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Move to compressor out of dsp.c and into its own source to reduce DSP clutter.

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A bit of a rough job for the moment but all works.

Change-Id: Id40852e0dec99caee02f943d0da8a1cdc16f022a
This commit is contained in:
Michael Sevakis 2012-02-08 14:55:37 -05:00
parent b0478726e4
commit 1ab9d14c77
5 changed files with 446 additions and 356 deletions
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1
apps/SOURCES
View file

@ -169,6 +169,7 @@ codec_thread.c
playback.c
codecs.c
dsp.c
compressor.c
#ifndef HAVE_HARDWARE_BEEP
beep.c
#endif

363
apps/compressor.c Normal file
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@ -0,0 +1,363 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2009 Jeffrey Goode
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "config.h"
#include "fixedpoint.h"
#include "fracmul.h"
#include "settings.h"
#include "dsp.h"
#include "compressor.h"
/* Define LOGF_ENABLE to enable logf output in this file */
/*#define LOGF_ENABLE*/
#include "logf.h"
static int32_t comp_rel_slope IBSS_ATTR; /* S7.24 format */
static int32_t comp_makeup_gain IBSS_ATTR; /* S7.24 format */
static int32_t comp_curve[66] IBSS_ATTR; /* S7.24 format */
static int32_t release_gain IBSS_ATTR; /* S7.24 format */
#define UNITY (1L << 24) /* unity gain in S7.24 format */
/** COMPRESSOR UPDATE
* Called via the menu system to configure the compressor process */
bool compressor_update(void)
{
static int curr_set[5];
int new_set[5] = {
global_settings.compressor_threshold,
global_settings.compressor_makeup_gain,
global_settings.compressor_ratio,
global_settings.compressor_knee,
global_settings.compressor_release_time};
/* make menu values useful */
int threshold = new_set[0];
bool auto_gain = (new_set[1] == 1);
const int comp_ratios[] = {2, 4, 6, 10, 0};
int ratio = comp_ratios[new_set[2]];
bool soft_knee = (new_set[3] == 1);
int release = new_set[4] * NATIVE_FREQUENCY / 1000;
bool changed = false;
bool active = (threshold < 0);
for (int i = 0; i < 5; i++)
{
if (curr_set[i] != new_set[i])
{
changed = true;
curr_set[i] = new_set[i];
#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
switch (i)
{
case 0:
logf(" Compressor Threshold: %d dB\tEnabled: %s",
threshold, active ? "Yes" : "No");
break;
case 1:
logf(" Compressor Makeup Gain: %s",
auto_gain ? "Auto" : "Off");
break;
case 2:
if (ratio)
{ logf(" Compressor Ratio: %d:1", ratio); }
else
{ logf(" Compressor Ratio: Limit"); }
break;
case 3:
logf(" Compressor Knee: %s", soft_knee?"Soft":"Hard");
break;
case 4:
logf(" Compressor Release: %d", release);
break;
}
#endif
}
}
if (changed && active)
{
/* configure variables for compressor operation */
static const int32_t db[] = {
/* positive db equivalents in S15.16 format */
0x000000, 0x241FA4, 0x1E1A5E, 0x1A94C8,
0x181518, 0x1624EA, 0x148F82, 0x1338BD,
0x120FD2, 0x1109EB, 0x101FA4, 0x0F4BB6,
0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E,
0x0C0A8C, 0x0B83BE, 0x0B04A5, 0x0A8C6C,
0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398,
0x0884F6, 0x082A30, 0x07D2FA, 0x077F0F,
0x072E31, 0x06E02A, 0x0694C8, 0x064BDF,
0x060546, 0x05C0DA, 0x057E78, 0x053E03,
0x04FF5F, 0x04C273, 0x048726, 0x044D64,
0x041518, 0x03DE30, 0x03A89B, 0x037448,
0x03412A, 0x030F32, 0x02DE52, 0x02AE80,
0x027FB0, 0x0251D6, 0x0224EA, 0x01F8E2,
0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC,
0x0128EB, 0x010190, 0x00DAE4, 0x00B4E1,
0x008F82, 0x006AC1, 0x004699, 0x002305};
struct curve_point
{
int32_t db; /* S15.16 format */
int32_t offset; /* S15.16 format */
} db_curve[5];
/** Set up the shape of the compression curve first as decibel
values */
