuf notifications */
+    pcmbuf_monitor_track_change(false);
+    audio_pcmbuf_track_change_clear();
+    /* Scrub the audio queue of the old mold */
+    while (queue_peek_ex(&audio_queue, NULL,
+                         filter_count | QPEEK_REMOVE_EVENTS,
+                         filter_list))
+    {
+        yield(); /* Not strictly needed, per se, ad infinitum, ra, ra */
    }
 }
-static void buffering_handle_rebuffer_callback(void *data)
+/* Takes actions based upon track load status codes */
+static void audio_handle_track_load_status(int trackstat)
 {
-    (void)data;
+    switch (trackstat)
-    LOGFQUEUE("audio >| audio Q_AUDIO_FLUSH");
+    {
-    queue_post(&audio_queue, Q_AUDIO_FLUSH, 0);
+    case LOAD_TRACK_ERR_NO_MORE:
+        if (track_list_count() > 0)
+            break;
+    case LOAD_TRACK_ERR_START_CODEC:
+        audio_queue_post(Q_AUDIO_CODEC_COMPLETE, CODEC_ERROR);
+        break;
+    default:
+        break;
+    }
 }
-static void buffering_handle_finished_callback(void *data)
+/* Announce the end of playing the current track */
+static void audio_playlist_track_finish(void)
 {
-    logf("handle %d finished buffering", *(int*)data);
+    struct mp3entry *id3 = valid_mp3entry(id3_get(PLAYING_ID3));
-    int hid = (*(int*)data);
-    if (hid == tracks[track_widx].id3_hid)
+    playlist_update_resume_info(filling == STATE_ENDED ? NULL : id3);
+    if (id3)
    {
-        int offset = ci.new_track + wps_offset;
+        send_event(PLAYBACK_EVENT_TRACK_FINISH, id3);
-        int next_idx = (track_ridx + offset + 1) & MAX_TRACK_MASK;
+        prev_track_elapsed = id3->elapsed;
-        /* The metadata handle for the last loaded track has been buffered.
-           We can ask the audio thread to load the rest of the track's data. */
-        LOGFQUEUE("audio > audio Q_AUDIO_FINISH_LOAD");
-        queue_post(&audio_queue, Q_AUDIO_FINISH_LOAD, 0);
-        if (tracks[next_idx].id3_hid == hid)
-            send_event(PLAYBACK_EVENT_NEXTTRACKID3_AVAILABLE, NULL);
    }
    else
    {
-        /* This is most likely an audio handle, so we strip the useless
+        prev_track_elapsed = 0;
-           trailing tags that are left. */
-        strip_tags(hid);
-        if (hid == tracks[track_widx-1].audio_hid
-            && filling == STATE_END_OF_PLAYLIST)
-        {
-            /* This was the last track in the playlist.
-               We now have all the data we need. */
-            logf("last track finished buffering");
-            filling = STATE_FINISHED;
-        }
    }
 }
+/* Announce the beginning of the new track */
+static void audio_playlist_track_change(void)
+{
+    struct mp3entry *id3 = valid_mp3entry(id3_get(PLAYING_ID3));
+    if (id3)
+        send_event(PLAYBACK_EVENT_TRACK_CHANGE, id3);
-/* --- Audio thread --- */
+    playlist_update_resume_info(id3);
+}
-static bool audio_have_tracks(void)
+/* Change the data for the next track and send the event */
+static void audio_update_and_announce_next_track(const struct mp3entry *id3_next)
 {
-    return (audio_track_count() != 0);
+    id3_write_locked(NEXTTRACK_ID3, id3_next);
+    send_event(PLAYBACK_EVENT_NEXTTRACKID3_AVAILABLE,
+               id3_get(NEXTTRACK_ID3));
 }
-static int audio_free_track_count(void)
+/* Bring the user current mp3entry up to date and set a new offset for the
+   buffered metadata */
+static void playing_id3_sync(struct track_info *user_info, size_t offset)
 {
-    /* Used tracks + free tracks adds up to MAX_TRACK - 1 */
+    id3_mutex_lock();
-    return MAX_TRACK - 1 - audio_track_count();
+    struct mp3entry *id3 = bufgetid3(user_info->id3_hid);
+    if (offset == (size_t)-1)
+    {
+        struct mp3entry *ply_id3 = id3_get(PLAYING_ID3);
+        size_t play_offset = ply_id3->offset;
+        long play_elapsed = ply_id3->elapsed;
+        id3_write(PLAYING_ID3, id3);
+        ply_id3->offset = play_offset;
+        ply_id3->elapsed = play_elapsed;
+        offset = 0;
+    }
+    else
+    {
+        id3_write(PLAYING_ID3, id3);
+    }
+    if (id3)
+        id3->offset = offset;
+    id3_mutex_unlock();
 }
-int audio_track_count(void)
+/* Wipe-out track metadata - current is optional */
+static void wipe_track_metadata(bool current)
 {
-    /* Calculate difference from track_ridx to track_widx
+    id3_mutex_lock();
-     * taking into account a possible wrap-around. */
-    return (MAX_TRACK + track_widx - track_ridx) & MAX_TRACK_MASK;
+    if (current)
+        id3_write(PLAYING_ID3, NULL);
+    id3_write(NEXTTRACK_ID3, NULL);
+    id3_write(UNBUFFERED_ID3, NULL);
+    id3_mutex_unlock();
 }
-long audio_filebufused(void)
+/* Called when buffering is completed on the last track handle */
+static void filling_is_finished(void)
 {
-    return (long) buf_used();
+    logf("last track finished buffering");
+    /* There's no more to load or watch for */
+    buf_set_watermark(0);
+    filling = STATE_FINISHED;
 }
-/* Update track info after successful a codec track change */
+/* Stop the codec decoding or waiting for its data to be ready - returns
-static void audio_update_trackinfo(void)
+   'false' if the codec ended up stopped */
+static bool halt_decoding_track(bool stop)
 {
-    bool resume = false;
+    /* If it was waiting for us to clear the buffer to make a rebuffer
+       happen, it should cease otherwise codec_stop could deadlock waiting
+       for the codec to go to its main loop - codec's request will now
+       force-fail */
+    bool retval = false;
-    /* Load the curent track's metadata into curtrack_id3 */
+    buf_signal_handle(ci.audio_hid, true);
-    if (CUR_TI->id3_hid >= 0)
-        bufreadid3(CUR_TI->id3_hid, thistrack_id3);
-    /* Reset current position */
+    if (stop)
-    thistrack_id3->elapsed = 0;
+        codec_stop();
+    else
+        retval = codec_pause();
-#ifdef HAVE_TAGCACHE
+    audio_clear_track_notifications();
-    /* Ignoring resume position for automatic track change if so configured */
-    resume = global_settings.autoresume_enable &&
-        (!automatic_skip     /* Resume all manually selected tracks */
-         || global_settings.autoresume_automatic == AUTORESUME_NEXTTRACK_ALWAYS
-         || (global_settings.autoresume_automatic != AUTORESUME_NEXTTRACK_NEVER
-                                                /* Not never resume? */
-             && autoresumable(thistrack_id3))); /* Pass Resume filter? */
-#endif
-    if (!resume)
+    /* We now know it's idle and not waiting for buffered data */
-    {
+    buf_signal_handle(ci.audio_hid, false);
-        thistrack_id3->offset = 0;
-    }
-    logf("audio_update_trackinfo: Set offset for %s to %lX\n",
+    codec_skip_pending = false;
-         thistrack_id3->title, thistrack_id3->offset);
+    codec_seeking = false;
-    /* Update the codec API */
+    return retval;
-    ci.filesize = CUR_TI->filesize;
-    ci.id3 = thistrack_id3;
-    ci.curpos = 0;
-    ci.taginfo_ready = &CUR_TI->taginfo_ready;
 }
-/* Clear tracks between write and read, non inclusive */
+/* Clear the PCM on a manual skip */
-static void audio_clear_track_entries(void)
+static void audio_clear_paused_pcm(void)
 {
-    int cur_idx = track_widx;
+    if (play_status == PLAY_PAUSED && !pcmbuf_is_crossfade_active())
+        pcmbuf_play_stop();
-    logf("Clearing tracks: r%d/w%d", track_ridx, track_widx);
+}
-    /* Loop over all tracks from write-to-read */
+/* End the ff/rw mode */
-    while (1)
+static void audio_ff_rewind_end(void)
+{
+    /* A seamless seek (not calling audio_pre_ff_rewind) skips this
+       section */
+    if (ff_rw_mode)
    {
-        cur_idx = (cur_idx + 1) & MAX_TRACK_MASK;
+        ff_rw_mode = false;
-        if (cur_idx == track_ridx)
+        if (codec_seeking)
-            break;
+        {
+            /* Clear the buffer */
+            pcmbuf_play_stop();
+        }
-        clear_track_info(&tracks[cur_idx]);
+        if (play_status != PLAY_PAUSED)
+        {
+            /* Seeking-while-playing, resume PCM playback */
+            pcmbuf_pause(false);
+        }
    }
 }
-/* Clear all tracks */
+/* Complete the codec seek */
-static bool audio_release_tracks(void)
+static void audio_complete_codec_seek(void)
 {
-    int i, cur_idx;
+    /* If a seek completed while paused, 'paused' is true.
+     * If seeking from seek mode, 'ff_rw_mode' is true. */
-    logf("releasing all tracks");
+    if (codec_seeking)
-    for(i = 0; i < MAX_TRACK; i++)
    {
-        cur_idx = (track_ridx + i) & MAX_TRACK_MASK;
+        audio_ff_rewind_end();
-        if (!clear_track_info(&tracks[cur_idx]))
+        codec_seeking = false; /* set _after_ the call! */
-            return false;
    }
+    /* else it's waiting and we must repond */
+}
-    return true;
+/* Get the current cuesheet pointer */
+static inline struct cuesheet * get_current_cuesheet(void)
+{
+    return audio_scratch_memory->curr_cue;
 }
-static bool audio_loadcodec(bool start_play)
+/* Read the cuesheet from the buffer */
+static void buf_read_cuesheet(int handle_id)
 {
-    int prev_track, hid;
+    struct cuesheet *cue = get_current_cuesheet();
-    char codec_path[MAX_PATH]; /* Full path to codec */
-    const struct mp3entry *id3, *prev_id3;
-    if (tracks[track_widx].id3_hid < 0) {
+    if (!cue || handle_id < 0)
-        return false;
+        return;
-    }
-    id3 = bufgetid3(tracks[track_widx].id3_hid);
+    bufread(handle_id, sizeof (struct cuesheet), cue);
-    if (!id3)
+}
-        return false;
-    const char *codec_fn = get_codec_filename(id3->codectype);
+/* Backend to peek/current/next track metadata interface functions -
-    if (codec_fn == NULL)
+   fill in the mp3entry with as much information as we may obtain about
+   the track at the specified offset from the user current track -
+   returns false if no information exists with us */
+static bool audio_get_track_metadata(int offset, struct mp3entry *id3)
+{
+    if (play_status == PLAY_STOPPED)
        return false;
-    tracks[track_widx].codec_hid = -1;
+    if (id3->path[0] != '\0')
+        return true; /* Already filled */
+    struct track_info *info = track_list_user_current(offset);
-    if (start_play)
+    if (!info)
    {
-        /* Load the codec directly from disk and save some memory. */
+        struct mp3entry *ub_id3 = id3_get(UNBUFFERED_ID3);
-        track_ridx = track_widx;
-        ci.filesize = CUR_TI->filesize;
+        if (offset > 0 && track_list_user_current(offset - 1))
-        ci.id3 = thistrack_id3;
+        {
-        ci.taginfo_ready = &CUR_TI->taginfo_ready;
+            /* Try the unbuffered id3 since we're moving forward */
-        ci.curpos = 0;
+            if (ub_id3->path[0] != '\0')
-        return codec_load(-1, id3->codectype);
+            {
+                copy_mp3entry(id3, ub_id3);
+                return true;
+            }
+        }
    }
-    else
+    else if (bufreadid3(info->id3_hid, id3))
    {
-        /* If we already have another track than this one buffered */
+        return true;
-        if (track_widx != track_ridx)
+    }
-        {
-            prev_track = (track_widx - 1) & MAX_TRACK_MASK;
-            id3 = bufgetid3(tracks[track_widx].id3_hid);
-            prev_id3 = bufgetid3(tracks[prev_track].id3_hid);
-            /* If the previous codec is the same as this one and the current
+    /* We didn't find the ID3 metadata, so we fill it with the little info we
-             * one is the correct one, there is no need to put another copy of
+       have and return that */
-             * it on the file buffer */
-            if (id3 && prev_id3)
-            {
-                int codt = get_codec_base_type(id3->codectype);
-                int prev_codt = get_codec_base_type(prev_id3->codectype);
-                int cod_loaded = get_codec_base_type(codec_loaded());
-                if (codt == prev_codt && codt == cod_loaded)
+    char path[MAX_PATH+1];
-                {
+    if (playlist_peek(offset, path, sizeof (path)))
-                    logf("Reusing prev. codec");
+    {
-                    return true;
+#if defined(HAVE_TC_RAMCACHE) && defined(HAVE_DIRCACHE)
-                }
+        /* Try to get it from the database */
-            }
+        if (!tagcache_fill_tags(id3, path))
+#endif
+        {
+            /* By now, filename is the only source of info */
+            fill_metadata_from_path(id3, path);
        }
+        return true;
    }
-    codec_get_full_path(codec_path, codec_fn);
+    wipe_mp3entry(id3);
-    hid = tracks[track_widx].codec_hid = bufopen(codec_path, 0, TYPE_CODEC, NULL);
+    return false;
+}
-    /* not an error if codec load it supported, will load it from disk
+/* Get a resume rewind adjusted offset from the ID3 */
-     * application builds don't support it
+unsigned long resume_rewind_adjusted_offset(const struct mp3entry *id3)
-     */
+{
-    if (hid < 0 && hid != ERR_UNSUPPORTED_TYPE)
+    unsigned long offset = id3->offset;
-        return false;
+    size_t resume_rewind = global_settings.resume_rewind *
+                           id3->bitrate * (1000/8);
-    if (hid >= 0)
+    if (offset < resume_rewind)
-        logf("Loaded codec");
+        offset = 0;
    else
-        logf("Buffering codec unsupported, load later from disk");
+        offset -= resume_rewind;
-    return true;
+    return offset;
 }
-/* Load metadata for the next track (with bufopen). The rest of the track
+/* Get the codec into ram and initialize it - keep it if it's ready */
-   loading will be handled by audio_finish_load_track once the metadata has been
+static bool audio_init_codec(struct track_info *track_info,
-   actually loaded by the buffering thread. */
+                             struct mp3entry *track_id3)
-static bool audio_load_track(size_t offset, bool start_play)
 {
-    char name_buf[MAX_PATH + 1];
+    int codt_loaded = get_audio_base_codec_type(codec_loaded());
-    const char *trackname;
+    int hid = ERR_HANDLE_NOT_FOUND;
-    int fd = -1;
-    if (track_load_started) {
+    if (codt_loaded != AFMT_UNKNOWN)
-        /* There is already a track load in progress, so track_widx hasn't been
+    {
-           incremented yet. Loading another track would overwrite the one that
+        int codt = get_audio_base_codec_type(track_id3->codectype);
-           hasn't finished loading. */
-        logf("audio_load_track(): a track load is already in progress");
+        if (codt == codt_loaded)
-        return false;
+        {
+            /* Codec is the same base type */
+            logf("Reusing prev. codec: %d", track_id3->codectype);
+#ifdef HAVE_CODEC_BUFFERING
+            /* Close any buffered codec (we could have skipped directly to a
+               format transistion that is the same format as the current track
+               and the buffered one is no longer needed) */
+            track_info_close_handle(&track_info->codec_hid);
+#endif
+            return true;
+        }
+        else
+        {
+            /* New codec - first make sure the old one's gone */
+            logf("Removing prev. codec: %d", codt_loaded);
+            codec_unload();
+        }
    }
-    start_play_g = start_play;  /* will be read by audio_finish_load_track */
+    logf("New codec: %d/%d", track_id3->codectype, codec_loaded());
-    /* Stop buffer filling if there is no free track entries.
