rockbox/lib/rbcodec/dsp/dsp_sample_input.c
Aidan MacDonald 34a092a997 rbcodec dsp: Replace enum dsp_ids arguments with unsigned int
Because casting to and from "enum dsp_id" just adds noise,
change everything to unsigned int.

Change-Id: I52a7ae55f406e673d5b811b29657fcdc4b62ab10
2022-12-22 18:00:37 -05:00

268 lines
7.6 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 "fixedpoint.h"
#include "gcc_extensions.h"
#include "dsp_core.h"
#include "dsp_sample_io.h"
#include "dsp_proc_entry.h"
#if 0
#undef DEBUGF
#define DEBUGF(...)
#endif
/* The internal format is 32-bit samples, non-interleaved, stereo. This
* format is similar to the raw output from several codecs, so no copying is
* needed for that case.
*
* Note that for mono, dst[0] equals dst[1], as there is no point in
* processing the same data twice nor should it be done when modifying
* samples in-place.
*
* When conversion is required:
* Updates source buffer to point past the samples "consumed" also consuming
* that portion of the input buffer and the destination is set to the buffer
* of samples for later stages to consume.
*
* Input operates similarly to how an out-of-place processing stage should
* behave.
*/
#define SAMPLE_BUF_COUNT 128 /* Per channel, per DSP */
/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
static int32_t sample_bufs[3][SAMPLE_BUF_COUNT] IBSS_ATTR;
/* inline helper to setup buffers when conversion is required */
static FORCE_INLINE int sample_input_setup(struct sample_io_data *this,
struct dsp_buffer **buf_p,
int channels,
struct dsp_buffer **src,
struct dsp_buffer **dst)
{
struct dsp_buffer *s = *buf_p;
struct dsp_buffer *d = *dst = &this->sample_buf;
*buf_p = d;
if (d->remcount > 0)
return 0; /* data still remains */
*src = s;
int count = MIN(s->remcount, SAMPLE_BUF_COUNT);
d->remcount = count;
d->p32[0] = this->sample_buf_p[0];
d->p32[1] = this->sample_buf_p[channels - 1];
d->proc_mask = s->proc_mask;
return count;
}
/* convert count 16-bit mono to 32-bit mono */
static void sample_input_mono16(struct sample_io_data *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *src, *dst;
int count = sample_input_setup(this, buf_p, 1, &src, &dst);
if (count <= 0)
return;
const int16_t *s = src->pin[0];
int32_t *d = dst->p32[0];
const int scale = WORD_SHIFT;
dsp_advance_buffer_input(src, count, sizeof (int16_t));
do
{
*d++ = *s++ << scale;
}
while (--count > 0);
}
/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
static void sample_input_i_stereo16(struct sample_io_data *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *src, *dst;
int count = sample_input_setup(this, buf_p, 2, &src, &dst);
if (count <= 0)
return;
const int16_t *s = src->pin[0];
int32_t *dl = dst->p32[0];
int32_t *dr = dst->p32[1];
const int scale = WORD_SHIFT;
dsp_advance_buffer_input(src, count, 2*sizeof (int16_t));
do
{
*dl++ = *s++ << scale;
*dr++ = *s++ << scale;
}
while (--count > 0);
}
/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
static void sample_input_ni_stereo16(struct sample_io_data *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *src, *dst;
int count = sample_input_setup(this, buf_p, 2, &src, &dst);
if (count <= 0)
return;
const int16_t *sl = src->pin[0];
const int16_t *sr = src->pin[1];
int32_t *dl = dst->p32[0];
int32_t *dr = dst->p32[1];
const int scale = WORD_SHIFT;
dsp_advance_buffer_input(src, count, sizeof (int16_t));
do
{
*dl++ = *sl++ << scale;
*dr++ = *sr++ << scale;
}
while (--count > 0);
}
/* convert count 32-bit mono to 32-bit mono */
static void sample_input_mono32(struct sample_io_data *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *dst = &this->sample_buf;
if (dst->remcount > 0)
{
*buf_p = dst;
return; /* data still remains */
}
/* else no buffer switch */
struct dsp_buffer *src = *buf_p;
src->p32[1] = src->p32[0];
}
/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
static void sample_input_i_stereo32(struct sample_io_data *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *src, *dst;
int count = sample_input_setup(this, buf_p, 2, &src, &dst);
if (count <= 0)
return;
const int32_t *s = src->pin[0];
int32_t *dl = dst->p32[0];
int32_t *dr = dst->p32[1];
dsp_advance_buffer_input(src, count, 2*sizeof (int32_t));
do
{
*dl++ = *s++;
*dr++ = *s++;
}
while (--count > 0);
}
/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
static void sample_input_ni_stereo32(struct sample_io_data *this,
struct dsp_buffer **buf_p)
{
struct dsp_buffer *dst = &this->sample_buf;
if (dst->remcount > 0)
*buf_p = dst; /* data still remains */
/* else no buffer switch */
}
/* set the to-native sample conversion function based on dsp sample
* parameters - depends upon stereo_mode and sample_depth */
void dsp_sample_input_format_change(struct sample_io_data *this,
struct sample_format *format)
{
static const sample_input_fn_type fns[STEREO_NUM_MODES][2] =
{
[STEREO_INTERLEAVED] =
{ sample_input_i_stereo16,
sample_input_i_stereo32 },
[STEREO_NONINTERLEAVED] =
{ sample_input_ni_stereo16,
sample_input_ni_stereo32 },
[STEREO_MONO] =
{ sample_input_mono16,
sample_input_mono32 },
};
if (this->sample_buf.remcount > 0)
return;
DSP_PRINT_FORMAT(DSP Input, this->format);
this->format_dirty = 0;
this->sample_buf.format = *format;
this->input_samples = fns[this->stereo_mode]
[this->sample_depth > NATIVE_DEPTH ? 1 : 0];
}
/* discard the sample buffer */
void dsp_sample_input_flush(struct sample_io_data *this)
{
this->sample_buf.remcount = 0;
}
void dsp_sample_input_init(struct sample_io_data *this, unsigned int dsp_id)
{
int32_t *lbuf, *rbuf;
switch (dsp_id)
{
case CODEC_IDX_AUDIO:
lbuf = sample_bufs[0];
rbuf = sample_bufs[1];
break;
case CODEC_IDX_VOICE:
lbuf = rbuf = sample_bufs[2]; /* Always mono */
break;
default:
/* orly */
DEBUGF("DSP Input- unknown dsp %u\n", dsp_id);
return;
}
this->sample_buf_p[0] = lbuf;
this->sample_buf_p[1] = rbuf;
}