/* db_curve[0] = bottom of knee
[1] = threshold
[2] = top of knee
[3] = 0 db input
[4] = ~+12db input (2 bits clipping overhead) */
db_curve[1].db = threshold << 16;
if (soft_knee)
{
/* bottom of knee is 3dB below the threshold for soft knee*/
db_curve[0].db = db_curve[1].db - (3 << 16);
/* top of knee is 3dB above the threshold for soft knee */
db_curve[2].db = db_curve[1].db + (3 << 16);
if (ratio)
/* offset = -3db * (ratio - 1) / ratio */
db_curve[2].offset = (int32_t)((long long)(-3 << 16)
* (ratio - 1) / ratio);
else
/* offset = -3db for hard limit */
db_curve[2].offset = (-3 << 16);
}
else
{
/* bottom of knee is at the threshold for hard knee */
db_curve[0].db = threshold << 16;
/* top of knee is at the threshold for hard knee */
db_curve[2].db = threshold << 16;
db_curve[2].offset = 0;
}
/* Calculate 0db and ~+12db offsets */
db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */
if (ratio)
{
/* offset = threshold * (ratio - 1) / ratio */
db_curve[3].offset = (int32_t)((long long)(threshold << 16)
* (ratio - 1) / ratio);
db_curve[4].offset = (int32_t)((long long)-db_curve[4].db
* (ratio - 1) / ratio) + db_curve[3].offset;
}
else
{
/* offset = threshold for hard limit */
db_curve[3].offset = (threshold << 16);
db_curve[4].offset = -db_curve[4].db + db_curve[3].offset;
}
/** Now set up the comp_curve table with compression offsets in the
form of gain factors in S7.24 format */
/* comp_curve[0] is 0 (-infinity db) input */
comp_curve[0] = UNITY;
/* comp_curve[1 to 63] are intermediate compression values
corresponding to the 6 MSB of the input values of a non-clipped
signal */
for (int i = 1; i < 64; i++)
{
/* db constants are stored as positive numbers;
make them negative here */
int32_t this_db = -db[i];
/* no compression below the knee */
if (this_db <= db_curve[0].db)
comp_curve[i] = UNITY;
/* if soft knee and below top of knee,
interpolate along soft knee slope */
else if (soft_knee && (this_db <= db_curve[2].db))
comp_curve[i] = fp_factor(fp_mul(
((this_db - db_curve[0].db) / 6),
db_curve[2].offset, 16), 16) << 8;
/* interpolate along ratio slope above the knee */
else
comp_curve[i] = fp_factor(fp_mul(
fp_div((db_curve[1].db - this_db), db_curve[1].db, 16),
db_curve[3].offset, 16), 16) << 8;
}
/* comp_curve[64] is the compression level of a maximum level,
non-clipped signal */
comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8;
/* comp_curve[65] is the compression level of a maximum level,
clipped signal */
comp_curve[65] = fp_factor(db_curve[4].offset, 16) << 8;
#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
logf("\n *** Compression Offsets ***");
/* some settings for display only, not used in calculations */
db_curve[0].offset = 0;
db_curve[1].offset = 0;
db_curve[3].db = 0;
for (int i = 0; i <= 4; i++)
{
logf("Curve[%d]: db: % 6.2f\toffset: % 6.2f", i,
(float)db_curve[i].db / (1 << 16),
(float)db_curve[i].offset / (1 << 16));
}
logf("\nGain factors:");
for (int i = 1; i <= 65; i++)
{
debugf("%02d: %.6f ", i, (float)comp_curve[i] / UNITY);
if (i % 4 == 0) debugf("\n");
}
debugf("\n");
#endif
/* if using auto peak, then makeup gain is max offset -
.1dB headroom */
comp_makeup_gain = auto_gain ?
fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY;
logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / UNITY);
/* calculate per-sample gain change a rate of 10db over release time
*/
comp_rel_slope = 0xAF0BB2 / release;
logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY);
release_gain = UNITY;
}
return active;
}
/** GET COMPRESSION GAIN
* Returns the required gain factor in S7.24 format in order to compress the
* sample in accordance with the compression curve. Always 1 or less.
