fce81a8a74
Change-Id: I79f44f0dcc1b23b33a5040795220713660a1d18a
300 lines
8.7 KiB
C
300 lines
8.7 KiB
C
/***************************************************************************
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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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#include "fracmul.h"
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#include "fixedpoint.h"
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#include "dsp_proc_entry.h"
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#include <string.h>
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/**
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* Linear interpolation resampling that introduces a one sample delay because
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* of our inability to look into the future at the end of a frame.
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*/
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#if 0 /* Set to '1' to enable debug messages */
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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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#define RESAMPLE_BUF_COUNT 192 /* Per channel, per DSP */
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/* CODEC_IDX_AUDIO = left and right, CODEC_IDX_VOICE = mono */
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static int32_t resample_out_bufs[3][RESAMPLE_BUF_COUNT] IBSS_ATTR;
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/* Data for each resampler on each DSP */
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static struct resample_data
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{
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uint32_t delta; /* 00h: Phase delta for each step */
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uint32_t phase; /* 04h: Current phase [pos16|frac16] */
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int32_t last_sample[2]; /* 08h: Last samples for interpolation (L+R) */
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/* 10h */
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int32_t frequency; /* Virtual samplerate */
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struct dsp_buffer resample_buf; /* Buffer descriptor for resampled data */
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int32_t *resample_out_p[2]; /* Actual output buffer pointers */
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} resample_data[DSP_COUNT] IBSS_ATTR;
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/* Actual worker function. Implemented here or in target assembly code. */
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int resample_linear(struct resample_data *data, struct dsp_buffer *src,
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struct dsp_buffer *dst);
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static void resample_flush_data(struct resample_data *data)
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{
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data->phase = 0;
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data->last_sample[0] = 0;
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data->last_sample[1] = 0;
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}
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static void resample_flush(struct dsp_proc_entry *this)
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{
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struct resample_data *data = (void *)this->data;
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data->resample_buf.remcount = 0;
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resample_flush_data(data);
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}
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static bool resample_new_delta(struct resample_data *data,
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struct sample_format *format)
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{
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int32_t frequency = format->frequency; /* virtual samplerate */
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data->frequency = frequency;
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data->delta = fp_div(frequency, NATIVE_FREQUENCY, 16);
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if (frequency == NATIVE_FREQUENCY)
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{
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/* NOTE: If fully glitch-free transistions from no resampling to
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resampling are desired, last_sample history should be maintained
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even when not resampling. */
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resample_flush_data(data);
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return false;
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}
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return true;
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}
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#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
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/* Where the real work is done */
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int resample_linear(struct resample_data *data, struct dsp_buffer *src,
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struct dsp_buffer *dst)
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{
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int ch = src->format.num_channels - 1;
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uint32_t count = MIN(src->remcount, 0x8000);
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uint32_t delta = data->delta;
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uint32_t phase, pos;
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int32_t *d;
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do
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{
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const int32_t *s = src->p32[ch];
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d = dst->p32[ch];
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int32_t *dmax = d + dst->bufcount;
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phase = data->phase;
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pos = phase >> 16;
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pos = MIN(pos, count);
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int32_t last = pos > 0 ? s[pos - 1] : data->last_sample[ch];
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if (pos < count)
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{
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while (1)
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{
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*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
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phase += delta;
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pos = phase >> 16;
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if (pos >= count || d >= dmax)
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break;
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if (pos > 0)
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last = s[pos - 1];
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}
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if (pos > 0)
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{
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pos = MIN(pos, count);
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last = s[pos - 1];
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}
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}
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data->last_sample[ch] = last;
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}
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while (--ch >= 0);
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/* Wrap phase accumulator back to start of next frame. */
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data->phase = phase - (pos << 16);
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dst->remcount = d - dst->p32[0];
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return pos;
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}
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#endif /* CPU */
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/* Resample count stereo samples or stop when the destination is full.