+#ifdef HAVE_CODEC_BUFFERING
-       Don't fill up the last track entry (we wan't to store next track
+    /* Codec thread will close the handle even if it fails and will load from
-       metadata there). */
+       storage if hid is not valid or the buffer load fails */
-    if (!audio_free_track_count())
+    hid = track_info->codec_hid;
-    {
+    track_info->codec_hid = ERR_HANDLE_NOT_FOUND;
-        logf("No free tracks");
+#endif
+    return codec_load(hid, track_id3->codectype);
+    (void)track_info; /* When codec buffering isn't supported */
+}
+/* Start the codec for the current track scheduled to be decoded */
+static bool audio_start_codec(bool auto_skip)
+{
+    struct track_info *info = track_list_current(0);
+    struct mp3entry *cur_id3 = valid_mp3entry(bufgetid3(info->id3_hid));
+    if (!cur_id3)
        return false;
-    }
-    last_peek_offset++;
+    buf_pin_handle(info->id3_hid, true);
-    tracks[track_widx].taginfo_ready = false;
-    logf("Buffering track: r%d/w%d", track_ridx, track_widx);
+    if (!audio_init_codec(info, cur_id3))
-    /* Get track name from current playlist read position. */
-    while ((trackname = playlist_peek(last_peek_offset, name_buf,
-        sizeof(name_buf))) != NULL)
    {
-        /* Handle broken playlists. */
+        buf_pin_handle(info->id3_hid, false);
-        fd = open(trackname, O_RDONLY);
+        return false;
-        if (fd < 0)
-        {
-            logf("Open failed");
-            /* Skip invalid entry from playlist. */
-            playlist_skip_entry(NULL, last_peek_offset);
-        }
-        else
-            break;
    }
-    if (!trackname)
+#ifdef HAVE_TAGCACHE
+    bool autoresume_enable = global_settings.autoresume_enable;
+    if (autoresume_enable && !cur_id3->offset)
    {
-        logf("End-of-playlist");
+        /* Resume all manually selected tracks */
-        memset(&unbuffered_id3, 0, sizeof(struct mp3entry));
+        bool resume = !auto_skip;
-        filling = STATE_END_OF_PLAYLIST;
+        /* Send the "buffer" event to obtain the resume position for the codec */
+        send_event(PLAYBACK_EVENT_TRACK_BUFFER, cur_id3);
-        if (thistrack_id3->length == 0 && thistrack_id3->filesize == 0)
+        if (!resume)
        {
-            /* Stop playback if no valid track was found. */
+            /* Automatic skip - do further tests to see if we should just
-            audio_stop_playback();
+               ignore any autoresume position */
+            int autoresume_automatic = global_settings.autoresume_automatic;
+            switch (autoresume_automatic)
+            {
+            case AUTORESUME_NEXTTRACK_ALWAYS:
+                /* Just resume unconditionally */
+                resume = true;
+                break;
+            case AUTORESUME_NEXTTRACK_NEVER:
+                /* Force-rewind it */
+                break;
+            default:
+                /* Not "never resume" - pass resume filter? */
+                resume = autoresumable(cur_id3);
+            }
        }
-        return false;
+        if (!resume)
+            cur_id3->offset = 0;
+        logf("%s: Set offset for %s to %lX\n", __func__,
+             cur_id3->title, cur_id3->offset);
    }
+#endif /* HAVE_TAGCACHE */
-    tracks[track_widx].filesize = filesize(fd);
+    /* Rewind the required amount - if an autoresume was done, this also rewinds
+       that by the setting's amount
-    if (offset > tracks[track_widx].filesize)
+       It would be best to have bookkeeping about whether or not the track
-        offset = 0;
+       sounded or not since skipping to it or else skipping to it while paused
+       and back again will cause accumulation of silent rewinds - that's not
+       our job to track directly nor could it be in any reasonable way
+     */
+    cur_id3->offset = resume_rewind_adjusted_offset(cur_id3);
+    /* Update the codec API with the metadata and track info */
+    id3_write(CODEC_ID3, cur_id3);
-    /* Set default values */
+    ci.audio_hid = info->audio_hid;
-    if (start_play)
+    ci.filesize = info->filesize;
+    buf_set_base_handle(info->audio_hid);
+    /* All required data is now available for the codec */
+    codec_go();
+#ifdef HAVE_TAGCACHE
+    if (!autoresume_enable || cur_id3->offset)
+#endif
    {
-        buf_set_watermark(filebuflen/2);
+        /* Send the "buffer" event now */
-        dsp_configure(ci.dsp, DSP_RESET, 0);
+        send_event(PLAYBACK_EVENT_TRACK_BUFFER, cur_id3);
-        playlist_update_resume_info(audio_current_track());
    }
-    /* Get track metadata if we don't already have it. */
+    buf_pin_handle(info->id3_hid, false);
-    if (tracks[track_widx].id3_hid < 0)
+    return true;
+    (void)auto_skip; /* ifndef HAVE_TAGCACHE */
+}
+/** --- Audio thread --- **/
+/* Load and parse a cuesheet for the file - returns false if the buffer
+   is full */
+static bool audio_load_cuesheet(struct track_info *info,
+                                struct mp3entry *track_id3)
+{
+    struct cuesheet *cue = get_current_cuesheet();
+    track_id3->cuesheet = NULL;
+    if (cue && info->cuesheet_hid == ERR_HANDLE_NOT_FOUND)
    {
-        tracks[track_widx].id3_hid = bufopen(trackname, 0, TYPE_ID3, NULL);
+        /* If error other than a full buffer, then mark it "unsupported" to
+           avoid reloading attempt */
+        int hid = ERR_UNSUPPORTED_TYPE;
+        char cuepath[MAX_PATH];
-        if (tracks[track_widx].id3_hid < 0)
+#ifdef HAVE_IO_PRIORITY
+        buf_back_off_storage(true);
+#endif
+        if (look_for_cuesheet_file(track_id3->path, cuepath))
        {
-            /* Buffer is full. */
+            hid = bufalloc(NULL, sizeof (struct cuesheet), TYPE_CUESHEET);
-            get_metadata(&unbuffered_id3, fd, trackname);
-            last_peek_offset--;
-            close(fd);
-            logf("buffer is full for now (get metadata)");
-            filling = STATE_FULL;
-            return false;
-        }
-        if (track_widx == track_ridx)
+            if (hid >= 0)
-        {
+            {
-            /* TODO: Superfluos buffering call? */
+                void *cuesheet = NULL;
-            buf_request_buffer_handle(tracks[track_widx].id3_hid);
+                bufgetdata(hid, sizeof (struct cuesheet), &cuesheet);
-            if (bufreadid3(tracks[track_widx].id3_hid, thistrack_id3))
-            {    
+                if (parse_cuesheet(cuepath, (struct cuesheet *)cuesheet))
-                thistrack_id3->offset = offset;
+                {
-                logf("audio_load_track: set offset for %s to %lX\n",
+                    /* Indicate cuesheet is present (while track remains
-                     thistrack_id3->title,
+                       buffered) */
-                     offset);
+                    track_id3->cuesheet = cue;
+                }
+                else
+                {
+                    bufclose(hid);
+                    hid = ERR_UNSUPPORTED_TYPE;
+                }
            }
-            else
-                memset(thistrack_id3, 0, sizeof(struct mp3entry));
        }
-        if (start_play)
+#ifdef HAVE_IO_PRIORITY
+        buf_back_off_storage(false);
+#endif
+        if (hid == ERR_BUFFER_FULL)
        {
-            playlist_update_resume_info(audio_current_track());
+            logf("buffer is full for now (%s)", __func__);
+            return false;
+        }
+        else
+        {
+            if (hid < 0)
+                logf("Cuesheet loading failed");
+            info->cuesheet_hid = hid;
        }
    }
-    close(fd);
-    track_load_started = true; /* Remember that we've started loading a track */
    return true;
 }
 #ifdef HAVE_ALBUMART
-/* Load any album art for the file */
+/* Load any album art for the file - returns false if the buffer is full */
-static void audio_load_albumart(struct mp3entry *track_id3)
+static bool audio_load_albumart(struct track_info *info,
+                                struct mp3entry *track_id3)
 {
    int i;
    FOREACH_ALBUMART(i)
    {
        struct bufopen_bitmap_data user_data;
-        int hid = ERR_HANDLE_NOT_FOUND;
+        int *aa_hid = &info->aa_hid[i];
+        int hid = ERR_UNSUPPORTED_TYPE;
        /* albumart_slots may change during a yield of bufopen,
         * but that's no problem */
-        if (tracks[track_widx].aa_hid[i] >= 0 || !albumart_slots[i].used)
+        if (*aa_hid >= 0 || *aa_hid == ERR_UNSUPPORTED_TYPE ||
+            !albumart_slots[i].used)
            continue;
        memset(&user_data, 0, sizeof(user_data));
-        user_data.dim = &(albumart_slots[i].dim);
+        user_data.dim = &albumart_slots[i].dim;
-        /* we can only decode jpeg for embedded AA */
+#ifdef HAVE_IO_PRIORITY
+        buf_back_off_storage(true);
+#endif
+        /* We can only decode jpeg for embedded AA */
        if (track_id3->embed_albumart && track_id3->albumart.type == AA_TYPE_JPG)
        {
-            user_data.embedded_albumart = &(track_id3->albumart);
+            user_data.embedded_albumart = &track_id3->albumart;
            hid = bufopen(track_id3->path, 0, TYPE_BITMAP, &user_data);
        }
        if (hid < 0 && hid != ERR_BUFFER_FULL)
        {
-            /* no embedded AA or it couldn't be loaded, try other sources */
+            /* No embedded AA or it couldn't be loaded - try other sources */
            char path[MAX_PATH];
            if (find_albumart(track_id3, path, sizeof(path),
-                &(albumart_slots[i].dim)))
+                              &albumart_slots[i].dim))
            {
                user_data.embedded_albumart = NULL;
                hid = bufopen(path, 0, TYPE_BITMAP, &user_data);
            }
        }
+#ifdef HAVE_IO_PRIORITY
+        buf_back_off_storage(false);
+#endif
        if (hid == ERR_BUFFER_FULL)
        {
-            filling = STATE_FULL;
+            logf("buffer is full for now (%s)", __func__);
-            logf("buffer is full for now (get album art)");
+            return false;
        }
-        else if (hid < 0)
+        else
        {
-            logf("Album art loading failed");
+            /* If error other than a full buffer, then mark it "unsupported"
-        }
+               to avoid reloading attempt */
+            if (hid < 0)
+            {
+                logf("Album art loading failed");
+                hid = ERR_UNSUPPORTED_TYPE;
+            }
-        tracks[track_widx].aa_hid[i] = hid;
+            *aa_hid = hid;
+        }
    }
+    return true;
 }
-#endif
+#endif /* HAVE_ALBUMART */
-/* Second part of the track loading: We now have the metadata available, so we
+#ifdef HAVE_CODEC_BUFFERING
-   can load the codec, the album art and finally the audio data.
+/* Load a codec for the file onto the buffer - assumes we're working from the
-   This is called on the audio thread after the buffering thread calls the
+   currently loading track - not called for the current track */
-   buffering_handle_finished_callback callback. */
+static bool audio_buffer_codec(struct track_info *track_info,                             
-static void audio_finish_load_track(void)
+                               struct mp3entry *track_id3)
 {
-    size_t file_offset = 0;
+    /* This will not be the current track -> it cannot be the first and the
-    size_t offset = 0;
+       current track cannot be ahead of buffering -> there is a previous
-    bool start_play = start_play_g;
+       track entry which is either current or ahead of the current */
+    struct track_info *prev_info = track_list_last(-1);
+    struct mp3entry *prev_id3 = bufgetid3(prev_info->id3_hid);
-    track_load_started = false;
+    /* If the previous codec is the same as this one, there is no need to
+       put another copy of it on the file buffer (in other words, only
+       buffer codecs at format transitions) */
+    if (prev_id3)
+    {
+        int codt = get_audio_base_codec_type(track_id3->codectype);
+        int prev_codt = get_audio_base_codec_type(prev_id3->codectype);
-    if (tracks[track_widx].id3_hid < 0) {
+        if (codt == prev_codt)
-        logf("No metadata");
+        {
-        return;
+            logf("Reusing prev. codec: %d", prev_id3->codectype);
+            return true;
+        }
    }
+    /* else just load it (harmless) */
-    struct mp3entry *track_id3;
+    /* Load the codec onto the buffer if possible */
+    const char *codec_fn = get_codec_filename(track_id3->codectype);
+    if (!codec_fn)
+        return false;
-    if (track_widx == track_ridx)
+    char codec_path[MAX_PATH+1]; /* Full path to codec */
-        track_id3 = thistrack_id3;
+    codec_get_full_path(codec_path, codec_fn);
-    else
-        track_id3 = bufgetid3(tracks[track_widx].id3_hid);
+    track_info->codec_hid = bufopen(codec_path, 0, TYPE_CODEC, NULL);
-    if (track_id3->length == 0 && track_id3->filesize == 0)
+    if (track_info->codec_hid >= 0)
    {
-        logf("audio_finish_load_track: invalid metadata");
+        logf("Buffered codec: %d", afmt);
+        return true;
+    }
-        /* Invalid metadata */
+    return false;
-        bufclose(tracks[track_widx].id3_hid);
+}
-        tracks[track_widx].id3_hid = -1;
+#endif /* HAVE_CODEC_BUFFERING */
-        /* Skip invalid entry from playlist. */
+/* Load metadata for the next track (with bufopen). The rest of the track
-        playlist_skip_entry(NULL, last_peek_offset--);
+   loading will be handled by audio_finish_load_track once the metadata has
+   been actually loaded by the buffering thread.
-        /* load next track */
+   Each track is arranged in the buffer as follows:
-        LOGFQUEUE("audio > audio Q_AUDIO_FILL_BUFFER %d", (int)start_play);
+        <id3|[cuesheet|][album art|][codec|]audio>
-        queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, start_play);
-        return;
+   The next will not be loaded until the previous succeeds if the buffer was
-    }
+   full at the time. To put any metadata after audio would make those handles
-    /* Try to load a cuesheet for the track */
+   unmovable.
-    if (curr_cue)
+*/
+static int audio_load_track(void)
+{
+    if (in_progress_id3_hid >= 0)
    {
-        char cuepath[MAX_PATH];
+        /* There must be an info pointer if the in-progress id3 is even there */
-        if (look_for_cuesheet_file(track_id3->path, cuepath))
+        struct track_info *info = track_list_last(0);
+        if (info->id3_hid == in_progress_id3_hid)
        {
-            void *temp;
+            if (filling == STATE_FILLING)
-            tracks[track_widx].cuesheet_hid = 
-                        bufalloc(NULL, sizeof(struct cuesheet), TYPE_CUESHEET);
-            if (tracks[track_widx].cuesheet_hid >= 0)
            {
-                bufgetdata(tracks[track_widx].cuesheet_hid,
+                /* Haven't finished the metadata but the notification is
-                           sizeof(struct cuesheet), &temp);
+                   anticipated to come soon */
-                struct cuesheet *cuesheet = (struct cuesheet*)temp;
+                logf("%s(): in progress ok: %d". __func__, info->id3_hid);
-                if (!parse_cuesheet(cuepath, cuesheet))
+                return LOAD_TRACK_OK;
-                {
+            }
-                    bufclose(tracks[track_widx].cuesheet_hid);
+            else if (filling == STATE_FULL)
-                    track_id3->cuesheet = NULL;
+            {
-                }
+                /* Buffer was full trying to complete the load after the
+                   metadata finished, so attempt to continue - older handles
+                   should have been cleared already */
+                logf("%s(): finishing load: %d". __func__, info->id3_hid);
+                filling = STATE_FILLING;
+                buffer_event_finished_callback(&info->id3_hid);
+                return LOAD_TRACK_OK;
            }
        }
+        /* Some old, stray buffering message */
+        logf("%s(): already in progress: %d". __func__, info->id3_hid);
+        return LOAD_TRACK_ERR_BUSY;
    }
-#ifdef HAVE_ALBUMART
+    filling = STATE_FILLING;
-    audio_load_albumart(track_id3);
-#endif
+    struct track_info *info = track_list_alloc_track();
+    if (info == NULL)
+    {
+        /* List is full so stop buffering tracks - however, attempt to obtain
+           metadata as the unbuffered id3 */
+        logf("No free tracks");
+        filling = STATE_FULL;
+    }
+    playlist_peek_offset++;
+    logf("Buffering track: s%u/c%u/e%u/p%d",
+         track_list.start, track_list.current, track_list.end,
+         playlist_peek_offset);
+    /* Get track name from current playlist read position */
+    int fd = -1;
+    char name_buf[MAX_PATH + 1];
+    const char *trackname;
-    /* Load the codec. */
+    while (1)
-    if (!audio_loadcodec(start_play))
    {
-        if (tracks[track_widx].codec_hid == ERR_BUFFER_FULL)
+        trackname = playlist_peek(playlist_peek_offset, name_buf,
+                                  sizeof (name_buf));
+        if (!trackname)
+            break;
+        /* Test for broken playlists by probing for the files */
+        fd = open(trackname, O_RDONLY);
+        if (fd >= 0)
+            break;
+        logf("Open failed");
+        /* Skip invalid entry from playlist */
+        playlist_skip_entry(NULL, playlist_peek_offset);
+        /* Sync the playlist if it isn't finished */
+        if (playlist_peek(playlist_peek_offset, NULL, 0))
+            playlist_next(0);
+    }
+    if (!trackname)
+    {
+        /* No track - exhausted the playlist entries */
+        logf("End-of-playlist");
+        id3_write_locked(UNBUFFERED_ID3, NULL);
+        if (filling != STATE_FULL)
+            track_list_unalloc_track(); /* Free this entry */
+        playlist_peek_offset--;         /* Maintain at last index */
+        /* We can end up here after the real last track signals its completion
+           and miss the transition to STATE_FINISHED esp. if dropping the last
+           songs of a playlist late in their load (2nd stage) */
+        info = track_list_last(0);
+        if (info && buf_handle_remaining(info->audio_hid) == 0)
+            filling_is_finished();
+        else
+            filling = STATE_END_OF_PLAYLIST;
+        return LOAD_TRACK_ERR_NO_MORE;
+    }
+    /* Successfully opened the file - get track metadata */
+    if (filling == STATE_FULL ||
+        (info->id3_hid = bufopen(trackname, 0, TYPE_ID3, NULL)) < 0)
+    {
+        /* Buffer or track list is full */
+        struct mp3entry *ub_id3;
+        playlist_peek_offset--;
+        /* Load the metadata for the first unbuffered track */
+        ub_id3 = id3_get(UNBUFFERED_ID3);
+        id3_mutex_lock();
+        get_metadata(ub_id3, fd, trackname);
+        id3_mutex_unlock();
+        if (filling != STATE_FULL)
        {
-            /* No space for codec on buffer, not an error */
+            track_list_unalloc_track();
            filling = STATE_FULL;
-            return;
        }
-        /* This is an error condition, either no codec was found, or reading
+        logf("%s: buffer is full for now (%u tracks)", __func__,
-         * the codec file failed part way through, either way, skip the track */
+             track_list_count());
-        /* FIXME: We should not use splashf from audio thread! */
+    }
-        splashf(HZ*2, "No codec for: %s", track_id3->path);
+    else
-        /* Skip invalid entry from playlist. */
+    {
-        playlist_skip_entry(NULL, last_peek_offset);
+        /* Successful load initiation */
-        return;
+        info->filesize = filesize(fd);
+        in_progress_id3_hid = info->id3_hid; /* Remember what's in-progress */
    }
-    track_id3->elapsed = 0;
+    close(fd);
-    offset = track_id3->offset;
+    return LOAD_TRACK_OK;
-    size_t resume_rewind = (global_settings.resume_rewind *
+}
-                            track_id3->bitrate * 1000) / 8;
-    if (offset < resume_rewind)
+/* Second part of the track loading: We now have the metadata available, so we
+   can load the codec, the album art and finally the audio data.
