Commit graph

50 commits

Author SHA1 Message Date
Chiwen Chang
3ae0f32ac3 three new DSPs
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
2015-01-19 19:34:01 +01:00
Ryan Billing
d0918b98fa DSP Compressor: Sidechain, Exponential Atk/Rls
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>
2013-12-15 22:24:08 +01:00
Michael Sevakis
b1209d4789 Fix FS#12889 : Audible pop right after setting Repeat/Shuffle
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
2013-08-16 09:28:36 -04:00
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 Giacomelli
d475dd36a3 Remove old EQ presets.
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
2013-06-21 16:53:02 +02:00
Michael Sevakis
46688a60db Missed removing a couple unwanted includes in previous commit.
Get those too.

Change-Id: Id2a39afe7a61d6ec0cea38633b94fe1b7122204f
2013-05-27 03:40:02 -04:00
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
6e211ab3ac Remove dsp_callback because DSP is now library code, not app code.
Yep, nope, not necessary anymore. Just call functions directly.

Change-Id: I21dc35f8d674c2a9c8379b7cebd5613c1f05b5eb
2013-05-23 14:25:37 -04:00
Michael Sevakis
a7dee7f447 Introduce new hermite polynomial resampler.
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>
2013-05-16 18:52:21 +02:00
Michael Sevakis
fce81a8a74 Rename all the "lin_resample..." stuff to simply "resample_...".
Change-Id: I79f44f0dcc1b23b33a5040795220713660a1d18a
2013-05-07 00:35:46 -04:00
Michael Sevakis
3fd25dcbed Purge the usage of DSP_SWITCH_FREQUENCY.
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
2013-05-05 00:48:40 -04:00
Michael Sevakis
1a4acc9d1e Fix missed optimization opportunity in dsp_process.
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
2013-05-04 14:23:21 -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
0c7b787398 Straighten out the mad twisted state of sound.c and related areas.
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
2013-04-15 12:02:05 -04:00
Michael Sevakis
f5a5b94686 Implement universal in-PCM-driver software volume control.
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>
2013-04-11 22:55:16 +02:00
Jonathan Gordon
1eb17dc9f4 EQ settings: Rework the settings to clean up the config file.
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>
2013-02-09 13:05:32 +01:00
Hayden Pearce
d73c20933b 10 Band EQ w/Presets
- 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>
2013-01-29 06:53:41 +01:00
Michael Sevakis
362ade3892 Fix FS#12794 - new EQ code does not compile for the Nokia N8x0
The old GCC version currently required (sbox-arm-linux-gcc 3.4.4
release) apparently has trouble with function pointers used as
static array initializers when using indexed initializers + ranges
(ie. [A ... B] = fn).

Change-Id: I494c2b607e4d93a9893264749d0ac257fb54ce3b
2012-12-28 14:12:18 -05:00
Bertrik Sikken
afc96087f8 New crossfeed algorithm for Rockbox: "Meier" crossfeed
Emulates the basic "Meier" crossfeed (2 capacitors, 3 resistors)
as discussed in
http://www.meier-audio.homepage.t-online.de/passivefilter.htm

This crossfeed blends a bit of low-pass filtered L signal into
the R signal (and vice versa) while adding about 300 us delay
to the crossfed-signal. A difference with the crossfeed already
present in rockbox, is that this algorithm keeps the total
spectrum flat (the one currently in rockbox accentuates
low-frequency signals, making it sound a bit muffled).

This implementation is quite lightweight, just 3 multiplies per
left-right pair of samples. Has a default C implementation and
optimized assembly versions for ARM and Coldfire.

The crossfeed effect is quite subtle and is noticeable mostly
one albums that have very strong left-right separation (e.g.
one instrument only on the left, another only on the right).

In the user interface, the new crossfeed option appears as
"Meier" and is not configureable. The existing crossfeed is
renamed to "Custom" as it allows itself to be customised.

There is no entry for the user manual yet.

Change-Id: Iaa100616fe0fcd7e16f08cdb9a7f41501973eee1
2012-05-28 11:34:15 +02:00
Michael Sevakis
29cfd29a6c Stop timestretch freezing things during format changes.
When it was inactive but enabled, the format change hook was dropping
through to code that it shouldn't execute without it also being active
in processing samples.

