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Add software based bass/treble controls for targets which have no such functionality in hardware (currently only X5). They can also be used on any other SWCODEC target by adding #define HAVE_SW_TONE_CONTROLS in the relevant config-*.h file. Also remove some now unneeded zero checks when using get_replaygain_int(). Comments on sound quality are welcome as some parameters can still be fine-tuned.

Browse source 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12489 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen 2007-02-26 00:41:26 +00:00
parent 1915c10994
commit a7fabf0741
9 changed files with 175 additions and 45 deletions
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71
apps/dsp.c
View file

@ -116,9 +116,10 @@ struct crossfeed_data
/* 8ch */
};
/* Current setup is one lowshelf filters, three peaking filters and one
highshelf filter. Varying the number of shelving filters make no sense,
but adding peaking filters is possible. */
/* Current setup is one lowshelf filters three peaking filters and one
* highshelf filter. Varying the number of shelving filters make no sense,
* but adding peaking filters is possible.
*/
struct eq_state
{
char enabled[5]; /* 00h - Flags for active filters */
@ -171,6 +172,13 @@ static long dither_bias IBSS_ATTR;
struct crossfeed_data crossfeed_data IBSS_ATTR; /* A */
/* Equalizer */
static struct eq_state eq_data; /* A/V */
#ifdef HAVE_SW_TONE_CONTROLS
static int prescale;
static int bass;
static int treble;
/* Filter struct for software bass/treble controls */
static struct eqfilter tone_filter;
#endif
/* Settings applicable to audio codec only */
static int pitch_ratio = 1000;
@ -704,11 +712,7 @@ void dsp_set_crossfeed(bool enable)
void dsp_set_crossfeed_direct_gain(int gain)
{
/* Work around bug in get_replaygain_int which returns 0 for 0 dB */
if (gain == 0)
crossfeed_data.gain = 0x7fffffff;
else
crossfeed_data.gain = get_replaygain_int(gain * -10) << 7;
crossfeed_data.gain = get_replaygain_int(gain * -10) << 7;
}
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
@ -716,8 +720,8 @@ void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
long g1 = get_replaygain_int(lf_gain * -10) << 3;
long g2 = get_replaygain_int(hf_gain * -10) << 3;
filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*cutoff, g1, g2,
crossfeed_data.coefs);
filter_shelf_coefs(0xffffffff/NATIVE_FREQUENCY*cutoff, g1, g2,
crossfeed_data.coefs);
}
/* Applies crossfeed to the stereo signal in src.
@ -985,6 +989,36 @@ static void channels_process_sound_chan_mono(int count, int32_t *buf[])
}
#endif /* DSP_HAVE_ASM_SOUND_CHAN_MONO */
#ifdef HAVE_SW_TONE_CONTROLS
static void set_tone_controls(void)
{
filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200,
0xffffffff/NATIVE_FREQUENCY*3500,
bass, treble, -prescale, tone_filter.coefs);
}
int dsp_callback(int msg, intptr_t param)
{
switch (msg) {
case DSP_CALLBACK_SET_PRESCALE:
prescale = param;
set_tone_controls();
break;
/* prescaler is always set after calling any of these, so we wait with
* calculating coefs until the above case is hit.
*/
case DSP_CALLBACK_SET_BASS:
bass = param;
break;
case DSP_CALLBACK_SET_TREBLE:
treble = param;
default:
break;
}
return 0;
}
#endif
#ifndef DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
static void channels_process_sound_chan_custom(int count, int32_t *buf[])
{
@ -1068,12 +1102,12 @@ int dsp_process(char *dst, const char *src[], int count)
int written = 0;
int samples;
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
/* set emac unit for dsp processing, and save old macsr, we're running in
codec thread context at this point, so can't clobber it */
unsigned long old_macsr = coldfire_get_macsr();
coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
#endif
#endif
while (count > 0)
{
@ -1085,8 +1119,17 @@ int dsp_process(char *dst, const char *src[], int count)
break; /* I'm pretty sure we're downsampling here */
if (dsp->apply_crossfeed)
dsp->apply_crossfeed(tmp, samples);
/* TODO: EQ and tone controls need separate structs for audio and voice
* DSP processing thanks to filter history. isn't really audible now, but
* might be the day we start handling voice more delicately.
*/
if (eq_enabled)
eq_process(samples, tmp);
#ifdef HAVE_SW_TONE_CONTROLS
if ((bass | treble) != 0)
eq_filter(tmp, &tone_filter, samples, dsp->data.num_channels,
FILTER_BISHELF_SHIFT);
#endif
if (dsp->channels_process)
dsp->channels_process(samples, tmp);
dsp->output_samples(samples, &dsp->data, tmp, (int16_t *)dst);
@ -1095,10 +1138,10 @@ int dsp_process(char *dst, const char *src[], int count)
yield();
}
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
/* set old macsr again */
coldfire_set_macsr(old_macsr);
#endif
#endif
return written;
}

