Playback needs to receive a couple of settings-related messages even
when not playing.
Put the message reply back where it was when loading an encoder for
recording.
Change-Id: I8cc80f46e42a0afd119991d698510e1ebef38ead
Eliminates the pcmrec thread and keeps playback and recording engine
operation mutually-exclusive.
audio_thread.c contains the audio thread which branches to the
correct engine depending upon the request. It also handles the main
audio initialization.
Moves pcm_init into main.c just before dsp_init because I don't want
that one in audio_init in the new file.
(Also makes revision df6e1bc pointless ;)
Change-Id: Ifc1db24404e6d8dd9ac42d9f4dfbc207aa9a26e1
...by default where they would be interpreted as valid but not actually
be which would cause calls to buffering while it was not initialized.
Add BUFFER_EVENT_BUFFER_RESET to inform users of buffering that the
buffer is being reinitialized. Basically, this wraps all the
functionality being provided by three events (...START_PLAYBACK,
RECORDING_EVENT_START, RECORDING_EVENT_STOP) into one for radioart.c,
the only user of those events (perhaps remove them?) and closes some
loopholes.
Change-Id: I99ec46b9b5fb4e36605db5944c60ed986163db3a
Buffers are not allocated and thread is not created until the first
call where voice is required.
Adds a different callback (sync_callback) to buflib so that other
sorts of synchonization are possible, such as briefly locking-out the
PCM callback for a buffer move. It's sort of a messy addition but it
is needed so voice decoding won't have to be stopped when its buffer
is moved.
Change-Id: I4d4d8c35eed5dd15fb7ee7df9323af3d036e92b3
Moved to playback.c, since it doesn't use metadata from the music file.
Change-Id: I5c3ad7750d94b36754f64eb302f96ec163785cb9
Reviewed-on: http://gerrit.rockbox.org/142
Reviewed-by: Nils Wallménius <nils@rockbox.org>
This function has been changed to rbcodec_format_is_atomic, which
doesn't require an enum from the kernel.
Change-Id: I1d537605087fe130a9b545509d7b8a340806dbf2
Reviewed-on: http://gerrit.rockbox.org/141
Reviewed-by: Nils Wallménius <nils@rockbox.org>
Tested-by: Nils Wallménius <nils@rockbox.org>
When enabled, if the user has set "Start File Browser Here" (config.cfg:
start directory) to anything other than root and "Auto-Change Directory"
is set to "Yes" or "Random", the directory returned when an auto change
is required will be constrained to the value of "start directory" or below.
Change-Id: Iaab773868c4cab5a54f6ae67bdb22e84642a9e4b
Reviewed-on: http://gerrit.rockbox.org/182
Reviewed-by: Nick Peskett <rockbox@peskett.co.uk>
Tested-by: Nick Peskett <rockbox@peskett.co.uk>
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
The simulator defines PLATFORM_HOSTED, as RaaA will do (RaaA will not define SIMULATOR).
The new define is to (de-)select code to compile on hosted platforms generally.
Should be no functional change to targets or the simulator.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27019 a1c6a512-1295-4272-9138-f99709370657
put your station images in .rockbox/fmpresets/<preset name>.bmp or .jpg. Must be in preset mode and the preset name must match the filename
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@26078 a1c6a512-1295-4272-9138-f99709370657
Change the way the wps playlist viewer gets the token values. All %i tokens are now supported (and a few others, experiment :) )
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24233 a1c6a512-1295-4272-9138-f99709370657
Fixing it because correcting the event api prototypes causes many warnings.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23301 a1c6a512-1295-4272-9138-f99709370657
The custom statusbar can be used as a WPS for the main UI, using .(r)sbs files. It's using the skin engine and knows all tags the WPS also knows.
The default folder for .sbs is the wps folder to reuse images used in the WPS.
As it can be shown in the WPS also, it's useful to move shared parts to the custom statusbar in order to save skin buffer space.
There are a few restrictions/TODOs:
*) Peak meter doesn't redraw nicely(not frequent enough), as very frequent updates would slow the UI down as hell (some targets fight with it in the WPS already: FS#10686)
*) No touchregion support as the statusbar doesn't have any action handling (it won't fail to parse though).
*) Drawing stuff into the default VP is forbidden (loading images in it is not). You *need* to use viewports for the displaying stuff (parsing fails if no viewport is used).
*) Themes that don't use a custom ui viewport can be fixed up using the new %Vi tag to avoid nasty redraw effectts (you must not draw into it as well, it's used to fix up the ui viewport). %Vi describes the viewport that the lists can use without getting in the way of the statusbar.
Otherwise, it behaves like the classic statusbar, it can be configured in the theme settings, and can be turned off in the wps using %wd.
Note to translaters: When translating LANG_STATUSBAR_CUSTOM, please consider using the same translation as for LANG_CHANNEL_CUSTOM if it's compatible. They could be combined later then.
Flyspray: FS#10566
Author: myself
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23258 a1c6a512-1295-4272-9138-f99709370657
Playback now has a few albumart slots. Anything (most importantly: skins) can obtain such a slot.
The slot has fields for the size which is passed to bufopen then to image_load to buffer the albumart with the proper size.
Currently there's 1 slot. We can increase it for remotes if we want. Custom statusbar will increase it.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23209 a1c6a512-1295-4272-9138-f99709370657
-add wrappers wps_data_load() and wps_data_init() so that other code doesn't need the structs for that
-change (and rename) gui_sync_wps_uses_albumart() to take points to be filled as parameter to get the AA size of a wps
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22139 a1c6a512-1295-4272-9138-f99709370657
Like changing AFMT_AAC to AFMT_MP4_AAC and AFMT_RAAC to AFMT_RM_AAC.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22066 a1c6a512-1295-4272-9138-f99709370657
also remove the audio_filename from the cuesheet struct as its useless
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@21982 a1c6a512-1295-4272-9138-f99709370657
swcodec: search for a .cue during buffering (with the possibility of adding embedded cuesheets later)
hwcodec: search for a .cue when the id3 info for the current track is requested for the first time (disk should be spining so non issue)
major beenfit from this is simplofy cuesheet handling code a bit... if mp3entry.cuesheet != NULL then there is a valid cuesheet.. no need to worry about if its enabled and preloaded.
There is the possibility of putting the next/prev subtrack handling inside the playback code (as well as the id3 updating stuff (see FS#9789 for more info), but thats probably not a good idea.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@21978 a1c6a512-1295-4272-9138-f99709370657
* Move strncpy() from core to the pluginlib
* Introduce strlcpy() and use that instead in most places (use memcpy in a few) in core and some plugins
* Drop strncpy() from the codec api as no codec used it
* Bump codec and plugin api versions
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@21863 a1c6a512-1295-4272-9138-f99709370657
The pcmbuffer wasn't reset after subsequent seeks (i.e. all but the first one) when seeking while paused. This caused the buffer to be filled only once and so the wrong sound was played upon resuming. Now we make sure the pcmbuffer is always reset when not playing (a more detailed explaination is in the task).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@21570 a1c6a512-1295-4272-9138-f99709370657
