rockbox/lib/rbcodec/dsp/dsp_sample_output.c
Aidan MacDonald 8165a6c245 rbcodec dsp: Remove INIT_ATTR from the DSP library
All of these are technically unsafe cross-section references but
most aren't reported by the linker, probably due to inlining. In
practice there was no problem because the affected code was only
run at init time anyway.

For now, remove INIT_ATTR until the init code can be refactored
to avoid the problematic references. This should also save code
size by moving more code to the init section.

dsp_init() gets to keep its attribute because it's already OK.

Change-Id: Idc9ac0e02cb07f31d186686e0382275c02a85dbb
2022-12-18 22:23:52 +00:00

213 lines
6.5 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
* 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.
*
****************************************************************************/
#include "rbcodecconfig.h"
#include "platform.h"
#include "dsp_core.h"
#include "dsp_sample_io.h"
#include "dsp_proc_entry.h"
#include "dsp-util.h"
#include <string.h>
#if 0
#include <debug.h>
#else
#undef DEBUGF
#define DEBUGF(...)
#endif
/* May be implemented in here or externally.*/
void sample_output_mono(struct sample_io_data *this,
struct dsp_buffer *src, struct dsp_buffer *dst);
void sample_output_stereo(struct sample_io_data *this,
struct dsp_buffer *src, struct dsp_buffer *dst);
void sample_output_dithered(struct sample_io_data *this,
struct dsp_buffer *src, struct dsp_buffer *dst);
/** Sample output **/
#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
/* write mono internal format to output format */
void sample_output_mono(struct sample_io_data *this,
struct dsp_buffer *src, struct dsp_buffer *dst)
{
int count = this->outcount;
const int32_t *s0 = src->p32[0];
int16_t *d = dst->p16out;
int scale = src->format.output_scale;
int32_t dc_bias = 1L << (scale - 1);
do
{
int32_t lr = clip_sample_16((*s0++ + dc_bias) >> scale);
*d++ = lr;
*d++ = lr;
}
while (--count > 0);
}
/* write stereo internal format to output format */
void sample_output_stereo(struct sample_io_data *this,
struct dsp_buffer *src, struct dsp_buffer *dst)
{
int count = this->outcount;
const int32_t *s0 = src->p32[0];
const int32_t *s1 = src->p32[1];
int16_t *d = dst->p16out;
int scale = src->format.output_scale;
int32_t dc_bias = 1L << (scale - 1);
do
{
*d++ = clip_sample_16((*s0++ + dc_bias) >> scale);
*d++ = clip_sample_16((*s1++ + dc_bias) >> scale);
}
while (--count > 0);
}
#endif /* CPU */
/**
* The "dither" code to convert the 24-bit samples produced by libmad was
* taken from the coolplayer project - coolplayer.sourceforge.net
*
* This function handles mono and stereo outputs.
*/
static struct dither_data
{
struct dither_state
{
long error[3]; /* 00h: error term history */
long random; /* 0ch: last random value */
} state[2]; /* 0=left, 1=right */
bool enabled; /* 20h: dithered output enabled */
/* 24h */
} dither_data IBSS_ATTR;
void sample_output_dithered(struct sample_io_data *this,
struct dsp_buffer *src, struct dsp_buffer *dst)
{
int count = this->outcount;
int channels = src->format.num_channels;
int scale = src->format.output_scale;
int32_t dc_bias = 1L << (scale - 1); /* 1/2 bit of significance */
int32_t mask = (1L << scale) - 1; /* Mask of bits quantized away */
for (int ch = 0; ch < channels; ch++)
{
struct dither_state *dither = &dither_data.state[ch];
const int32_t *s = src->p32[ch];
int16_t *d = &dst->p16out[ch];
for (int i = 0; i < count; i++, s++, d += 2)
{
/* Noise shape and bias (for correct rounding later) */
int32_t sample = *s;
sample += dither->error[0] - dither->error[1] + dither->error[2];
dither->error[2] = dither->error[1];
dither->error[1] = dither->error[0] / 2;
int32_t output = sample + dc_bias;
/* Dither, highpass triangle PDF */
int32_t random = dither->random*0x0019660dL + 0x3c6ef35fL;
output += (random & mask) - (dither->random & mask);
dither->random = random;
/* Quantize sample to output range */
output >>= scale;
/* Error feedback of quantization */
dither->error[0] = sample - (output << scale);
/* Clip and store */
*d = clip_sample_16(output);
}
}
if (channels > 1)
return;
/* Have to duplicate left samples into the right channel since
output is interleaved stereo */
int16_t *d = dst->p16out;
do
{
int16_t s = *d++;
*d++ = s;
}
while (--count > 0);
}
/* Initialize the output function for settings and format */
void dsp_sample_output_format_change(struct sample_io_data *this,
struct sample_format *format)
{
static const sample_output_fn_type fns[2][2] =
{
{ sample_output_mono, /* DC-biased quantizing */
sample_output_stereo },
{ sample_output_dithered, /* Tri-PDF dithering */
sample_output_dithered },
};
bool dither = dsp_get_id((void *)this) == CODEC_IDX_AUDIO &&
dither_data.enabled;
int channels = format->num_channels;
DSP_PRINT_FORMAT(DSP Output, *format);
this->output_samples = fns[dither ? 1 : 0][channels - 1];
this->output_version = format->version;
}
void dsp_sample_output_init(struct sample_io_data *this)
{
this->output_version = 0;
this->output_samples = sample_output_stereo;
}
/* Flush the dither history */
void dsp_sample_output_flush(struct sample_io_data *this)
{
if (dsp_get_id((void *)this) == CODEC_IDX_AUDIO)
memset(dither_data.state, 0, sizeof (dither_data.state));
}
/** Output settings **/
/* Set the tri-pdf dithered output */
void dsp_dither_enable(bool enable)
{
if (enable == dither_data.enabled)
return;
dither_data.enabled = enable;
struct sample_io_data *data = (void *)dsp_get_config(CODEC_IDX_AUDIO);
if (enable)
dsp_sample_output_flush(data);
data->output_version = 0; /* Force format update */
}