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rockbox/lib/rbcodec/dsp/crossfeed.c

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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>
2012-03-27 23:52:15 +00:00
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2006 Thom Johansen
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
2012-05-01 07:58:27 +00:00
* Copyright (C) 2010 Bertrik Sikken
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>
2012-03-27 23:52:15 +00:00
* Copyright (C) 2012 Michael Sevakis
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
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 07:23:33 +00:00
#include "rbcodecconfig.h"
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>
2012-03-27 23:52:15 +00:00
#include "fixedpoint.h"
#include "fracmul.h"
#include "replaygain.h"
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>
2013-05-23 17:58:51 +00:00
#include "dsp_misc.h"
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>
2012-03-27 23:52:15 +00:00
#include "dsp_proc_entry.h"
Make rbcodec/dsp includes more specific. Change-Id: Idb6af40df26f5b8499a40e8b98602261ef227044
2012-04-29 21:31:30 +00:00
#include "dsp_filter.h"
#include "crossfeed.h"
#include <string.h>
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>
2012-03-27 23:52:15 +00:00
/* Implemented here or in target assembly code */
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
2012-05-01 07:58:27 +00:00
void crossfeed_process(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
void crossfeed_meier_process(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
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>
2012-03-27 23:52:15 +00:00
/**
* Applies crossfeed to the stereo signal.
*
* Crossfeed is a process where listening over speakers is simulated. This
* is good for old hard panned stereo records, which might be quite fatiguing
* to listen to on headphones with no crossfeed.
*/
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>
2013-05-23 17:58:51 +00:00
#define DELAY_LEN(fs) ((300*(fs) / 1000000)*2) /* ~300 uS */
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>
2012-03-27 23:52:15 +00:00
/* Crossfeed */
static struct crossfeed_state
{
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
2012-05-01 07:58:27 +00:00
union
{
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>
2013-05-23 17:58:51 +00:00
struct /* Data for meier crossfeed */
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
2012-05-01 07:58:27 +00:00
{
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>
2013-05-23 17:58:51 +00:00
int32_t reserved; /* 00h: Reserved: overlaps gain */
int32_t vcl; /* 04h: Left filter output */
int32_t vcr; /* 08h: Right filter output */
int32_t vdiff; /* 0ch: L-R difference signal */
int32_t coef1; /* 10h: Left/right filter coef */
int32_t coef2; /* 14h: Crossfeed filter coef */
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
2012-05-01 07:58:27 +00:00
};
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>
2013-05-23 17:58:51 +00:00
struct /* Data for custom crossfeed */
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
2012-05-01 07:58:27 +00:00
{
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>
2013-05-23 17:58:51 +00:00
int32_t gain; /* 00h: Direct path gain */
int32_t coefs[3]; /* 04h: Filter coefficients: b0, b1, a1 */
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
2012-05-01 07:58:27 +00:00
int32_t history[4]; /* 10h: Format is x[n - 1], y[n - 1] (L + R) */
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>
2013-05-23 17:58:51 +00:00
int32_t *index; /* 20h: Current pointer into the delay line */
int32_t *index_max; /* 24h: Current max pointer of delay line */
/* 28h: Delay line buffer (L + R interleaved) */
int32_t delay[DELAY_LEN(DSP_OUT_MAX_HZ)]; /* Target-dependent size */
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
2012-05-01 07:58:27 +00:00
};
};
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>
2012-03-27 23:52:15 +00:00
} crossfeed_state IBSS_ATTR;
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
2012-05-01 07:58:27 +00:00
static int crossfeed_type = CROSSFEED_TYPE_NONE;
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>
2013-05-23 17:58:51 +00:00
/* Cached custom settings */
static long crossfeed_lf_gain;
static long crossfeed_hf_gain;
static long crossfeed_cutoff;
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
2012-05-01 07:58:27 +00:00
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>
2012-03-27 23:52:15 +00:00
/* Discard the sample histories */
static void crossfeed_flush(struct dsp_proc_entry *this)
{
struct crossfeed_state *state = (void *)this->data;
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
2012-05-01 07:58:27 +00:00
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
2012-12-19 22:34:57 +00:00
if (crossfeed_type != CROSSFEED_TYPE_CUSTOM)
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
2012-05-01 07:58:27 +00:00
{
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
2012-12-19 22:34:57 +00:00
state->vcl = state->vcr = state->vdiff = 0;
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
2012-05-01 07:58:27 +00:00
}
else
{
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>
2013-05-23 17:58:51 +00:00
memset(state->history, 0, sizeof (state->history));
memset(state->delay, 0, sizeof (state->delay));
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
2012-12-19 22:34:57 +00:00
state->index = state->delay;
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
2012-05-01 07:58:27 +00:00
}
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>
2012-03-27 23:52:15 +00:00
}
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>
2013-05-23 17:58:51 +00:00
static void crossfeed_meier_update_filter(struct crossfeed_state *state,
unsigned int fout)
{
/* 1 / (F.Rforward.C) */
state->coef1 = fp_div(2128, fout, 31);
/* 1 / (F.Rcross.C) */
state->coef2 = fp_div(1000, fout, 31);
}
static void crossfeed_custom_update_filter(struct crossfeed_state *state,
unsigned int fout)
{
long lf_gain = crossfeed_lf_gain;
long hf_gain = crossfeed_hf_gain;
long cutoff = crossfeed_cutoff;
int32_t *c = state->coefs;
long scaler = get_replaygain_int(lf_gain * 10) << 7;
cutoff = fp_div(cutoff, fout, 32);
hf_gain -= lf_gain;
/* Divide cutoff by sqrt(10^(hf_gain/20)) to place cutoff at the -3 dB
* point instead of shelf midpoint. This is for compatibility with the old
* crossfeed shelf filter and should be removed if crossfeed settings are
* ever made incompatible for any other good reason.
