ffmpeg_bitstream.c is included in libcodec, so there doesn't seem to
be any reason for individual codecs to also compile it (and clobber
any previous copy while they're at it, leading to broken builds)
Change-Id: I2bedc277ab109f44a6e8feb3d12ed01a720e00a6
Fixes a buffer overflow present when MP3 is encoded at 32000 Hz sample
rate, affected bitrates are 320 and 256 kbps.
Change-Id: I7634e70409be9d675d47be316a42630dd3147636
Reorganization
- Separated iBasso devices from PLATFORM_ANDROID. These are now standlone
hosted targets. Most device specific code is in the
firmware/target/hosted/ibasso directory.
- No dependency on Android SDK, only the Android NDK is needed.
32 bit Android NDK and Android API Level 16.
- Separate implementation for each device where feasible.
Code cleanup
- Rewrite of existing code, from simple reformat to complete reimplementation.
- New backlight interface, seperating backlight from touchscreen.
- Rewrite of device button handler, removing unneeded code and fixing memory
leaks.
- New Debug messages interface logging to Android adb logcat (DEBUGF, panicf,
logf).
- Rewrite of lcd device handler, removing unneeded code and fixing memory leaks.
- Rewrite of audiohw device handler/pcm interface, removing unneeded code and
fixing memory leaks, enabling 44.1/48kHz pthreaded playback.
- Rewrite of power and powermng, proper shutdown, using batterylog results
(see http://gerrit.rockbox.org/r/#/c/1047/).
- Rewrite of configure (Android NDK) and device specific config.
- Rewrite of the Android NDK specific Makefile.
Misc
- All plugins/games/demos activated.
- Update tinyalsa to latest from https://github.com/tinyalsa/tinyalsa.
Includes
- http://gerrit.rockbox.org/r/#/c/993/
- http://gerrit.rockbox.org/r/#/c/1010/
- http://gerrit.rockbox.org/r/#/c/1035/
Does not include http://gerrit.rockbox.org/r/#/c/1007/ due to new backlight
interface and new option for hold switch, touchscreen, physical button
interaction.
Rockbox needs the iBasso DX50/DX90 loader for startup, see
http://gerrit.rockbox.org/r/#/c/1099/
The loader expects Rockbox to be installed in /mnt/sdcard/.rockbox/. If
/mnt/sdcard/ is accessed as USB mass storage device, Rockbox will exit
gracefully and the loader will restart Rockbox on USB disconnect.
Tested on iBasso DX50.
Compiled (not tested) for iBasso DX90.
Compiled (not tested) for PLATFORM_ANDROID.
Change-Id: I5f5e22e68f5b4cf29c28e2b40b2c265f2beb7ab7
surround_enabled was never true, end up dsp_surround_flush didn't work; Thats why a cracking noise occurs in right channel when moving track positions.
redo pbe/surround flush in a much simpler way suits the current single buffer style.
Change-Id: I394054ddfb164b82c90b3dcf49df4442db87d8d2
perceptual bass enhancement
- a bbe-ish group delay corrction with Biophonic EQ boost.
- precut
auditory fatigue reduction
-reduce signal in frequency that may trigger temporary threshold shift
haas surround
-frequency between f(x1) and f(x2) is always bypassed.
-can apply to side only.
Change-Id: Icb6355ce9b1c99bf2c58c9385c3c411c0ae209d3
The origin of the register value was never
moved in the desired register state due to a typo ('rhs' vs. 'rhd').
While looking at the code, I noticed the action taken
for the register value is another copy'n'paste error
from the ADD opcode above -> it added to the register value
instead of MOVing the current value.
Patch submitted upstream.
cppcheck reported:
[lib/unwarminder/unwarm_thumb.c:473]: (warning) Redundant assignment of 'state.regData[rhd].o' to itself.
Change-Id: I78cdbf37a191007a3bddbaa350b906dbce2fe671
Also fix a wrong format specifier for an unsigned variable.
Detected by cppcheck, patch submitted upstream.