+   This is called on the audio thread after the buffering thread calls the
+   buffering_handle_finished_callback callback. */
+static int audio_finish_load_track(struct track_info *info)
+{
+    int trackstat = LOAD_TRACK_OK;
+    if (info->id3_hid != in_progress_id3_hid)
    {
-        offset = 0;
+        /* We must not be here if not! */
+        logf("%s: wrong track %d/%d", __func__, info->id3_hid,
+             in_progress_id3_hid);
+        return LOAD_TRACK_ERR_BUSY;
    }
-    else
+    /* The current track for decoding (there is always one if the list is
+       populated) */
+    struct track_info *cur_info = track_list_current(0);
+    struct mp3entry *track_id3 = valid_mp3entry(bufgetid3(info->id3_hid));
+    if (!track_id3)
    {
-        offset -= resume_rewind;
+        /* This is an error condition. Track cannot be played without valid
+           metadata; skip the track. */
+        logf("No metadata for: %s", track_id3->path);
+        trackstat = LOAD_TRACK_ERR_FINISH_FAILED;
+        goto audio_finish_load_track_exit;
+    }
+    /* Try to load a cuesheet for the track */
+    if (!audio_load_cuesheet(info, track_id3))
+    {
+        /* No space for cuesheet on buffer, not an error */
+        filling = STATE_FULL;
+        goto audio_finish_load_track_exit;
    }
-    enum data_type type = TYPE_PACKET_AUDIO;
+#ifdef HAVE_ALBUMART
+    /* Try to load album art for the track */
+    if (!audio_load_albumart(info, track_id3))
+    {
+        /* No space for album art on buffer, not an error */
+        filling = STATE_FULL;
+        goto audio_finish_load_track_exit;
+    }
+#endif
-    switch (track_id3->codectype) {
+#ifdef HAVE_CODEC_BUFFERING
-    case AFMT_MPA_L1:
+    /* Try to buffer a codec for the track */
-    case AFMT_MPA_L2:
+    if (info != cur_info && !audio_buffer_codec(info, track_id3))
-    case AFMT_MPA_L3:
+    {
-        if (offset > 0) {
+        if (info->codec_hid == ERR_BUFFER_FULL)
-            file_offset = offset;
+        {
+            /* No space for codec on buffer, not an error */
+            filling = STATE_FULL;
+            logf("buffer is full for now (%s)", __func__);
        }
-        break;
+        else
+        {
-    case AFMT_WAVPACK:
+            /* This is an error condition, either no codec was found, or
-        if (offset > 0) {
+               reading the codec file failed part way through, either way,
-            file_offset = offset;
+               skip the track */
-            track_id3->elapsed = track_id3->length / 2;
+            logf("No codec for: %s", track_id3->path);
+            trackstat = LOAD_TRACK_ERR_FINISH_FAILED;
        }
-        break;
-    case AFMT_NSF:
+        goto audio_finish_load_track_exit;
-    case AFMT_SPC:
-    case AFMT_SID:
-        logf("Loading atomic %d",track_id3->codectype);
-        type = TYPE_ATOMIC_AUDIO;
-        break;
-    
-    default:
-        /* no special treatment needed */
-        break;
    }
+#endif /* HAVE_CODEC_BUFFERING */
-    track_id3->offset = offset;
+    /** Finally, load the audio **/
+    size_t file_offset = 0;
+    track_id3->elapsed = 0;
+    if (track_id3->offset >= info->filesize)
+        track_id3->offset = 0;
+    logf("%s: set offset for %s to %lu\n", __func__,
+         id3->title, (unsigned long)offset);
+    /* Adjust for resume rewind so we know what to buffer - starting the codec
+       calls it again, so we don't save it (and they shouldn't accumulate) */
+    size_t offset = resume_rewind_adjusted_offset(track_id3);
+    enum data_type audiotype = get_audio_base_data_type(track_id3->codectype);
+    if (audiotype == TYPE_ATOMIC_AUDIO)
+        logf("Loading atomic %d", track_id3->codectype);
+    if (format_buffers_with_offset(track_id3->codectype))
+    {
+        /* This format can begin buffering from any point */
+        file_offset = offset;
+    }
    logf("load track: %s", track_id3->path);
    if (file_offset > AUDIO_REBUFFER_GUESS_SIZE)
+    {
+        /* We can buffer later in the file, adjust the hunt-and-peck margin */
        file_offset -= AUDIO_REBUFFER_GUESS_SIZE;
-    else if (track_id3->first_frame_offset)
+    }
-        file_offset = track_id3->first_frame_offset;
    else
-        file_offset = 0;
+    {
+        /* No offset given or it is very minimal - begin at the first frame
+           according to the metadata */
+        file_offset = track_id3->first_frame_offset;
+    }
-    tracks[track_widx].audio_hid = bufopen(track_id3->path, file_offset, type,
+    int hid = bufopen(track_id3->path, file_offset, audiotype, NULL);
-                                            NULL);
-    /* No space left, not an error */
+    if (hid >= 0)
-    if (tracks[track_widx].audio_hid == ERR_BUFFER_FULL)
    {
-        filling = STATE_FULL;
+        info->audio_hid = hid;
-        logf("buffer is full for now (load audio)");
-        return;
+        if (info == cur_info)
+        {
+            /* This is the current track to decode - should be started now */
+            trackstat = LOAD_TRACK_READY;
+        }
    }
-    else if (tracks[track_widx].audio_hid < 0)
+    else
    {
-        /* another error, do not continue either */
+        /* Buffer could be full but not properly so if this is the only
-        logf("Could not add audio data handle");
+           track! */
-        return;
+        if (hid == ERR_BUFFER_FULL && audio_track_count() > 1)
+        {
+            filling = STATE_FULL;
+            logf("Buffer is full for now (%s)", __func__);
+        }
+        else
+        {
+            /* Nothing to play if no audio handle - skip this */
+            logf("Could not add audio data handle");
+            trackstat = LOAD_TRACK_ERR_FINISH_FAILED;
+        }
    }
-    /* All required data is now available for the codec -- unless the
+audio_finish_load_track_exit:
-       autoresume feature is in effect.  In the latter case, the codec
+    if (trackstat < LOAD_TRACK_OK)
-       must wait until after PLAYBACK_EVENT_TRACK_BUFFER, which may
-       generate a resume position.  */
-#ifdef HAVE_TAGCACHE
-    if (!global_settings.autoresume_enable || offset)
-#endif
-        tracks[track_widx].taginfo_ready = true;
-    if (start_play)
    {
-        ci.curpos=file_offset;
+        playlist_skip_entry(NULL, playlist_peek_offset);
-        buf_request_buffer_handle(tracks[track_widx].audio_hid);
+        track_info_close(info);
-    }
+        track_list_unalloc_track();
-    send_event(PLAYBACK_EVENT_TRACK_BUFFER, track_id3);
+        if (playlist_peek(playlist_peek_offset, NULL, 0))
+            playlist_next(0);
-#ifdef HAVE_TAGCACHE
+        playlist_peek_offset--;
-    /* In case the autoresume feature has been enabled, finally all
+    }
-       required data is available for the codec. */
-    if (global_settings.autoresume_enable && !offset)
-        tracks[track_widx].taginfo_ready = true;
-#endif
-    track_widx = (track_widx + 1) & MAX_TRACK_MASK;
-    /* load next track */
+    if (filling != STATE_FULL)
-    LOGFQUEUE("audio > audio Q_AUDIO_FILL_BUFFER");
+    {
-    queue_post(&audio_queue, Q_AUDIO_FILL_BUFFER, 0);
+        /* Load next track - error or not */
+        in_progress_id3_hid = ERR_HANDLE_NOT_FOUND;
+        LOGFQUEUE("audio > audio Q_AUDIO_FILL_BUFFER");
+        audio_queue_post(Q_AUDIO_FILL_BUFFER, 0);
+    }
+    else
+    {
+        /* Full */
+        trackstat = LOAD_TRACK_ERR_FINISH_FULL;
+    }
-    return;
+    return trackstat;
 }
-static void audio_fill_file_buffer(bool start_play, size_t offset)
+/* Start a new track load */
+static int audio_fill_file_buffer(void)
 {
-    trigger_cpu_boost();
+    if (play_status == PLAY_STOPPED)
+        return LOAD_TRACK_ERR_FAILED;
-    /* No need to rebuffer if there are track skips pending,
+    trigger_cpu_boost();
-     * however don't cancel buffering on skipping while filling. */
-    if (ci.new_track != 0 && filling != STATE_FILLING)
-        return;
-    filling = STATE_FILLING;
    /* Must reset the buffer before use if trashed or voice only - voice
       file size shouldn't have changed so we can go straight from
@@ -1511,772 +1852,1848 @@ static void audio_fill_file_buffer(bool start_play, size_t offset)
    logf("Starting buffer fill");
-    if (!start_play)
+    int trackstat = audio_load_track();
-        audio_clear_track_entries();
-    /* Save the current resume position once. */
+    if (trackstat >= LOAD_TRACK_OK)
-    playlist_update_resume_info(audio_current_track());
+    {
+        if (track_list_current(0) == track_list_user_current(0))
+            playlist_next(0);
-    audio_load_track(offset, start_play);
+        if (filling == STATE_FULL && !track_list_user_current(1))
+        {
+            /* There are no user tracks on the buffer after this therefore
+               this is the next track */
+            audio_update_and_announce_next_track(id3_get(UNBUFFERED_ID3));
+        }
+    }
+    return trackstat;
 }
-static void audio_rebuffer(void)
+/* Discard unwanted tracks and start refill from after the specified playlist
+   offset */
+static int audio_reset_and_rebuffer(
+    enum track_clear_action action, int peek_offset)
 {
-    logf("Forcing rebuffer");
+    logf("Forcing rebuffer: 0x%X, %d", flags, peek_offset);
-    clear_track_info(CUR_TI);
+    id3_write_locked(UNBUFFERED_ID3, NULL);
-    /* Reset track pointers */
+    /* Remove unwanted tracks - caller must have ensured codec isn't using
-    track_widx = track_ridx;
+       any */
-    audio_clear_track_entries();
+    track_list_clear(action);
-    /* Reset a possibly interrupted track load */
+    /* Refill at specified position (-1 starts at index offset 0) */
-    track_load_started = false;
+    playlist_peek_offset = peek_offset;
    /* Fill the buffer */
-    last_peek_offset = -1;
+    return audio_fill_file_buffer();
-    ci.curpos = 0;
+}
-    if (!CUR_TI->taginfo_ready)
+/* Handle buffering events
-        memset(thistrack_id3, 0, sizeof(struct mp3entry));
+   (Q_AUDIO_BUFFERING) */
+static void audio_on_buffering(int event)
+{
+    enum track_clear_action action;
+    int peek_offset;
+    if (track_list_empty())
+        return;
+    
+    switch (event)
+    {
+    case BUFFER_EVENT_BUFFER_LOW:
+        if (filling != STATE_FULL && filling != STATE_END_OF_PLAYLIST)
+            return; /* Should be nothing left to fill */
+        /* Clear old tracks and continue buffering where it left off */
+        action = TRACK_LIST_KEEP_NEW;
+        peek_offset = playlist_peek_offset;
+        break;
-    audio_fill_file_buffer(false, 0);
+    case BUFFER_EVENT_REBUFFER:
+        /* Remove all but the currently decoding track and redo buffering
+           after that */
+        action = TRACK_LIST_KEEP_CURRENT;
+        peek_offset = (skip_pending == TRACK_SKIP_AUTO) ? 1 : 0;
+        break;
+    default:
+        return;
+    }
+    switch (skip_pending)
+    {
+    case TRACK_SKIP_NONE:
+    case TRACK_SKIP_AUTO:
+    case TRACK_SKIP_AUTO_NEW_PLAYLIST:
+        audio_reset_and_rebuffer(action, peek_offset);
+        break;
+    case TRACK_SKIP_AUTO_END_PLAYLIST:
+        /* Already finished */
+        break;
+    default:
+        /* Invalid */
+        logf("Buffering call, inv. state: %d", (int)skip_pending);
+    }
 }
-/* Called on request from the codec to get a new track. This is the codec part
+/* Handle starting the next track load
-   of the track transition. */
+   (Q_AUDIO_FILL_BUFFER) */
-static void audio_last_track(bool automatic)
+static void audio_on_fill_buffer(void)
 {
-    if (automatic)
+    audio_handle_track_load_status(audio_fill_file_buffer());
+}
+/* Handle posted load track finish event
+   (Q_AUDIO_FINISH_LOAD_TRACK) */
+static void audio_on_finish_load_track(int id3_hid)
+{
+    struct track_info *info = track_list_last(0);
+    if (!info || !buf_is_handle(id3_hid))
+        return;
+    if (info == track_list_user_current(1))
    {
-        ci.new_track = 0;
+        /* Just loaded the metadata right after the current position */
-        automatic_skip = false;
+        audio_update_and_announce_next_track(bufgetid3(info->id3_hid));
+    }
-        if (filling != STATE_ENDING)
+    if (audio_finish_load_track(info) != LOAD_TRACK_READY)
+        return; /* Not current track */
+    bool is_user_current = info == track_list_user_current(0);
+    if (is_user_current)
+    {
+        /* Copy cuesheet */
+        buf_read_cuesheet(info->cuesheet_hid);
+    }
+    if (audio_start_codec(automatic_skip))
+    {
+        if (is_user_current)
        {
-            /* Monitor remaining PCM before stopping */
+            /* Be sure all tagtree info is synchronized; it will be needed for the
-            filling = STATE_ENDING;
+               track finish event - the sync will happen when finalizing a track
-            pcmbuf_monitor_track_change(true);
+               change otherwise */
-        }
+            bool was_valid = valid_mp3entry(id3_get(PLAYING_ID3));
-        codec_stop();
+            playing_id3_sync(info, -1);
+            if (!was_valid)
+            {
+                /* Playing id3 hadn't been updated yet because no valid track
+                   was yet available - treat like the first track */
+                audio_playlist_track_change();
+            }
+        }
    }
    else
    {
+        audio_handle_track_load_status(LOAD_TRACK_ERR_START_CODEC);
+    }
+}
+/* Called when handles other than metadata handles have finished buffering
+   (Q_AUDIO_HANDLE_FINISHED) */
+static void audio_on_handle_finished(int hid)
+{
+    /* Right now, only audio handles should end up calling this */
+    if (filling == STATE_END_OF_PLAYLIST)
+    {
+        struct track_info *info = track_list_last(0);
+        /* Really we don't know which order the handles will actually complete
+           to zero bytes remaining since another thread is doing it - be sure
+           it's the right one */
+        if (info && info->audio_hid == hid)
+        {
+            /* This was the last track in the playlist and we now have all the
+               data we need */
+            filling_is_finished();
+        }
+    }
+}
+/* Called to make an outstanding track skip the current track and to send the
+   transition events */
+static void audio_finalise_track_change(bool delayed)
+{
+    switch (skip_pending)
+    {
+    case TRACK_SKIP_NONE: /* Manual skip */
+        break;
+    case TRACK_SKIP_AUTO:
+    case TRACK_SKIP_AUTO_NEW_PLAYLIST:
+    {
+        int playlist_delta = skip_pending == TRACK_SKIP_AUTO ? 1 : 0;
+        audio_playlist_track_finish();
+        if (!playlist_peek(playlist_delta, NULL, 0))
+        {
+            /* Track ended up rejected - push things ahead like the codec blew
+               it (because it was never started and now we're here where it
+               should have been decoding the next track by now) - next, a
+               directory change or end of playback will most likely happen */
+            skip_pending = TRACK_SKIP_NONE;
+            audio_handle_track_load_status(LOAD_TRACK_ERR_START_CODEC);
+            return;
+        }
+        if (!playlist_delta)
+            break;
+        playlist_peek_offset -= playlist_delta;
+        if (playlist_next(playlist_delta) >= 0)
+            break;
+            /* What!? Disappear? Hopeless bleak despair */
+        }
+        /* Fallthrough */
+    case TRACK_SKIP_AUTO_END_PLAYLIST:
+    default:            /* Invalid */
+        filling = STATE_ENDED;
        audio_stop_playback();
+        return;
+    }
+    struct track_info *info = track_list_current(0);
+    struct mp3entry *track_id3 = NULL;
+    id3_mutex_lock();
+    /* Update the current cuesheet if any and enabled */
+    if (info)
+    {
+        buf_read_cuesheet(info->cuesheet_hid);
+        track_id3 = bufgetid3(info->id3_hid);
+    }
+    id3_write(PLAYING_ID3, track_id3);
+    if (delayed)
+    {
+        /* Delayed skip where codec is ahead of user's current track */
+        struct mp3entry *ci_id3 = id3_get(CODEC_ID3);
+        struct mp3entry *ply_id3 = id3_get(PLAYING_ID3);
+        ply_id3->elapsed = ci_id3->elapsed;
+        ply_id3->offset = ci_id3->offset;
    }
+    /* The skip is technically over */
+    skip_pending = TRACK_SKIP_NONE;
+    /* Sync the next track information */
+    info = track_list_current(1);
+    id3_write(NEXTTRACK_ID3, info ? bufgetid3(info->id3_hid) :
+                                    id3_get(UNBUFFERED_ID3));
+    id3_mutex_unlock();
+    audio_playlist_track_change();
 }
-static void audio_check_new_track(void)
+/* Actually begin a transition and take care of the codec change - may complete
+   it now or ask pcmbuf for notification depending on the type and what pcmbuf
+   has to say */
+static void audio_begin_track_change(bool auto_skip, int trackstat)
 {
-    int track_count;
+    /* Even if the new track is bad, the old track must be finished off */
-    int old_track_ridx;
+    bool finalised = pcmbuf_start_track_change(auto_skip);
-    int i, idx;
-    bool forward;
-    struct mp3entry *temp;
-    if (ci.new_track == 0)
+    if (finalised)
    {
-        ci.new_track++;
+        /* pcmbuf says that the transition happens now - complete it */
-        automatic_skip = true;
+        audio_finalise_track_change(false);
+        if (play_status == PLAY_STOPPED)
+            return; /* Stopped us */
+    }
+    if (!auto_skip)
+        audio_clear_paused_pcm();
+    if (trackstat >= LOAD_TRACK_OK)
+    {
+        struct track_info *info = track_list_current(0);
+        if (info->audio_hid < 0)
+            return;
+        /* Everything needed for the codec is ready - start it */
+        if (audio_start_codec(auto_skip))
+        {
+            if (finalised)
+                playing_id3_sync(info, -1);
+            return;
+        }
+        trackstat = LOAD_TRACK_ERR_START_CODEC;
    }
-    track_count = audio_track_count();
+    audio_handle_track_load_status(trackstat);
-    old_track_ridx = track_ridx;
+}
+/* Transition to end-of-playlist state and begin wait for PCM to finish */
+static void audio_monitor_end_of_playlist(void)
+{
+    skip_pending = TRACK_SKIP_AUTO_END_PLAYLIST;
+    filling = STATE_ENDING;
+    pcmbuf_monitor_track_change(true);
+}
+/* Codec has completed decoding the track
+   (usually Q_AUDIO_CODEC_COMPLETE) */
+static void audio_on_codec_complete(int status)
+{
+    logf("%s(%d)", __func__, status);
-    /* Now it's good time to send track finish events. */
+    if (play_status == PLAY_STOPPED)
-    send_event(PLAYBACK_EVENT_TRACK_FINISH, thistrack_id3);
+        return;
-    /* swap the mp3entry pointers */
-    temp = thistrack_id3;
-    thistrack_id3 = othertrack_id3;
-    othertrack_id3 = temp;
-    ci.id3 = thistrack_id3;
-    memset(thistrack_id3, 0, sizeof(struct mp3entry));
-    if (dir_skip)
+    /* If it didn't notify us first, don't expect "seek complete" message
+       since the codec can't post it now - do things like it would have
+       done */
+    audio_complete_codec_seek();
+    if (play_status == PLAY_PAUSED || skip_pending != TRACK_SKIP_NONE)
    {
-        dir_skip = false;
+        /* Old-hay on the ip-skay - codec has completed decoding
-        /* regardless of the return value we need to rebuffer.
