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Michael Sevakis
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Move to compressor out of dsp.c and into its own source to reduce DSP clutter.
A bit of a rough job for the moment but all works. Change-Id: Id40852e0dec99caee02f943d0da8a1cdc16f022a
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apps/SOURCES

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Original file line numberDiff line numberDiff line change
@@ -169,6 +169,7 @@ codec_thread.c
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playback.c
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codecs.c
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dsp.c
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compressor.c
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#ifndef HAVE_HARDWARE_BEEP
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beep.c
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#endif

apps/compressor.c

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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) 2009 Jeffrey Goode
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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.h"
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#include "compressor.h"
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/* Define LOGF_ENABLE to enable logf output in this file */
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/*#define LOGF_ENABLE*/
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#include "logf.h"
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static int32_t comp_rel_slope IBSS_ATTR; /* S7.24 format */
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static int32_t comp_makeup_gain IBSS_ATTR; /* S7.24 format */
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static int32_t comp_curve[66] IBSS_ATTR; /* S7.24 format */
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static int32_t release_gain IBSS_ATTR; /* S7.24 format */
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#define UNITY (1L << 24) /* unity gain in S7.24 format */
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/** COMPRESSOR UPDATE
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* Called via the menu system to configure the compressor process */
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bool compressor_update(void)
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{
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static int curr_set[5];
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int new_set[5] = {
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global_settings.compressor_threshold,
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global_settings.compressor_makeup_gain,
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global_settings.compressor_ratio,
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global_settings.compressor_knee,
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global_settings.compressor_release_time};
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/* make menu values useful */
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int threshold = new_set[0];
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bool auto_gain = (new_set[1] == 1);
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const int comp_ratios[] = {2, 4, 6, 10, 0};
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int ratio = comp_ratios[new_set[2]];
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bool soft_knee = (new_set[3] == 1);
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int release = new_set[4] * NATIVE_FREQUENCY / 1000;
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bool changed = false;
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bool active = (threshold < 0);
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for (int i = 0; i < 5; i++)
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{
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if (curr_set[i] != new_set[i])
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{
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changed = true;
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curr_set[i] = new_set[i];
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#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
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switch (i)
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{
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case 0:
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logf(" Compressor Threshold: %d dB\tEnabled: %s",
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threshold, active ? "Yes" : "No");
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break;
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case 1:
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logf(" Compressor Makeup Gain: %s",
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auto_gain ? "Auto" : "Off");
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break;
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case 2:
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if (ratio)
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{ logf(" Compressor Ratio: %d:1", ratio); }
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else
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{ logf(" Compressor Ratio: Limit"); }
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break;
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case 3:
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logf(" Compressor Knee: %s", soft_knee?"Soft":"Hard");
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break;
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case 4:
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logf(" Compressor Release: %d", release);
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break;
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}
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#endif
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}
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}
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if (changed && active)
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{
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/* configure variables for compressor operation */
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static const int32_t db[] = {
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/* positive db equivalents in S15.16 format */
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0x000000, 0x241FA4, 0x1E1A5E, 0x1A94C8,
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0x181518, 0x1624EA, 0x148F82, 0x1338BD,
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0x120FD2, 0x1109EB, 0x101FA4, 0x0F4BB6,
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0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E,
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0x0C0A8C, 0x0B83BE, 0x0B04A5, 0x0A8C6C,
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0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398,
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0x0884F6, 0x082A30, 0x07D2FA, 0x077F0F,
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0x072E31, 0x06E02A, 0x0694C8, 0x064BDF,
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0x060546, 0x05C0DA, 0x057E78, 0x053E03,
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0x04FF5F, 0x04C273, 0x048726, 0x044D64,
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0x041518, 0x03DE30, 0x03A89B, 0x037448,
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0x03412A, 0x030F32, 0x02DE52, 0x02AE80,
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0x027FB0, 0x0251D6, 0x0224EA, 0x01F8E2,
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0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC,
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0x0128EB, 0x010190, 0x00DAE4, 0x00B4E1,
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0x008F82, 0x006AC1, 0x004699, 0x002305};
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struct curve_point
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{
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int32_t db; /* S15.16 format */
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int32_t offset; /* S15.16 format */
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} db_curve[5];
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/** Set up the shape of the compression curve first as decibel
