2012-03-27 23:52:15 +00:00
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/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2005 Miika Pekkarinen
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* Copyright (C) 2012 Michael Sevakis
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "config.h"
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#include "system.h"
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2012-04-29 21:31:30 +00:00
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#include "dsp_core.h"
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2012-03-27 23:52:15 +00:00
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#include "dsp_sample_io.h"
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#if 1
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#include <debug.h>
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#else
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#undef DEBUGF
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#define DEBUGF(...)
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#endif
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/* The internal format is 32-bit samples, non-interleaved, stereo. This
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* format is similar to the raw output from several codecs, so no copying is
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* needed for that case.
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*
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* Note that for mono, dst[0] equals dst[1], as there is no point in
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* processing the same data twice nor should it be done when modifying
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* samples in-place.
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*
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* When conversion is required:
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* Updates source buffer to point past the samples "consumed" also consuming
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* that portion of the input buffer and the destination is set to the buffer
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* of samples for later stages to consume.
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*
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* Input operates similarly to how an out-of-place processing stage should
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* behave.
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*/
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extern void dsp_sample_output_init(struct sample_io_data *this);
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extern void dsp_sample_output_flush(struct sample_io_data *this);
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/* convert count 16-bit mono to 32-bit mono */
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static void sample_input_mono16(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *src = *buf_p;
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struct dsp_buffer *dst = &this->sample_buf;
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*buf_p = dst;
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if (dst->remcount > 0)
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return; /* data still remains */
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int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
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dst->remcount = count;
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dst->p32[0] = this->sample_buf_arr[0];
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dst->p32[1] = this->sample_buf_arr[0];
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dst->proc_mask = src->proc_mask;
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if (count <= 0)
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return; /* purged sample_buf */
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const int16_t *s = src->pin[0];
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int32_t *d = dst->p32[0];
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const int scale = WORD_SHIFT;
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dsp_advance_buffer_input(src, count, sizeof (int16_t));
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do
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{
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*d++ = *s++ << scale;
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}
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while (--count > 0);
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}
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/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
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static void sample_input_i_stereo16(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *src = *buf_p;
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struct dsp_buffer *dst = &this->sample_buf;
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*buf_p = dst;
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if (dst->remcount > 0)
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return; /* data still remains */
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int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
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dst->remcount = count;
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dst->p32[0] = this->sample_buf_arr[0];
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dst->p32[1] = this->sample_buf_arr[1];
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dst->proc_mask = src->proc_mask;
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if (count <= 0)
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return; /* purged sample_buf */
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const int16_t *s = src->pin[0];
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int32_t *dl = dst->p32[0];
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int32_t *dr = dst->p32[1];
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const int scale = WORD_SHIFT;
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dsp_advance_buffer_input(src, count, 2*sizeof (int16_t));
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do
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{
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*dl++ = *s++ << scale;
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*dr++ = *s++ << scale;
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}
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while (--count > 0);
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}
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/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
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static void sample_input_ni_stereo16(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *src = *buf_p;
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struct dsp_buffer *dst = &this->sample_buf;
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*buf_p = dst;
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if (dst->remcount > 0)
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return; /* data still remains */
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int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
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dst->remcount = count;
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dst->p32[0] = this->sample_buf_arr[0];
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dst->p32[1] = this->sample_buf_arr[1];
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dst->proc_mask = src->proc_mask;
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if (count <= 0)
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return; /* purged sample_buf */
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const int16_t *sl = src->pin[0];
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const int16_t *sr = src->pin[1];
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int32_t *dl = dst->p32[0];
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int32_t *dr = dst->p32[1];
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const int scale = WORD_SHIFT;
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dsp_advance_buffer_input(src, count, sizeof (int16_t));
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do
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{
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*dl++ = *sl++ << scale;
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*dr++ = *sr++ << scale;
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}
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while (--count > 0);
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}
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/* convert count 32-bit mono to 32-bit mono */
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static void sample_input_mono32(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *dst = &this->sample_buf;
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if (dst->remcount > 0)
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{
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*buf_p = dst;
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return; /* data still remains */
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}
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/* else no buffer switch */
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struct dsp_buffer *src = *buf_p;
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src->p32[1] = src->p32[0];
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}
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/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
