Commit graph

10 commits

Author SHA1 Message Date
Aidan MacDonald
34a092a997 rbcodec dsp: Replace enum dsp_ids arguments with unsigned int
Because casting to and from "enum dsp_id" just adds noise,
change everything to unsigned int.

Change-Id: I52a7ae55f406e673d5b811b29657fcdc4b62ab10
2022-12-22 18:00:37 -05:00
Aidan MacDonald
b96b7640de rbcodec dsp: Move dsp_sample_io_configure() to its own file
Makes dsp_sample_input.c a bit less messy, and dependencies
are more explicit. There's possibly a minor loss of inlining
but it isn't a big deal.

Change-Id: I30f923a0ca758f2b113d32852d1f65586dff0cd1
2022-12-22 17:20:14 -05:00
Aidan MacDonald
8165a6c245 rbcodec dsp: Remove INIT_ATTR from the DSP library
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
2022-12-18 22:23:52 +00:00
Michael Sevakis
d37bf24d90 Enable setting of global output samplerate on certain targets.
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>
2013-07-06 04:22:04 +02:00
Michael Sevakis
b5a6517e9d Remove explicit config.h and system.h includes from DSP code.
Replace with rbcodecconfig.h and platform.h includes. Remove now-
unneeded ones as well.

Change-Id: I6111b71e90bf86d9fe272a7916f2d34a5c6dd724
2013-05-27 03:23:33 -04:00
Michael Sevakis
78a45b47de Cleanup and simplify latest DSP code incarnation.
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
2013-05-04 13:43:33 -04:00
Michael Sevakis
87a9951cf8 Consolidate some sample input code.
Input functions have common setup sequences that can be placed
into an inline function instead of repeating it all repeatedly.

Change-Id: I9e62904ff0948651c64ddf160ed4400ed6dc81ff
2012-05-08 21:27:43 -04:00
Michael Sevakis
88aeef9127 Remove pointless IRAM allocation from voice DSP.
It's always used in MONO mode and doesn't need the IRAM sample/
resample buffers and 1280 bytes can be freed.

M5 can now have its PCM mixer downmix buffer in IRAM.

Change-Id: I0af08be5b212b7dfe382bba588a6585eb328a038
2012-05-04 22:00:44 -04:00
Michael Sevakis
56f17c4164 Make rbcodec/dsp includes more specific.
Change-Id: Idb6af40df26f5b8499a40e8b98602261ef227044
2012-04-29 17:31:30 -04:00
Michael Sevakis
c9bcbe202d Fundamentally rewrite much of the audio DSP.
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>
2012-04-29 10:00:56 +02:00