This complements offset-based resume and playback start funcionality.
The implementation is global on both HWCODEC and SWCODEC.
Basically, if either the specified elapsed or offset are non-zero,
it indicates a mid-track resume.
To resume by time only, set elapsed to nonzero and offset to zero.
To resume by offset only, set offset to nonzero and elapsed to zero.
Which one the codec uses and which has priority is up to the codec;
however, using an elapsed time covers more cases:
* Codecs not able to use an offset such as VGM or other atomic
formats
* Starting playback at a nonzero elapsed time from a source that
contains no offset, such as a cuesheet
The change re-versions pretty much everything from tagcache to nvram.
Change-Id: Ic7aebb24e99a03ae99585c5e236eba960d163f38
Reviewed-on: http://gerrit.rockbox.org/516
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested: Michael Sevakis <jethead71@rockbox.org>
talk_init() is called by settings_apply() already which is called on boot.
Doing it again means loading the voicefile a second time which isn't necessary.
Change-Id: I4decd16401e63bf87338d3163c99d06d69fe3a3c
When the policy is not set, it'll by default not give the clip buffer away.
Callers of core_alloc_maximum() suffer from this. However, the thumbnail
buffer can be easily freed when needed because nothing needs to be
reloaded from disk when it is reallocated (thumbnail clips are loaded on
demand, when in the file browser). Do this to give core_alloc_maximum() callers
a better chance to succeed with the default talk buffer policy.
Change-Id: I8c0da29c520612ca903f6c930bd7c74ae97eca3b
This fixes the radioart crash that was the result of buffering.c working
on a freed buffer at the same time as buflib (radioart uses buffering.c for the
images). With this change the buffer is owned by buflib exclusively so this
cannot happen.
As a result, audio_get_buffer() doesn't exist anymore. Callers should call
core_alloc_maximum() directly. This buffer needs to be protected as usual
against movement if necessary (previously it was not protected at all which
cased the radioart crash), To get most of it they can adjust the willingness of
the talk engine to give its buffer away (at the expense of disabling voice
interface) with the new talk_buffer_set_policy() function.
Change-Id: I52123012208d04967876a304451d634e2bef3a33
* Remove explicit tracking of elapsed time of previous track.
* Remove function to obtain auto skip flag.
* Most playback events now carry the extra information instead and
pass 'struct track_event *' for data.
* Tweak scrobbler to use PLAYBACK_EVENT_TRACK_FINISH, which makes
it cleaner and removes the struct mp3entry.
Change-Id: I500d2abb4056a32646496efc3617406e36811ec5
Use generic void * and size_t and make mp3_play_data and its callback
agree on types. Use mp3_play_callback_t instead of prototyping
right in the function call (so it's not so messy to look at). Change
doesn't appear to require plugin API version increment.
Change-Id: Idcab2740ee316a2beb6e0a87b8f4934d9d6b3dd8
* 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
if the voice is bigger than the audiobuffer. NOTE: This is the case on the sim
so voice doesn't appear to work currently on hwcodec. Someone needs to verify
on a real target.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30910 a1c6a512-1295-4272-9138-f99709370657
Some functions must only be called when audio is already initialized, due to talk <-> audio interdependency, same as on swcodec.
This makes hwcodec boot and play music again. Voice menus also working again, talk clips not yet.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30738 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
The buffer_offset paramter of audio_init_recording() is removed as it
was unused in both implementations.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30310 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
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
* Use events to notify things when the track has changed instead of the nasty has_track_changed()
* Event for when the mp3entry for the next track is avilable (which allows alot more tags to be static which means less redrawing in the WPS)
* virtually guarentee that the mp3entry sturct returned by audio_current/next_track() is going to be valid for the duration of the current track. The only time it wont be now is during the time between the codec finishing the previous track and the next track actually starting (~2s), but this is not an issue as long as it is called again when the TRACK_CHANGED event happens (or just use the pointer that gives)
It is still possible to confuse the WPS with the next tracks id3 info being displayed but this should fix itself up faster than it used to (and be harder to do)
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@20633 a1c6a512-1295-4272-9138-f99709370657
This should be a good first step to allow multi-driver targets, like the Elio (ATA/SD), or the D2 (NAND/SD).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18960 a1c6a512-1295-4272-9138-f99709370657
