'swcodec' is now always set (and recording_swcodec for recording-capable
units) in feature.txt so the manual and language strings don't need to
all be fixed up.
Change-Id: Ib2c9d5d157af8d33653e2d4b4a12881b9aa6ddb0
Unifies time formatting in settings_list.c allows time format to
display as HH:MM:SS.MSS or any consecutive combination thereof
(hh:mm:ss, mm:ss, mm:ss.mss, ss.mss, hh, mm, ss ,mss)
works in INT and TABLE settings with the addition of flag 'F_TIME_SETTING'
Time is auto-ranged dependent on value
Adds talk_time_intervals to allow time values to be spoken similar to
display format: x Hours, x Minutes, x Seconds, x Milliseconds
Table lookups merged or removed from recording, clip meter and lcd timeout
-String_Choice replaced with TABLE_SETTING or INT_SETTING for these
functions as well, cleaned-up cfg_vals that get saved to cfgfile
RTL Languages ARE supported
Negative values ARE supported
Backlight on/off are now Always and Never to share formatter with LCD
Timeout
Added flag to allow ranged units to be locked to a minimum index
Added flag to allow leading zero to be supressed from the largest unit
merged talk_time_unit() and talk_time_intervals()
optimized time_split()
optimized format_time_auto()
Backlight time-out list same as original
Change-Id: I59027c62d3f2956bd16fdcc1a48b2ac32c084abd
The implementation is not very complicated but there are a few things worth
noting. There was a previous "speaker enable" setting but it was a boolean.
I decided to replace it with a choice setting that has 2 options (on, off)
if headphones cannot be detect on this target, or 3 options (on, off, auto)
if we can detect headphones. This will break the old setting on target that
cannot detect jack but it makes the code more uniform and avoid maintaining
two settings with more #ifdef. The third option (auto) uses the LANG_AUTO
text, which I think is clear enough (disable speaker on jack plug).
In order to avoid code duplication (both in apps and firmware), I decided to
keep the audiohw_enable_speaker function as-is: it takes a boolean and doesn't
care about the speaker policy. I introduced a new audio_enable_speaker that
takes directly the mode (which follows the setting encoding): 0=off, 1=on
and 2=auto. This way one calls audio_enable_speaker and it changes the speaker
once to reflect the request mode. The apps code then uses this function in the
places where it makes sense: on setting load, setting change and jack (un)plug
event.
Change-Id: I027873f698eb4bc365d7c02b515297806355d9e2
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>
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
Basically, just give it a good rewrite.
Software codec recording can be implemented in a more straightforward
and simple manner and made more robust through the better codec
control now available.
Encoded audio buffer uses a packed format instead of fixed-size
chunks and uses smaller data headers leading to more efficient usage.
The greatest benefit is with a VBR format like wavpack which needs
to request a maximum size but only actually ends up committing part
of that request.
No guard buffers are used for either PCM or encoded audio. PCM is
read into the codec's provided buffer and mono conversion done at
that time in the core if required. Any highly-specialized sample
conversion is still done within the codec itself, such as 32-bit
(wavpack) or interleaved mono (mp3).
There is no longer a separate filename array. All metadata goes
onto the main encoded audio buffer, eliminating any predermined
file limit on the buffer as well as not wasting the space for
unused path queue slots.
The core and codec interface is less awkward and a bit more sensible.
Some less useful interface features were removed. Threads are kept
on narrow code paths ie. the audio thread never calls encoding
functions and the codec thread never calls file functions as before.
Codecs no longer call file functions directly. Writes are buffered
in the core and data written to storage in larger chunks to speed up
flushing of data. In fact, codecs are no longer aware of the stream
being a file at all and have no access to the fd.
SPDIF frequency detection no longer requires a restart of recording
or plugging the source before entering the screen. It will poll
for changes and update when stopped or prerecording (which does
discard now-invalid prerecorded data).
I've seen to it that writing a proper header on full disk works
when the format makes it reasonably practical to do so. Other cases
may have incorrect data sizes but sample info will be in tact. File
left that way may play anyway.
mp3_enc.codec acquires the ability to write 'Info' headers with LAME
tags to make it gapless (bonus).
Change-Id: I670685166d5eb32ef58ef317f50b8af766ceb653
Reviewed-on: http://gerrit.rockbox.org/493
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
* 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
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
Use host's functions for file i/o directly (open(), close() ,etc.), not the sim_* variants.
Some dir functions need to be wrapped still because we need to cache the parents dir's path (host's dirent doesn't let us know).
For the same reason (incompatibility) with host's dirent) detach some members from Rockbox' dirent struct and put it into an extra one,
the values can be retrieved via the new dir_get_info().
Get rid of the sim_ prefix for sleep as well and change the signature to unix sleep().
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27968 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
This is to a) to cleanup firmware/common and firmware/include a bit, but also b) for Rockbox as an application which should use the host system's c library and headers, separating makes it easy to exclude our files from the build.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25850 a1c6a512-1295-4272-9138-f99709370657
Introduce a new .init section for initialisation code, so that it can be copied to an area which is later overwritten before calling. The stack/bss can then overwrite that code, effectively freeing the code size that the initialisation routines need. Gives a few kB ram usage back.
Only implemented for PP and as3525 so far. More targets could be added, as well as more functions.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25013 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
implemented but manual tuning works nicely. Thanks to Rafaël Carré,
Bertrik Sikken and Robert Menes for suggestions and debugging help.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19372 a1c6a512-1295-4272-9138-f99709370657
The cover pictures are loaded from external bitmaps. JPEG and embedded art are not supported. The pictures will only be drawn on the main display. There is no resizing but it is possible to specify the WPS bitmap size in the bitmap names (e.g. cover.100x100.bmp).
The bitmaps are stored in the main buffer and read directly from there. Currently, duplicate bitmaps will simply be present several times in the buffer, but this will be improved.
To enable for a target, #define HAVE_ALBUMART in its config file.
For more information, see the wiki page: http://www.rockbox.org/wiki/AlbumArt.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15572 a1c6a512-1295-4272-9138-f99709370657
