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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
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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>
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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
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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
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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
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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>
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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>
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Change-Id: Ie3f43e43076e0dcae9a10f1b0b9e4698b398acee
Reviewed-on: http://gerrit.rockbox.org/492
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
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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
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Change-Id: I8d1278b8cfaa376d2ad5a99dd552dc980c66e1da
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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
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Get those too.
Change-Id: Id2a39afe7a61d6ec0cea38633b94fe1b7122204f
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Replace with rbcodecconfig.h and platform.h includes. Remove now-
unneeded ones as well.
Change-Id: I6111b71e90bf86d9fe272a7916f2d34a5c6dd724
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Fix wrapping hazard which did eventually manifest on the right file.
Change-Id: I996a6efd3181b56fd172b5c3a526c7434f88bbbe
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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
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Yep, nope, not necessary anymore. Just call functions directly.
Change-Id: I21dc35f8d674c2a9c8379b7cebd5613c1f05b5eb
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Why? Why not? Cuts a few MHz.
Change-Id: Ied5c70b1aedd255cbe5d42b7d3028bbe47aad01d
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Most SoCs are these days are fast enough for realtime BRR, gaussian
interpolation and echo processing.
Change-Id: I180ce8ad45242c67b5e573a406b9522098a3f12b
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Change-Id: I6c72f4d1c79b1a99a11fb28e7d46886c08a56a75
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'Nuff said. Last update wasn't quite right.
Change-Id: I082a79c4e0c82b968fe2375cb82ee5c3a64a208b
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Change-Id: I1519f3bf2cdf74f3d4741951973352b2678b7722
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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
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Anyway, that's true now.
Change-Id: I247ea9a10543a8b65f3e73495f0e2ea725ec533e
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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
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Change-Id: I8a8604d6726304d04281671b475b2f75f9bfc0e5
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Fixes playback of files with large embedded album art.
Change-Id: I94d336e3da968a93047dd00a5fa65e4c3423a7da
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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
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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>
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Change-Id: I79f44f0dcc1b23b33a5040795220713660a1d18a
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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
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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 850491a043a9341ef31ffb8c02eedd2ae7b2c53e.
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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
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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
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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
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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
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Change-Id: I34b77287ba0b1a0002db3d52e893a52c50593362
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Thanks to Frank Gevaerts.
Change-Id: I6ca1d0258bfc70950d0ad5c2975d2bd88060b8a3
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Change-Id: Icc10d6e85f890c432f191233a4d64e09f00be43d
Reviewed-on: http://gerrit.rockbox.org/456
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
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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
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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>
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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
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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
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Change-Id: I03055d045c0a8e0e63e17b290cc71c54a8dc3634
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Change-Id: Ie3e35292ba8d74f0ff3d1bb3483a5e83aae0e6b6
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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
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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>
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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>
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The skin_parser now needs __PCTOOL__ set to build libskin_parser.a properly for
use with the Theme Editor.
Change-Id: I48a518fa296cc8ec5d0e3022baaedd796afe7c5f
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- 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>
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Instead specify explicit library dependencies for codecs and plugins.
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