7418ec5009
I think the assumption is that during playback, the DSP is already running so it assumes PBE needs to be flushed before it is properly enabled. Change-Id: I2bac3d02c80f97c8d9ce26a575175f6344a8e86a
243 lines
6.6 KiB
C
243 lines
6.6 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 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 dsp_pbe_flush(void)
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{
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if (handle < 0)
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return;
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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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int32_t *b0[2], *b2[2], *b3[2];
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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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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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