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values */
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/* db_curve[0] = bottom of knee
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[1] = threshold
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[2] = top of knee
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[3] = 0 db input
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[4] = ~+12db input (2 bits clipping overhead) */
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db_curve[1].db = threshold << 16;
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if (soft_knee)
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{
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/* bottom of knee is 3dB below the threshold for soft knee*/
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db_curve[0].db = db_curve[1].db - (3 << 16);
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/* top of knee is 3dB above the threshold for soft knee */
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db_curve[2].db = db_curve[1].db + (3 << 16);
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if (ratio)
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/* offset = -3db * (ratio - 1) / ratio */
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db_curve[2].offset = (int32_t)((long long)(-3 << 16)
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* (ratio - 1) / ratio);
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else
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/* offset = -3db for hard limit */
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db_curve[2].offset = (-3 << 16);
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}
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else
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{
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/* bottom of knee is at the threshold for hard knee */
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db_curve[0].db = threshold << 16;
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/* top of knee is at the threshold for hard knee */
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db_curve[2].db = threshold << 16;
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db_curve[2].offset = 0;
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}
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/* Calculate 0db and ~+12db offsets */
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db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */
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if (ratio)
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{
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/* offset = threshold * (ratio - 1) / ratio */
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db_curve[3].offset = (int32_t)((long long)(threshold << 16)
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* (ratio - 1) / ratio);
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db_curve[4].offset = (int32_t)((long long)-db_curve[4].db
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* (ratio - 1) / ratio) + db_curve[3].offset;
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}
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else
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{
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/* offset = threshold for hard limit */
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db_curve[3].offset = (threshold << 16);
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db_curve[4].offset = -db_curve[4].db + db_curve[3].offset;
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}
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/** Now set up the comp_curve table with compression offsets in the
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form of gain factors in S7.24 format */
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/* comp_curve[0] is 0 (-infinity db) input */
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comp_curve[0] = UNITY;
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/* comp_curve[1 to 63] are intermediate compression values
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corresponding to the 6 MSB of the input values of a non-clipped
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signal */
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for (int i = 1; i < 64; i++)
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{
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/* db constants are stored as positive numbers;
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make them negative here */
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int32_t this_db = -db[i];
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/* no compression below the knee */
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if (this_db <= db_curve[0].db)
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comp_curve[i] = UNITY;
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/* if soft knee and below top of knee,
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interpolate along soft knee slope */
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else if (soft_knee && (this_db <= db_curve[2].db))
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comp_curve[i] = fp_factor(fp_mul(
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((this_db - db_curve[0].db) / 6),
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db_curve[2].offset, 16), 16) << 8;
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/* interpolate along ratio slope above the knee */
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else
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comp_curve[i] = fp_factor(fp_mul(
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fp_div((db_curve[1].db - this_db), db_curve[1].db, 16),
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db_curve[3].offset, 16), 16) << 8;
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}
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/* comp_curve[64] is the compression level of a maximum level,
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non-clipped signal */
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comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8;
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/* comp_curve[65] is the compression level of a maximum level,
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clipped signal */
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comp_curve[65] = fp_factor(db_curve[4].offset, 16) << 8;
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#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
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logf("\n *** Compression Offsets ***");
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/* some settings for display only, not used in calculations */
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db_curve[0].offset = 0;
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db_curve[1].offset = 0;
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db_curve[3].db = 0;
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for (int i = 0; i <= 4; i++)
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{
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logf("Curve[%d]: db: % 6.2f\toffset: % 6.2f", i,
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(float)db_curve[i].db / (1 << 16),
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(float)db_curve[i].offset / (1 << 16));
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}
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logf("\nGain factors:");
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for (int i = 1; i <= 65; i++)
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{
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debugf("%02d: %.6f ", i, (float)comp_curve[i] / UNITY);
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if (i % 4 == 0) debugf("\n");
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}
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debugf("\n");
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#endif
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/* if using auto peak, then makeup gain is max offset -
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.1dB headroom */
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comp_makeup_gain = auto_gain ?
