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
It seems removing this causes a crash on the Clip+ when playing
any file. Appears to be a timing-related issue as replacing the
loop with an mdelay() also fixes it. Needs further investigation
to identify the real cause of the problem, see FS#13386.
Change-Id: Ia93a2887a79b376de50563d6bb3bbc79cee11a1c
Refactor DSP init routines so there is a dedicated init function
for the stages that need it. Remove the DSP_INIT configure message.
This allows the init code to be safely marked INIT_ATTR, saving a
bit of code size, and allowing the linker to verify that there are
no unsafe references to the init routines.
Change-Id: I1702f0f579bbb300a6fe7d0e67b13aa2e9dd7f8a
All of these are technically unsafe cross-section references but
most aren't reported by the linker, probably due to inlining. In
practice there was no problem because the affected code was only
run at init time anyway.
For now, remove INIT_ATTR until the init code can be refactored
to avoid the problematic references. This should also save code
size by moving more code to the init section.
dsp_init() gets to keep its attribute because it's already OK.
Change-Id: Idc9ac0e02cb07f31d186686e0382275c02a85dbb
haas surround is causing a seg fault
it appears process is null see https://www.rockbox.org/tracker/task/13382
for details
when the low_latency_callback is enabled it happens less frequently
lets default to an empty process that way there are no NULL pointers to call
Change-Id: Ib72ba1a58cbb20cef04b5ea50964adadeee74a75
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>
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
Use them to move tick counting, yielding and coldfire macsr handling
code to a rockbox specific file.
Change-Id: Id7417dc98c08a342eba45ba56b044a276e50564b
Reviewed-on: http://gerrit.rockbox.org/229
Tested-by: Nils Wallménius <nils@rockbox.org>
Reviewed-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>
