1) Adds way to pop activity without refreshing the skin at
the same time.
Activities are sometimes popped in immediate succession,
or one activity is popped before another one is pushed right
away. This can lead to the UI appearing glitchy, due to an
activity only appearing for a split-second, which is especially
noticeable with complex skins that change the dimensions
of the UI viewport depending on the current activity
To fix this, prevent superfluous skin updates
* when switching between:
- WPS and browser
- WPS and Playlist Catalogue
- WPS and playlist
- WPS and Settings/System/Plugins
* when accessing Track Info or when displaying
bookmarks using the context menu on the WPS
* when switching from QuickScreen to Shortcuts Menu
2) The playlist viewer activity was pushed & popped
redundantly by playlist_view.
----
NB:
Behavior has remained unchanged in all instances of the
code where pop_current_activity() has been replaced by
pop_current_activity(ACTIVITY_REFRESH_NOW).
Change-Id: I56b517b8c9dba823a9fed3a3f558d7469dcea9fd
Removing the "list_wrap" argument is actually pretty easy.
In practice, almost all lists are using LIST_WRAP_UNLESS_HELD
behavior so we can make that the default. A couple of lists
disable wraparound with LIST_WRAP_OFF; this is now achieved
by setting the list "wraparound" flag to false when setting
up the list. LIST_WRAP_ON was unused and is of questionable
value, so it has been removed entirely.
This makes list wraparound behavior a property of the list,
controlled solely by the "wraparound" flag. The result is a
simpler list API and implementation, without changing the
behavior of any lists.
Change-Id: Ib55d17519e6d92fc95ae17b84ab0aaf4233bcb5a
Allow a clean shutdown to end in either power off or reboot. Add a
new event SYS_REBOOT to signal it and sys_reboot() to trigger the
event. SYS_REBOOT signals a reboot request and should be listened
for alongside SYS_POWEROFF events.
Change-Id: I99ba7fb5feed2bb5a0a40a274e8466ad74fe3a43
This controls the playback volume, which is not useful for
the M3K because it's impossible to play back and record at
the same time.
Change-Id: I5b0931c22b484f3ff6441d6c88079f18ec70f7f8
Remove the use of constants and ifdefs in favor of a
single enum and build the menu with a counter. This
simplifies the source and optimizes to the same code.
Change-Id: Ib07f0634ac8f663452c3e6e3c3d92fb6da1a1b4c
There were some hardcoded branches handling left/right gain for line-in
or FM radio inputs. If the target only has a microphone, these bits are
useless and cause compile errors due to missing audiohw settings, etc.
This patch #ifdef's them out.
Change-Id: If6a6918b2fabb256261c84651803bbf44deacb1a
'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
output_dyn_value now requires the count for number of units
Binary scale now shows Kibibytes instead of kilobytes (g#1742)
Fixes output for negative values as well
Change-Id: I8aa896860e97d2453fa35069e2dfe1caac60109f
Forgot static for recording_step_levels()
There is no need to re-calculate split_time or split_bytes each loop
you can't change them within recording screen -- these become constants
We don't need longs to store the conversion of minutes to seconds
or MB to bytes these are now shorts and the conversion takes place within
rec_timesplit_seconds() and rec_sizesplit_bytes() respectively
Change-Id: Ie1514085e92e00e67eb1942c0f4f8820714587e9
Limit buf2 to enough characters(16) to display signed int & label
consolidate gain/volume settings
Change-Id: I9ef6aa1e16ea2a17acc311a9b7e1c0556f58eff7
The new vuprintf makes unnecessary workarounds due to formatting
limitations. I checked grep output for whatever appeared to fit
but it's possible I missed some instances because they weren't
so obvious.
Also, this means sound settings can dynamically work with any
number of decimals rather than the current assumption of one or
two. Add an ipow() function to help and take advantage of dynamic
field width and precision. Consolidate string formatting of sound
settings.
Change-Id: I46caf534859dfd1916cd440cd25e5206b192fcd8
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 patch redoes the filesystem code from the FAT driver up to the
clipboard code in onplay.c.
Not every aspect of this is finished therefore it is still "WIP". I
don't wish to do too much at once (haha!). What is left to do is get
dircache back in the sim and find an implementation for the dircache
indicies in the tagcache and playlist code or do something else that
has the same benefit. Leaving these out for now does not make anything
unusable. All the basics are done.
Phone app code should probably get vetted (and app path handling
just plain rewritten as environment expansions); the SDL app and
Android run well.
Main things addressed:
1) Thread safety: There is none right now in the trunk code. Most of
what currently works is luck when multiple threads are involved or
multiple descriptors to the same file are open.
2) POSIX compliance: Many of the functions behave nothing like their
counterparts on a host system. This leads to inconsistent code or very
different behavior from native to hosted. One huge offender was
rename(). Going point by point would fill a book.
3) Actual running RAM usage: Many targets will use less RAM and less
stack space (some more RAM because I upped the number of cache buffers
for large memory). There's very little memory lying fallow in rarely-used
areas (see 'Key core changes' below). Also, all targets may open the same
number of directory streams whereas before those with less than 8MB RAM
were limited to 8, not 12 implying those targets will save slightly
less.