7
apps/dsp.h
View file

@ -51,6 +51,12 @@ enum
DSP_CROSSFEED
};
enum {
DSP_CALLBACK_SET_PRESCALE = 0,
DSP_CALLBACK_SET_BASS,
DSP_CALLBACK_SET_TREBLE
};
/* A bunch of fixed point assembler helper macros */
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
/* These macros use the Coldfire EMAC extension and need the MACSR flags set
@ -209,6 +215,7 @@ void dsp_set_eq_precut(int precut);
void dsp_set_eq_coefs(int band);
void sound_set_pitch(int r);
int sound_get_pitch(void);
int dsp_callback(int msg, intptr_t param);
void channels_set(int value);
void stereo_width_set(int value);
void dsp_dither_enable(bool enable);

60
apps/eq.c
View file

@ -7,7 +7,7 @@
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2006 Thom Johansen
* Copyright (C) 2006-2007 Thom Johansen
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
@ -127,7 +127,7 @@ static long fsincos(unsigned long phase, long *cos) {
* @param an gain at Nyquist frequency. s3.27 fixed point.
* @param c pointer to coefficient storage. The coefs are s0.31 format.
*/
void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
void filter_shelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
{
const long one = 1 << 27;
long a0, a1;
@ -137,7 +137,7 @@ void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
cs = one + (cs >> 4);
/* For max A = 4 (24 dB) */
b0 = FRACMUL_SHL(an, cs, 4) + FRACMUL_SHL(ad, s, 4);
b0 = FRACMUL_SHL(ad, s, 4) + FRACMUL_SHL(an, cs, 4);
b1 = FRACMUL_SHL(ad, s, 4) - FRACMUL_SHL(an, cs, 4);
a0 = s + cs;
a1 = s - cs;
@ -147,6 +147,58 @@ void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
c[2] = -DIV64(a1, a0, 31);
}
/**
* Calculate second order section filter consisting of one low-shelf and one
* high-shelf section.
* @param cutoff_low low-shelf midpoint frequency. See eq_pk_coefs for format.
* @param cutoff_high high-shelf midpoint frequency.
* @param A_low decibel value multiplied by ten, describing gain/attenuation of
* low-shelf part. Max value is 24 dB.
* @param A_high decibel value multiplied by ten, describing gain/attenuation of
* high-shelf part. Max value is 24 dB.
* @param A decibel value multiplied by ten, describing additional overall gain.
* @param c pointer to coefficient storage. Coefficients are s4.27 format.
*/
void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
long A_low, long A_high, long A, int32_t *c)
{
long sin1, cos2; /* s0.31 */
long cos1, sin2; /* s3.28 */
int32_t b0, b1, b2, b3; /* s3.28 */
int32_t a0, a1, a2, a3;
const long gd = get_replaygain_int(A_low*5) << 4; /* 10^(db/40), s3.28 */
const long gn = get_replaygain_int(A_high*5) << 4; /* 10^(db/40), s3.28 */
const long g = get_replaygain_int(A*10) << 7; /* 10^(db/20), s0.31 */
sin1 = fsincos(cutoff_low/2, &cos1);
sin2 = fsincos(cutoff_high/2, &cos2) >> 3;
cos1 >>= 3;
/* lowshelf filter, ranges listed are for all possible cutoffs */
b0 = FRACMUL(sin1, gd) + cos1; /* 0.25 .. 4.10 */
b1 = FRACMUL(sin1, gd) - cos1; /* -1 .. 3.98 */
a0 = DIV64(sin1, gd, 25) + cos1; /* 0.25 .. 4.10 */
a1 = DIV64(sin1, gd, 25) - cos1; /* -1 .. 3.98 */
/* highshelf filter */
b2 = sin2 + FRACMUL(cos2, gn); /* 0.25 .. 4.10 */
b3 = sin2 - FRACMUL(cos2, gn); /* -3.98 .. 1 */
a2 = sin2 + DIV64(cos2, gn, 25); /* 0.25 .. 4.10 */
a3 = sin2 - DIV64(cos2, gn, 25); /* -3.98 .. 1 */
/* now we cascade the two first order filters to one second order filter
* which can be used by eq_filter(). these resulting coefficients have a
* really wide numerical range, so we use a fixed point format which will
* work for the selected cutoff frequencies (in dsp.c) only.
*/
const int32_t rcp_a0 = DIV64(1, FRACMUL(a0, a2), 53); /* s3.28 */
*c++ = FRACMUL(g, FRACMUL_SHL(FRACMUL(b0, b2), rcp_a0, 5));
*c++ = FRACMUL(g, FRACMUL_SHL(FRACMUL(b0, b3) + FRACMUL(b1, b2), rcp_a0, 5));
*c++ = FRACMUL(g, FRACMUL_SHL(FRACMUL(b1, b3), rcp_a0, 5));
*c++ = -FRACMUL_SHL(FRACMUL(a0, a3) + FRACMUL(a1, a2), rcp_a0, 5);
*c++ = -FRACMUL_SHL(FRACMUL(a1, a3), rcp_a0, 5);
}
/* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
* Slightly faster calculation can be done by deriving forms which use tan()
* instead of cos() and sin(), but the latter are far easier to use when doing
@ -162,7 +214,7 @@ void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
* @param Q Q factor value multiplied by ten. Lower bound is artificially set
* at 0.5.
* @param db decibel value multiplied by ten, describing gain/attenuation at
* peak freq.
* peak freq. Max value is 24 dB.
* @param c pointer to coefficient storage. Coefficients are s3.28 format.
*/
void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)