Change-Id: I9b84d91eeb242ed77b53ecc16252c5b35190bb9f
Either by mistake or because its meaning changed, usb_powered() doesn't mean
what the name suggest, so clarify its meaning by renaming it to usb_powered_only.
So use of usb_powered() are replaced by usb_inserted() when it makes more sense.
Change-Id: I112887e2d8560e84587bee5f55c826dde8c806d8
Reviewed-on: http://gerrit.rockbox.org/1097
Reviewed-by: Amaury Pouly <amaury.pouly@gmail.com>
If any of those functions should be (unused) API functions,
they can easily be turned back once really needed.
Detected using a new cppcheck check that
uses the internal symbol database to catch
functions that are only used in the current file.
Change-Id: Ic2b1e5b8020b76397f11cefc4e205f3b7ac1f184
- Leave original ptr untouched if allocation fails
(bail out early)
- Behave like malloc() in case ptr is NULL
Change-Id: Ib854ca19bd0e069999b7780d2d9a533ece705add
This file revealed several problems with our ASF parser:
1) The packet count in the ASF was actually a 64 bit value,
leading to overflow in very long files.
2) Seeking blindly trusted the bitrate listed in the ASF header
rather than computing it from the packet size and number of packets.
Fix these problems and fix a few minor issues.
Change-Id: Ie0f68734e6423e837757528ddb155f3bdcc979f3
Forgot to (void) an unused parameter when priorityless.
usb-drv-rl27xx.c was using a compound init to initialize a semaphore
but the structure changed so that it is no longer correct. Use
designated initializers to avoid having to complete all fields.
Forgot to break compatibility on all plugins and codecs since the
kernel objects are now different. Take care of that too and do the
sort thing.
Change-Id: Ie2ab8da152d40be0c69dc573ced8d697d94b0674
including
BAR_PARAMS, %xl, %dr, %T,%St, %xl and %Cl
Change-Id: I0811ebfff5f83085481dcbf08f97b7223f677bfe
Reviewed-on: http://gerrit.rockbox.org/900
Reviewed-by: Jonathan Gordon <rockbox@jdgordon.info>
Speeds up decoding of the 64 kbps test file by 2.59 MHz and the
128 kbps test file by 4.31 MHz on H300 (cf). Decoding the same
files on c200 is sped up by 0.33 MHz and 0.55 MHz respectively.
Change-Id: I0f9f9ef6a7293581cf45e3201b33c65504c95c81
The recent merge of upstream changed the fft to use C_MUL which
wasn't implemented in asm for coldfire.
Speeds up decoding 64 kbps test file by 2.68 MHz and 128 kbps
test file by 2.80 MHz on H300.
Change-Id: I8b61fc0f9568d6350431e311a12e44fe4f60f72e
Sync to commit bb4b6885a139644cf3ac14e7deda9f633ec2d93c
This brings in a bunch of optimizations to decode speed
and memory usage. Allocations are switched from using
the pseudostack to using the real stack. Enabled hacks
to reduce stack usage.
This should fix crashes on sansa clip, although some
files will not play due to failing allocations in the
codec buffer.
Speeds up decoding of the following test files:
H300 (cf) C200 (arm7tdmi) ipod classic (arm9e)
16 kbps (silk) 14.28 MHz 4.00 MHz 2.61 MHz
64 kbps (celt) 4.09 MHz 8.08 MHz 6.24 MHz
128 kbps (celt) 1.93 MHz 8.83 MHz 6.53 MHz
Change-Id: I851733a8a5824b61feb363a173091bc7e6629b58
Implicit promotion of integer literals to unsigned long introduced a subtle bug
on 64-bit systems due to weird sign extensions (leads to audible glitches in a
few files). The table is originally designed for unsigned 32bit integers, and
it works with those so use them. As a consequence the lookup table size is
halved as well.
Change-Id: I35d878d6df03300387f0e403e0f3c3bdc73eea00
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 is an improvement to the current compressor which I have added
to my own Sansa Fuze V2 build. I am submitting here in case others
find it interesting.
Features added to the existing compressor:
Attack, Look-ahead, Sidechain Filtering.
Exponential attack and release characteristic response.
