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2020-07-24[3/4] Completely remove HWCODEC supportSolomon Peachy1-96/+3
'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
2018-12-22Auto-Ranging Time Formatting For Menus (hh:mm:ss:mss)William Wilgus1-1/+1
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
2017-02-04Implement speaker enable/disable on jack (un)plugAmaury Pouly1-0/+9
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
2014-03-10Implement time-based resume and playback start.Michael Sevakis1-1/+1
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>
2013-12-23playback,talk: Share audiobuffer via core_alloc_maximum().Thomas Martitz1-4/+0
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
2013-07-13Get rid of some superfluous single-purpose functions in playback.Michael Sevakis1-3/+17
* 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
2013-06-30Update software recording engine to latest codec interface.Michael Sevakis1-4/+1
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>
2013-06-30Fix some whitespace in files changed in following commit.Michael Sevakis1-11/+11
Change-Id: Ie3f43e43076e0dcae9a10f1b0b9e4698b398acee Reviewed-on: http://gerrit.rockbox.org/492 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2011-11-17Buflib: Make shrinking and buflib_available() smarter.Thomas Martitz1-0/+4
* 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
2011-08-30GSoC/Buflib: Add buflib memory alocator to the core.Thomas Martitz1-0/+1
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
2011-08-14Fix errors and warnings.Thomas Martitz1-1/+1
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
2011-08-14GSoC/Buflib: Replace all direct accesses to audiobuf with buffer API functions.Thomas Martitz1-2/+4
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
2011-02-10audio_peek_track should copy the struct mp3entry instead of pointing ↵Michael Sevakis1-1/+1
directly into the buffer. Despite the dire warning, caller does in fact yield/sleep and its usage is too nonlocalized to control that reliably. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29275 a1c6a512-1295-4272-9138-f99709370657
2010-09-01Ged rid of uisimulator/common/io.c for android builds.Thomas Martitz1-5/+0
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
2010-06-21Rockbox as an application: Replace many occurences of #ifdef SIMULATOR with ↵Thomas Martitz1-1/+1
#if (CONFIG_PLATFORM & PLATFORM_HOSTED) (or equivalently). 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
2010-05-06Move c/h files implementing/defining standard library stuff into a new libc ↵Thomas Martitz1-1/+1
directory, also standard'ify some parts of the code base (almost entirely #include fixes). 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
2010-04-18Move audio_peek_track declaration from skin_display.c to audio.hBertrik Sikken1-0/+1
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25670 a1c6a512-1295-4272-9138-f99709370657
2010-03-03FS#10756 - Free unused init codeThomas Martitz1-1/+1
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
2009-11-16mpeg.h/c cleanupJeffrey Goode1-0/+7
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23652 a1c6a512-1295-4272-9138-f99709370657
2009-11-16Cleanup audio.h, related functionsJeffrey Goode1-33/+28
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23651 a1c6a512-1295-4272-9138-f99709370657
2009-04-06FS9795 - some playback cleanup. Jonathan Gordon1-1/+0
* 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
2008-12-13Keyclick fixup take two. It is only 88 samples (2msec) long so keep a small ↵Michael Sevakis1-1/+0
static buffer around for beeps less than or equal to keyclick duration. This way it operates no matter the buffer state and still won't interfere with alternate PCM operations like recording or plugin playback. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19415 a1c6a512-1295-4272-9138-f99709370657
2008-12-13PCM buffer beeping needs the audio buffer reset in order to work after ↵Michael Sevakis1-0/+1
operations that trash the buffer or keyclick won't be audible until playback is restarted. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19413 a1c6a512-1295-4272-9138-f99709370657
2008-12-13Fix FS#8660-Recording hardlocks with keyclick enabled and FS#9388-keyclicks ↵Michael Sevakis1-0/+10
too late. Introduce audio_buffer_state to check whether PCM buffer is useable or disabled (trashed). git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19411 a1c6a512-1295-4272-9138-f99709370657
2008-12-09FS#9609 FM radio support for the Gigabeat S, seeking/scanning is not yet Nils Wallménius1-0/+1
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
2008-10-14Moved pcm_record from firmware to apps. Cleaned up some. Now all code using ↵Björn Stenberg1-2/+0
struct mp3entry is in apps. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18807 a1c6a512-1295-4272-9138-f99709370657
2008-10-08New recording setting to configure how mono recordings are made. Previously, ↵Peter D'Hoye1-0/+1
this was always L+R, which was kinda silly if your signal was on L only. This setting allows for L, R or L+R. SWCODEC only for now, to be added for HWCODEC (although that will only be L and L+R probably) git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18745 a1c6a512-1295-4272-9138-f99709370657
2008-10-07add HAVE_DISK_STORAGE, and use that instead of HAVE_FLASH_STORAGE when ↵Frank Gevaerts1-1/+1
checking for an actual disk (i.e. related to spindown, skipping,...) git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18735 a1c6a512-1295-4272-9138-f99709370657
2008-06-28Updated our source code header to explicitly mention that we are GPL v2 orDaniel Stenberg1-2/+4
later. We still need to hunt down snippets used that are not. 1324 modified files... http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
2008-03-28Revert my earlier const madness, we'll keep the parameter lists simple.Steve Bavin1-5/+5
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@16863 a1c6a512-1295-4272-9138-f99709370657
2008-03-28The const police raid playback.c, should be no change to behaviour.Steve Bavin1-24/+6
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@16860 a1c6a512-1295-4272-9138-f99709370657
2007-11-19Make sure initial menu item is voiced at startup. Voice thread must wait for ↵Michael Sevakis1-0/+1
the audio thread to finish initializing hardware. A known issue at speex commit time (perhaps only by myself ;). The behavioral refinement time begins. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15682 a1c6a512-1295-4272-9138-f99709370657