4) Performance: The test_disk plugin shows markedly improved performance,
particularly in the area of (uncached) directory scanning, due partly to
more optimal directory reading and to a better sector cache algorithm.
Uncached times tend to be better while there is a bit of a slowdown in
dircache due to it being a bit heavier of an implementation. It's not
noticeable by a human as far as I can say.
Key core changes:
1) Files and directories share core code and data structures.
2) The filesystem code knows which descriptors refer to same file.
This ensures that changes from one stream are appropriately reflected
in every open descriptor for that file (fileobj_mgr.c).
3) File and directory cache buffers are borrowed from the main sector
cache. This means that when they are not in use by a file, they are not
wasted, but used for the cache. Most of the time, only a few of them
are needed. It also means that adding more file and directory handles
is less expensive. All one must do in ensure a large enough cache to
borrow from.
4) Relative path components are supported and the namespace is unified.
It does not support full relative paths to an implied current directory;
what is does support is use of "." and "..". Adding the former would
not be very difficult. The namespace is unified in the sense that
volumes may be specified several times along with relative parts, e.g.:
"/<0>/foo/../../<1>/bar" :<=> "/<1>/bar".
5) Stack usage is down due to sharing of data, static allocation and
less duplication of strings on the stack. This requires more
serialization than I would like but since the number of threads is
limited to a low number, the tradoff in favor of the stack seems
reasonable.
6) Separates and heirarchicalizes (sic) the SIM and APP filesystem
code. SIM path and volume handling is just like the target. Some
aspects of the APP file code get more straightforward (e.g. no path
hashing is needed).
Dircache:
Deserves its own section. Dircache is new but pays homage to the old.
The old one was not compatible and so it, since it got redone, does
all the stuff it always should have done such as:
1) It may be update and used at any time during the build process.
No longer has one to wait for it to finish building to do basic file
management (create, remove, rename, etc.).
2) It does not need to be either fully scanned or completely disabled;
it can be incomplete (i.e. overfilled, missing paths), still be
of benefit and be correct.
3) Handles mounting and dismounting of individual volumes which means
a full rebuild is not needed just because you pop a new SD card in the
slot. Now, because it reuses its freed entry data, may rebuild only
that volume.
4) Much more fundamental to the file code. When it is built, it is
the keeper of the master file list whether enabled or not ("disabled"
is just a state of the cache). Its must always to ready to be started
and bind all streams opened prior to being enabled.
5) Maintains any short filenames in OEM format which means that it does
not need to be rebuilt when changing the default codepage.
Miscellaneous Compatibility:
1) Update any other code that would otherwise not work such as the
hotswap mounting code in various card drivers.
2) File management: Clipboard needed updating because of the behavioral
changes. Still needs a little more work on some finer points.
3) Remove now-obsolete functionality such as the mutex's "no preempt"
flag (which was only for the prior FAT driver).
4) struct dirinfo uses time_t rather than raw FAT directory entry
time fields. I plan to follow up on genericizing everything there
(i.e. no FAT attributes).
5) unicode.c needed some redoing so that the file code does not try
try to load codepages during a scan, which is actually a problem with
the current code. The default codepage, if any is required, is now
kept in RAM separarately (bufalloced) from codepages specified to
iso_decode() (which must not be bufalloced because the conversion
may be done by playback threads).
Brings with it some additional reusable core code:
1) Revised file functions: Reusable code that does things such as
safe path concatenation and parsing without buffer limitations or
data duplication. Variants that copy or alter the input path may be
based off these.
To do:
1) Put dircache functionality back in the sim. Treating it internally
as a different kind of file system seems the best approach at this
time.
2) Restore use of dircache indexes in the playlist and database or
something effectively the same. Since the cache doesn't have to be
complete in order to be used, not getting a hit on the cache doesn't
unambiguously say if the path exists or not.
Change-Id: Ia30f3082a136253e3a0eae0784e3091d138915c8
Reviewed-on: http://gerrit.rockbox.org/566
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
Those who have keyclick enabled and are really eagar to record upon
boot can get the samplerate messed up because who gets to set the sample
rate last (recording or mixer) is not currently deterministic.
Change-Id: Icc43ed789cf23f928ca49657cb146445b0c558cb
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>
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
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
1) fix bug in fmt_gain()
2) take into account steps field of sound_settings_info struct when inc/dec gain
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29593 a1c6a512-1295-4272-9138-f99709370657
For RaaA it evaluates user paths at runtime. For everything but codecs/plugins it will give the path under $HOME/.config/rockbox.org if write access is needed or if the file/folder in question exists there (otherwise it gives /usr/local/share/rockbox).
This allows for installing themes under $HOME as well as having config.cfg and other important files there while installing the application (and default themes) under /usr/local.
On the DAPs it's a no-op, returing /.rockbox directly.
Not converted to use get_user_file_path() are plugins themselves, because RaaA doesn't build plugins yet.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27656 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