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fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY;
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logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / UNITY);
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/* calculate per-sample gain change a rate of 10db over release time
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*/
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comp_rel_slope = 0xAF0BB2 / release;
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logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY);
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release_gain = UNITY;
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}
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return active;
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}
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/** GET COMPRESSION GAIN
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* Returns the required gain factor in S7.24 format in order to compress the
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* sample in accordance with the compression curve. Always 1 or less.
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*/
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static inline int32_t get_compression_gain(struct dsp_data *data,
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int32_t sample)
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{
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const int frac_bits_offset = data->frac_bits - 15;
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/* sample must be positive */
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if (sample < 0)
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sample = -(sample + 1);
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/* shift sample into 15 frac bit range */
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if (frac_bits_offset > 0)
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sample >>= frac_bits_offset;
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if (frac_bits_offset < 0)
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sample <<= -frac_bits_offset;
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/* normal case: sample isn't clipped */
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if (sample < (1 << 15))
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{
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/* index is 6 MSB, rem is 9 LSB */
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int index = sample >> 9;
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int32_t rem = (sample & 0x1FF) << 22;
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/* interpolate from the compression curve:
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higher gain - ((rem / (1 << 31)) * (higher gain - lower gain)) */
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return comp_curve[index] - (FRACMUL(rem,
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(comp_curve[index] - comp_curve[index + 1])));
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}
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/* sample is somewhat clipped, up to 2 bits of overhead */
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if (sample < (1 << 17))
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{
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/* straight interpolation:
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higher gain - ((clipped portion of sample * 4/3
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/ (1 << 31)) * (higher gain - lower gain)) */
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return comp_curve[64] - (FRACMUL(((sample - (1 << 15)) / 3) << 16,
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(comp_curve[64] - comp_curve[65])));
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}
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/* sample is too clipped, return invalid value */
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return -1;
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}
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/** COMPRESSOR PROCESS
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* Changes the gain of the samples according to the compressor curve
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*/
300+
void compressor_process(int count, struct dsp_data *data, int32_t *buf[])
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{
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const int num_chan = data->num_channels;
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int32_t *in_buf[2] = {buf[0], buf[1]};
304+
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while (count-- > 0)
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{
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int ch;
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/* use lowest (most compressed) gain factor of the output buffer
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sample pair for both samples (mono is also handled correctly here)
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*/
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int32_t sample_gain = UNITY;
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for (ch = 0; ch < num_chan; ch++)
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{
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int32_t this_gain = get_compression_gain(data, *in_buf[ch]);
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if (this_gain < sample_gain)
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sample_gain = this_gain;
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}
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/* perform release slope; skip if no compression and no release slope
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*/
321+
if ((sample_gain != UNITY) || (release_gain != UNITY))
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{
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/* if larger offset than previous slope, start new release slope
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*/
325+
if ((sample_gain <= release_gain) && (sample_gain > 0))
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{
327+
release_gain = sample_gain;
328+
}
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else
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/* keep sloping towards unity gain (and ignore invalid value) */
331+
{
332+
release_gain += comp_rel_slope;
333+
if (release_gain > UNITY)
334+
{
335+
release_gain = UNITY;
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}
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}
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}
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/* total gain factor is the product of release gain and makeup gain,
341+
but avoid computation if possible */
342+
int32_t total_gain = ((release_gain == UNITY) ? comp_makeup_gain :
343+
(comp_makeup_gain == UNITY) ? release_gain :
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FRACMUL_SHL(release_gain, comp_makeup_gain, 7));
345+
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/* Implement the compressor: apply total gain factor (if any) to the
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output buffer sample pair/mono sample */
348+
if (total_gain != UNITY)
349+
{
350+
for (ch = 0; ch < num_chan; ch++)
351+
{
352+
*in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7);
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}
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}
355+
in_buf[0]++;
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in_buf[1]++;
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}
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}
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void compressor_reset(void)
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{
362+
release_gain = UNITY;
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}

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