2007-11-18Make speex the new voice format for SWCODEC targets (non-Archos). Remove ↵Michael Sevakis1-4/+10
codec swapping and build speex voice decoding directly into the core binary. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15668 a1c6a512-1295-4272-9138-f99709370657
2007-11-11Album art support. Based on FS#3045, but heavily modified to adapt to MoB ↵Nicolas Pennequin1-0/+3
and for cleanness. 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
2007-10-06Unify PCM interface just above the hardware driver level for all targets ↵Michael Sevakis1-1/+1
including the sims. Perform lockout of audio callback when changing states. Weird new playback or recording trouble? Check before and after this revision first though things seem quite sound. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15006 a1c6a512-1295-4272-9138-f99709370657
2007-08-02Even more optimizing for the recording name change...Peter D'Hoye1-0/+1
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@14148 a1c6a512-1295-4272-9138-f99709370657
2007-08-01Disable 'Disk spindown' and 'Anti skip buffer' settings and some related ↵Nils Wallménius1-0/+2
stuff for flash storage targets as they make no sense git-svn-id: svn://svn.rockbox.org/rockbox/trunk@14110 a1c6a512-1295-4272-9138-f99709370657
2007-06-08Straighten out some audio path APIs and misc. audio stuff. Having recording ↵Michael Sevakis1-7/+13
is not a prerequisite to having input/output source selection which is probably most useful when adding a audio input features like FM to a new port without forcing recording to be implemented first. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@13599 a1c6a512-1295-4272-9138-f99709370657
2007-06-05e200: Add recording. Just from MIC right now and FM integration will happen ↵Michael Sevakis1-0/+3
soon. Most every bit of weirdness is nescessary and no problems seem to exist that the retailos doesn't exhibit too (namely noise when LCD is on when recording from MIC). git-svn-id: svn://svn.rockbox.org/rockbox/trunk@13557 a1c6a512-1295-4272-9138-f99709370657
2007-05-20Use bitmasks to define which inputs are available. Makes it easier to remove ↵Michael Sevakis1-21/+26
old assumptions of which are available. Inspired by e200 being unique in having FM Radio and Mic but no Line. Doesn't remove the assumption that Mic is available or that one of Mic and/or Line is available just to avoid excessive #ifdef'ing until needed. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@13448 a1c6a512-1295-4272-9138-f99709370657
2007-02-18Fix remaining CONFIG_TUNER checks.Jens Arnold1-4/+4
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12385 a1c6a512-1295-4272-9138-f99709370657
2006-12-17SWCODEC recording: More cleanup of stuff after queue additions and audio ↵Michael Sevakis1-0/+5
driver unification. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11783 a1c6a512-1295-4272-9138-f99709370657
2006-12-10Place all recording functionality on pcmrec thread to serialize all ↵Michael Sevakis1-8/+9
recording operations. Button mash problems should be ruled out of pcm_record.c. Add additional lightweight checks by default and display any warnings that occurred during recording in first line of recording screen when they occur by blinking back and forth from warning display to normal line (Warning: <hex bitmask>). Warnings are cleared when beginning a new recording so write the number down if you see it and file a report. Add heavier checks when PCMREC_PARANOID is defined in the player config header (encoders and pcm_record must be aware of the define since it changes the chunk header format). These checks are mainly concerned with things that may cause skipping but also add unwanted overhead for normal operation. Best used with logf enabled. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11705 a1c6a512-1295-4272-9138-f99709370657
2006-11-23iRiver/iAudio: Added audio_set_recording gain and sound_default to plugin ↵Michael Sevakis1-0/+13
API. Simplified plugin recording by target/-ing some audio functions. UDA1380 records with WSPLL as a result. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11577 a1c6a512-1295-4272-9138-f99709370657
2006-11-18Changed #ifdef USE_IRAM to #ifdef IRAM_STEAL. Also want to check if results ↵Michael Sevakis1-1/+1
are the same again since another person also built a red one successfully. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11545 a1c6a512-1295-4272-9138-f99709370657
2006-11-18SWCODEC/IRAM: Save voice IRAM when a plugin initializes its IRAM. Defines ↵Michael Sevakis1-0/+5
two macros for declaring and initializing IRAM. Plugins should use these instead. See mp3_encoder, doom, etc. for details. Further tweaks in buffer restoration after other use. Hiding of some interfaces that should only be used by buffer management. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11544 a1c6a512-1295-4272-9138-f99709370657
2006-11-13H100/General: HAL for S/PDIF and refinement/bufixes in optical output ↵Michael Sevakis1-4/+0
powering/source selection. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11523 a1c6a512-1295-4272-9138-f99709370657
2006-11-06Big Patch adds primarily: Samplerate and format selection to recording for ↵Michael Sevakis1-30/+76
SWCODEC. Supprort for samplerates changing in playback (just goes with the recording part inseparably). Samplerates to all encoders. Encoders can be configured individually on a menu specific to the encoder in the recording menu. File creation is delayed until flush time to reduce spinups when splitting. Misc: statusbar icons for numbers are individual digits to display any number. Audio buffer was rearranged to maximize memory available to recording and properly reinitialized when trashed. ColdFire PCM stuff moved to target tree to avoid a complicated mess when adding samplerate switching. Some needed API changes and to neaten up growing gap between hardware and software codecs. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11452 a1c6a512-1295-4272-9138-f99709370657
2006-10-19Patch #5166 by Robert Keevil - Last.fm loggingLinus Nielsen Feltzing1-0/+1
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11269 a1c6a512-1295-4272-9138-f99709370657
2006-10-12Minor simplification #2; replace unneeded variable filebufused with a macroSteve Bavin1-0/+1
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11201 a1c6a512-1295-4272-9138-f99709370657