Change-Id: Ie7899df0395d3f0d10f2bf2b55ea549dd06749a7
2012-05-21 17:12:04 -04:00
Michael Giacomelli
b154e51168 Revert "Work in progress hermite resampler."
This reverts commit f358228ea1.
2012-05-20 01:11:52 -04:00
Michael Giacomelli
f358228ea1 Work in progress hermite resampler.
Based on http://src.gnu-darwin.org/ports/multimedia/helixplayer/work/hxplay-1.0.7/audio/resampler/hermite.c

Change-Id: Id87565a060aa2383701e7c2f3ea023c7555ad9ef
2012-05-20 01:05:47 -04:00
Thomas Martitz
c9d082f056 dsp_arm: Fix up some .section directives to fix crash on app targets.
This is needed on app targets as e.g. ".section .icode" leads to the
code getting linked to incorrect locations (0x0 in this case).

Change-Id: Ic28c5ae6d4f8001d211d685b5ca92d5ffff0c7b2
2012-05-13 22:27:18 +02:00
Nils Wallménius
2202ed3535 TDSpeed: Fix crackling on some systems
Use memmove instead of memcpy for overlapping copy, fixes
crackling in sims and warble on my system. Native targets
seem to have been unaffected.

Change-Id: I265d4ce373e224581bd2f5ba15c75b473ec231f2
2012-05-12 08:47:10 +02:00
Michael Sevakis
fbe9ccc85c TDSpeed settings to setup call need to be recorded, always.
If the settings, like samplerate, were to go out of range where
timestretch drops out of processing and then go back to the same as
when they were valid, it would fail to switch back on by itelf.

Change-Id: Ic5bcb268540b0db8e0483117b8a5a0ce5c5a9db0
2012-05-11 06:56:16 -04:00
Michael Sevakis
dd59e1d789 TDSpeed: Minor assembly optimization to frame fade on Coldfire.
Makes quite a huge difference to get rid of 64-bit math in a hot
area. Cuts about 12 MHz. Generic routine generates good code on
ARM and asm cuts no instructions there.

Change-Id: I4ac647406006c42004f9f5ab396cbf4e85688854
2012-05-11 03:31:29 -04:00
Michael Sevakis
0e5dd0a9cf TDSpeed: Fix up samples consumed return (FS#12666) + other stuff like...
Wrap up the the stereo case into loops and remove unused calculations
hanging out in tdspeed_update().

A wee little bit of code style and column policing.

Change-Id: I8dd3ab4b3e7e56b55dc00c00f3e32996228cc457
2012-05-10 21:41:26 -04:00
Nils Wallménius
d29a11b7a8 Rename HAVE_PITCHSCREEN to HAVE_PITCHCONTROL
Also move the definition to config.h

Change-Id: I36bb5020c5e06b2344292bc05e8c13ccc7a6a1ff
Reviewed-on: http://gerrit.rockbox.org/234
Reviewed-by: Nils Wallménius <nils@rockbox.org>
2012-05-09 14:32:38 +02:00
Michael Sevakis
d26a35d10b Tweak dsp_format_change_process (default format handler).
Just stop searching if the entry is found (as it should have been).

Change-Id: Id968694e825282d58c8ca4a7789c236f98643a5f
2012-05-08 22:47:51 -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
Nils Wallménius
3f61caa0cd rbcodec: abstract tdspeed buffer allocation
Move code dealing with rockbox specific buflib allocations into a
rockbox specific file and implement buffer allocation with
malloc/free for warble/stand alone lib.
Based on patch by Sean Bartell.

Change-Id: I8cb85dad5890fbd34c1bb26abbb89c0b0f6b55cf
Reviewed-on: http://gerrit.rockbox.org/144
Tested-by: Nils Wallménius <nils@rockbox.org>
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Reviewed-by: Nils Wallménius <nils@rockbox.org>
2012-05-07 10:29:07 +02:00
Michael Sevakis
6fd4780ac4 Correct comments in lin_resample re: what is actually used by asm.
Change-Id: Idd457f3c645c5d469ebc6fab0bfc85e7b8dd56da
2012-05-06 18:20:11 -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
dbe5e5f2df rbcodec: Hooks for target specific functions in dsp_process loop
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>
2012-05-03 23:47:46 +02:00
Michael Sevakis
b4eec0dd42 Make INITDATA_ATTR work on everything that has INIT_ATTR enabled for code.
Change-Id: If9936bfbbd3bc3eb2a3e3e290701b8517eabfb13
2012-05-01 01:28:50 -04:00
Michael Sevakis
f5d9a45e3f Should've had dsp_replaygain_set_gains as static for now...
...because currently gains are only set through dsp_configure.