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* Updates the src buffer and changes to its own output buffer to refer to
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* the resampled data. */
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static void resample_process(struct dsp_proc_entry *this,
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struct dsp_buffer **buf_p)
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{
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struct resample_data *data = (void *)this->data;
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struct dsp_buffer *src = *buf_p;
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struct dsp_buffer *dst = &data->resample_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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dst->remcount = 0;
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dst->p32[0] = data->resample_out_p[0];
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dst->p32[1] = data->resample_out_p[1];
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if (src->remcount > 0)
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{
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dst->bufcount = RESAMPLE_BUF_COUNT;
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int consumed = resample_linear(data, src, dst);
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/* Advance src by consumed amount */
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if (consumed > 0)
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dsp_advance_buffer32(src, consumed);
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}
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/* else purged resample_buf */
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/* Inherit in-place processed mask from source buffer */
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dst->proc_mask = src->proc_mask;
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}
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/* Finish draining old samples then switch format or shut off */
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static intptr_t resample_new_format(struct dsp_proc_entry *this,
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struct dsp_config *dsp,
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struct sample_format *format)
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{
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struct resample_data *data = (void *)this->data;
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struct dsp_buffer *dst = &data->resample_buf;
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if (dst->remcount > 0)
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return PROC_NEW_FORMAT_TRANSITION;
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DSP_PRINT_FORMAT(DSP_PROC_RESAMPLE, *format);
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int32_t frequency = data->frequency;
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bool active = dsp_proc_active(dsp, DSP_PROC_RESAMPLE);
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if (format->frequency != frequency)
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{
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DEBUGF(" DSP_PROC_RESAMPLE- new delta\n");
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active = resample_new_delta(data, format);
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dsp_proc_activate(dsp, DSP_PROC_RESAMPLE, active);
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}
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/* Everything after us is NATIVE_FREQUENCY */
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dst->format = *format;
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dst->format.frequency = NATIVE_FREQUENCY;
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dst->format.codec_frequency = NATIVE_FREQUENCY;
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if (active)
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return PROC_NEW_FORMAT_OK;
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/* No longer needed */
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DEBUGF(" DSP_PROC_RESAMPLE- deactivated\n");
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return PROC_NEW_FORMAT_DEACTIVATED;
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}
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static void INIT_ATTR resample_dsp_init(struct dsp_config *dsp,
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enum dsp_ids dsp_id)
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{
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int32_t *lbuf, *rbuf;
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switch (dsp_id)
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{
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case CODEC_IDX_AUDIO:
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lbuf = resample_out_bufs[0];
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rbuf = resample_out_bufs[1];
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break;
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case CODEC_IDX_VOICE:
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lbuf = rbuf = resample_out_bufs[2]; /* Always mono */
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break;
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default:
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/* huh? */
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DEBUGF("DSP_PROC_RESAMPLE- unknown DSP %d\n", (int)dsp_id);
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return;
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}
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/* Always enable resampler so that format changes may be monitored and
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* it self-activated when required */
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dsp_proc_enable(dsp, DSP_PROC_RESAMPLE, true);
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resample_data[dsp_id].resample_out_p[0] = lbuf;
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resample_data[dsp_id].resample_out_p[1] = rbuf;
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}
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static void INIT_ATTR resample_proc_init(struct dsp_proc_entry *this,
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struct dsp_config *dsp)
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{
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dsp_proc_set_in_place(dsp, DSP_PROC_RESAMPLE, false);
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this->data = (intptr_t)&resample_data[dsp_get_id(dsp)];
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this->process = resample_process;
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}
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/* DSP message hook */
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static intptr_t resample_configure(struct dsp_proc_entry *this,
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struct dsp_config *dsp,
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unsigned int setting,
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intptr_t value)
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{
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intptr_t retval = 0;
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switch (setting)
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{
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case DSP_INIT:
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resample_dsp_init(dsp, (enum dsp_ids)value);
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break;
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case DSP_FLUSH:
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resample_flush(this);
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break;
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case DSP_PROC_INIT:
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resample_proc_init(this, dsp);
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break;
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case DSP_PROC_CLOSE:
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/* This stage should be enabled at all times */
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DEBUGF("DSP_PROC_RESAMPLE- Error: Closing!\n");
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break;
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case DSP_PROC_NEW_FORMAT:
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retval = resample_new_format(this, dsp, (struct sample_format *)value);
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break;
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}
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return retval;
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}
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/* Database entry */
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DSP_PROC_DB_ENTRY(RESAMPLE,
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resample_configure);
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