-           if it fails the old playlist will resume, else the
+           Paused: We're not sounding it, so just remember that it happened
-           next dir will start playing */
+                   and the resume will begin the transition
-        playlist_next_dir(ci.new_track);
-        ci.new_track = 0;
+           Skipping: There was already a skip in progress, remember it and
-        audio_rebuffer();
+                     allow no further progress until the PCM from the previous
-        goto skip_done;
+                     song has finished
+         */
+        codec_skip_pending = true;
+        codec_skip_status = status;
+        return;
    }
-    if (new_playlist)
+    codec_skip_pending = false;
-        ci.new_track = 0;
-    /* If the playlist isn't that big */
+    if (status >= 0)
-    if (automatic_skip)
    {
-        while (!playlist_check(ci.new_track))
+        /* Normal automatic skip */
+        ab_end_of_track_report();
+    }
+    int trackstat = LOAD_TRACK_OK;
+    automatic_skip = true;
+    skip_pending = TRACK_SKIP_AUTO;
+    /* Does this track have an entry allocated? */
+    struct track_info *info = track_list_advance_current(1);
+    if (!info || info->audio_hid < 0)
+    {
+        bool end_of_playlist = false;
+        if (info)
        {
-            if (ci.new_track >= 0)
+            /* Track load is not complete - it might have stopped on a
+               full buffer without reaching the audio handle or we just
+               arrived at it early
+               If this type is atomic and we couldn't get the audio,
+               perhaps it would need to wrap to make the allocation and
+               handles are in the way - to maximize the liklihood it can
+               be allocated, clear all handles to reset the buffer and
+               its indexes to 0 - for packet audio, this should not be an
+               issue and a pointless full reload of all the track's
+               metadata may be avoided */
+            struct mp3entry *track_id3 = bufgetid3(info->id3_hid);
+            if (track_id3 &&
+                get_audio_base_data_type(track_id3->codectype)
+                            == TYPE_PACKET_AUDIO)
            {
-                audio_last_track(true);
+                /* Continue filling after this track */
+                audio_reset_and_rebuffer(TRACK_LIST_KEEP_CURRENT, 1);
+                audio_begin_track_change(true, trackstat);
                return;
            }
-            ci.new_track++;
+            /* else rebuffer at this track; status applies to the track we
+               want */
+        }
+        else if (!playlist_peek(1, NULL, 0))
+        {
+            /* Play sequence is complete - directory change or other playlist
+               resequencing - the playlist must now be advanced in order to
+               continue since a peek ahead to the next track is not possible */
+            skip_pending = TRACK_SKIP_AUTO_NEW_PLAYLIST;
+            end_of_playlist = playlist_next(1) < 0;
+        }
+        if (!end_of_playlist)
+        {
+            trackstat = audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL,
+                            skip_pending == TRACK_SKIP_AUTO ? 0 : -1);
+            if (trackstat == LOAD_TRACK_ERR_NO_MORE)
+            {
+                /* Failed to find anything afterall - do playlist switchover
+                   instead */
+                skip_pending = TRACK_SKIP_AUTO_NEW_PLAYLIST;
+                end_of_playlist = playlist_next(1) < 0;
+            }
+        }
+        if (end_of_playlist)
+        {
+            audio_monitor_end_of_playlist();
+            return;
+        }
+    }
+    audio_begin_track_change(true, trackstat);
+}
+/* Called when codec completes seek operation
+   (usually Q_AUDIO_CODEC_SEEK_COMPLETE) */
+static void audio_on_codec_seek_complete(void)
+{
+    logf("%s()", __func__);
+    audio_complete_codec_seek();
+    codec_go();
+}
+/* Called when PCM track change has completed
+   (Q_AUDIO_TRACK_CHANGED) */
+static void audio_on_track_changed(void)
+{
+    /* Finish whatever is pending so that the WPS is in sync */
+    audio_finalise_track_change(true);
+    if (codec_skip_pending)
+    {
+        /* Codec got ahead completing a short track - complete the
+           codec's skip and begin the next */
+        codec_skip_pending = false;
+        audio_on_codec_complete(codec_skip_status);
+    }
+}
+/* Begin playback from an idle state, transition to a new playlist or
+   invalidate the buffer and resume (if playing).
+   (usually Q_AUDIO_PLAY, Q_AUDIO_REMAKE_AUDIO_BUFFER) */
+static void audio_start_playback(size_t offset, unsigned int flags)
+{
+    enum play_status old_status = play_status;
+    if (flags & AUDIO_START_NEWBUF)
+    {
+        /* Mark the buffer dirty - if not playing, it will be reset next
+           time */
+        if (buffer_state == AUDIOBUF_STATE_INITIALIZED)
+            buffer_state = AUDIOBUF_STATE_VOICED_ONLY;
+    }
+    if (old_status != PLAY_STOPPED)
+    {
+        logf("%s(%lu): skipping", __func__, (unsigned long)offset);
+        halt_decoding_track(true);
+        automatic_skip = false;
+        ff_rw_mode = false;
+        if (flags & AUDIO_START_RESTART)
+        {
+            /* Clear out some stuff to resume the current track where it
+               left off */
+            pcmbuf_play_stop();
+            offset = id3_get(PLAYING_ID3)->offset;
+            track_list_clear(TRACK_LIST_CLEAR_ALL);
+        }
+        else
+        {
+            /* This is more-or-less treated as manual track transition */
+            /* Save resume information for current track */
+            audio_playlist_track_finish();
+            track_list_clear(TRACK_LIST_CLEAR_ALL);
+            /* Indicate manual track change */
+            pcmbuf_start_track_change(false);
+            audio_clear_paused_pcm();
+            wipe_track_metadata(true);
        }
+        /* Set after track finish event in case skip was in progress */
+        skip_pending = TRACK_SKIP_NONE;
+    }
+    else
+    {
+        if (flags & AUDIO_START_RESTART)
+            return; /* Must already be playing */
+        /* Cold playback start from a stopped state */
+        logf("%s(%lu): starting", __func__, offset);
+        /* Set audio parameters */
+#if INPUT_SRC_CAPS != 0
+        audio_set_input_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
+        audio_set_output_source(AUDIO_SRC_PLAYBACK);
+#endif
+#ifndef PLATFORM_HAS_VOLUME_CHANGE
+        sound_set_volume(global_settings.volume);
+#endif
+        /* Update our state */
+        play_status = PLAY_PLAYING;
+    }
+    /* Start fill from beginning of playlist */
+    playlist_peek_offset = -1;
+    buf_set_base_handle(-1);
+    /* Officially playing */
+    queue_reply(&audio_queue, 1);
+    /* Add these now - finish event for the first id3 will most likely be sent
+       immediately */
+    add_event(BUFFER_EVENT_REBUFFER, false, buffer_event_rebuffer_callback);
+    add_event(BUFFER_EVENT_FINISHED, false, buffer_event_finished_callback);
+    if (old_status == PLAY_STOPPED)
+    {
+        /* Send coldstart event */
+        send_event(PLAYBACK_EVENT_START_PLAYBACK, NULL);
    }
-    /* Update the playlist */
+    /* Fill the buffer */
-    last_peek_offset -= ci.new_track;
+    int trackstat = audio_fill_file_buffer();
+    if (trackstat >= LOAD_TRACK_OK)
+    {
+        /* This is the currently playing track - get metadata, stat */
+        playing_id3_sync(track_list_current(0), offset);
-    if (playlist_next(ci.new_track) < 0)
+        if (valid_mp3entry(id3_get(PLAYING_ID3)))
+        {
+            /* Only if actually changing tracks... */
+            if (!(flags & AUDIO_START_RESTART))
+                audio_playlist_track_change();
+        }
+    }
+    else
    {
-        /* End of list */
+        /* Found nothing playable */
-        audio_last_track(automatic_skip);
+        audio_handle_track_load_status(trackstat);
+    }
+}
+/* Stop playback and enter an idle state
+   (usually Q_AUDIO_STOP) */
+static void audio_stop_playback(void)
+{
+    logf("%s()", __func__);
+    if (play_status == PLAY_STOPPED)
+        return;
+    /* Stop the codec and unload it */
+    halt_decoding_track(true);
+    pcmbuf_play_stop();
+    codec_unload();
+    /* Save resume information  - "filling" might have been set to
+       "STATE_ENDED" by caller in order to facilitate end of playlist */
+    audio_playlist_track_finish();
+    skip_pending = TRACK_SKIP_NONE;
+    automatic_skip = false;
+    /* Close all tracks and mark them NULL */
+    remove_event(BUFFER_EVENT_REBUFFER, buffer_event_rebuffer_callback);
+    remove_event(BUFFER_EVENT_FINISHED, buffer_event_finished_callback);
+    remove_event(BUFFER_EVENT_BUFFER_LOW, buffer_event_buffer_low_callback);
+    track_list_clear(TRACK_LIST_CLEAR_ALL);
+    /* Update our state */
+    ff_rw_mode = false;
+    play_status = PLAY_STOPPED;
+    wipe_track_metadata(true);
+    /* Go idle */
+    filling = STATE_IDLE;
+    cancel_cpu_boost();
+}
+/* Pause the playback of the current track
+   (Q_AUDIO_PAUSE) */
+static void audio_on_pause(bool pause)
+{
+    logf("%s(%s)", __func__, pause ? "true" : "false");
+    if (play_status == PLAY_STOPPED || pause == (play_status == PLAY_PAUSED))
        return;
+    if (!ff_rw_mode)
+    {
+        /* Not in ff/rw mode - may set the state (otherwise this could make
+           old data play because seek hasn't completed and cleared it) */
+        pcmbuf_pause(pause);
    }
-    
-    if (new_playlist)
+    play_status = pause ? PLAY_PAUSED : PLAY_PLAYING;
+    if (!pause && codec_skip_pending)
    {
-        ci.new_track = 1;
+        /* Actually do the skip that is due - resets the status flag */
-        new_playlist = false;
+        audio_on_codec_complete(codec_skip_status);
    }
+}
+/* Skip a certain number of tracks forwards or backwards
+   (Q_AUDIO_SKIP) */
+static void audio_on_skip(void)
+{
+    id3_mutex_lock();
+    /* Eat the delta to keep it synced, even if not playing */
+    int toskip = skip_offset;
+    skip_offset = 0;
-    /* Save a pointer to the old track to allow later clearing */
+    logf("%s(): %d", __func__, toskip);
-    prev_ti = CUR_TI;
+    id3_mutex_unlock();
+    if (play_status == PLAY_STOPPED)
+        return;
-    for (i = 0; i < ci.new_track; i++)
+    /* Force codec to abort this track */
+    halt_decoding_track(true);
+    /* Kill the ff/rw halt */
+    ff_rw_mode = false;
+    /* Manual skip */
+    automatic_skip = false;
+    /* If there was an auto skip in progress, there will be residual
+       advancement of the playlist and/or track list so compensation will be
+       required in order to end up in the right spot */
+    int track_list_delta = toskip;
+    int playlist_delta = toskip;
+    if (skip_pending != TRACK_SKIP_NONE)
    {
-        idx = (track_ridx + i) & MAX_TRACK_MASK;
+        if (skip_pending != TRACK_SKIP_AUTO_END_PLAYLIST)
-        struct mp3entry *id3 = bufgetid3(tracks[idx].id3_hid);
+            track_list_delta--;
-        ssize_t offset = buf_handle_offset(tracks[idx].audio_hid);
-        if (!id3 || offset < 0 || (unsigned)offset > id3->first_frame_offset)
+        if (skip_pending == TRACK_SKIP_AUTO_NEW_PLAYLIST)
+            playlist_delta--;
+    }
+    audio_playlist_track_finish();
+    skip_pending = TRACK_SKIP_NONE;
+    /* Update the playlist current track now */
+    while (playlist_next(playlist_delta) < 0)
+    {
+        /* Manual skip out of range (because the playlist wasn't updated
+           yet by us and so the check in audio_skip returned 'ok') - bring
+           back into range */
+        int d = toskip < 0 ? 1 : -1;
+        while (!playlist_check(playlist_delta))
        {
-            /* We don't have all the audio data for that track, so clear it,
+            if (playlist_delta == d)
-               but keep the metadata. */
-            if (tracks[idx].audio_hid >= 0 && bufclose(tracks[idx].audio_hid))
            {
-                tracks[idx].audio_hid = -1;
+                /* Had to move the opposite direction to correct, which is
-                tracks[idx].filesize = 0;
+                   wrong - this is the end */
+                filling = STATE_ENDED;
+                audio_stop_playback();
+                return;
            }
+            playlist_delta += d;
+            track_list_delta += d;
        }
    }
-    /* Move to the new track */
+    /* Adjust things by how much the playlist was manually moved */
-    track_ridx = (track_ridx + ci.new_track) & MAX_TRACK_MASK;
+    playlist_peek_offset -= playlist_delta;
-    buf_set_base_handle(CUR_TI->audio_hid);
-    if (automatic_skip)
+    struct track_info *info = track_list_advance_current(track_list_delta);
-    {
+    int trackstat = LOAD_TRACK_OK;
-        wps_offset = -ci.new_track;
-    }
-    /* If it is not safe to even skip this many track entries */
+    if (!info || info->audio_hid < 0)
-    if (ci.new_track >= track_count || ci.new_track <= track_count - MAX_TRACK)
    {
-        ci.new_track = 0;
+        /* We don't know the next track thus we know we don't have it */
-        audio_rebuffer();
+        trackstat = audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL, -1);
-        goto skip_done;
    }
-    forward = ci.new_track > 0;
+    audio_begin_track_change(false, trackstat);
-    ci.new_track = 0;
+}
+/* Skip to the next/previous directory
+   (Q_AUDIO_DIR_SKIP) */
+static void audio_on_dir_skip(int direction)
+{
+    logf("%s(%d)", __func__, direction);
+    id3_mutex_lock();
+    skip_offset = 0;
+    id3_mutex_unlock();
+    if (play_status == PLAY_STOPPED)
+        return;
+    /* Force codec to abort this track */
+    halt_decoding_track(true);
+    /* Kill the ff/rw halt */
+    ff_rw_mode = false;
+    /* Manual skip */
+    automatic_skip = false;
+    audio_playlist_track_finish();
+    /* Unless automatic and gapless, skips do not pend */
+    skip_pending = TRACK_SKIP_NONE;
+    /* Regardless of the return value we need to rebuffer. If it fails the old
+       playlist will resume, else the next dir will start playing. */
+    playlist_next_dir(direction);
+    wipe_track_metadata(false);
-    /* If the target track is clearly not in memory */
+    int trackstat = audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL, -1);
-    if (CUR_TI->filesize == 0 || !CUR_TI->taginfo_ready)
+    if (trackstat == LOAD_TRACK_ERR_NO_MORE)
    {
-        audio_rebuffer();
+        /* The day the music died - finish-off whatever is playing and call it
-        goto skip_done;
+           quits */
+        audio_monitor_end_of_playlist();
+        return;
    }
-    /* When skipping backwards, it is possible that we've found a track that's
+    audio_begin_track_change(false, trackstat);
-     * buffered, but which is around the track-wrap and therefore not the track
+}
-     * we are looking for */
-    if (!forward)
+/* Enter seek mode in order to start a seek
+   (Q_AUDIO_PRE_FF_REWIND) */
+static void audio_on_pre_ff_rewind(void)
+{
+    logf("%s()", __func__);
+    if (play_status == PLAY_STOPPED || ff_rw_mode)
+        return;
+    ff_rw_mode = true;
+    if (play_status == PLAY_PAUSED)
+        return;
+    pcmbuf_pause(true);
+}
+/* Seek the playback of the current track to the specified time
+   (Q_AUDIO_FF_REWIND) */
+static void audio_on_ff_rewind(long time)
+{
+    logf("%s(%ld)", __func__, time);
+    if (play_status == PLAY_STOPPED)
+        return;
+    enum track_skip_type pending = skip_pending;
+    switch (pending)
+    {
+    case TRACK_SKIP_NONE:              /* The usual case */
+    case TRACK_SKIP_AUTO:              /* Have to back it out (fun!) */
+    case TRACK_SKIP_AUTO_END_PLAYLIST: /* Still have the last codec used */
    {
-        int cur_idx = track_ridx;
+        struct mp3entry *id3 = id3_get(PLAYING_ID3);
-        bool taginfo_ready = true;
+        struct mp3entry *ci_id3 = id3_get(CODEC_ID3);
-        /* We've wrapped the buffer backwards if new > old */
-        bool wrap = track_ridx > old_track_ridx;
+        automatic_skip = false;
+        /* Send event before clobbering the time */