*/
static inline int32_t get_compression_gain(struct dsp_data *data,
int32_t sample)
{
const int frac_bits_offset = data->frac_bits - 15;
/* sample must be positive */
if (sample < 0)
sample = -(sample + 1);
/* shift sample into 15 frac bit range */
if (frac_bits_offset > 0)
sample >>= frac_bits_offset;
if (frac_bits_offset < 0)
sample <<= -frac_bits_offset;
/* normal case: sample isn't clipped */
if (sample < (1 << 15))
{
/* index is 6 MSB, rem is 9 LSB */
int index = sample >> 9;
int32_t rem = (sample & 0x1FF) << 22;
/* interpolate from the compression curve:
higher gain - ((rem / (1 << 31)) * (higher gain - lower gain)) */
return comp_curve[index] - (FRACMUL(rem,
(comp_curve[index] - comp_curve[index + 1])));
}
/* sample is somewhat clipped, up to 2 bits of overhead */
if (sample < (1 << 17))
{
/* straight interpolation:
higher gain - ((clipped portion of sample * 4/3
/ (1 << 31)) * (higher gain - lower gain)) */
return comp_curve[64] - (FRACMUL(((sample - (1 << 15)) / 3) << 16,
(comp_curve[64] - comp_curve[65])));
}
/* sample is too clipped, return invalid value */
return -1;
}
/** COMPRESSOR PROCESS
* Changes the gain of the samples according to the compressor curve
*/
void compressor_process(int count, struct dsp_data *data, int32_t *buf[])
{
const int num_chan = data->num_channels;
int32_t *in_buf[2] = {buf[0], buf[1]};
while (count-- > 0)
{
int ch;
/* use lowest (most compressed) gain factor of the output buffer
sample pair for both samples (mono is also handled correctly here)
*/
int32_t sample_gain = UNITY;
for (ch = 0; ch < num_chan; ch++)
{
int32_t this_gain = get_compression_gain(data, *in_buf[ch]);
if (this_gain < sample_gain)
sample_gain = this_gain;
}
/* perform release slope; skip if no compression and no release slope
*/
if ((sample_gain != UNITY) || (release_gain != UNITY))
{
/* if larger offset than previous slope, start new release slope
*/
if ((sample_gain <= release_gain) && (sample_gain > 0))
{
release_gain = sample_gain;
}
else
/* keep sloping towards unity gain (and ignore invalid value) */
{
release_gain += comp_rel_slope;
if (release_gain > UNITY)
{
release_gain = UNITY;
}
}
}
/* total gain factor is the product of release gain and makeup gain,
but avoid computation if possible */
int32_t total_gain = ((release_gain == UNITY) ? comp_makeup_gain :
(comp_makeup_gain == UNITY) ? release_gain :
FRACMUL_SHL(release_gain, comp_makeup_gain, 7));
/* Implement the compressor: apply total gain factor (if any) to the
output buffer sample pair/mono sample */
if (total_gain != UNITY)
{
for (ch = 0; ch < num_chan; ch++)
{
*in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7);
}
}
in_buf[0]++;
in_buf[1]++;
}
}
void compressor_reset(void)
{
release_gain = UNITY;
}

29
apps/compressor.h Normal file
View file

@ -0,0 +1,29 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2009 Jeffrey Goode
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#ifndef COMPRESSOR_H
#define COMPRESSOR_H
void compressor_process(int count, struct dsp_data *data, int32_t *buf[]);
bool compressor_update(void);
void compressor_reset(void);
#endif /* COMPRESSOR_H */

371
apps/dsp.c
View file

@ -24,6 +24,7 @@
#include "dsp.h"
#include "dsp-util.h"
#include "eq.h"
#include "compressor.h"
#include "kernel.h"
#include "settings.h"
#include "replaygain.h"
@ -66,42 +67,6 @@ enum
SAMPLE_OUTPUT_DITHERED_STEREO
};
/****************************************************************************
* NOTE: Any assembly routines that use these structures must be updated
* if current data members are moved or changed.
*/
struct resample_data
{
uint32_t delta; /* 00h */
uint32_t phase; /* 04h */
int32_t last_sample[2]; /* 08h */
/* 10h */
};
/* This is for passing needed data to assembly dsp routines. If another
* dsp parameter needs to be passed, add to the end of the structure
* and remove from dsp_config.
* If another function type becomes assembly optimized and requires dsp
* config info, add a pointer paramter of type "struct dsp_data *".
