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/***************************************************************************
* __________ __ ___.
* 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 "config.h"
#include "system.h"
#include "dsp_core.h"
#include "dsp_sample_io.h"
#if 1
#include <debug.h>
#else
#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.
*/
extern void dsp_sample_output_init(struct sample_io_data *this);
extern void dsp_sample_output_flush(struct sample_io_data *this);
/* 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 = *buf_p;
struct dsp_buffer *dst = &this->sample_buf;
*buf_p = dst;
if (dst->remcount > 0)
return; /* data still remains */
int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
dst->remcount = count;
dst->p32[0] = this->sample_buf_arr[0];
dst->p32[1] = this->sample_buf_arr[0];
dst->proc_mask = src->proc_mask;
if (count <= 0)
return; /* purged sample_buf */
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 = *buf_p;
struct dsp_buffer *dst = &this->sample_buf;
*buf_p = dst;
if (dst->remcount > 0)
return; /* data still remains */
int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
dst->remcount = count;
dst->p32[0] = this->sample_buf_arr[0];
dst->p32[1] = this->sample_buf_arr[1];
dst->proc_mask = src->proc_mask;
if (count <= 0)
return; /* purged sample_buf */
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 = *buf_p;
struct dsp_buffer *dst = &this->sample_buf;
*buf_p = dst;
if (dst->remcount > 0)
return; /* data still remains */
int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
dst->remcount = count;
dst->p32[0] = this->sample_buf_arr[0];
dst->p32[1] = this->sample_buf_arr[1];
dst->proc_mask = src->proc_mask;
if (count <= 0)
return; /* purged sample_buf */
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 = *buf_p;
struct dsp_buffer *dst = &this->sample_buf;
*buf_p = dst;
if (dst->remcount > 0)
return; /* data still remains */
int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
dst->remcount = count;
dst->p32[0] = this->sample_buf_arr[0];
dst->p32[1] = this->sample_buf_arr[1];
dst->proc_mask = src->proc_mask;
if (count <= 0)
return; /* purged sample_buf */
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 */
static void dsp_sample_input_format_change(struct sample_io_data *this,
struct dsp_buffer **buf_p)
{
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 },
};
struct dsp_buffer *src = *buf_p;
struct dsp_buffer *dst = &this->sample_buf;
/* Ack configured format change */
format_change_ack(&this->format);
if (dst->remcount > 0)
{
*buf_p = dst;
return; /* data still remains */
}
DSP_PRINT_FORMAT(DSP Input, -1, src->format);
/* new format - remember it and pass it along */
dst->format = src->format;
this->input_samples[0] = fns[this->stereo_mode]
[this->sample_depth > NATIVE_DEPTH ? 1 : 0];
this->input_samples[0](this, buf_p);
if (*buf_p == dst) /* buffer switch? */
format_change_ack(&src->format);
}
static void dsp_sample_input_init(struct sample_io_data *this)
{
this->input_samples[0] = sample_input_ni_stereo32;
this->input_samples[1] = dsp_sample_input_format_change;
}
/* discard the sample buffer */
static void dsp_sample_input_flush(struct sample_io_data *this)
{
this->sample_buf.remcount = 0;
}
void dsp_sample_io_configure(struct sample_io_data *this,
unsigned int setting,
intptr_t value)
{
switch (setting)
{
case DSP_INIT:
dsp_sample_input_init(this);
dsp_sample_output_init(this);
break;
case DSP_RESET:
/* Reset all sample descriptions to default */
format_change_set(&this->format);
this->format.num_channels = 2;
this->format.frac_bits = WORD_FRACBITS;
this->format.output_scale = WORD_FRACBITS + 1 - NATIVE_DEPTH;
this->format.frequency = NATIVE_FREQUENCY;
this->format.codec_frequency = NATIVE_FREQUENCY;
this->sample_depth = NATIVE_DEPTH;
this->stereo_mode = STEREO_NONINTERLEAVED;
break;
case DSP_SET_FREQUENCY:
value = value > 0 ? value : NATIVE_FREQUENCY;
format_change_set(&this->format);
this->format.frequency = value;
this->format.codec_frequency = value;
break;
case DSP_SET_SAMPLE_DEPTH:
format_change_set(&this->format);
this->format.frac_bits =
value <= NATIVE_DEPTH ? WORD_FRACBITS : value;
this->format.output_scale =
this->format.frac_bits + 1 - NATIVE_DEPTH;
this->sample_depth = value;
break;
case DSP_SET_STEREO_MODE:
format_change_set(&this->format);
this->format.num_channels = value == STEREO_MONO ? 1 : 2;
this->stereo_mode = value;
break;
case DSP_FLUSH:
dsp_sample_input_flush(this);
dsp_sample_output_flush(this);
break;
}
}
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