+        /* FIXME: Nasty, but the tagtree expects this so that rewinding and
+           then skipping back to this track resumes properly. Something else
+           should be sent. We're not _really_ finishing the track are we? */
+        if (time == 0)
+            send_event(PLAYBACK_EVENT_TRACK_FINISH, id3);
+        /* Prevent user codec time update - coerce to something that is
+           innocuous concerning lookaheads */
+        if (pending == TRACK_SKIP_NONE)
+            skip_pending = TRACK_SKIP_AUTO_END_PLAYLIST;
+        id3->elapsed = time;
+        queue_reply(&audio_queue, 1);
+        bool haltres = halt_decoding_track(pending == TRACK_SKIP_AUTO);
+        /* Need this set in case ff/rw mode + error but _after_ the codec
+           halt that will reset it */
+        codec_seeking = true;
-        while (1)
+        if (pending == TRACK_SKIP_AUTO)
        {
-            cur_idx = (cur_idx + 1) & MAX_TRACK_MASK;
+            if (!track_list_advance_current(-1))
+            {
+                /* Not in list - must rebuffer at the current playlist index */
+                if (audio_reset_and_rebuffer(TRACK_LIST_CLEAR_ALL, -1)
+                        < LOAD_TRACK_OK)
+                {
+                    /* Codec is stopped */
+                    break;
+                }
+            }
+        }
-            /* if we've advanced past the wrap when cur_idx is zeroed */
+        /* Set after audio_fill_file_buffer to disable playing id3 clobber if
-            if (!cur_idx)
+           rebuffer is needed */
-                wrap = false;
+        skip_pending = TRACK_SKIP_NONE;
+        struct track_info *cur_info = track_list_current(0);
-            /* if we aren't still on the wrap and we've caught the old track */
+        /* Track must complete the loading _now_ since a codec and audio
-            if (!(wrap || cur_idx < old_track_ridx))
+           handle are needed in order to do the seek */
-                break;
+        if (cur_info->audio_hid < 0 &&
+            audio_finish_load_track(cur_info) != LOAD_TRACK_READY)
+        {
+            /* Call above should push any load sequence - no need for
+               halt_decoding_track here if no skip was pending here because
+               there would not be a codec started if no audio handle was yet
+               opened */
+            break;
+        }
-            /* If we hit a track in between without valid tag info, bail */
+        if (pending == TRACK_SKIP_AUTO)
-            if (!tracks[cur_idx].taginfo_ready)
+        {
+            if (!bufreadid3(cur_info->id3_hid, ci_id3) ||
+                !audio_init_codec(cur_info, ci_id3))
            {
-                taginfo_ready = false;
+                /* We should have still been able to get it - skip it and move
+                   onto the next one - like it or not this track is borken */
                break;
            }
+            /* Set the codec API to the correct metadata and track info */
+            ci.audio_hid = cur_info->audio_hid;
+            ci.filesize = cur_info->filesize;
+            buf_set_base_handle(cur_info->audio_hid);
        }
-        if (!taginfo_ready)
+        if (!haltres)
        {
-            audio_rebuffer();
+            /* If codec must be (re)started, reset the offset */
+            ci_id3->offset = 0;
        }
-    }
-skip_done:
+        codec_seek(time);
-    audio_update_trackinfo();
+        return;
-    pcmbuf_start_track_change(automatic_skip);
+        }
-    if (get_codec_base_type(codec_loaded()) ==
+    case TRACK_SKIP_AUTO_NEW_PLAYLIST:
-        get_codec_base_type(thistrack_id3->codectype))
    {
-        /* codec is the same base type */
+        /* We cannot do this because the playlist must be reversed by one
-        logf("New track loaded");
+           and it doesn't always return the same song when going backwards
-        codec_ack_msg(Q_CODEC_REQUEST_COMPLETE, false);
+           across boundaries as forwards (either because of randomization
+           or inconsistency in deciding what the previous track should be),
+           therefore the whole operation would often end up as nonsense -
+           lock out seeking for a couple seconds */
+        /* Sure as heck cancel seek mode too! */
+        audio_ff_rewind_end();
+        return;
+        }
+    default:
+        /* Won't see this */
+        return;
    }
-    else
+    if (play_status == PLAY_STOPPED)
    {
-        /* a codec change is required */
+        /* Playback ended because of an error completing a track load */
-        logf("New codec: %d/%d", thistrack_id3->codectype, codec_loaded());
+        return;
-        codec_ack_msg(Q_CODEC_REQUEST_COMPLETE, true);
-        codec_load(tracks[track_ridx].codec_hid, thistrack_id3->codectype);
-        tracks[track_ridx].codec_hid = -1; /* Codec thread will close it */
    }
+    /* Always fake it as a codec start error which will handle mode
+       cancellations and skip to the next track */
+    audio_handle_track_load_status(LOAD_TRACK_ERR_START_CODEC);
 }
-unsigned long audio_prev_elapsed(void)
+/* Invalidates all but currently playing track
+   (Q_AUDIO_FLUSH) */
+static void audio_on_audio_flush(void)
 {
-    return prev_track_elapsed;
+    logf("%s", __func__);
+    if (track_list_empty())
+        return; /* Nothing to flush out */
+    switch (skip_pending)
+    {
+    case TRACK_SKIP_NONE:
+    case TRACK_SKIP_AUTO_END_PLAYLIST:
+        /* Remove all but the currently playing track from the list and
+           refill after that */
+        track_list_clear(TRACK_LIST_KEEP_CURRENT);
+        playlist_peek_offset = 0;
+        id3_write_locked(UNBUFFERED_ID3, NULL);
+        audio_update_and_announce_next_track(NULL);
+        /* Ignore return since it's about the next track, not this one */        
+        audio_fill_file_buffer();
+        if (skip_pending == TRACK_SKIP_NONE)
+            break;
+        /* There's now a track after this one now - convert to auto skip -
+           no skip should pend right now because multiple flush messages can
+           be fired which would cause a restart in the below cases */
+        skip_pending = TRACK_SKIP_NONE;
+        audio_clear_track_notifications();
+        audio_queue_post(Q_AUDIO_CODEC_COMPLETE, CODEC_OK);
+        break;
+    case TRACK_SKIP_AUTO:
+    case TRACK_SKIP_AUTO_NEW_PLAYLIST:
+        /* Precisely removing what it already decoded for the next track is
+           not possible so a restart is required in order to continue the
+           currently playing track without the now invalid future track
+           playing */
+        audio_start_playback(0, AUDIO_START_RESTART);
+        break;
+    default: /* Nothing else is a state */
+        break;
+    }
 }
-void audio_set_prev_elapsed(unsigned long setting)
+#ifdef AUDIO_HAVE_RECORDING
+/* Load the requested encoder type
+   (Q_AUDIO_LOAD_ENCODER) */
+static void audio_on_load_encoder(int afmt)
 {
-    prev_track_elapsed = setting;
+    bool res = true;
+    if (play_status != PLAY_STOPPED)
+        audio_stop_playback(); /* Can't load both types at once */
+    else
+        codec_unload(); /* Encoder still loaded, stop and unload it */
+    if (afmt != AFMT_UNKNOWN)
+    {
+        res = codec_load(-1, afmt | CODEC_TYPE_ENCODER);
+        if (res)
+            codec_go(); /* These are run immediately */
+    }
+    queue_reply(&audio_queue, res);
 }
+#endif /* AUDIO_HAVE_RECORDING */
-/* Stop the codec and reset the PCM buffer */
+static void audio_thread(void)
-static void audio_stop_codec_flush(void)
 {
-    bool pcm_playing;
+    struct queue_event ev;
-    pcmbuf_pause(true);
+    pcm_postinit();
-    codec_stop();
+    filling = STATE_IDLE;
-    pcm_play_lock();
+    while (1)
+    {
+        switch (filling)
+        {
+        /* Active states */
+        case STATE_FULL:
+        case STATE_END_OF_PLAYLIST:
+            if (buf_get_watermark() == 0)
+            {
+                /* End of buffering for now, let's calculate the watermark,
+                   register for a low buffer event and unboost */
+                audio_update_filebuf_watermark(0);
+                add_event(BUFFER_EVENT_BUFFER_LOW, true,
+                          buffer_event_buffer_low_callback);
+            }
+            /* Fall-through */
+        case STATE_FINISHED:
+            /* All data was buffered */
+            cancel_cpu_boost();
+            /* Fall-through */
+        case STATE_FILLING:
+        case STATE_ENDING:
+            if (audio_pcmbuf_track_change_scan())
+            {
+                /* Transfer notification to audio queue event */
+                ev.id = Q_AUDIO_TRACK_CHANGED;
+                ev.data = 1;
+            }
+            else
+            {
+                /* If doing auto skip, poll pcmbuf track notifications a bit
+                   faster to promply detect the transition */
+                queue_wait_w_tmo(&audio_queue, &ev,
+                                 skip_pending == TRACK_SKIP_NONE ?
+                                    HZ/2 : HZ/10);
+            }
+            break;
-    pcm_playing = pcm_is_playing();
+        /* Idle states */
+        default:
+            queue_wait(&audio_queue, &ev);
-    pcmbuf_play_stop();
+#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
-    queue_clear(&pcmbuf_queue);
+            switch (ev.id)
+            {
+#ifdef AUDIO_HAVE_RECORDING
+            /* Must monitor the encoder message for recording so it can remove
+               it if we process the insertion before it does. It cannot simply
+               be removed from under recording however. */
+            case Q_AUDIO_LOAD_ENCODER:
+                break;
+#endif
+            case SYS_USB_DISCONNECTED:
+                filling = STATE_IDLE;
+                break;
+            default:
+                if (filling == STATE_USB)
+                    continue;
+            }
+#endif /* CONFIG_PLATFORM */
+        }
+        switch (ev.id)
+        {
+        /** Codec and track change messages **/
+        case Q_AUDIO_CODEC_COMPLETE:
+            /* Codec is done processing track and has gone idle */
+            LOGFQUEUE("audio < Q_AUDIO_CODEC_COMPLETE: %ld", (long)ev.data);
+            audio_on_codec_complete(ev.data);
+            break;
+        case Q_AUDIO_CODEC_SEEK_COMPLETE:
+            /* Codec is done seeking */
+            LOGFQUEUE("audio < Q_AUDIO_SEEK_COMPLETE");
+            audio_on_codec_seek_complete();
+            break;
+        case Q_AUDIO_TRACK_CHANGED:
+            /* PCM track change done */
+            LOGFQUEUE("audio < Q_AUDIO_TRACK_CHANGED");
+            audio_on_track_changed();
+            break;
+        /** Control messages **/
+        case Q_AUDIO_PLAY:
+            LOGFQUEUE("audio < Q_AUDIO_PLAY");
+            audio_start_playback(ev.data, 0);
+            break;
+        case Q_AUDIO_STOP:
+            LOGFQUEUE("audio < Q_AUDIO_STOP");
+            audio_stop_playback();
+            if (ev.data != 0)
+                queue_clear(&audio_queue);
+            break;
+        case Q_AUDIO_PAUSE:
+            LOGFQUEUE("audio < Q_AUDIO_PAUSE");
+            audio_on_pause(ev.data);
+            break;
+        case Q_AUDIO_SKIP:
+            LOGFQUEUE("audio < Q_AUDIO_SKIP");
+            audio_on_skip();
+            break;
-    if (pcm_playing)
+        case Q_AUDIO_DIR_SKIP:
-        pcmbuf_pause(paused);
+            LOGFQUEUE("audio < Q_AUDIO_DIR_SKIP");
+            audio_on_dir_skip(ev.data);
+            break;
+        case Q_AUDIO_PRE_FF_REWIND:
+            LOGFQUEUE("audio < Q_AUDIO_PRE_FF_REWIND");
+            audio_on_pre_ff_rewind();
+            break;
+        case Q_AUDIO_FF_REWIND:
+            LOGFQUEUE("audio < Q_AUDIO_FF_REWIND");
+            audio_on_ff_rewind(ev.data);
+            break;
+        case Q_AUDIO_FLUSH:
+            LOGFQUEUE("audio < Q_AUDIO_FLUSH: %d", (int)ev.data);
+            audio_on_audio_flush();
+            break;
-    pcm_play_unlock();
+        /** Buffering messages **/
+        case Q_AUDIO_BUFFERING:
+            /* some buffering event */
+            LOGFQUEUE("audio < Q_AUDIO_BUFFERING: %d", (int)ev.data);
+            audio_on_buffering(ev.data);
+            break;
+        case Q_AUDIO_FILL_BUFFER:
+            /* continue buffering next track */
+            LOGFQUEUE("audio < Q_AUDIO_FILL_BUFFER");
+            audio_on_fill_buffer();
+            break;
+        case Q_AUDIO_FINISH_LOAD_TRACK:
+            /* metadata is buffered */
+            LOGFQUEUE("audio < Q_AUDIO_FINISH_LOAD_TRACK");
+            audio_on_finish_load_track(ev.data);
+            break;
+        case Q_AUDIO_HANDLE_FINISHED:
+            /* some other type is buffered */
+            LOGFQUEUE("audio < Q_AUDIO_HANDLE_FINISHED");
+            audio_on_handle_finished(ev.data); 
+            break;
+        /** Miscellaneous messages **/
+        case Q_AUDIO_REMAKE_AUDIO_BUFFER:
+            /* buffer needs to be reinitialized */
+            LOGFQUEUE("audio < Q_AUDIO_REMAKE_AUDIO_BUFFER");
+            audio_start_playback(0, AUDIO_START_RESTART | AUDIO_START_NEWBUF);
+            break;
+#ifdef HAVE_DISK_STORAGE
+        case Q_AUDIO_UPDATE_WATERMARK:
+            /* buffering watermark needs updating */
+            LOGFQUEUE("audio < Q_AUDIO_UPDATE_WATERMARK: %d", (int)ev.data);
+            audio_update_filebuf_watermark(ev.data);
+            break;
+#endif /* HAVE_DISK_STORAGE */
+#ifdef AUDIO_HAVE_RECORDING
+        case Q_AUDIO_LOAD_ENCODER:
+            /* load an encoder for recording */
+            LOGFQUEUE("audio < Q_AUDIO_LOAD_ENCODER: %d", (int)ev.data);
+            audio_on_load_encoder(ev.data);
+            break;
+#endif /* AUDIO_HAVE_RECORDING */
+#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
+        case SYS_USB_CONNECTED:
+            LOGFQUEUE("audio < SYS_USB_CONNECTED");
+            audio_stop_playback();
+#ifdef PLAYBACK_VOICE
+            voice_stop();
+#endif
+            filling = STATE_USB;
+            usb_acknowledge(SYS_USB_CONNECTED_ACK);
+            break;
+#endif /* (CONFIG_PLATFORM & PLATFORM_NATIVE) */
+        case SYS_TIMEOUT:
+            LOGFQUEUE_SYS_TIMEOUT("audio < SYS_TIMEOUT");
+            break;
+        default:
+            /* LOGFQUEUE("audio < default : %08lX", ev.id); */
+            break;
+        } /* end switch */
+    } /* end while */
 }
-static void audio_stop_playback(void)
+/* --- Buffering callbacks --- */
+/* Called when fullness is below the watermark level */
+static void buffer_event_buffer_low_callback(void *data)
 {
-    if (playing)
+    logf("low buffer callback");
-    {
+    LOGFQUEUE("buffering > audio Q_AUDIO_BUFFERING: buffer low");
-        /* If we were playing, save resume information */
+    audio_queue_post(Q_AUDIO_BUFFERING, BUFFER_EVENT_BUFFER_LOW);
-        struct mp3entry *id3 = NULL;
+    (void)data;
+}
-        if (!ci.stop_codec)
+/* Called when handles must be discarded in order to buffer new data */
-            id3 = audio_current_track();
+static void buffer_event_rebuffer_callback(void *data)
+{
+    logf("rebuffer callback");
+    LOGFQUEUE("buffering > audio Q_AUDIO_BUFFERING: rebuffer");
+    audio_queue_post(Q_AUDIO_BUFFERING, BUFFER_EVENT_REBUFFER);
+    (void)data;
+}
-        /* Save the current playing spot, or NULL if the playlist has ended */
+/* A handle has completed buffering and all required data is available */
-        playlist_update_resume_info(id3);
+static void buffer_event_finished_callback(void *data)
+{
+    int hid = *(const int *)data;
+    const enum data_type htype = buf_handle_data_type(hid);
-        /* Now it's good time to send track finish events.  Do this
+    logf("handle %d finished buffering (type:%u)", hid, (unsigned)htype);
-           only if this hasn't been done already as part of a track
-           switch. */
-        if (id3 == thistrack_id3)
-            send_event(PLAYBACK_EVENT_TRACK_FINISH, thistrack_id3);
-        /* TODO: Create auto bookmark too? */
+    /* Limit queue traffic */
+    switch (htype)
+    {
+    case TYPE_ID3:
+        /* The metadata handle for the last loaded track has been buffered.