Benefits from adding missing features:
Attack:
Preserve perceived "brightness" of tone by letting onset transients
come through at a higher level than the rest of the compressed program
material.
Look-ahead:
With Attack comes clipping on the leading several cycles of a transient
onset. With look-ahead function, this can be pre-emptively mitigated with
a slower gain change (less distortion). Look-ahead limiting is implemented
to prevent clipping while keeping gain change ramp to an interval near 3ms
instead of instant attack.
The existing compressor implementation distorts the leading edge of a
transient by causing instant gain change, resulting in log() distortion.
This sounds "woofy" to me.
Exponential Attack/Release:
eMore natural sounding. On attack, this is a true straight line of 10dB per
attack interval. Release is a little different, however, sounds natural as
an analog compressor.
Sidechain Filtering:
Mild high-pass filter reduces response to low frequency onsets. For example,
a hard kick drum is less likely to make the whole of the program material
appear to fade in and out. Combined with a moderate attack time, such a
transient will ride through with minimal audible artifact.
Overall these changes make dynamic music sound more "open", more natural. The
goal of a compressor is to make dyanamic music sound louder without necessarily
sounding as though it has been compressed. I believe these changes come closer to this goal.
Enjoy. If not, I am enjoying it
Change-Id: I664eace546c364b815b4dc9ed4a72849231a0eb2
Reviewed-on: http://gerrit.rockbox.org/626
Tested: Purling Nayuki <cyq.yzfl@gmail.com>
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
As comment in code states:
"It is possible for our seek to land in the middle of audio
data that looks exactly like a frame header from a future
version of an encoder. When that happens, frame_sync() will
return false. But there is a remote possibility that it is
properly synced at such a "future-codec frame", so to make sure,
we wait to see several "unparseable" errors in a row before
bailing out."
Currently we wait for 10 "unparseable" errors. libFLAC waits for 20.
But I've got a valid flac+cue, wherein switching to certain track
gave me 24 "unparsaeable" errors. Therefore I increased
unparseable_count to 30.
Change-Id: I4e97a5385c729adf3d5075d41ea312622c69e548
Reviewed-on: http://gerrit.rockbox.org/658
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Reviewed-by: Boris Gjenero <boris.gjenero@gmail.com>
Tested-by: Andrew Ryabinin <ryabinin.a.a@gmail.com>
Reviewed-by: Andrew Ryabinin <ryabinin.a.a@gmail.com>
Although Linux accepts several implicit definitions of SEEK_END found in
stdio.h, the compiler on FreeBSD won't. Rockbox compilation will fail
without stdio.h included.
There is a precedent for including this header, see
lib/rbcodec/codecs/libtremor/ivorbisfile.h.
Change-Id: I58510101b59a354cd6601cb3f323f385a824d2e8
Reviewed-on: http://gerrit.rockbox.org/639
Tested-by: Kevin Zheng <kevinz5000@gmail.com>
Reviewed-by: Frank Gevaerts <frank@gevaerts.be>
This enables the encoders - i.e. to record audio -
to be loaded also on the simulator.
Change-Id: I54fdbeb75b89023c0d7824a34cf76301c02c3150
Reviewed-on: http://gerrit.rockbox.org/632
Reviewed-by: Thomas Martitz <kugel@rockbox.org>
Sync opus codec to upstream commit
02fed471a4568852d6618e041c4f2af0d7730ee2 (August 30 2013)
This brings in a lot of optimizations but also makes the diff
between our codec and the upstream much smaller as most of our
optimizations have been upstreamed or supeceded.
Speedups across the board for CELT mode files:
64kbps 128kbps
H300 9.82MHz 15.48MHz
c200 4.86MHz 9.63MHz
fuze v1 10.32MHz 15.92MHz
For the silk mode test file (16kbps) arm targets get a speedup
of about 2MHz while the H300 is 7.8MHz slower, likely because it's
now using the pseudostack more rather than the real stack which
is in iram. Patches to get around that are upcomming.
Change-Id: Ifecf963e461c51ac42e09dac1e91bc4bc3b12fa3
Instead of providing yet another memory allocator implementation
use tlsf and simply link tlsf library.