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static void sample_input_i_stereo32(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *src = *buf_p;
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struct dsp_buffer *dst = &this->sample_buf;
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*buf_p = dst;
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if (dst->remcount > 0)
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return; /* data still remains */
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int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
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dst->remcount = count;
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dst->p32[0] = this->sample_buf_arr[0];
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dst->p32[1] = this->sample_buf_arr[1];
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dst->proc_mask = src->proc_mask;
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if (count <= 0)
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return; /* purged sample_buf */
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const int32_t *s = src->pin[0];
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int32_t *dl = dst->p32[0];
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int32_t *dr = dst->p32[1];
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dsp_advance_buffer_input(src, count, 2*sizeof (int32_t));
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do
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{
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*dl++ = *s++;
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*dr++ = *s++;
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}
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while (--count > 0);
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}
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/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
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static void sample_input_ni_stereo32(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *dst = &this->sample_buf;
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if (dst->remcount > 0)
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*buf_p = dst; /* data still remains */
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/* else no buffer switch */
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}
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/* set the to-native sample conversion function based on dsp sample
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* parameters */
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static void dsp_sample_input_format_change(struct sample_io_data *this,
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struct dsp_buffer **buf_p)
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{
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static const sample_input_fn_type fns[STEREO_NUM_MODES][2] =
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{
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[STEREO_INTERLEAVED] =
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{ sample_input_i_stereo16,
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sample_input_i_stereo32 },
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[STEREO_NONINTERLEAVED] =
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{ sample_input_ni_stereo16,
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sample_input_ni_stereo32 },
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[STEREO_MONO] =
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{ sample_input_mono16,
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sample_input_mono32 },
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};
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struct dsp_buffer *src = *buf_p;
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struct dsp_buffer *dst = &this->sample_buf;
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/* Ack configured format change */
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format_change_ack(&this->format);
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if (dst->remcount > 0)
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{
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*buf_p = dst;
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return; /* data still remains */
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}
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DSP_PRINT_FORMAT(DSP Input, -1, src->format);
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/* new format - remember it and pass it along */
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dst->format = src->format;
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this->input_samples[0] = fns[this->stereo_mode]
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[this->sample_depth > NATIVE_DEPTH ? 1 : 0];
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this->input_samples[0](this, buf_p);
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if (*buf_p == dst) /* buffer switch? */
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format_change_ack(&src->format);
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}
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static void dsp_sample_input_init(struct sample_io_data *this)
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{
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this->input_samples[0] = sample_input_ni_stereo32;
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this->input_samples[1] = dsp_sample_input_format_change;
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}
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/* discard the sample buffer */
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static void dsp_sample_input_flush(struct sample_io_data *this)
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{
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this->sample_buf.remcount = 0;
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}
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void dsp_sample_io_configure(struct sample_io_data *this,
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unsigned int setting,
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intptr_t value)
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{
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switch (setting)
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{
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case DSP_INIT:
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dsp_sample_input_init(this);
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dsp_sample_output_init(this);
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break;
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case DSP_RESET:
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/* Reset all sample descriptions to default */
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format_change_set(&this->format);
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this->format.num_channels = 2;
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this->format.frac_bits = WORD_FRACBITS;
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this->format.output_scale = WORD_FRACBITS + 1 - NATIVE_DEPTH;
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this->format.frequency = NATIVE_FREQUENCY;
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this->format.codec_frequency = NATIVE_FREQUENCY;
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this->sample_depth = NATIVE_DEPTH;
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this->stereo_mode = STEREO_NONINTERLEAVED;
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break;
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case DSP_SET_FREQUENCY:
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value = value > 0 ? value : NATIVE_FREQUENCY;
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format_change_set(&this->format);
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this->format.frequency = value;
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this->format.codec_frequency = value;
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break;
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case DSP_SET_SAMPLE_DEPTH:
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format_change_set(&this->format);
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this->format.frac_bits =
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value <= NATIVE_DEPTH ? WORD_FRACBITS : value;
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this->format.output_scale =
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this->format.frac_bits + 1 - NATIVE_DEPTH;
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this->sample_depth = value;
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break;
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case DSP_SET_STEREO_MODE:
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format_change_set(&this->format);
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this->format.num_channels = value == STEREO_MONO ? 1 : 2;
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this->stereo_mode = value;
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break;
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case DSP_FLUSH:
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dsp_sample_input_flush(this);
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dsp_sample_output_flush(this);
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break;
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}
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}
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