Change-Id: I2866473a82fdd5f41de4705b45928daa7e43f8eb
2012-04-30 17:51:05 -04:00
Michael Sevakis
8f9e3b10a5 Still need settings.h in dsp_misc.c for now for software volume.
Change-Id: I824e8f9935013f6e2a1db6ccd2db4bd406257057
2012-04-30 17:18:26 -04:00
Michael Sevakis
57a20d2d63 Make DSP's replaygain independent of global_settings.
Moves replaygain definitions to lib/rbcodec/dsp/dsp_misc.h.
Intermediate functions in misc.c handle any adjustment and calling
the rbcodec APIs.

Change-Id: I9f03561bca9aedd13760cf19c4e19aa3c68e7024
Reviewed-on: http://gerrit.rockbox.org/140
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
2012-04-30 22:47:37 +02:00
Michael Sevakis
ae5656a142 Put the <string.h> channel_mode.c for memcpy.
Hmmm, how'd I miss that?

Change-Id: I30d9a1b5f053aad069687aa0a01ebdf40a6b2d76
2012-04-29 17:44:57 -04:00
Michael Sevakis
56f17c4164 Make rbcodec/dsp includes more specific.
Change-Id: Idb6af40df26f5b8499a40e8b98602261ef227044
2012-04-29 17:31:30 -04:00
Michael Sevakis
23b5f3e5e1 Make compressor_update static.
Change-Id: Ic29242b4c397e82c2bee3808492a2d0a9ffebbe6
2012-04-29 14:47:01 -04:00
Michael Sevakis
230f6f4326 Lower IRAM footprint on ARM.
Move a few functions to .text that probably don't see a huge benefit
from being .icode. Will scrutinize later.

Change-Id: I7bdffc326076c5cd7e6a1c57d25d31e653920327
2012-04-29 14:10:14 -04:00
Michael Sevakis
3b578f018c Fix 3g warning in dsp_arm.S (which showed as an error).
Change-Id: Iccbeca66e809413dda90fec36439b4a180b8a879
2012-04-29 04:57:57 -04:00
Michael Sevakis
7cc8bbdaaf Fix no newline at end warning.
Change-Id: I9edb1ebb34f91893b6290d7640fcdaede3434b40
2012-04-29 04:14:11 -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
Michael Sevakis
0048e5b8ce Some whitespace correction in dsp_*.S
Change-Id: I6ee14c0adc9dd456c8a2d171952cbaedb3752428
2012-04-27 16:55:16 -04:00
Sean Bartell
a6dea9e13d rbcodec refactoring: dsp_set_eq_coefs
dsp_set_eq_coefs now has parameters for the band settings, so it doesn't
need to access global_settings.

Change-Id: I29ac19fc353b15a79cb25f0e45132aef0881e4c9
Reviewed-on: http://gerrit.rockbox.org/138
Reviewed-by: Nils Wallménius <nils@rockbox.org>
2012-04-27 16:33:27 +02:00
Michael Sevakis
0842d7f7e1 Consolidate compressor settings into a struct.
Doing that makes things cleaner for later on.

Change-Id: I4e279aa57ace16a348acc0fc09059592325ec95f
2012-04-26 17:19:16 -04:00
Michael Sevakis
e5c3327cef Add a more correct absolute difference function to dsp-util.
Differences between signed samples cover the entire unsigned 32-bit
range. "abs" will think any difference exceeding INT32_MAX is negative
which is not corrent. Test which argument is greater and subtract the
lesser from it, outputting unsigned difference.

Change-Id: I73a8e5e418d49ff73d1a7c98eeb4731946dcfe84
2012-04-26 16:04:43 -04:00
Sean Bartell
b5716df4cb Build librbcodec with DSP and metadata.
All associated files are moved to /lib/rbcodec.

Change-Id: I572ddd2b8a996aae1e98c081d06b1ed356dce222
2012-03-18 12:00:39 +01:00