ARMv7-M has hardware division, so it doesn't require __div0
or any support functions for 32-bit division.
Change-Id: I840683a1a77d737f378899ca4bcf858216b81014
This reverts commit 4ec34f6986.
Reason for revert:
-fshort-enums makes them smaller if the explicitrly enumerated values will fit in a smaller type
Change-Id: I834dfd2b2039eda91bc02c9cf95a0f9dfc5783f6
Codecs mostly use custom LOGF define for logging (i.e. see aac.c). Now such logging can be enabled in single file with #define LOGF_ENABLE
Change-Id: I36312fbcd2d9166fb1fe5ead31e7354342d8828d
Replace the minimum version bound with a check on the size of
the API struct. The version only needs to be incremented for
ABI breaking changes. Additions to the API won't need to touch
the version number, resulting in fewer merge conflicts.
Change-Id: I916a04a7bf5890dcf5d615ce30087643165f8e1f
It turns out removing DSP_INIT broke the codec ABI and caused
old codecs to crash; the loop and mdelay() was a red herring.
This reverts commit 541960a110.
Change-Id: I020d826e7b4beb006d093d9c3d4f45fa5eaac717
'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
Just use long so the compiler potentially doesn't complain about
use of other values not in the enum. It's also the type used
around the system for event ids.
Increase min codec API version.
No functional changes.
Change-Id: If4419b42912f5e4ef673adcdeb69313e503f94cc
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
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>
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>
librbcodec users must provide these two files when the library is built.
rbcodecconfig.h provides configuration #defines and basic types, and
will be included by public librbcodec headers, so it must not conflict
with the user's code. rbcodecplatform.h provides various OS functions,
and will only be included by source files and private headers. This
system is intended to provide maximum flexibility for use on embedded
systems, where no operating system headers are included. Unix systems
can just copy rbcodecconfig-example.h and rbcodecplatform-unix.h with
minimal changes.
Change-Id: I350a2274d173da391fd1ca00c4202e9760d91def
Reviewed-on: http://gerrit.rockbox.org/143
Reviewed-by: Nils Wallménius <nils@rockbox.org>
Tested-by: Nils Wallménius <nils@rockbox.org>
Creates a standard buffer passing, local data passing and messaging
system for processing stages. Stages can be moved to their own source
files to reduce clutter and ease assimilation of new ones. dsp.c
becomes dsp_core.c which supports an engine and framework for effects.
Formats and change notifications are passed along with the buffer so
that they arrive at the correct time at each stage in the chain
regardless of the internal delays of a particular one.
Removes restrictions on the number of samples that can be processed at
a time and it pays attention to destination buffer size restrictions
without having to limit input count, which also allows pcmbuf to
remain fuller and safely set its own buffer limits as it sees fit.
There is no longer a need to query input/output counts given a certain
number of input samples; just give it the sizes of the source and
destination buffers.
Works in harmony with stages that are not deterministic in terms of
sample input/output ratio (like both resamplers but most notably
the timestretch). As a result it fixes quirks with timestretch hanging
up with certain settings and it now operates properly throughout its
full settings range.
Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734
Reviewed-on: http://gerrit.rockbox.org/200
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