* If removing something from other than the end, reserve the spot or
* else update every implementation for every target.
* Be sure to add the offset of the new member for easy viewing as well. :)
* It is the first member of dsp_config and all members can be accessesed
* through the main aggregate but this is intended to make a safe haven
* for these items whereas the c part can be rearranged at will. dsp_data
* could even moved within dsp_config without disurbing the order.
*/
struct dsp_data
{
int output_scale; /* 00h */
int num_channels; /* 04h */
struct resample_data resample_data; /* 08h */
int32_t clip_min; /* 18h */
int32_t clip_max; /* 1ch */
int32_t gain; /* 20h - Note that this is in S8.23 format. */
/* 24h */
};
/* No asm...yet */
struct dither_data
{
@ -154,7 +119,7 @@ typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data,
struct dsp_config
{
struct dsp_data data; /* Config members for use in asm routines */
struct dsp_data data; /* Config members for use in external routines */
long codec_frequency; /* Sample rate of data coming from the codec */
long frequency; /* Effective sample rate after pitch shift (if any) */
int sample_depth;
@ -164,7 +129,6 @@ struct dsp_config
#ifdef HAVE_PITCHSCREEN
bool tdspeed_active; /* Timestretch is in use */
#endif
int frac_bits;
#ifdef HAVE_SW_TONE_CONTROLS
/* Filter struct for software bass/treble controls */
struct eqfilter tone_filter;
@ -180,7 +144,7 @@ struct dsp_config
channels_process_fn_type apply_crossfeed;
channels_process_fn_type eq_process;
channels_process_fn_type channels_process;
channels_process_fn_type compressor_process;
channels_process_dsp_fn_type compressor_process;
};
/* General DSP config */
@ -249,15 +213,6 @@ static int32_t *sample_buf[2] = { small_sample_buf[0], small_sample_buf[1] };
static int resample_buf_count = SMALL_RESAMPLE_BUF_COUNT;
static int32_t *resample_buf[2] = { small_resample_buf[0], small_resample_buf[1] };
/* compressor */
static int32_t comp_rel_slope IBSS_ATTR; /* S7.24 format */
static int32_t comp_makeup_gain IBSS_ATTR; /* S7.24 format */
static int32_t comp_curve[66] IBSS_ATTR; /* S7.24 format */
static int32_t release_gain IBSS_ATTR; /* S7.24 format */
#define UNITY (1L << 24) /* unity gain in S7.24 format */
static void compressor_process(int count, int32_t *buf[]);
#ifdef HAVE_PITCHSCREEN
int32_t sound_get_pitch(void)
{
@ -813,8 +768,8 @@ static inline int resample(struct dsp_config *dsp, int count, int32_t *src[])
static void dither_init(struct dsp_config *dsp)
{
memset(dither_data, 0, sizeof (dither_data));
dither_bias = (1L << (dsp->frac_bits - NATIVE_DEPTH));
dither_mask = (1L << (dsp->frac_bits + 1 - NATIVE_DEPTH)) - 1;
dither_bias = (1L << (dsp->data.frac_bits - NATIVE_DEPTH));
dither_mask = (1L << (dsp->data.frac_bits + 1 - NATIVE_DEPTH)) - 1;
}
void dsp_dither_enable(bool enable)
@ -1319,7 +1274,7 @@ int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
dsp->channels_process(chunk, t2);
if (dsp->compressor_process)
dsp->compressor_process(chunk, t2);
dsp->compressor_process(chunk, &dsp->data, t2);
dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
@ -1453,20 +1408,20 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
if (dsp->sample_depth <= NATIVE_DEPTH)
{
dsp->frac_bits = WORD_FRACBITS;
dsp->data.frac_bits = WORD_FRACBITS;
dsp->sample_bytes = sizeof (int16_t); /* samples are 16 bits */
dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
dsp->data.clip_min = -((1 << WORD_FRACBITS));
}
else
{
dsp->frac_bits = value;
dsp->data.frac_bits = value;
dsp->sample_bytes = sizeof (int32_t); /* samples are 32 bits */
dsp->data.clip_max = (1 << value) - 1;
