Nyaaori/rockbox
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
rockbox/lib/rbcodec/dsp/channel_mode.c

252 lines
7.3 KiB
C
Raw Normal View History

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
* 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 "sound.h"
#include "fixedpoint.h"
#include "fracmul.h"
#include "dsp_proc_entry.h"
Make rbcodec/dsp includes more specific. Change-Id: Idb6af40df26f5b8499a40e8b98602261ef227044
2012-04-29 21:31:30 +00:00
#include "channel_mode.h"
Put the <string.h> channel_mode.c for memcpy. Hmmm, how'd I miss that? Change-Id: I30d9a1b5f053aad069687aa0a01ebdf40a6b2d76
2012-04-29 21:44:57 +00:00
#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
#if 0
/* SOUND_CHAN_STEREO mode is a noop so has no function - just outline one for
* completeness. */
void channel_mode_proc_stereo(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
#endif
void channel_mode_proc_mono(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
void channel_mode_proc_mono_left(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
void channel_mode_proc_mono_right(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
void channel_mode_proc_custom(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
void channel_mode_proc_karaoke(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p);
static struct channel_mode_data
{
long sw_gain; /* 00h: for mode: custom */
long sw_cross; /* 04h: for mode: custom */
int mode;
} channel_mode_data =
{
.sw_gain = 0,
.sw_cross = 0,
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
.mode = SOUND_CHAN_STEREO
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 0
/* SOUND_CHAN_STEREO mode is a noop so has no function - just outline one for
* completeness. */
void channel_mode_proc_stereo(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
/* The channels are each just themselves */
(void)this; (void)buf_p;
}
#endif
#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
/* Unoptimized routines */
void channel_mode_proc_mono(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *buf = *buf_p;
int32_t *sl = buf->p32[0];
int32_t *sr = buf->p32[1];
int count = buf->remcount;
do
{
int32_t lr = *sl / 2 + *sr / 2;
*sl++ = lr;
*sr++ = lr;
}
while (--count > 0);
(void)this;
}
void channel_mode_proc_custom(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
struct channel_mode_data *data = (void *)this->data;
struct dsp_buffer *buf = *buf_p;
int32_t *sl = buf->p32[0];
int32_t *sr = buf->p32[1];
int count = buf->remcount;
const int32_t gain = data->sw_gain;
const int32_t cross = data->sw_cross;
do
{
int32_t l = *sl;
int32_t r = *sr;
*sl++ = FRACMUL(l, gain) + FRACMUL(r, cross);
*sr++ = FRACMUL(r, gain) + FRACMUL(l, cross);
}
while (--count > 0);
}
void channel_mode_proc_karaoke(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *buf = *buf_p;
int32_t *sl = buf->p32[0];
int32_t *sr = buf->p32[1];
int count = buf->remcount;
do
{
int32_t ch = *sl / 2 - *sr / 2;
*sl++ = ch;
*sr++ = -ch;
}
while (--count > 0);
(void)this;
}
#endif /* CPU */
void channel_mode_proc_mono_left(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
/* Just copy over the other channel */
struct dsp_buffer *buf = *buf_p;
memcpy(buf->p32[1], buf->p32[0], buf->remcount * sizeof (int32_t));
(void)this;
}
void channel_mode_proc_mono_right(struct dsp_proc_entry *this,
struct dsp_buffer **buf_p)
{
/* Just copy over the other channel */
struct dsp_buffer *buf = *buf_p;
memcpy(buf->p32[0], buf->p32[1], buf->remcount * sizeof (int32_t));
(void)this;
}
void channel_mode_set_config(int value)
{
if (value < 0 || value >= SOUND_CHAN_NUM_MODES)
value = SOUND_CHAN_STEREO; /* Out of range */
if (value == channel_mode_data.mode)
return;
channel_mode_data.mode = value;
dsp_proc_enable(dsp_get_config(CODEC_IDX_AUDIO), DSP_PROC_CHANNEL_MODE,
value != SOUND_CHAN_STEREO);
}
void channel_mode_custom_set_width(int value)
{
long width, straight, cross;
width = value * 0x7fffff / 100;
if (value <= 100)
{
straight = (0x7fffff + width) / 2;
cross = straight - width;
}
else
{
/* straight = (1 + width) / (2 * width) */
straight = fp_div(0x7fffff + width, width, 22);
cross = straight - 0x7fffff;
}
channel_mode_data.sw_gain = straight << 8;
channel_mode_data.sw_cross = cross << 8;
}
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
static void update_process_fn(struct dsp_proc_entry *this)
{
static const dsp_proc_fn_type fns[SOUND_CHAN_NUM_MODES] =
{
[SOUND_CHAN_STEREO] = NULL,
[SOUND_CHAN_MONO] = channel_mode_proc_mono,
[SOUND_CHAN_CUSTOM] = channel_mode_proc_custom,
[SOUND_CHAN_MONO_LEFT] = channel_mode_proc_mono_left,
[SOUND_CHAN_MONO_RIGHT] = channel_mode_proc_mono_right,
[SOUND_CHAN_KARAOKE] = channel_mode_proc_karaoke,
};
this->process = fns[((struct channel_mode_data *)this->data)->mode];
}
/* Handle format changes and verify the format compatibility */
static intptr_t channel_mode_new_format(struct dsp_proc_entry *this,
struct dsp_config *dsp,
struct sample_format *format)
{
rbcodec: fix compilation in debug mode Change-Id: I124cf59c641c2e161cc147b031d9bef5ef773dfb
2017-07-30 12:31:05 +00:00
DSP_PRINT_FORMAT(DSP_PROC_CHANNEL_MODE, *format);
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
bool active = format->num_channels >= 2;
dsp_proc_activate(dsp, DSP_PROC_CHANNEL_MODE, active);
if (active)
return PROC_NEW_FORMAT_OK;
/* Can't do this. Sleep until next change. */
DEBUGF(" DSP_PROC_CHANNEL_MODE- deactivated\n");
return PROC_NEW_FORMAT_DEACTIVATED;
(void)this;
}
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 channel_mode_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:
if (value == 0)
this->data = (intptr_t)&channel_mode_data;
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);
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
case DSP_PROC_NEW_FORMAT:
retval = channel_mode_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(
CHANNEL_MODE,
channel_mode_configure);
Reference in a new issue Copy permalink