rockbox/lib/rbcodec/dsp/dsp.h
Michael Sevakis c9bcbe202d Fundamentally rewrite much of the audio DSP.
Creates a standard buffer passing, local data passing and messaging
system for processing stages. Stages can be moved to their own source
files to reduce clutter and ease assimilation of new ones. dsp.c
becomes dsp_core.c which supports an engine and framework for effects.

Formats and change notifications are passed along with the buffer so
that they arrive at the correct time at each stage in the chain
regardless of the internal delays of a particular one.

Removes restrictions on the number of samples that can be processed at
a time and it pays attention to destination buffer size restrictions
without having to limit input count, which also allows pcmbuf to
remain fuller and safely set its own buffer limits as it sees fit.
There is no longer a need to query input/output counts given a certain
number of input samples; just give it the sizes of the source and
destination buffers.

Works in harmony with stages that are not deterministic in terms of
sample input/output ratio (like both resamplers but most notably
the timestretch). As a result it fixes quirks with timestretch hanging
up with certain settings and it now operates properly throughout its
full settings range.
Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734
Reviewed-on: http://gerrit.rockbox.org/200
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-04-29 10:00:56 +02:00

176 lines
6 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
*
* 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.
*
****************************************************************************/
#ifndef _DSP_H
#define _DSP_H
struct dsp_config;
/* Include all this junk here for now */
#include "dsp_proc_settings.h"
enum dsp_ids
{
CODEC_IDX_AUDIO,
CODEC_IDX_VOICE,
DSP_COUNT,
};
enum dsp_settings
{
DSP_INIT, /* For dsp_init */
DSP_RESET,
DSP_SET_FREQUENCY,
DSP_SWITCH_FREQUENCY = DSP_SET_FREQUENCY, /* deprecated */
DSP_SET_SAMPLE_DEPTH,
DSP_SET_STEREO_MODE,
DSP_FLUSH,
DSP_PROC_INIT,
DSP_PROC_CLOSE,
DSP_PROC_SETTING, /* stage-specific should be this + id */
};
#define NATIVE_FREQUENCY 44100 /* internal/output sample rate */
enum dsp_stereo_modes
{
STEREO_INTERLEAVED,
STEREO_NONINTERLEAVED,
STEREO_MONO,
STEREO_NUM_MODES,
};
/* Format into for the buffer (if .valid == true) */
struct sample_format
{
uint8_t changed; /* 00h: 0=no change, 1=changed (is also index) */
uint8_t num_channels; /* 01h: number of channels of data */
uint8_t frac_bits; /* 02h: number of fractional bits */
uint8_t output_scale; /* 03h: output scaling shift */
int32_t frequency; /* 04h: pitch-adjusted sample rate */
int32_t codec_frequency; /* 08h: codec-specifed sample rate */
/* 0ch */
};
/* Compare format data only */
#define EQU_SAMPLE_FORMAT(f1, f2) \
(!memcmp(&(f1).num_channels, &(f2).num_channels, \
sizeof (f1) - sizeof ((f1).changed)))
static inline void format_change_set(struct sample_format *f)
{ f->changed = 1; }
static inline void format_change_ack(struct sample_format *f)
{ f->changed = 0; }
/* Used by ASM routines - keep field order or else fix the functions */
struct dsp_buffer
{
int32_t remcount; /* 00h: Samples in buffer (In, Int, Out) */
union
{
const void *pin[2]; /* 04h: Channel pointers (In) */
int32_t *p32[2]; /* 04h: Channel pointers (Int) */
int16_t *p16out; /* 04h: DSP output buffer (Out) */
};
union
{
uint32_t proc_mask; /* 0Ch: In-place effects already appled to buffer
in order to avoid double-processing. Set
to zero on new buffer before passing to
DSP. */
int bufcount; /* 0Ch: Buffer length/dest buffer remaining
Basically, pay no attention unless it's
*your* new buffer and is used internally
or is specifically the final output
buffer. */
};
struct sample_format format; /* 10h: Buffer format data */
/* 1ch */
};
/* Remove samples from input buffer (In). Sample size is specified.
Provided to dsp_process(). */
static inline void dsp_advance_buffer_input(struct dsp_buffer *buf,
int by_count,
size_t size_each)
{
buf->remcount -= by_count;
buf->pin[0] += by_count * size_each;
buf->pin[1] += by_count * size_each;
}
/* Add samples to output buffer and update remaining space (Out).
Provided to dsp_process() */
static inline void dsp_advance_buffer_output(struct dsp_buffer *buf,
int by_count)
{
buf->bufcount -= by_count;
buf->remcount += by_count;
buf->p16out += 2 * by_count; /* Interleaved stereo */
}
/* Remove samples from internal input buffer (In, Int).
Provided to dsp_process() or by another processing stage. */
static inline void dsp_advance_buffer32(struct dsp_buffer *buf,
int by_count)
{
buf->remcount -= by_count;
buf->p32[0] += by_count;
buf->p32[1] += by_count;
}
/** For use by processing stages **/
#define DSP_PRINT_FORMAT(name, id, format) \
DEBUGF("DSP format- " #name "\n" \
" id:%d chg:%c ch:%u fb:%u os:%u hz:%u chz:%u\n", \
(int)id, \
(format).changed ? 'y' : 'n', \
(unsigned int)(format).num_channels, \
(unsigned int)(format).frac_bits, \
(unsigned int)(format).output_scale, \
(unsigned int)(format).frequency, \
(unsigned int)(format).codec_frequency);
/* Get DSP pointer */
struct dsp_config * dsp_get_config(enum dsp_ids id);
/* Get DSP id */
enum dsp_ids dsp_get_id(const struct dsp_config *dsp);
#if 0 /* Not needed now but enable if something must know this */
/* Is the DSP processing a buffer? */
bool dsp_is_busy(const struct dsp_config *dsp);
#endif /* 0 */
/** General DSP processing **/
/* Process the given buffer - see implementation in dsp.c for more */
void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src,
struct dsp_buffer *dst);
/* Change DSP settings */
intptr_t dsp_configure(struct dsp_config *dsp, unsigned int setting,
intptr_t value);
/* One-time startup init that must come before settings reset/apply */
void dsp_init(void);
#endif /* _DSP_H */