*/
cutoff = fp_div(cutoff, get_replaygain_int(hf_gain*5), 24);
filter_shelf_coefs(cutoff, hf_gain, false, c);
/* Scale coefs by LF gain and shift them to s0.31 format. We have no gains
* over 1 and can do this safely
*/
c[0] = FRACMUL_SHL(c[0], scaler, 4);
c[1] = FRACMUL_SHL(c[1], scaler, 4);
c[2] <<= 4;
}
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>
2012-03-27 23:52:15 +00:00
/** DSP interface **/
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
2012-05-01 07:58:27 +00:00
/* Set the type of crossfeed to use */
void dsp_set_crossfeed_type(int type)
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>
2012-03-27 23:52:15 +00:00
{
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
2012-05-01 07:58:27 +00:00
if (type == crossfeed_type)
return; /* No change */
crossfeed_type = type;
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>
2012-03-27 23:52:15 +00:00
struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
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
2012-05-01 07:58:27 +00:00
dsp_proc_enable(dsp, DSP_PROC_CROSSFEED, type != CROSSFEED_TYPE_NONE);
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>
2012-03-27 23:52:15 +00:00
}
/* Set the gain of the dry mix */
void dsp_set_crossfeed_direct_gain(int gain)
{
uint32_t gain32 = get_replaygain_int(gain * 10);
crossfeed_state.gain =
gain32 >= (0x80000000ul >> 7) ? 0x7ffffffful: (gain32 << 7);
}
/* Both gains should be below 0 dB */
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
{
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>
2013-05-23 17:58:51 +00:00
crossfeed_lf_gain = lf_gain;
crossfeed_hf_gain = hf_gain;
crossfeed_cutoff = cutoff;
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>
2012-03-27 23:52:15 +00:00
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 13:28:36 +00:00
if (crossfeed_type != CROSSFEED_TYPE_CUSTOM)
return;
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>
2013-05-23 17:58:51 +00:00
struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
crossfeed_custom_update_filter(&crossfeed_state,
dsp_get_output_frequency(dsp));
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>
2012-03-27 23:52:15 +00:00
}
#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
/* Apply the crossfade to the buffer in place */
void crossfeed_process(struct dsp_proc_entry *this, struct dsp_buffer **buf_p)
{
struct crossfeed_state *state = (void *)this->data;
struct dsp_buffer *buf = *buf_p;
int32_t *hist_l = &state->history[0];
int32_t *hist_r = &state->history[2];
int32_t *delay = state->delay;
int32_t *coefs = &state->coefs[0];
int32_t gain = state->gain;
int32_t *di = state->index;
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>
2013-05-23 17:58:51 +00:00
int32_t *di_max = state->index_max;
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>
2012-03-27 23:52:15 +00:00
int count = buf->remcount;
for (int i = 0; i < count; i++)
{
int32_t left = buf->p32[0][i];
int32_t right = buf->p32[1][i];
/* Filter delayed sample from left speaker */
int32_t acc = FRACMUL(*di, coefs[0]);
acc += FRACMUL(hist_l[0], coefs[1]);
acc += FRACMUL(hist_l[1], coefs[2]);
/* Save filter history for left speaker */
hist_l[1] = acc;
hist_l[0] = *di;
*di++ = left;
/* Filter delayed sample from right speaker */
acc = FRACMUL(*di, coefs[0]);
acc += FRACMUL(hist_r[0], coefs[1]);
acc += FRACMUL(hist_r[1], coefs[2]);
/* Save filter history for right speaker */
hist_r[1] = acc;
hist_r[0] = *di;
*di++ = right;
/* Now add the attenuated direct sound and write to outputs */
buf->p32[0][i] = FRACMUL(left, gain) + hist_r[1];
buf->p32[1][i] = FRACMUL(right, gain) + hist_l[1];
/* Wrap delay line index if bigger than delay line size */
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>
2013-05-23 17:58:51 +00:00
if (di >= di_max)
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>
2012-03-27 23:52:15 +00:00
di = delay;
}
/* Write back local copies of data we've modified */
state->index = di;
}
#endif /* CPU */
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
2012-05-01 07:58:27 +00:00
#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
/**
* Implementation of the "simple" passive crossfeed circuit by Jan Meier.