Another small improvement is to *grow* memory pool by grabbing
audiobuffer instead of just switching to use audiobuf exclusively.
Tested with simple lua 'memory eater' script.
This patch extends tlsf lib slightly. You can provide
void *get_new_area(size_t * size) function which will override
weak dummy implementation provided in lib itself. This allows to
automaticaly initialize memory pool as well as grow memory
pool if needed (for example grab audiobuffer when pluginbuffer
is exhaused).
Change-Id: I841af6b6b5bbbf546c14cbf139a7723fbb982f1b
codec makefiles larger freedom in what they can do to it.
Use this in libopus to prepend the libopus searchpaths to
CODECFLAGS so that its internal config.h will be picked up before
our global one. This avoids having to do a s/config.h/opus_config.h/
when syncing which will be handy soon.
Change-Id: I018d729aa0c8300fa3149f22a5a8c5668b339dfa
Reviewed-on: http://gerrit.rockbox.org/496
Reviewed-by: Nils Wallménius <nils@rockbox.org>
The quickscreen calls settings_apply() and the crossfeed code wasn't
checking that the right crossfeed was set before updating the filter
for the custom setting, which was overwriting the Meier crossfeed
data (custom and Meier share the same data space).
Change-Id: Ifaa2f46fe062d4497681a2dd0d5068ec906c96a3
Distractions make logic fail. It only needs one more loop and should
not trigger further compression cycles after not feeding more data.
Change-Id: Ie0dbb34af92e0ca5718480dd4ab4719a141717ff
For mp3_enc, split encoding duties between COP and CPU.
For wavpack_enc, simply run the encoding on COP (splitting that one
needs more consideration) which keeps the it and the UI from running
on the same core.
As a result, at least they are now useable on PP at "normal" sample
rates.
mp3_enc in all this gets an extensive renovation and some optimizations
for speed, to reduce IRAM requirements and remove unneeded stuff.
Change-Id: I215578dbe36f14e516b05a5ca70880eb01ca0ec2
Replaces the NATIVE_FREQUENCY constant with a configurable frequency.
The user may select 48000Hz if the hardware supports it. The default is
still 44100Hz and the minimum is 44100Hz. The setting is located in the
playback settings, under "Frequency".
"Frequency" was duplicated in english.lang for now to avoid having to
fix every .lang file for the moment and throwing everything out of sync
because of the new play_frequency feature in features.txt. The next
cleanup should combine it with the one included for recording and
generalize the ID label.
If the hardware doesn't support 48000Hz, no setting will be available.
On particular hardware where very high rates are practical and desireable,
the upper bound can be extended by patching.
The PCM mixer can be configured to play at the full hardware frequency
range. The DSP core can configure to the hardware minimum up to the
maximum playback setting (some buffers must be reserved according to
the maximum rate).
If only 44100Hz is supported or possible on a given target for playback,
using the DSP and mixer at other samperates is possible if the hardware
offers them.
Change-Id: I6023cf0c0baa8bc6292b6919b4dd3618a6a25622
Reviewed-on: http://gerrit.rockbox.org/479
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
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>
The old presets never made sense for Rockbox's EQ. They were apparently
copied from some other software. We have a parametric EQ, that means that
EQ bands can be made wider or narrower. Putting two identical bands side
by side just wastes battery life and adds rounding error. Replacement
presets are on gerrit but they need more work. In the mean time, users
should probably not be using these.
Change-Id: I85213100129fafd3ac0fa1a9438cb4d651bb94cb
Rockbox only uses the first album art image (APIC / PIC frame) found in id3v2
tags. When a file contains more than one image the second one is ignored but
the parsealbumart() callback overwrites the already set data. This causes the
metadata structure to contain an invalid pointer to the image data, resulting
in no image shown.
Make parsealbumart() aware of this and skip parsing when an albumart image has
already been found. Fixes FS#12870.
Change-Id: Id8164f319cd5e1ee868b581f8f4ad3ea69c17f77
When writing a value to PC, execution continues at that location,
so subtracting 4 returns to the next instruction. Previously, two
instructions after the faulting instruction were being skipped, causing
safe_read functions to return true even if a data abort happened.
