Now all threads need to ack the connection like on real target, dircache is unloaded and playback stops accordingly.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31009 a1c6a512-1295-4272-9138-f99709370657
* shrinking now considers freespace just before the alloc-to-be-shrinked,
that means less (or sometimes none at all) is taken from the audio buffer.
* core_available() now searches for the best free space, instead of simply the end,
i.e. it will not return 0 if the audio buffer is allocated and there's free space
before it. It also runs a compaction to ensure maximum contiguous memory.
audio_buffer_available() is also enhanced. It now considers the 256K reserve buffer,
and returns free buflib space instead if the audio buffer is short.
This all fixes the root problem of FS#12344 (Sansa Clip+: PANIC occurred when
dircache is enabled), that alloced from the audio buffer, even if it was very
short and buflib had many more available as free space before it.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31006 a1c6a512-1295-4272-9138-f99709370657
Very frequent start-stop cycles (as caused by frequent core_alloc() calls)
of audio makes the codecs lose the resume position, and this causes playback
from the beginning.
To work around, use queue_post() instead of queue_send() to delay the resume
so that it only resumes once per core_alloc() set.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30900 a1c6a512-1295-4272-9138-f99709370657
audio playback. If it goes below 256K new buflib allocations fail.
This prevents buffer underruns as the new buffer size wasn't actually
checked at all.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30893 a1c6a512-1295-4272-9138-f99709370657
The buflib metadata gets corrupted at the new loation between core_shrink()
and actually applying, the new buffer boundaries (most probably due to yield()).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30574 a1c6a512-1295-4272-9138-f99709370657
Reuse playback's Q_AUDIO_REMAKE_AUDIO_BUFFER capabilities to set the new
playback buffer, instead of stopping/restarting manual. This strongly
reduces the visibility of the short audio stop.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30480 a1c6a512-1295-4272-9138-f99709370657
This enables the ability to allocate (and free) memory dynamically
without fragmentation, through compaction. This means allocations can move
and fragmentation be reduced. Most changes are preparing Rockbox for this,
which many times means adding a move callback which can temporarily disable
movement when the corresponding code is in a critical section.
For now, the audio buffer allocation has a central role, because it's the one
having allocated most. This buffer is able to shrink itself, for which it
needs to stop playback for a very short moment. For this,
audio_buffer_available() returns the size of the audio buffer which can
possibly be used by other allocations because the audio buffer can shrink.
lastfm scrobbling and timestretch can now be toggled at runtime without
requiring a reboot.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30381 a1c6a512-1295-4272-9138-f99709370657
The buflib memory allocator is handle based and can free and
compact, move or resize memory on demand. This allows to effeciently
allocate memory dynamically without an MMU, by avoiding fragmentation
through memory compaction.
This patch adds the buflib library to the core, along with
convinience wrappers to omit the context parameter. Compaction is
not yet enabled, but will be in a later patch. Therefore, this acts as a
replacement for buffer_alloc/buffer_get_buffer() with the benifit of a debug
menu.
See buflib.h for some API documentation.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30380 a1c6a512-1295-4272-9138-f99709370657
buffer chunks.
* Samples and position indication is closely associated with audio data
instead of compensating by a latency constant. Alleviates problems with
using the elapsed as a track indicator where it could be off by several
steps.
* Timing is accurate throughout track even if resampling for pitch shift,
whereas before it updated during transition latency at the normal 1:1 rate.
* Simpler PCM buffer with a constant chunk size, no linked lists.
In converting crossfade, a minor change was made to not change the WPS until
the fade-in of the incoming track, whereas before it would change upon the
start of the fade-out of the outgoing track possibly having the WPS change
with far too much lead time.
Codec changes are to set elapsed times *before* writing next PCM frame because
time and position data last set are saved in the next committed PCM chunk.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657
Namely, introduce buffer_get_buffer() and buffer_release_buffer().
buffer_get_buffer() aquires all available and grabs a lock, attempting to
call buffer_alloc() or buffer_get_buffer() while this lock is locked will cause
a panicf() (doesn't actually happen, but is for debugging purpose).
buffer_release_buffer() unlocks that lock and can additionally increment the
audiobuf buffer to make an allocation. Pass 0 to only unlock if buffer was
used temporarily only.
buffer_available() is a replacement function to query audiobuflen, i.e. what's
left in the buffer.
Buffer init is moved up in the init chain and handles ipodvideo64mb internally.
Further changes happened to mp3data.c and talk.c as to not call the above API
functions, but get the buffer from callers. The caller is the audio system
which has the buffer lock while mp3data.c and talk mess with the buffer.
mpeg.c now implements some buffer related functions of playback.h, especially
audio_get_buffer(), allowing to reduce #ifdef hell a tiny bit.
audiobuf and audiobufend are local to buffer.c now.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30308 a1c6a512-1295-4272-9138-f99709370657
This is done to make reboot more transparent. If a playlist has ended, there should be no difference between the player doing nothing for ten minutes and the player shutting down after the idle timeout and being restarted.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30226 a1c6a512-1295-4272-9138-f99709370657
The resume position coming from these sources takes precedence over
autoresume. If we have such a resume offset, we don't have to wait
for the database to produce a resume offset before we can start the
codec.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29470 a1c6a512-1295-4272-9138-f99709370657
It's not useful to do it since you need to write back the code to disk to be able to load it from memory, it also requires writing to an executable directory.
Keep it for the simulator for the sake of simulating.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29261 a1c6a512-1295-4272-9138-f99709370657
- Support is limited to non-desync jpeg in id3v2 tags. Other formats (hopefully) follow in the future.
- Embedded album art takes precedence over files in album art files.
- No additional buffers are used, the jpeg is read directly from the audio file.
Flyspray: FS#11216
Author: Yoshihisa Uchida and I
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29259 a1c6a512-1295-4272-9138-f99709370657
Move autoresume setting into its own menu. Add option to customize
which tracks should be resumed on automatic track change. Tracks can
be selected based on their their file location or genre tag
(comma-separated list of filename / genre substrings).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29251 a1c6a512-1295-4272-9138-f99709370657
track is interrupted within the first 15 seconds.
Regard a rewind to 0:00 as a track restart (updating resume position /
playback statistics before the rewind and starting the 15 s delay).
This allows skipping forward across an unplayed track without changing
its resume offset. Also, it is possible to skip backward to the
previous track after rewinding to the current track to 0:00 (pressing
Left twice) without losing the current track's resume position.
Initially contributed by Dave Slusher
Caveats:
* Works only for SWCODEC
* Skipping forward without altering the resume position does not work
when skip to outro has been turned on.
* The 15-second window in which the resume offset will not be updated
should start at the initial resume position, not at 0:00. This
would allow skipping over partially played tracks without altering
the resume position.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29250 a1c6a512-1295-4272-9138-f99709370657
Fix Runtime statistics data not gathered when playback stops
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@28644 a1c6a512-1295-4272-9138-f99709370657
(DEBUG builds do not work/build anyway)
Also use a shorter message
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27951 a1c6a512-1295-4272-9138-f99709370657
ideally all targets should define CACHEALIGN_BITS, for now we default it
to 16 bytes if it's not specified
Since the buffer is already aligned in playback.c no need to align it
again in buffering.c
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27073 a1c6a512-1295-4272-9138-f99709370657