dsp->data.clip_min = -(1 << value);
}
dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
dsp->data.output_scale = dsp->data.frac_bits + 1 - NATIVE_DEPTH;
sample_input_new_format(dsp);
dither_init(dsp);
break;
@ -1484,9 +1439,9 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
dsp->stereo_mode = STEREO_NONINTERLEAVED;
dsp->data.num_channels = 2;
dsp->sample_depth = NATIVE_DEPTH;
dsp->frac_bits = WORD_FRACBITS;
dsp->data.frac_bits = WORD_FRACBITS;
dsp->sample_bytes = sizeof (int16_t);
dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
dsp->data.output_scale = dsp->data.frac_bits + 1 - NATIVE_DEPTH;
dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
dsp->data.clip_min = -((1 << WORD_FRACBITS));
dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY;
@ -1506,7 +1461,7 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
tdspeed_setup(dsp);
#endif
if (dsp == &AUDIO_DSP)
release_gain = UNITY;
compressor_reset();
break;
case DSP_FLUSH:
@ -1518,7 +1473,7 @@ intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
tdspeed_setup(dsp);
#endif
if (dsp == &AUDIO_DSP)
release_gain = UNITY;
compressor_reset();
break;
case DSP_SET_TRACK_GAIN:
@ -1616,303 +1571,7 @@ void dsp_set_replaygain(void)
* Called by the menu system to configure the compressor process */
void dsp_set_compressor(void)
{
static int curr_set[5];
int new_set[5] = {
global_settings.compressor_threshold,
global_settings.compressor_makeup_gain,
global_settings.compressor_ratio,
global_settings.compressor_knee,
global_settings.compressor_release_time};
/* make menu values useful */
int threshold = new_set[0];
bool auto_gain = (new_set[1] == 1);
const int comp_ratios[] = {2, 4, 6, 10, 0};
int ratio = comp_ratios[new_set[2]];
bool soft_knee = (new_set[3] == 1);
int release = new_set[4] * NATIVE_FREQUENCY / 1000;
bool changed = false;
bool active = (threshold < 0);
for (int i = 0; i < 5; i++)
{
if (curr_set[i] != new_set[i])
{
changed = true;
curr_set[i] = new_set[i];
#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
switch (i)
{
case 0:
logf(" Compressor Threshold: %d dB\tEnabled: %s",
threshold, active ? "Yes" : "No");
break;
case 1:
logf(" Compressor Makeup Gain: %s",
auto_gain ? "Auto" : "Off");
break;
case 2:
if (ratio)
{ logf(" Compressor Ratio: %d:1", ratio); }
else
{ logf(" Compressor Ratio: Limit"); }
break;
case 3:
logf(" Compressor Knee: %s", soft_knee?"Soft":"Hard");
break;
case 4:
logf(" Compressor Release: %d", release);
break;
}
#endif
}
}
if (changed && active)
{
/* configure variables for compressor operation */
const int32_t db[] ={0x000000, /* positive db equivalents in S15.16 format */
0x241FA4, 0x1E1A5E, 0x1A94C8, 0x181518, 0x1624EA, 0x148F82, 0x1338BD, 0x120FD2,
0x1109EB, 0x101FA4, 0x0F4BB6, 0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E, 0x0C0A8C,
0x0B83BE, 0x0B04A5, 0x0A8C6C, 0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398, 0x0884F6,
0x082A30, 0x07D2FA, 0x077F0F, 0x072E31, 0x06E02A, 0x0694C8, 0x064BDF, 0x060546,
0x05C0DA, 0x057E78, 0x053E03, 0x04FF5F, 0x04C273, 0x048726, 0x044D64, 0x041518,
0x03DE30, 0x03A89B, 0x037448, 0x03412A, 0x030F32, 0x02DE52, 0x02AE80, 0x027FB0,
0x0251D6, 0x0224EA, 0x01F8E2, 0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC, 0x0128EB,
0x010190, 0x00DAE4, 0x00B4E1, 0x008F82, 0x006AC1, 0x004699, 0x002305};
struct curve_point
{
int32_t db; /* S15.16 format */
int32_t offset; /* S15.16 format */
} db_curve[5];
/** Set up the shape of the compression curve first as decibel values*/
/* db_curve[0] = bottom of knee
[1] = threshold
[2] = top of knee
[3] = 0 db input
[4] = ~+12db input (2 bits clipping overhead) */