Change-Id: I3fd02d54646323ea2050d0504e38f6d22f09c749
Most SoCs are these days are fast enough for realtime BRR, gaussian
interpolation and echo processing.
Change-Id: I180ce8ad45242c67b5e573a406b9522098a3f12b
Affected BRR cached waveforms but not realtime BRR decode as far as
I could ascertain. BRR cached waves required loop points to be inside
the initial waveform but this change removes that restriction.
Change-Id: I0ef4db720e5c28bd7b2fb9ae255d27c0a7213f79
CPU optimization gets its own files in which to fill-in optimizable
routines.
Some pointless #if 0's for profiling need removal. Those macros are
empty if not profiling.
Force some functions that are undesirable to be force-inlined by the
compiler to be not inlined.
Change-Id: Ia7b7e45380d7efb20c9b1a4d52e05db3ef6bbaab
Use the tlsf malloc and friends instead of the silly
codec_malloc to get actually working free and saner
realloc that doesn't leak memory.
Makes files with moderately sized embedded AA play
on targets with large enough codec buffers and files
with too large AA are now skipped rather than crashing.
Fixes crash when playing example file in FS#12842.
Change-Id: I06562955c4d9a95bd90f55738214fba462092b71
Uses the Catmull-Rom case of Hermite cubic splines.
Vastly improves the quality and accuracy of audio resampling with a
rather minor additional overhead compared to the previous linear
implementation.
ARM and Coldfire assembly implementations included.
Change-Id: Ic45d84bc66c5b312ef373198297a952167a4be26
Reviewed-on: http://gerrit.rockbox.org/304
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
DSP_SWITCH_FREQUENCY has been deprecated and the same enumerated value
as DSP_SET_FREQUENCY since major DSP revisions were committed. This
task should have been performed much earlier but, oh well, do it now.
Change-Id: I3f30d651b894136a07c7e17f78fc16a7d98631ff
While it made the mini2g not crash during startup anymore further tests showed
that other mini2g devices still exhibit the crash, or end up with a "No
partition found" error; furthermore the device tested first still crashes on
USB disconnect. Therefore the change doesn't really help with the problem, and
at the expense of increasing binary size for all other targets there is no
point in keeping it for now.
This reverts commit 850491a043.
Input type can only change once per call because the DSP parameters
are only copied at the start and input is always taken from the src
buffer which means sample input format switching can be once per call
instead of once per loop.
Change-Id: Ifa3521753428fb0e6997e4934f24a3b915628cc7
Some things can just be a bit simpler in handling the list of stages
and some things, especially format change handling, can be simplified
for each stage implementation. Format changes are sent through the
configure() callback.
Hide some internal details and variables from processing stages and
let the core deal with it.
Do some miscellaneous cleanup and keep things a bit better factored.
Change-Id: I19dd8ce1d0b792ba914d426013088a49a52ecb7e
fixedpoint.make is not a subdir Makefile but a lib Makefile. Setting
CORE_GCSECTIONS in it will affect the final link and make it always use
--gc-sections (for SWCODEC Rockbox), since libfixedpoint is always needed
(bootloader and HWCODEC don't use libfixedpoint).
Fixes FS#12857.
Change-Id: Ib30bd03cbcea9c339a73daf7b673868aa3cc7a88
When seeking to the next id3v2 frame we need to consider if the tag has the
unsync flag set. Not doing so will likely make parsing end up in the middle of
the current frame if the frame size exceeds the upper limit set during read.
The latter usually happens for album art frames.
Fixes FS#12849.
Change-Id: Ic92853eef4374508d84df347bcc66b6661d5037d
This is going right in since it's long overdue. If anything is goofed,
drop me a line or just tweak it yourself if you know what's wrong. :-)
Make HW/SW codec interface more uniform when emulating HW functionality
on SWCODEC for functions such as "audiohw_set_pitch". The firmware-to-
DSP plumbing is in firmware/drivers/audiohw-swcodec.c. "sound_XXX"
APIs are all in sound.c with none in DSP code any longer.