+           We can ask the audio thread to load the rest of the track's data. */
+        LOGFQUEUE("buffering > audio Q_AUDIO_FINISH_LOAD_TRACK: %d", hid);
+        audio_queue_post(Q_AUDIO_FINISH_LOAD_TRACK, hid);
+        break;
-        prev_track_elapsed = othertrack_id3->elapsed;
+    case TYPE_PACKET_AUDIO:
+        /* Strip any useless trailing tags that are left. */
+        strip_tags(hid);
+        /* Fall-through */
+    case TYPE_ATOMIC_AUDIO:
+        LOGFQUEUE("buffering > audio Q_AUDIO_HANDLE_FINISHED: %d", hid);
+        audio_queue_post(Q_AUDIO_HANDLE_FINISHED, hid);
+        break;
-        remove_event(BUFFER_EVENT_BUFFER_LOW, buffering_low_buffer_callback);
+    default:
+        /* Don't care to know about these */
+        break;
    }
+}
-    audio_stop_codec_flush();
-    paused = false;
-    playing = false;
-    track_load_started = false;
-    filling = STATE_IDLE;
+/** -- Codec callbacks -- **/
+/* Update elapsed times with latency-adjusted values */
+void audio_codec_update_elapsed(unsigned long value)
+{
+#ifdef AB_REPEAT_ENABLE
+    ab_position_report(value);
+#endif
+    unsigned long latency = pcmbuf_get_latency();
+    if (LIKELY(value >= latency))
+    {
+        unsigned long elapsed = value - latency;
+        if (elapsed > value || elapsed < value - 2)
+            value = elapsed;
+    }
+    else
+    {
+        value = 0;
+    }
-    /* Mark all entries null. */
+    /* Track codec: used later when updating the playing at the user
-    audio_clear_track_entries();
+       transition */
+    id3_get(CODEC_ID3)->elapsed = value;
-    /* Close all tracks */
+    /* If a skip is pending, the PCM buffer is updating the time on the
-    audio_release_tracks();
+       previous song */
+    if (LIKELY(skip_pending == TRACK_SKIP_NONE))
+        id3_get(PLAYING_ID3)->elapsed = value;
 }
-static void audio_play_start(size_t offset)
+/* Update offsets with latency-adjusted values */
+void audio_codec_update_offset(size_t value)
 {
-    int i;
+    struct mp3entry *ci_id3 = id3_get(CODEC_ID3);
+    unsigned long latency = pcmbuf_get_latency() * ci_id3->bitrate / 8;
-    send_event(PLAYBACK_EVENT_START_PLAYBACK, NULL);
+    if (LIKELY(value >= latency))
-#if INPUT_SRC_CAPS != 0
+    {
-    audio_set_input_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
+        value -= latency;
-    audio_set_output_source(AUDIO_SRC_PLAYBACK);
+    }
-#endif
+    else
+    {
+        value = 0;
+    }
-    paused = false;
+    /* Track codec: used later when updating the playing id3 at the user
-    audio_stop_codec_flush();
+       transition */
+    ci_id3->offset = value;
-    playing = true;
+    /* If a skip is pending, the PCM buffer is updating the time on the
-    track_load_started = false;
+       previous song */
+    if (LIKELY(skip_pending == TRACK_SKIP_NONE))
+        id3_get(PLAYING_ID3)->offset = value;
+}
-    ci.new_track = 0;
-    ci.seek_time = 0;
-    wps_offset = 0;
-#ifndef PLATFORM_HAS_VOLUME_CHANGE
+/** --- Pcmbuf callbacks --- **/
-    sound_set_volume(global_settings.volume);
-#endif
-    track_widx = track_ridx = 0;
-    buf_set_base_handle(-1);
-    /* Clear all track entries. */
+/* Between the codec and PCM track change, we need to keep updating the
-    for (i = 0; i < MAX_TRACK; i++) {
+ * "elapsed" value of the previous (to the codec, but current to the
-        clear_track_info(&tracks[i]);
+ * user/PCM/WPS) track, so that the progressbar reaches the end. */
+void audio_pcmbuf_position_callback(unsigned int time)
+{
+    struct mp3entry *id3 = id3_get(PLAYING_ID3);
+    time += id3->elapsed;
+    id3->elapsed = MIN(time, id3->length);
+}
+/* Post message from pcmbuf that the end of the previous track has just
+ * been played */
+void audio_pcmbuf_track_change(bool pcmbuf)
+{
+    if (pcmbuf)
+    {
+        /* Notify of the change in special-purpose semaphore object */
+        LOGFQUEUE("pcmbuf > pcmbuf Q_AUDIO_TRACK_CHANGED");
+        audio_pcmbuf_track_change_post();
    }
+    else
+    {
+        /* Safe to post directly to the queue */
+        LOGFQUEUE("pcmbuf > audio Q_AUDIO_TRACK_CHANGED");
+        audio_queue_post(Q_AUDIO_TRACK_CHANGED, 0);
+    }
+}
-    last_peek_offset = -1;
+/* May pcmbuf start PCM playback when the buffer is full enough? */
+bool audio_pcmbuf_may_play(void)
+{
+    return play_status == PLAY_PLAYING && !ff_rw_mode;
+}
-    /* Officially playing */
-    queue_reply(&audio_queue, 1);
-    audio_fill_file_buffer(true, offset);
+/** -- External interfaces -- **/
-    add_event(BUFFER_EVENT_BUFFER_LOW, false, buffering_low_buffer_callback);
+/* Return the playback and recording status */
+int audio_status(void)
+{
+    unsigned int ret = play_status;
+#ifdef AUDIO_HAVE_RECORDING
+    /* Do this here for constitency with mpeg.c version */
+    ret |= pcm_rec_status();
+#endif
-    LOGFQUEUE("audio > audio Q_AUDIO_TRACK_CHANGED");
+    return (int)ret;
-    queue_post(&audio_queue, Q_AUDIO_TRACK_CHANGED, 0);
 }
+/* Clear all accumulated audio errors for playback and recording */
+void audio_error_clear(void)
+{
+#ifdef AUDIO_HAVE_RECORDING
+    pcm_rec_error_clear();
+#endif
+}
-/* Invalidates all but currently playing track. */
+/* Get a copy of the id3 data for the for current track + offset + skip delta */
-static void audio_invalidate_tracks(void)
+bool audio_peek_track(struct mp3entry *id3, int offset)
 {
-    if (audio_have_tracks())
+    bool retval = false;
+    id3_mutex_lock();
+    if (play_status != PLAY_STOPPED)
    {
-        last_peek_offset = 0;
+        id3->path[0] = '\0'; /* Null path means it should be filled now */
-        track_widx = track_ridx;
+        retval = audio_get_track_metadata(offset + skip_offset, id3) &&
+                id3->path[0] != '\0';
+    }
-        /* Mark all other entries null (also buffered wrong metadata). */
+    id3_mutex_unlock();
-        audio_clear_track_entries();
-        track_widx = (track_widx + 1) & MAX_TRACK_MASK;
+    return retval;
+}
+/* Return the mp3entry for the currently playing track */
+struct mp3entry * audio_current_track(void)
+{
+    struct mp3entry *id3;
-        audio_fill_file_buffer(false, 0);
+    id3_mutex_lock();
-        send_event(PLAYBACK_EVENT_TRACK_CHANGE, thistrack_id3);
+#ifdef AUDIO_FAST_SKIP_PREVIEW
+    if (skip_offset != 0)
+    {
+        /* This is a peekahead */
+        id3 = id3_get(PLAYING_PEEK_ID3);
+        audio_peek_track(id3, 0);
    }
+    else
+#endif
+    {
+        /* Normal case */
+        id3 = id3_get(PLAYING_ID3);
+        audio_get_track_metadata(0, id3);
+    }
+    id3_mutex_unlock();
+    return id3;
 }
-static void audio_new_playlist(void)
+/* Obtains the mp3entry for the next track from the current */
+struct mp3entry * audio_next_track(void)
 {
-    /* Prepare to start a new fill from the beginning of the playlist */
+    struct mp3entry *id3 = id3_get(NEXTTRACK_ID3);
-    last_peek_offset = -1;
-    /* Signal the codec to initiate a track change forward */
+    id3_mutex_lock();
-    new_playlist = true;
-    ci.new_track = 1;
-    if (audio_have_tracks())
+#ifdef AUDIO_FAST_SKIP_PREVIEW
+    if (skip_offset != 0)
+    {
+        /* This is a peekahead */
+        if (!audio_peek_track(id3, 1))
+            id3 = NULL;
+    }
+    else
+#endif
    {
-        if (paused)
+        /* Normal case */
-            skipped_during_pause = true;
+        if (!audio_get_track_metadata(1, id3))
-        track_widx = track_ridx;
+            id3 = NULL;
-        audio_clear_track_entries();
+    }
-        track_widx = (track_widx + 1) & MAX_TRACK_MASK;
+    id3_mutex_unlock();
-        /* Mark the current track as invalid to prevent skipping back to it */
+    return id3;
-        CUR_TI->taginfo_ready = false;
+}
-    }
-    /* Officially playing */
+/* Start playback at the specified offset */
-    queue_reply(&audio_queue, 1);
+void audio_play(long offset)
+{
+    logf("audio_play");
-    audio_fill_file_buffer(false, 0);
+#ifdef PLAYBACK_VOICE
+    /* Truncate any existing voice output so we don't have spelling
+     * etc. over the first part of the played track */
+    talk_force_shutup();
+#endif
+    LOGFQUEUE("audio >| audio Q_AUDIO_PLAY: %ld", offset);
+    audio_queue_send(Q_AUDIO_PLAY, offset);
 }
-/* Called on manual track skip */
+/* Stop playback if playing */
-static void audio_initiate_track_change(long direction)
+void audio_stop(void)
 {
-    logf("audio_initiate_track_change(%ld)", direction);
+    LOGFQUEUE("audio >| audio Q_AUDIO_STOP");
+    audio_queue_send(Q_AUDIO_STOP, 0);
+}
-    ci.new_track += direction;
+/* Pause playback if playing */
-    wps_offset -= direction;
+void audio_pause(void)
-    if (paused)
+{
-        skipped_during_pause = true;
+    LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE");
+    audio_queue_send(Q_AUDIO_PAUSE, true);
+}
+/* This sends a stop message and the audio thread will dump all its
+   subsequent messages */
+void audio_hard_stop(void)
+{
+    /* Stop playback */
+    LOGFQUEUE("audio >| audio Q_AUDIO_STOP: 1");
+    audio_queue_send(Q_AUDIO_STOP, 1);
+#ifdef PLAYBACK_VOICE
+    voice_stop();
+#endif
 }
-/* Called on manual dir skip */
+/* Resume playback if paused */
-static void audio_initiate_dir_change(long direction)
+void audio_resume(void)
 {
-    dir_skip = true;
+    LOGFQUEUE("audio >| audio Q_AUDIO_PAUSE resume");
-    ci.new_track = direction;
+    audio_queue_send(Q_AUDIO_PAUSE, false);
-    if (paused)
-        skipped_during_pause = true;
 }
-/* Called when PCM track change is complete */
+/* Skip the specified number of tracks forward or backward from the current */
-static void audio_finalise_track_change(void)
+void audio_skip(int offset)
 {
-    logf("audio_finalise_track_change");
+    id3_mutex_lock();
+    /* If offset has to be backed-out to stay in range, no skip is done */
+    int accum = skip_offset + offset;
-    if (automatic_skip)
+    while (offset != 0 && !playlist_check(accum))
    {
-        wps_offset = 0;
+        offset += offset < 0 ? 1 : -1;
-        automatic_skip = false;
+        accum = skip_offset + offset;
+    }
-        /* Invalidate prevtrack_id3 */
+    if (offset != 0)
-        memset(othertrack_id3, 0, sizeof(struct mp3entry));
+    {
+        /* Accumulate net manual skip count since the audio thread last
+           processed one */
+        skip_offset = accum;
-        if (prev_ti && prev_ti->audio_hid < 0)
+        if (global_settings.beep)
-        {
+            pcmbuf_beep(2000, 100, 2500*global_settings.beep);
-            /* No audio left so we clear all the track info. */
-            clear_track_info(prev_ti);
+        LOGFQUEUE("audio > audio Q_AUDIO_SKIP %d", offset);
-        }
+#ifdef AUDIO_FAST_SKIP_PREVIEW
+        /* Do this before posting so that the audio thread can correct us
+           when things settle down - additionally, if audio gets a message
+           and the delta is zero, the Q_AUDIO_SKIP handler (audio_on_skip)
+           handler a skip event with the correct info but doesn't skip */
+        send_event(PLAYBACK_EVENT_TRACK_SKIP, NULL);
+#endif /* AUDIO_FAST_SKIP_PREVIEW */
+        /* Playback only needs the final state even if more than one is
+           processed because it wasn't removed in time */
+        queue_remove_from_head(&audio_queue, Q_AUDIO_SKIP);
+        audio_queue_post(Q_AUDIO_SKIP, 0);
    }
-    send_event(PLAYBACK_EVENT_TRACK_CHANGE, thistrack_id3);
+    else
-    playlist_update_resume_info(audio_current_track());
+    {
+        /* No more tracks */
+        if (global_settings.beep)
+            pcmbuf_beep(1000, 100, 1500*global_settings.beep);
+    }
+    id3_mutex_unlock();
 }
-static void audio_seek_complete(void)
+/* Skip one track forward from the current */
+void audio_next(void)
 {
-    logf("audio_seek_complete");
+    audio_skip(1);
+}
-    if (!playing)
+/* Skip one track backward from the current */
-        return;
+void audio_prev(void)
+{
+    audio_skip(-1);
+}
-    /* If seeking-while-playing, pcm_is_paused() is true.
+/* Move one directory forward */
-     * If seeking-while-paused, audio_status PAUSE is true.
+void audio_next_dir(void)
-     * A seamless seek skips this section. */
+{
-    ci.seek_time = 0;
+    LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP 1");
+    audio_queue_post(Q_AUDIO_DIR_SKIP, 1);
+}
-    pcm_play_lock();
+/* Move one directory backward */
+void audio_prev_dir(void)
+{
+    LOGFQUEUE("audio > audio Q_AUDIO_DIR_SKIP -1");
+    audio_queue_post(Q_AUDIO_DIR_SKIP, -1);
+}
-    if (pcm_is_paused() || paused)
+/* Pause playback in order to start a seek that flushes the old audio */
-    {
+void audio_pre_ff_rewind(void)
-        /* Clear the buffer */
+{
-        pcmbuf_play_stop();
+    LOGFQUEUE("audio > audio Q_AUDIO_PRE_FF_REWIND");
+    audio_queue_post(Q_AUDIO_PRE_FF_REWIND, 0);
+}
-        /* If seeking-while-playing, resume PCM playback */
+/* Seek to the new time in the current track */
-        if (!paused)
+void audio_ff_rewind(long time)
-            pcmbuf_pause(false);
+{
-    }
+    LOGFQUEUE("audio > audio Q_AUDIO_FF_REWIND");
+    audio_queue_post(Q_AUDIO_FF_REWIND, time);
+}
-    pcm_play_unlock();
+/* Clear all but the currently playing track then rebuffer */
+void audio_flush_and_reload_tracks(void)
+{
+    LOGFQUEUE("audio > audio Q_AUDIO_FLUSH");
+    audio_queue_post(Q_AUDIO_FLUSH, 0);
 }
-static void audio_codec_status_message(long reason, int status)
+/* Return the pointer to the main audio buffer, optionally preserving
+   voicing */
+unsigned char * audio_get_buffer(bool talk_buf, size_t *buffer_size)
 {
-    /* TODO: Push the errors up to the normal UI somewhere */
+    unsigned char *buf, *end;
-    switch (reason)
+    if (audio_is_initialized)
    {
-    case Q_CODEC_LOAD_DISK:
+        audio_hard_stop();
-    case Q_CODEC_LOAD:
+    }
-        if (!playing)
+    /* else buffer_state will be AUDIOBUF_STATE_TRASHED at this point */
-            return;
-        if (status < 0)
+    if (buffer_size == NULL)
+    {
+        /* Special case for talk_init to use since it already knows it's
+           trashed */
+        buffer_state = AUDIOBUF_STATE_TRASHED;
+        return NULL;
+    }
+    if (talk_buf || buffer_state == AUDIOBUF_STATE_TRASHED
+           || !talk_voice_required())
+    {
+        logf("get buffer: talk, audio");
+        /* Ok to use everything from audiobuf to audiobufend - voice is loaded,
+           the talk buffer is not needed because voice isn't being used, or
+           could be AUDIOBUF_STATE_TRASHED already. If state is
+           AUDIOBUF_STATE_VOICED_ONLY, no problem as long as memory isn't written
+           without the caller knowing what's going on. Changing certain settings
+           may move it to a worse condition but the memory in use by something
+           else will remain undisturbed.
