5eee28e37d
Most importantly is surround shouldn't operate in mono mode. Have it watch and (de)activate itself on relevant format changes as it should. Other changes to better handle buffer allocation failure. PBE was set internally at 100 by default; SBZ. Change-Id: I328e0b674e56751a255eae817d7892d685796b06
243 lines
6.7 KiB
C
243 lines
6.7 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) 2014 by Chiwen Chang
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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 "fixedpoint.h"
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#include "fracmul.h"
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#include "settings.h"
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#include "dsp_proc_entry.h"
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#include "dsp_misc.h"
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#include "dsp_filter.h"
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#include "core_alloc.h"
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/* Perceptual bass enhancement */
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#define B3_DLY 72 /* ~800 uS */
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#define B2_DLY 180 /* ~2000 uS*/
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#define B0_DLY 276 /* ~3050 uS */
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#define B3_SIZE (B3_DLY+1)
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#define B2_SIZE (B2_DLY+1)
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#define B0_SIZE (B0_DLY+1)
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static int pbe_strength = 0;
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static int pbe_precut = 0;
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static int32_t tcoef1, tcoef2, tcoef3;
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static int32_t *b0[2], *b2[2], *b3[2];
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static int b0_r[2],b2_r[2],b3_r[2],b0_w[2],b2_w[2],b3_w[2];
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int32_t temp_buffer;
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static struct dsp_filter pbe_filter[5];
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static int handle = -1;
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#define PBE_BUFSIZE ((B0_SIZE + B2_SIZE + B3_SIZE)*2*sizeof(int32_t))
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static int pbe_buffer_alloc(void)
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{
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handle = core_alloc("dsp_pbe_buffer", PBE_BUFSIZE);
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return handle;
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}
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static void pbe_buffer_free(void)
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{
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if (handle < 0)
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return;
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core_free(handle);
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handle = -1;
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}
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static void pbe_buffer_get_data(void)
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{
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if (handle < 0)
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return;
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b0[0] = core_get_data(handle);
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b0[1] = b0[0] + B0_SIZE;
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b2[0] = b0[1] + B0_SIZE;
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b2[1] = b2[0] + B2_SIZE;
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b3[0] = b2[1] + B2_SIZE;
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b3[1] = b3[0] + B3_SIZE;
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}
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static void dsp_pbe_flush(void)
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{
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memset(core_get_data(handle), 0, PBE_BUFSIZE);
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b0_r[0] = 0; b0_w[0] = 0;
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b0_r[1] = 0; b0_w[1] = 0;
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b2_r[0] = 0; b2_w[0] = 0;
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b2_r[1] = 0; b2_w[1] = 0;
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b3_r[0] = 0; b3_w[0] = 0;
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b3_r[1] = 0; b3_w[1] = 0;
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for (int i = 0; i < 5; i++)
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filter_flush(&pbe_filter[i]);
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}
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static void pbe_update_filter(unsigned int fout)
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{
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tcoef1 = fp_div(160, fout, 31);
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tcoef2 = fp_div(500, fout, 31);
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tcoef3 = fp_div(1150, fout, 31);
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/* Biophonic EQ */
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filter_bishelf_coefs(fp_div(20, fout, 32),
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fp_div(16000, fout, 32),
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0, 53, -5 + pbe_precut,
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&pbe_filter[0]);
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filter_pk_coefs(fp_div(64, fout, 32), 28, 53,
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&pbe_filter[1]);
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filter_pk_coefs(fp_div(2000, fout, 32), 28, 58,
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&pbe_filter[2]);
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filter_pk_coefs(fp_div(7500, fout, 32), 43, -82,
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&pbe_filter[3]);
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filter_pk_coefs(fp_div(10000, fout, 32), 43, -29,
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&pbe_filter[4]);
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}
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void dsp_pbe_precut(int var)
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{
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if (var == pbe_precut)
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return; /* No change */
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pbe_precut = var;
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struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
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if (!dsp_proc_enabled(dsp, DSP_PROC_PBE))
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return; /* Not currently enabled */
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pbe_update_filter(dsp_get_output_frequency(dsp));
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}
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void dsp_pbe_enable(int var)
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{
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if (var == pbe_strength)
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return; /* No change */
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pbe_strength = var;
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struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
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bool was_enabled = dsp_proc_enabled(dsp, DSP_PROC_PBE);
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bool now_enabled = var > 0;
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if (now_enabled == was_enabled)
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return; /* No change in enabled status */
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dsp_proc_enable(dsp, DSP_PROC_PBE, now_enabled);
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}
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static void pbe_process(struct dsp_proc_entry *this,
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struct dsp_buffer **buf_p)
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{
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struct dsp_buffer *buf = *buf_p;
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int count = buf->remcount;
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int num_channels = buf->format.num_channels;
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int b2_level = (B2_DLY * pbe_strength) / 100;
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int b0_level = (B0_DLY * pbe_strength) / 100;
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int32_t x;
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pbe_buffer_get_data();
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for(int ch = 0; ch < num_channels; ch++)
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{
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for (int i = 0; i < count; i++)
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{
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/* 160hz - 500hz no delay */
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temp_buffer = FRACMUL(buf->p32[ch][i], tcoef1) -
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FRACMUL(buf->p32[ch][i], tcoef2);
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/* delay below 160hz*/
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x = buf->p32[ch][i] -
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FRACMUL(buf->p32[ch][i], tcoef1);
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temp_buffer += dequeue(b0[ch], &b0_r[ch], b0_level);
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enqueue(x, b0[ch], &b0_w[ch], b0_level );
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/* delay 500-1150hz */
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x = FRACMUL(buf->p32[ch][i], tcoef2) -
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FRACMUL(buf->p32[ch][i], tcoef3);
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temp_buffer += dequeue(b2[ch], &b2_r[ch], b2_level);
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enqueue(x, b2[ch], &b2_w[ch], b2_level );
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/* delay anything beyond 1150hz */
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x = FRACMUL(buf->p32[ch][i], tcoef3);
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temp_buffer += dequeue(b3[ch], &b3_r[ch], B3_DLY);
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enqueue(x, b3[ch], &b3_w[ch], B3_DLY );
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buf->p32[ch][i] = temp_buffer;
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}
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}
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/* apply Biophonic EQ */
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for (int i = 0; i < 5; i++)
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filter_process(&pbe_filter[i], buf->p32, buf->remcount,
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buf->format.num_channels);
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(void)this;
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}
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/* DSP message hook */
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static intptr_t pbe_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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/* This only attaches to the audio (codec) DSP */
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intptr_t retval = 0;
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switch (setting)
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{
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case DSP_PROC_INIT:
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/* Coming online; was disabled */
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retval = pbe_buffer_alloc();
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if (retval < 0)
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break;
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this->process = pbe_process;
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dsp_pbe_flush();
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/* Wouldn't have been getting frequency updates */
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pbe_update_filter(dsp_get_output_frequency(dsp));
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dsp_proc_activate(dsp, DSP_PROC_PBE, true);
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break;
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case DSP_PROC_CLOSE:
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/* Being disabled (called also if init fails) */
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pbe_buffer_free();
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break;
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case DSP_FLUSH:
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/* Discontinuity; clear filters */
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dsp_pbe_flush();
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break;
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case DSP_SET_OUT_FREQUENCY:
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/* New output frequency */
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pbe_update_filter(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(
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PBE,
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pbe_configure);
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