db_curve[1].db = threshold << 16;
if (soft_knee)
{
/* bottom of knee is 3dB below the threshold for soft knee*/
db_curve[0].db = db_curve[1].db - (3 << 16);
/* top of knee is 3dB above the threshold for soft knee */
db_curve[2].db = db_curve[1].db + (3 << 16);
if (ratio)
/* offset = -3db * (ratio - 1) / ratio */
db_curve[2].offset = (int32_t)((long long)(-3 << 16)
* (ratio - 1) / ratio);
else
/* offset = -3db for hard limit */
db_curve[2].offset = (-3 << 16);
}
else
{
/* bottom of knee is at the threshold for hard knee */
db_curve[0].db = threshold << 16;
/* top of knee is at the threshold for hard knee */
db_curve[2].db = threshold << 16;
db_curve[2].offset = 0;
}
/* Calculate 0db and ~+12db offsets */
db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */
if (ratio)
{
/* offset = threshold * (ratio - 1) / ratio */
db_curve[3].offset = (int32_t)((long long)(threshold << 16)
* (ratio - 1) / ratio);
db_curve[4].offset = (int32_t)((long long)-db_curve[4].db
* (ratio - 1) / ratio) + db_curve[3].offset;
}
else
{
/* offset = threshold for hard limit */
db_curve[3].offset = (threshold << 16);
db_curve[4].offset = -db_curve[4].db + db_curve[3].offset;
}
/** Now set up the comp_curve table with compression offsets in the form
of gain factors in S7.24 format */
/* comp_curve[0] is 0 (-infinity db) input */
comp_curve[0] = UNITY;
/* comp_curve[1 to 63] are intermediate compression values corresponding
to the 6 MSB of the input values of a non-clipped signal */
for (int i = 1; i < 64; i++)
{
/* db constants are stored as positive numbers;
make them negative here */
int32_t this_db = -db[i];
/* no compression below the knee */
if (this_db <= db_curve[0].db)
comp_curve[i] = UNITY;
/* if soft knee and below top of knee,
interpolate along soft knee slope */
else if (soft_knee && (this_db <= db_curve[2].db))
comp_curve[i] = fp_factor(fp_mul(
((this_db - db_curve[0].db) / 6),
db_curve[2].offset, 16), 16) << 8;
/* interpolate along ratio slope above the knee */
else
comp_curve[i] = fp_factor(fp_mul(
fp_div((db_curve[1].db - this_db), db_curve[1].db, 16),
db_curve[3].offset, 16), 16) << 8;
}
/* comp_curve[64] is the compression level of a maximum level,
non-clipped signal */
comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8;
/* comp_curve[65] is the compression level of a maximum level,
clipped signal */
comp_curve[65] = fp_factor(db_curve[4].offset, 16) << 8;
#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
logf("\n *** Compression Offsets ***");
/* some settings for display only, not used in calculations */
db_curve[0].offset = 0;
db_curve[1].offset = 0;
db_curve[3].db = 0;
for (int i = 0; i <= 4; i++)
{
logf("Curve[%d]: db: % 6.2f\toffset: % 6.2f", i,
(float)db_curve[i].db / (1 << 16),
(float)db_curve[i].offset / (1 << 16));
}
logf("\nGain factors:");
for (int i = 1; i <= 65; i++)
{
debugf("%02d: %.6f ", i, (float)comp_curve[i] / UNITY);
if (i % 4 == 0) debugf("\n");
}
debugf("\n");
#endif
/* if using auto peak, then makeup gain is max offset - .1dB headroom */
comp_makeup_gain = auto_gain ?
fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY;
logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / UNITY);
/* calculate per-sample gain change a rate of 10db over release time */
comp_rel_slope = 0xAF0BB2 / release;
logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY);
release_gain = UNITY;
}
/* enable/disable the compressor */
AUDIO_DSP.compressor_process = active ? compressor_process : NULL;
}
/** GET COMPRESSION GAIN
* Returns the required gain factor in S7.24 format in order to compress the
* sample in accordance with the compression curve. Always 1 or less.