+         */
+        if (buffer_state != AUDIOBUF_STATE_TRASHED)
        {
-            splash(HZ*2, "Codec failure");
+            talk_buffer_steal();
-            audio_check_new_track();
+            buffer_state = AUDIOBUF_STATE_TRASHED;
        }
-        break;
-#ifdef AUDIO_HAVE_RECORDING
+        buf = audiobuf;
-    case Q_ENCODER_LOAD_DISK:
+        end = audiobufend;
-        if (status < 0)
-            splash(HZ*2, "Encoder failure");
-        break;
-#endif /* AUDIO_HAVE_RECORDING */
    }
+    else
+    {
+        /* Safe to just return this if already AUDIOBUF_STATE_VOICED_ONLY or
+           still AUDIOBUF_STATE_INITIALIZED */
+        /* Skip talk buffer and move pcm buffer to end to maximize available
+           contiguous memory - no audio running means voice will not need the
+           swap space */
+        logf("get buffer: audio");
+        buf = audiobuf + talk_get_bufsize();
+        end = audiobufend - pcmbuf_init(audiobufend);
+        buffer_state = AUDIOBUF_STATE_VOICED_ONLY;
+    }
+    *buffer_size = end - buf;
+    return buf;
 }
-/*
+#ifdef HAVE_RECORDING
- * Layout audio buffer as follows - iram buffer depends on target:
+/* Stop audio, voice and obtain all available buffer space */
- * [|SWAP:iram][|TALK]|FILE|GUARD|PCM|[SWAP:dram[|iram]|]
+unsigned char * audio_get_recording_buffer(size_t *buffer_size)
- */
-static void audio_reset_buffer(void)
 {
-    /* see audio_get_recording_buffer if this is modified */
+    audio_hard_stop();
-    logf("audio_reset_buffer");
+    talk_buffer_steal();
-    /* If the setup of anything allocated before the file buffer is
+    unsigned char *end = audiobufend;
-       changed, do check the adjustments after the buffer_alloc call
+    buffer_state = AUDIOBUF_STATE_TRASHED;
-       as it will likely be affected and need sliding over */
+    *buffer_size = end - audiobuf;
-    /* Initially set up file buffer as all space available */
+    return (unsigned char *)audiobuf;
-    filebuf = audiobuf + talk_get_bufsize();
+}
-    filebuflen = audiobufend - filebuf;
+#endif /* HAVE_RECORDING */
-    ALIGN_BUFFER(filebuf, filebuflen, sizeof (intptr_t));
+/* Restore audio buffer to a particular state (one more valid than the current
+   state) */
+bool audio_restore_playback(int type)
+{
+    switch (type)
+    {
+    case AUDIO_WANT_PLAYBACK:
+        if (buffer_state != AUDIOBUF_STATE_INITIALIZED)
+            audio_reset_buffer();
+        return true;
+    case AUDIO_WANT_VOICE:
+        if (buffer_state == AUDIOBUF_STATE_TRASHED)
+            audio_reset_buffer();
+        return true;
+    default:
+        return false;
+    }
+}
-    /* Subtract whatever the pcm buffer says it used plus the guard buffer */
+/* Has the playback buffer been completely claimed? */
-    size_t pcmbuf_size = pcmbuf_init(filebuf + filebuflen) + GUARD_BUFSIZE;
+bool audio_buffer_state_trashed(void)
+{
+    return buffer_state == AUDIOBUF_STATE_TRASHED;
+}
-    /* Make sure filebuflen is a pointer sized multiple after adjustment */
-    pcmbuf_size = ALIGN_UP(pcmbuf_size, sizeof (intptr_t));
-    if(pcmbuf_size > filebuflen)
+/** --- Miscellaneous public interfaces --- **/
-        panicf("%s(): EOM (%zu > %zu)", __func__, pcmbuf_size, filebuflen);
-    filebuflen -= pcmbuf_size;
+#ifdef HAVE_ALBUMART
-    buffering_reset(filebuf, filebuflen);
+/* Return which album art handle is current for the user in the given slot */
+int playback_current_aa_hid(int slot)
+{
+    if ((unsigned)slot < MAX_MULTIPLE_AA)
+    {
+        struct track_info *info = track_list_user_current(skip_offset);
-    /* Clear any references to the file buffer */
+        if (!info && abs(skip_offset) <= 1)
-    buffer_state = AUDIOBUF_STATE_INITIALIZED;
+        {
+            /* Give the actual position a go */
+            info = track_list_user_current(0);
+        }
-#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
+        if (info)
-    /* Make sure everything adds up - yes, some info is a bit redundant but
+            return info->aa_hid[slot];
-       aids viewing and the sumation of certain variables should add up to
-       the location of others. */
-    {
-        size_t pcmbufsize;
-        const unsigned char *pcmbuf = pcmbuf_get_meminfo(&pcmbufsize);
-        logf("fbuf:   %08X", (unsigned)filebuf);
-        logf("fbufe:  %08X", (unsigned)(filebuf + filebuflen));
-        logf("gbuf:   %08X", (unsigned)(filebuf + filebuflen));
-        logf("gbufe:  %08X", (unsigned)(filebuf + filebuflen + GUARD_BUFSIZE));
-        logf("pcmb:   %08X", (unsigned)pcmbuf);
-        logf("pcmbe:  %08X", (unsigned)(pcmbuf + pcmbufsize));
    }
-#endif
+    return ERR_HANDLE_NOT_FOUND;
 }
-static void audio_thread(void)
+/* Find an album art slot that doesn't match the dimensions of another that
+   is already claimed - increment the use count if it is */
+int playback_claim_aa_slot(struct dim *dim)
 {
-    struct queue_event ev;
+    int i;
-    pcm_postinit();
-    audio_thread_ready = true;
-    while (1)
+    /* First try to find a slot already having the size to reuse it since we
+       don't want albumart of the same size buffered multiple times */
+    FOREACH_ALBUMART(i)
    {
-        switch (filling) {
+        struct albumart_slot *slot = &albumart_slots[i];
-            case STATE_IDLE:
-                queue_wait(&audio_queue, &ev);
-                break;
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
+        if (slot->dim.width == dim->width &&
-            case STATE_USB:
+            slot->dim.height == dim->height)
-                queue_wait(&audio_queue, &ev);
+        {
-                switch (ev.id) {
+            slot->used++;
-#ifdef AUDIO_HAVE_RECORDING
+            return i;
-                    /* Must monitor the encoder message for recording so it can
+        }
-                       remove it if we process the insertion before it does. It
+    }
-                       cannot simply be removed from under recording however. */
-                    case Q_AUDIO_LOAD_ENCODER:
-                        break;
-#endif
-                    case SYS_USB_DISCONNECTED:
-                        filling = STATE_IDLE;
-                    default:
-                        continue;
-                }
-                break;
-#endif /* CONFIG_PLATFORM */
-            default:
+    /* Size is new, find a free slot */
-                /* End of buffering, let's calculate the watermark and
+    FOREACH_ALBUMART(i)
-                   unboost */
+    {
-                set_filebuf_watermark();
+        if (!albumart_slots[i].used)
-                cancel_cpu_boost();
+        {
-                /* Fall-through */
+            albumart_slots[i].used++;
-            case STATE_FILLING:
+            albumart_slots[i].dim = *dim;
-            case STATE_ENDING:
+            return i;
-                if (!pcmbuf_queue_scan(&ev))
-                    queue_wait_w_tmo(&audio_queue, &ev, HZ/2);
-                break;
        }
+    }
-        switch (ev.id) {
+    /* Sorry, no free slot */
+    return -1;
+}
-            case Q_AUDIO_FILL_BUFFER:
+/* Invalidate the albumart_slot - decrement the use count if > 0 */
-                LOGFQUEUE("audio < Q_AUDIO_FILL_BUFFER %d", (int)ev.data);
+void playback_release_aa_slot(int slot)
-                audio_fill_file_buffer((bool)ev.data, 0);
+{
-                break;
+    if ((unsigned)slot < MAX_MULTIPLE_AA)
+    {
+        struct albumart_slot *aa_slot = &albumart_slots[slot];
-            case Q_AUDIO_FINISH_LOAD:
+        if (aa_slot->used > 0)
-                LOGFQUEUE("audio < Q_AUDIO_FINISH_LOAD");
+            aa_slot->used--;
-                audio_finish_load_track();
+    }
-                buf_set_base_handle(CUR_TI->audio_hid);
+}
-                break;
+#endif /* HAVE_ALBUMART */
-            case Q_AUDIO_PLAY:
-                LOGFQUEUE("audio < Q_AUDIO_PLAY");
-                if (playing && ev.data <= 0)
-                    audio_new_playlist();
-                else
-                {
-                    audio_stop_playback();
-                    audio_play_start((size_t)ev.data);
-                }
-                break;
-            case Q_AUDIO_STOP:
+#ifdef HAVE_RECORDING
-                LOGFQUEUE("audio < Q_AUDIO_STOP");
+/* Load an encoder and run it */
-                if (playing)
+bool audio_load_encoder(int afmt)
-                    audio_stop_playback();
+{
-                if (ev.data != 0)
+#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
-                    queue_clear(&audio_queue);
+    LOGFQUEUE("audio >| Q_AUDIO_LOAD_ENCODER: %d", afmt);
-                break;
+    return audio_queue_send(Q_AUDIO_LOAD_ENCODER, afmt) != 0;
+#else
+    (void)afmt;
+    return true;
+#endif
+}
-            case Q_AUDIO_PAUSE:
+/* Stop an encoder and unload it */
-                LOGFQUEUE("audio < Q_AUDIO_PAUSE");
+void audio_remove_encoder(void)
-                if (!(bool) ev.data && skipped_during_pause
+{
-#ifdef HAVE_CROSSFADE
+#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
-                    && !pcmbuf_is_crossfade_active()
+    LOGFQUEUE("audio >| Q_AUDIO_LOAD_ENCODER: NULL");
+    audio_queue_send(Q_AUDIO_LOAD_ENCODER, AFMT_UNKNOWN);
 #endif
-                    )
+}
-                    pcmbuf_play_stop(); /* Flush old track on resume after skip */
+#endif /* HAVE_RECORDING */
-                skipped_during_pause = false;
-                if (!playing)
-                    break;
-                pcmbuf_pause((bool)ev.data);
-                paused = (bool)ev.data;
-                break;
-            case Q_AUDIO_SKIP:
+/* Is an automatic skip in progress? If called outside transistion callbacks,
-                LOGFQUEUE("audio < Q_AUDIO_SKIP");
+   indicates the last skip type at the time it was processed and isn't very
-                audio_initiate_track_change((long)ev.data);
+   meaningful. */
-                break;
+bool audio_automatic_skip(void)
+{
+    return automatic_skip;
+}
-            case Q_AUDIO_PRE_FF_REWIND:
+/* Would normally calculate byte offset from an elapsed time but is not
-                LOGFQUEUE("audio < Q_AUDIO_PRE_FF_REWIND");
+   used on SWCODEC */
-                if (!playing)
+int audio_get_file_pos(void)
-                    break;
+{
-                pcmbuf_pause(true);
+    return 0;
-                break;
+}
-            case Q_AUDIO_FF_REWIND:
+/* Return the elasped time of the track previous to the current */
-                LOGFQUEUE("audio < Q_AUDIO_FF_REWIND");
+unsigned long audio_prev_elapsed(void)
-                if (!playing)
+{
-                    break;
+    return prev_track_elapsed;
+}
-                if (filling == STATE_ENDING)
+/* Is the audio thread ready to accept commands? */
-                {
+bool audio_is_thread_ready(void)
-                    /* Temp workaround: There is no codec available */
+{
-                    if (!paused)
+    return filling != STATE_BOOT;
-                        pcmbuf_pause(false);
+}
-                    break;
-                }
-                if ((long)ev.data == 0)
+/* Return total file buffer length after accounting for the talk buf */
-                {
+size_t audio_get_filebuflen(void)
-                    /* About to restart the track - send track finish
+{
-                       events if not already done. */
+    return buf_length();
-                    if (thistrack_id3 == audio_current_track())
+}
-                        send_event(PLAYBACK_EVENT_TRACK_FINISH, thistrack_id3);
-                }
-                if (automatic_skip)
+/* How many tracks exist on the buffer - full or partial */
-                {
+int audio_track_count(void)
-                    /* An automatic track skip is in progress. Finalize it,
+    __attribute__((alias("track_list_count")));
-                       then go back to the previous track */
-                    audio_finalise_track_change();
-                    ci.new_track = -1;
-                }
-                ci.seek_time = (long)ev.data+1;
-                break;
-            case Q_AUDIO_CHECK_NEW_TRACK:
+/* Return total ringbuffer space occupied - ridx to widx */
-                LOGFQUEUE("audio < Q_AUDIO_CHECK_NEW_TRACK");
+long audio_filebufused(void)
-                audio_check_new_track();
+{
-                break;
+    return buf_used();
+}
-            case Q_AUDIO_DIR_SKIP:
-                LOGFQUEUE("audio < Q_AUDIO_DIR_SKIP");
-                audio_initiate_dir_change(ev.data);
-                break;
-            case Q_AUDIO_FLUSH:
+/** -- Settings -- **/
-                LOGFQUEUE("audio < Q_AUDIO_FLUSH");
-                audio_invalidate_tracks();
-                break;
-            case Q_AUDIO_TRACK_CHANGED:
+/* Enable or disable cuesheet support and allocate/don't allocate the
-                /* PCM track change done */
+   extra associated resources */
-                LOGFQUEUE("audio < Q_AUDIO_TRACK_CHANGED");
+void audio_set_cuesheet(int enable)
-                /* Set new playlist position for resuming. */
+{
-                playlist_update_resume_index();
+    if (play_status == PLAY_STOPPED || !enable != !get_current_cuesheet())
-                if (filling != STATE_ENDING)
+    {
-                    audio_finalise_track_change();
+        LOGFQUEUE("audio >| audio Q_AUDIO_REMAKE_AUDIO_BUFFER");
-                else if (playing)
+        audio_queue_send(Q_AUDIO_REMAKE_AUDIO_BUFFER, 0);
-                    audio_stop_playback();
+    }
-                break;
+}
-            case Q_AUDIO_SEEK_COMPLETE:
+#ifdef HAVE_DISK_STORAGE
-                /* Codec seek done */
+/* Set the audio antiskip buffer margin by index */
-                LOGFQUEUE("audio < Q_AUDIO_SEEK_COMPLETE");
+void audio_set_buffer_margin(int setting)
-                audio_seek_complete();
+{
-                codec_ack_msg(Q_AUDIO_SEEK_COMPLETE, false);
+    static const unsigned short lookup[] =
-                break;
+        { 5, 15, 30, 60, 120, 180, 300, 600 };
-            case Q_CODEC_LOAD:
+    if ((unsigned)setting >= ARRAYLEN(lookup))
-            case Q_CODEC_LOAD_DISK:
+        setting = 0;
-#ifdef AUDIO_HAVE_RECORDING
-            case Q_ENCODER_LOAD_DISK:
-#endif
-                /* These are received when a codec has finished normally or
-                   upon a codec error */
-                audio_codec_status_message(ev.id, ev.data);
-                break;
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
+    logf("buffer margin: %u", (unsigned)lookup[setting]);
-            case SYS_USB_CONNECTED:
-                LOGFQUEUE("audio < SYS_USB_CONNECTED");
-                if (playing)
-                    audio_stop_playback();
-#ifdef PLAYBACK_VOICE
-                voice_stop();
-#endif
-                filling = STATE_USB;
-                usb_acknowledge(SYS_USB_CONNECTED_ACK);
-                break;
-#endif
-#ifdef AUDIO_HAVE_RECORDING
+    LOGFQUEUE("audio > audio Q_AUDIO_UPDATE_WATERMARK: %u",
-            case Q_AUDIO_LOAD_ENCODER:
+              (unsigned)lookup[setting]);
-                if (playing)
+    audio_queue_post(Q_AUDIO_UPDATE_WATERMARK, lookup[setting]);
-                    audio_stop_playback();
+}
-                else
+#endif /* HAVE_DISK_STORAGE */
-                    codec_stop(); /* If encoder still loaded, stop it */
-                if (ev.data == AFMT_UNKNOWN)
+#ifdef HAVE_CROSSFADE
-                    break;
+/* Take necessary steps to enable or disable the crossfade setting */
+void audio_set_crossfade(int enable)
+{
+    /* Tell it the next setting to use */
+    pcmbuf_request_crossfade_enable(enable);
-                queue_reply(&audio_queue,
+    /* Return if size hasn't changed or this is too early to determine
-                            codec_load(-1, ev.data | CODEC_TYPE_ENCODER));
+       which in the second case there's no way we could be playing
-                break;
+       anything at all */
-#endif /* AUDIO_HAVE_RECORDING */
+    if (!pcmbuf_is_same_size())
+    {
+        LOGFQUEUE("audio >| audio Q_AUDIO_REMAKE_AUDIO_BUFFER");
+        audio_queue_send(Q_AUDIO_REMAKE_AUDIO_BUFFER, 0);
+    }
+}
+#endif /* HAVE_CROSSFADE */
-            case SYS_TIMEOUT:
-                LOGFQUEUE_SYS_TIMEOUT("audio < SYS_TIMEOUT");
-                break;
-            default:
+/** -- Startup -- **/
-                /* LOGFQUEUE("audio < default : %08lX", ev.id); */
-                break;
-        } /* end switch */
-    } /* end while */
-}
-/* Initialize the audio system - called from init() in main.c.