Reduce number of settings definitions needed by each codec by providing
defaults for common ones like balance, channels and SW tone controls.
Remove need for separate SIM code and tables and add virtual codec header
for hosted targets.
Change-Id: I3f23702bca054fc9bda40f49824ce681bb7f777b
Implements double-buffered volume, balance and prescaling control in
the main PCM driver when HAVE_SW_VOLUME_CONTROL is defined ensuring
that all PCM is volume controlled and level changes are low in latency.
Supports -73 to +6 dB using a 15-bit factor so that no large-integer
math is needed.
Low-level hardware drivers do not have to implement it themselves but
parameters can be changed (currently defined in pcm-internal.h) to work
best with a particular SoC or to provide different volume ranges.
Volume and prescale calls should be made in the codec driver. It should
appear as a normal hardware interface. PCM volume calls expect .1 dB
units.
Change-Id: Idf6316a64ef4fb8abcede10707e1e6c6d01d57db
Reviewed-on: http://gerrit.rockbox.org/423
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Will need it soon enough.
Combine the contents of all the various fixedpoint.h files.
Not moving fixedpoint.c for now since I'm not sure where it
should be and it causes some dependency issues.
Change-Id: Ideacbca2ca78f9158c2b114b113c274f68e908d5
Prevents cutoff of tracks, especially short ones:
* Extend looped tracks by fade length to fade at start of loop repeat.
* No fade occurs for non-repeating track only having an intro.
* Uses id3.tail_trim field to store fade duration.
Use libGME built-in elapsed time reporting instead of custom calculation:
* libGME already reports in milliseconds.
* Don't advance time counter when Repeat == One. It just runs the progress
over the length limit.
Fix a comment about sample rate and set the reported bitrate to be
accurate for 44.1 kHz stereo.
Change-Id: I3ede22bda0f9a941a3fef751f4d678eb0027344c
libtlsf used not to be built for HWCODEC, but now that the gif
viewer uses libtlsf instead of building its own copy, libtlsf
is needed everywhere.
Change-Id: I730719c6a20e749adb8597056d2049b7758620e4
Flush decoder state and frame out buffer upon a forced stop to prevent
a short burst of stale audio from the previously decoding track from
playing when skipping from one WMA track to another.
Change-Id: I24c910c5dbd83caed2510db68d9e39a474332a79
Reviewed-on: http://gerrit.rockbox.org/406
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Instead of 3 cfg lines per eq band there is now a single line
for each:
<config name>: <cutoff/center freq>, <q>, <gain>
In addition, the config value names make a bit more sense.
The old settings are still readable but config.cfg and any new
settings files will be written with the new config values. (The
old settings will be removed completly sometime after the next
stable release).
Also a slight rework of the advanced EQ menu UI
Change-Id: I9008658d36ded442a5f2f825916df42a3934cbef
Reviewed-on: http://gerrit.rockbox.org/394
Reviewed-by: Jonathan Gordon <rockbox@jdgordon.info>
The skin_parser now needs __PCTOOL__ set to build libskin_parser.a properly for
use with the Theme Editor.
Change-Id: I48a518fa296cc8ec5d0e3022baaedd796afe7c5f
- A 10 Band EQ for Rockbox w/ presets adapted
from VLC
- frequency stepping at 32, 64, 125, 250, 500
1K, 2K, 4K, 8K, 16K
Change-Id: I85ad84d70a534edfc66c6ad9af8a76f022a02ec7
Reviewed-on: http://gerrit.rockbox.org/386
Reviewed-by: Jonathan Gordon <rockbox@jdgordon.info>
Speeds up decoding of 128k opus files by 1.2MHz on AMSv2. Rounding
error is 1 bit due to KissFFT using a 15 bit shift instead of a 16 bit shift.
Also, change an LDMIA in the armv4 code to LDM as the pointer should not
increment.
Change-Id: I626a207c6a056a1984e33cfe89415c35d0caed93
Reviewed-on: http://gerrit.rockbox.org/377
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Tested-by: Michael Giacomelli <giac2000@hotmail.com>