*/
static inline int32_t get_compression_gain(int32_t sample)
{
const int frac_bits_offset = AUDIO_DSP.frac_bits - 15;
/* sample must be positive */
if (sample < 0)
sample = -(sample + 1);
/* shift sample into 15 frac bit range */
if (frac_bits_offset > 0)
sample >>= frac_bits_offset;
if (frac_bits_offset < 0)
sample <<= -frac_bits_offset;
/* normal case: sample isn't clipped */
if (sample < (1 << 15))
{
/* index is 6 MSB, rem is 9 LSB */
int index = sample >> 9;
int32_t rem = (sample & 0x1FF) << 22;
/* interpolate from the compression curve:
higher gain - ((rem / (1 << 31)) * (higher gain - lower gain)) */
return comp_curve[index] - (FRACMUL(rem,
(comp_curve[index] - comp_curve[index + 1])));
}
/* sample is somewhat clipped, up to 2 bits of overhead */
if (sample < (1 << 17))
{
/* straight interpolation:
higher gain - ((clipped portion of sample * 4/3
/ (1 << 31)) * (higher gain - lower gain)) */
return comp_curve[64] - (FRACMUL(((sample - (1 << 15)) / 3) << 16,
(comp_curve[64] - comp_curve[65])));
}
/* sample is too clipped, return invalid value */
return -1;
}
/** COMPRESSOR PROCESS
* Changes the gain of the samples according to the compressor curve
*/
static void compressor_process(int count, int32_t *buf[])
{
const int num_chan = AUDIO_DSP.data.num_channels;
int32_t *in_buf[2] = {buf[0], buf[1]};
while (count-- > 0)
{
int ch;
/* use lowest (most compressed) gain factor of the output buffer
sample pair for both samples (mono is also handled correctly here) */
int32_t sample_gain = UNITY;
for (ch = 0; ch < num_chan; ch++)
{
int32_t this_gain = get_compression_gain(*in_buf[ch]);
if (this_gain < sample_gain)
sample_gain = this_gain;
}
/* perform release slope; skip if no compression and no release slope */
if ((sample_gain != UNITY) || (release_gain != UNITY))
{
/* if larger offset than previous slope, start new release slope */
if ((sample_gain <= release_gain) && (sample_gain > 0))
{
release_gain = sample_gain;
}
else
/* keep sloping towards unity gain (and ignore invalid value) */
{
release_gain += comp_rel_slope;
if (release_gain > UNITY)
{
release_gain = UNITY;
}
}
}
/* total gain factor is the product of release gain and makeup gain,
but avoid computation if possible */
int32_t total_gain = ((release_gain == UNITY) ? comp_makeup_gain :
(comp_makeup_gain == UNITY) ? release_gain :
FRACMUL_SHL(release_gain, comp_makeup_gain, 7));
/* Implement the compressor: apply total gain factor (if any) to the
output buffer sample pair/mono sample */
if (total_gain != UNITY)
{
for (ch = 0; ch < num_chan; ch++)
{
*in_buf[ch] = FRACMUL_SHL(total_gain, *in_buf[ch], 7);
}
}
in_buf[0]++;
in_buf[1]++;
}
AUDIO_DSP.compressor_process = compressor_update() ?
compressor_process : NULL;
}

38
apps/dsp.h
View file

@ -57,6 +57,44 @@ enum
DSP_CROSSFEED
};
/****************************************************************************
* NOTE: Any assembly routines that use these structures must be updated
* if current data members are moved or changed.
*/
struct resample_data
{
uint32_t delta; /* 00h */
uint32_t phase; /* 04h */
int32_t last_sample[2]; /* 08h */
/* 10h */
};
/* This is for passing needed data to external dsp routines. If another
* dsp parameter needs to be passed, add to the end of the structure
* and remove from dsp_config.
* If another function type becomes assembly/external and requires dsp
* config info, add a pointer paramter of type "struct dsp_data *".
* If removing something from other than the end, reserve the spot or
* else update every implementation for every target.
* Be sure to add the offset of the new member for easy viewing as well. :)
* It is the first member of dsp_config and all members can be accessesed
* through the main aggregate but this is intended to make a safe haven
* for these items whereas the c part can be rearranged at will. dsp_data
* could even moved within dsp_config without disurbing the order.
*/
struct dsp_data
{
int output_scale; /* 00h */
int num_channels; /* 04h */
struct resample_data resample_data; /* 08h */
int32_t clip_min; /* 18h */
int32_t clip_max; /* 1ch */
int32_t gain; /* 20h - Note that this is in S8.23 format. */
int frac_bits; /* 24h */
/* 28h */
};
struct dsp_config;
int dsp_process(struct dsp_config *dsp, char *dest,