+/* Initialize the audio system - called from init() in main.c */
- * Last function because of all the references to internal symbols
- */
 void audio_init(void)
 {
-    unsigned int audio_thread_id;
    /* Can never do this twice */
    if (audio_is_initialized)
    {
@@ -2290,31 +3707,20 @@ void audio_init(void)
       to send messages. Thread creation will be delayed however so nothing
       starts running until ready if something yields such as talk_init. */
    queue_init(&audio_queue, true);
-    queue_init(&pcmbuf_queue, false);
+    mutex_init(&id3_mutex);
    pcm_init();
    codec_init_codec_api();
    
-    thistrack_id3 = &mp3entry_buf[0];
-    othertrack_id3 = &mp3entry_buf[1];
-    
-    /* cuesheet support */
-    if (global_settings.cuesheet)
-        curr_cue = (struct cuesheet*)buffer_alloc(sizeof(struct cuesheet));
-    
-    /* initialize the buffer */
-    filebuf = audiobuf;
-    /* audio_reset_buffer must to know the size of voice buffer so init
-       talk first */
-    talk_init();
    make_codec_thread();
+    /* This thread does buffer, so match its priority */
    audio_thread_id = create_thread(audio_thread, audio_stack,
                  sizeof(audio_stack), CREATE_THREAD_FROZEN,
-                  audio_thread_name IF_PRIO(, PRIORITY_USER_INTERFACE)
+                  audio_thread_name
+                  IF_PRIO(, MIN(PRIORITY_BUFFERING, PRIORITY_USER_INTERFACE))
                  IF_COP(, CPU));
    queue_enable_queue_send(&audio_queue, &audio_queue_sender_list,
@@ -2324,39 +3730,21 @@ void audio_init(void)
    voice_thread_init();
 #endif
+    /* audio_reset_buffer must to know the size of voice buffer so init
+       talk first */
+    talk_init();
 #ifdef HAVE_CROSSFADE
    /* Set crossfade setting for next buffer init which should be about... */
    pcmbuf_request_crossfade_enable(global_settings.crossfade);
 #endif
-    /* initialize the buffering system */
+    /* Initialize the buffering system */
+    track_list_init();
    buffering_init();
    /* ...now! Set up the buffers */
    audio_reset_buffer();
-    int i;
-    for(i = 0; i < MAX_TRACK; i++)
-    {
-        tracks[i].audio_hid = -1;
-        tracks[i].id3_hid = -1;
-        tracks[i].codec_hid = -1;
-        tracks[i].cuesheet_hid = -1;
-    }
-#ifdef HAVE_ALBUMART
-    FOREACH_ALBUMART(i)
-    {
-        int j;
-        for (j = 0; j < MAX_TRACK; j++)
-        {
-            tracks[j].aa_hid[i] = -1;
-        }
-    }
-#endif
-    add_event(BUFFER_EVENT_REBUFFER, false, buffering_handle_rebuffer_callback);
-    add_event(BUFFER_EVENT_FINISHED, false, buffering_handle_finished_callback);
    /* Probably safe to say */
    audio_is_initialized = true;
@@ -2365,26 +3753,10 @@ void audio_init(void)
    audio_set_buffer_margin(global_settings.buffer_margin);
 #endif
-    /* it's safe to let the threads run now */
+    /* It's safe to let the threads run now */
 #ifdef PLAYBACK_VOICE
    voice_thread_resume();
 #endif
    codec_thread_resume();
    thread_thaw(audio_thread_id);
-} /* audio_init */
-bool audio_is_thread_ready(void)
-{
-    return audio_thread_ready;
-}
-size_t audio_get_filebuflen(void)
-{
-    return filebuflen;
-}
-int get_audio_hid()
-{
-    return CUR_TI->audio_hid;
 }
diff --git a/apps/playback.h b/apps/playback.h
index 76c394603f..225946cfaf 100644
--- a/apps/playback.h
+++ b/apps/playback.h
@@ -26,6 +26,16 @@
 #include <stdlib.h>
 #include "config.h"
+#if CONFIG_CODEC == SWCODEC
+/* Including the code for fast previews is entirely optional since it
+   does add two more mp3entry's - for certain targets it may be less
+   beneficial such as flash-only storage */
+#if MEMORYSIZE > 2
+#define AUDIO_FAST_SKIP_PREVIEW
+#endif
+#endif /* CONFIG_CODEC == SWCODEC */
 #ifdef HAVE_ALBUMART
 #include "bmp.h"
@@ -67,6 +77,8 @@ long audio_filebufused(void);
 void audio_pre_ff_rewind(void);
 void audio_skip(int direction);
 void audio_hard_stop(void); /* Stops audio from serving playback */
+void audio_set_cuesheet(int enable);
 #ifdef HAVE_CROSSFADE
 void audio_set_crossfade(int enable);
 #endif
@@ -78,11 +90,10 @@ enum
 };
 bool audio_restore_playback(int type); /* Restores the audio buffer to handle the requested playback */
 size_t audio_get_filebuflen(void);
-void audio_pcmbuf_position_callback(unsigned int time) ICODE_ATTR;
-void audio_post_track_change(bool pcmbuf);
-int get_audio_hid(void);
-void audio_set_prev_elapsed(unsigned long setting);
 bool audio_buffer_state_trashed(void);
+/* Automatic transition? Only valid to call during the track change events,
+   otherwise the result is undefined. */
 bool audio_automatic_skip(void);
 /* Define one constant that includes recording related functionality */
@@ -91,35 +102,62 @@ bool audio_automatic_skip(void);
 #endif
 enum {
-    Q_NULL = 0,
+    Q_NULL = 0,                 /* reserved */
+    /* -> audio */
    Q_AUDIO_PLAY = 1,
    Q_AUDIO_STOP,
    Q_AUDIO_PAUSE,
    Q_AUDIO_SKIP,
    Q_AUDIO_PRE_FF_REWIND,
    Q_AUDIO_FF_REWIND,
-    Q_AUDIO_CHECK_NEW_TRACK,
    Q_AUDIO_FLUSH,
-    Q_AUDIO_TRACK_CHANGED,
-    Q_AUDIO_SEEK_COMPLETE,
    Q_AUDIO_DIR_SKIP,
-    Q_AUDIO_POSTINIT,
-    Q_AUDIO_FILL_BUFFER,
-    Q_AUDIO_FINISH_LOAD,
-    Q_CODEC_REQUEST_COMPLETE,
-    Q_CODEC_REQUEST_FAILED,
+    /* pcmbuf -> audio */
+    Q_AUDIO_TRACK_CHANGED,
+    /* audio -> audio */
+    Q_AUDIO_FILL_BUFFER,        /* continue buffering next track */
+    /* buffering -> audio */
+    Q_AUDIO_BUFFERING,          /* some buffer event */
+    Q_AUDIO_FINISH_LOAD_TRACK,  /* metadata is buffered */
+    Q_AUDIO_HANDLE_FINISHED,    /* some other type is buffered */
+    /* codec -> audio (*) */
+    Q_AUDIO_CODEC_SEEK_COMPLETE,
+    Q_AUDIO_CODEC_COMPLETE,
+    /* audio -> codec */
    Q_CODEC_LOAD,
-    Q_CODEC_LOAD_DISK,
+    Q_CODEC_RUN,
+    Q_CODEC_PAUSE,
+    Q_CODEC_SEEK,
+    Q_CODEC_STOP,
+    Q_CODEC_UNLOAD,
+    /*- miscellanous -*/
 #ifdef AUDIO_HAVE_RECORDING
-    Q_AUDIO_LOAD_ENCODER,
+    /* -> codec */
-    Q_ENCODER_LOAD_DISK,
+    Q_AUDIO_LOAD_ENCODER,       /* load an encoder for recording */
-    Q_ENCODER_RECORD,
 #endif
+    /* -> codec */
    Q_CODEC_DO_CALLBACK,
-    Q_CODEC_ACK,
-};
+    /*- settings -*/
+#ifdef HAVE_DISK_STORAGE
+    /* -> audio */
+    Q_AUDIO_UPDATE_WATERMARK,   /* buffering watermark needs updating */
 #endif
+    /* -> audio */
+    Q_AUDIO_REMAKE_AUDIO_BUFFER, /* buffer needs to be reinitialized */
+};
+/* (*) If you change these, you must check audio_clear_track_notifications
+       in playback.c for correctness */
+#endif /* _PLAYBACK_H */
diff --git a/apps/playlist.c b/apps/playlist.c
index 14ebb7a198..d17bf230a5 100644
--- a/apps/playlist.c
+++ b/apps/playlist.c
@@ -822,9 +822,6 @@ static int add_track_to_playlist(struct playlist_info* playlist,
    playlist->amount++;
    playlist->num_inserted_tracks++;
    
-    /* Update index for resume. */
-    playlist_update_resume_index();
    return insert_position;
 }
@@ -925,9 +922,6 @@ static int remove_track_from_playlist(struct playlist_info* playlist,
        sync_control(playlist, false);
    }
    
-    /* Update index for resume. */
-    playlist_update_resume_index();
    return 0;
 }
@@ -987,9 +981,6 @@ static int randomise_playlist(struct playlist_info* playlist,
            playlist->first_index, NULL, NULL, NULL);
    }
    
-    /* Update index for resume. */
-    playlist_update_resume_index();
    return 0;
 }
@@ -1030,9 +1021,6 @@ static int sort_playlist(struct playlist_info* playlist, bool start_current,
            playlist->first_index, -1, NULL, NULL, NULL);
    }
    
-    /* Update index for resume. */
-    playlist_update_resume_index();
    return 0;
 }
@@ -1205,9 +1193,6 @@ static void find_and_set_playlist_index(struct playlist_info* playlist,
            break;
        }
    }
-    
-    /* Update index for resume. */
-    playlist_update_resume_index();
 }
 /*
@@ -2486,6 +2471,12 @@ const char* playlist_peek(int steps, char* buf, size_t buf_size)
    if (index < 0)
        return NULL;
+#if CONFIG_CODEC == SWCODEC
+    /* Just testing - don't care about the file name */
+    if (!buf || !buf_size)
+        return "";
+#endif
    control_file = playlist->indices[index] & PLAYLIST_INSERT_TYPE_MASK;
    seek = playlist->indices[index] & PLAYLIST_SEEK_MASK;
@@ -2632,30 +2623,17 @@ int playlist_get_resume_info(int *resume_index)
    return 0;
 }
-/* Get current playlist index. */
-int playlist_get_index(void)
-{
-    return current_playlist.index;
-}
-/* Update resume index within playlist_info structure. */
-void playlist_update_resume_index(void)
-{
-    struct playlist_info* playlist = &current_playlist;
-    playlist->resume_index = playlist->index;
-}
 /* Update resume info for current playing song.  Returns -1 on error. */
 int playlist_update_resume_info(const struct mp3entry* id3)
 {
    struct playlist_info* playlist = &current_playlist;
-    
    if (id3)
    {
-        if (global_status.resume_index  != playlist->resume_index ||
+        if (global_status.resume_index  != playlist->index ||
            global_status.resume_offset != id3->offset)
        {
-            global_status.resume_index  = playlist->resume_index;
+            global_status.resume_index  = playlist->index;
            global_status.resume_offset = id3->offset;
            status_save();
        }
@@ -3203,9 +3181,6 @@ int playlist_move(struct playlist_info* playlist, int index, int new_index)
    queue_post(&playlist_queue, PLAYLIST_LOAD_POINTERS, 0);
 #endif
-    /* Update index for resume. */
-    playlist_update_resume_index();
    return result;
 }
diff --git a/apps/playlist.h b/apps/playlist.h
index a0e3b579f7..9c45769981 100644
--- a/apps/playlist.h
+++ b/apps/playlist.h
@@ -90,7 +90,6 @@ struct playlist_info
    int  buffer_end_pos; /* last position where buffer was written  */
    int  index;          /* index of current playing track          */
    int  first_index;    /* index of first song in playlist         */
-    int  resume_index;   /* index of playing track to resume        */
    int  amount;         /* number of tracks in the index           */
    int  last_insert_pos; /* last position we inserted a track      */
    int  seed;           /* shuffle seed                            */
@@ -132,7 +131,6 @@ const char *playlist_peek(int steps, char* buf, size_t buf_size);
 int playlist_next(int steps);
 bool playlist_next_dir(int direction);
 int playlist_get_resume_info(int *resume_index);
-int playlist_get_index(void);
 int playlist_update_resume_info(const struct mp3entry* id3);
 int playlist_get_display_index(void);
 int playlist_amount(void);
@@ -176,6 +174,5 @@ int playlist_directory_tracksearch(const char* dirname, bool recurse,
                                   int (*callback)(char*, void*),
                                   void* context);
 int playlist_remove_all_tracks(struct playlist_info *playlist);
-void playlist_update_resume_index(void);
 #endif /* __PLAYLIST_H__ */
diff --git a/apps/plugin.c b/apps/plugin.c
index ea290c89a7..bb326d937b 100644
--- a/apps/plugin.c
+++ b/apps/plugin.c
@@ -692,7 +692,7 @@ static const struct plugin_api rockbox_api = {
 #if CONFIG_CODEC == SWCODEC
    codec_thread_do_callback,
    codec_load_file,
-    codec_begin,
+    codec_run_proc,
    codec_close,
    get_codec_filename,
    find_array_ptr,
diff --git a/apps/plugin.h b/apps/plugin.h
index 4537c6670b..cdf34e28b1 100644
--- a/apps/plugin.h
+++ b/apps/plugin.h
@@ -145,12 +145,12 @@ void* plugin_get_buffer(size_t *buffer_size);
 #define PLUGIN_MAGIC 0x526F634B /* RocK */
 /* increase this every time the api struct changes */
-#define PLUGIN_API_VERSION 203
+#define PLUGIN_API_VERSION 204
 /* update this to latest version if a change to the api struct breaks
   backwards compatibility (and please take the opportunity to sort in any
   new function which are "waiting" at the end of the function table) */
-#define PLUGIN_MIN_API_VERSION 203
+#define PLUGIN_MIN_API_VERSION 204
 /* plugin return codes */
 /* internal returns start at 0x100 to make exit(1..255) work */
@@ -799,9 +799,9 @@ struct plugin_api {
 #if CONFIG_CODEC == SWCODEC
    void (*codec_thread_do_callback)(void (*fn)(void),
                                     unsigned int *audio_thread_id);
-    void * (*codec_load_file)(const char* codec, struct codec_api *api);
+    int (*codec_load_file)(const char* codec, struct codec_api *api);
-    int (*codec_begin)(void *handle);
+    int (*codec_run_proc)(void);
-    void (*codec_close)(void *handle);
+    int (*codec_close)(void);
    const char *(*get_codec_filename)(int cod_spec);
    void ** (*find_array_ptr)(void **arr, void *ptr);
    int (*remove_array_ptr)(void **arr, void *ptr);
diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c
index 855503a0ec..4bde1ba39d 100644
--- a/apps/plugins/test_codec.c
+++ b/apps/plugins/test_codec.c
@@ -127,7 +127,6 @@ struct test_track_info {
 };
 static struct test_track_info track;
-static bool taginfo_ready = true;
 static bool use_dsp;
@@ -433,6 +432,7 @@ static void pcmbuf_insert_wav_checksum(const void *ch1, const void *ch2, int cou
 static void set_elapsed(unsigned long value)
 {
    elapsed = value;
+    ci.id3->elapsed = value;
 }
@@ -482,6 +482,7 @@ static void* request_buffer(size_t *realsize, size_t reqsize)
 static void advance_buffer(size_t amount)
 {
    ci.curpos += amount;
+    ci.id3->offset = ci.curpos;
 }
@@ -499,20 +500,17 @@ static void seek_complete(void)
    /* Do nothing */
 }
-/* Request file change from file buffer. Returns true is next
+/* Codec calls this to know what it should do next. */
-   track is available and changed. If return value is false,
+static enum codec_command_action get_command(intptr_t *param)
-   codec should exit immediately with PLUGIN_OK status. */
-static bool request_next_track(void)
 {
-    /* We are only decoding a single track */
+    rb->yield();
-    return false;
+    return CODEC_ACTION_NULL; /* just continue processing */
+    (void)param;
 }
 static void set_offset(size_t value)
 {
-    /* ??? */
+    ci.id3->offset = value;
-    (void)value;
 }
@@ -546,6 +544,9 @@ static void init_ci(void)
 {
    /* --- Our "fake" implementations of the codec API functions. --- */
+    ci.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
+                                                    CODEC_IDX_AUDIO);
    ci.codec_get_buffer = codec_get_buffer;
    if (wavinfo.fd >= 0 || checksum) {
@@ -560,11 +561,9 @@ static void init_ci(void)
    ci.advance_buffer = advance_buffer;
    ci.seek_buffer = seek_buffer;
    ci.seek_complete = seek_complete;
-    ci.request_next_track = request_next_track;
    ci.set_offset = set_offset;
    ci.configure = configure;
-    ci.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
+    ci.get_command = get_command;
-                                                    CODEC_IDX_AUDIO);
    /* --- "Core" functions --- */
@@ -620,20 +619,22 @@ static void init_ci(void)
 static void codec_thread(void)
 {
    const char* codecname;
-    void *handle;
+    int res;
-    int res = CODEC_ERROR;
    codecname = rb->get_codec_filename(track.id3.codectype);
-    /* Load the codec and start decoding. */
+    /* Load the codec */
-    handle = rb->codec_load_file(codecname,&ci);
+    res = rb->codec_load_file(codecname, &ci);
-    if (handle != NULL)
+    if (res >= 0)
    {
-        res = rb->codec_begin(handle);
+        /* Decode the file */
-        rb->codec_close(handle);
+        res = rb->codec_run_proc();
    }
+    /* Clean up */
+    rb->codec_close();
    /* Signal to the main thread that we are done */
    endtick = *rb->current_tick - rebuffertick;
    codec_playing = false;
@@ -705,11 +706,7 @@ static enum plugin_status test_track(const char* filename)
    /* Prepare the codec struct for playing the whole file */
    ci.filesize = track.filesize;
    ci.id3 = &track.id3;
-    ci.taginfo_ready = &taginfo_ready;
    ci.curpos = 0;
-    ci.stop_codec = false;
-    ci.new_track = 0;
-    ci.seek_time = 0;
    if (use_dsp)
        rb->dsp_configure(ci.dsp, DSP_RESET, 0);