* See also: http://www.meier-audio.homepage.t-online.de/passivefilter.htm
*/
void crossfeed_meier_process(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *buf = *buf_p;
/* Get filter state */
struct crossfeed_state *state = (struct crossfeed_state *)this->data;
int32_t vcl = state->vcl;
int32_t vcr = state->vcr;
int32_t vdiff = state->vdiff;
int32_t coef1 = state->coef1;
int32_t coef2 = state->coef2;
int count = buf->remcount;
for (int i = 0; i < count; i++)
{
/* Calculate new output */
int32_t lout = buf->p32[0][i] + vcl;
int32_t rout = buf->p32[1][i] + vcr;
buf->p32[0][i] = lout;
buf->p32[1][i] = rout;
/* Update filter state */
int32_t common = FRACMUL(vdiff, coef2);
vcl -= FRACMUL(vcl, coef1) + common;
vcr -= FRACMUL(vcr, coef1) - common;
vdiff = lout - rout;
}
/* Store filter state */
state->vcl = vcl;
state->vcr = vcr;
state->vdiff = vdiff;
}
#endif /* CPU */
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
2012-12-19 22:34:57 +00:00
/* Update the processing function according to crossfeed type */
static void update_process_fn(struct dsp_proc_entry *this,
struct dsp_config *dsp)
{
struct crossfeed_state *state = (struct crossfeed_state *)this->data;
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>
2013-05-23 17:58:51 +00:00
dsp_proc_fn_type fn;
unsigned int fout = dsp_get_output_frequency(dsp);
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
2012-12-19 22:34:57 +00:00
if (crossfeed_type != CROSSFEED_TYPE_CUSTOM)
{
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>
2013-05-23 17:58:51 +00:00
crossfeed_meier_update_filter(state, fout);
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
2012-12-19 22:34:57 +00:00
fn = crossfeed_meier_process;
}
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>
2013-05-23 17:58:51 +00:00
else
{
state->index_max = state->delay + DELAY_LEN(fout);
crossfeed_custom_update_filter(state, fout);
fn = crossfeed_process;
}
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
2012-12-19 22:34:57 +00:00
if (this->process != fn)
{
this->process = fn; /* Set proper function */
if (dsp_proc_active(dsp, DSP_PROC_CROSSFEED))
crossfeed_flush(this);
}
}
/* Crossfeed boot/format change function */
static intptr_t crossfeed_new_format(struct dsp_proc_entry *this,
struct dsp_config *dsp,
struct sample_format *format)
{
DSP_PRINT_FORMAT(DSP_PROC_CROSSFEED, format);
bool was_active = dsp_proc_active(dsp, DSP_PROC_CROSSFEED);
bool active = format->num_channels >= 2;
dsp_proc_activate(dsp, DSP_PROC_CROSSFEED, active);
if (active)
{
if (!was_active)
crossfeed_flush(this); /* Going online */
return PROC_NEW_FORMAT_OK;
}
/* Can't do this. Sleep until next change */
DEBUGF(" DSP_PROC_CROSSFEED- deactivated\n");
return PROC_NEW_FORMAT_DEACTIVATED;
}
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>
2012-03-27 23:52:15 +00:00
/* DSP message hook */
static intptr_t crossfeed_configure(struct dsp_proc_entry *this,
struct dsp_config *dsp,
unsigned int setting,
intptr_t value)
{
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
2012-12-19 22:34:57 +00:00
intptr_t retval = 0;
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>
2012-03-27 23:52:15 +00:00
switch (setting)
{
case DSP_PROC_INIT:
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
2012-05-01 07:58:27 +00:00
if (value == 0)
this->data = (intptr_t)&crossfeed_state;
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>
2013-05-23 17:58:51 +00:00
case DSP_SET_OUT_FREQUENCY:
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
2012-12-19 22:34:57 +00:00
update_process_fn(this, dsp);
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
2012-05-01 07:58:27 +00:00
break;
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>
2012-03-27 23:52:15 +00:00
case DSP_FLUSH:
crossfeed_flush(this);
break;
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
2012-12-19 22:34:57 +00:00
case DSP_PROC_NEW_FORMAT:
retval = crossfeed_new_format(this, dsp,
(struct sample_format *)value);
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>
2012-03-27 23:52:15 +00:00
break;
}
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
2012-12-19 22:34:57 +00:00
return retval;
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>
2012-03-27 23:52:15 +00:00
}
/* Database entry */
DSP_PROC_DB_ENTRY(
CROSSFEED,
crossfeed_configure);
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