7
apps/eq.h
View file

@ -7,7 +7,7 @@
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2006 Thom Johansen
* Copyright (C) 2006-2007 Thom Johansen
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
@ -25,6 +25,7 @@
/* These depend on the fixed point formats used by the different filter types
and need to be changed when they change.
*/
#define FILTER_BISHELF_SHIFT 5
#define EQ_PEAK_SHIFT 4
#define EQ_SHELF_SHIFT 6
@ -33,7 +34,9 @@ struct eqfilter {
int32_t history[2][4];
};
void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c);
void filter_shelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c);
void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
long A_low, long A_high, long A, int32_t *c);
void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c);
void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c);
void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c);

10
apps/menus/sound_menu.c
View file

@ -55,12 +55,8 @@ int soundmenu_callback(int action,const struct menu_item_ex *this_item)
#endif
MENUITEM_SETTING(volume, &global_settings.volume, soundmenu_callback);
#ifndef HAVE_TLV320
MENUITEM_SETTING(bass, &global_settings.bass, soundmenu_callback);
MENUITEM_SETTING(treble, &global_settings.treble, soundmenu_callback);
#endif
MENUITEM_SETTING(bass, &global_settings.bass, soundmenu_callback);
MENUITEM_SETTING(treble, &global_settings.treble, soundmenu_callback);
MENUITEM_SETTING(balance, &global_settings.balance, soundmenu_callback);
MENUITEM_SETTING(channel_config, &global_settings.channel_config, soundmenu_callback);
MENUITEM_SETTING(stereo_width, &global_settings.stereo_width, soundmenu_callback);
@ -99,9 +95,7 @@ MENUITEM_SETTING(stereo_width, &global_settings.stereo_width, soundmenu_callback
MAKE_MENU(sound_settings, ID2P(LANG_SOUND_SETTINGS), NULL, bitmap_icons_6x8[Icon_Audio],
&volume,
#ifndef HAVE_TLV320
&bass,&treble,
#endif
&balance,&channel_config,&stereo_width
#if CONFIG_CODEC == SWCODEC
,&crossfeed_menu, &equalizer_menu, &dithering_enabled

5
apps/settings.c
View file

@ -642,6 +642,9 @@ void settings_apply_pm_range(void)
void sound_settings_apply(void)
{
#ifdef HAVE_SW_TONE_CONTROLS
sound_set_dsp_callback(dsp_callback);
#endif
sound_set(SOUND_BASS, global_settings.bass);
sound_set(SOUND_TREBLE, global_settings.treble);
sound_set(SOUND_BALANCE, global_settings.balance);
@ -967,7 +970,7 @@ bool set_sound(const unsigned char * string,
talkunit = UNIT_PERCENT;
else if (*unit == 'H')
talkunit = UNIT_HERTZ;
if(!numdec)
if (!numdec)
#if CONFIG_CODEC == SWCODEC
/* We need to hijack this one and send it off to apps/dsp.c instead of
firmware/sound.c */

3
firmware/export/config-iaudiox5.h
View file

@ -84,6 +84,9 @@
#define HAVE_TLV320
/* TLV320 has no tone controls, so we use the software ones */
#define HAVE_SW_TONE_CONTROLS
#ifndef SIMULATOR
/* Define this if your LCD can set contrast */

2
firmware/export/sound.h
View file

@ -19,6 +19,7 @@
#ifndef SOUND_H
#define SOUND_H
#include <inttypes.h>
#ifdef HAVE_UDA1380
#include "uda1380.h"
#elif defined(HAVE_WM8975) || defined(HAVE_WM8751)
@ -76,6 +77,7 @@ int sound_max(int setting);
int sound_default(int setting);
sound_set_type* sound_get_fn(int setting);
void sound_set_dsp_callback(int (*func)(int, intptr_t));
void sound_set_volume(int value);
void sound_set_balance(int value);
void sound_set_bass(int value);

51
firmware/sound.c
View file

@ -86,6 +86,11 @@ static const struct sound_settings_info sound_settings_table[] = {
[SOUND_VOLUME] = {"dB", 0, 1, -78, 18, -18, sound_set_volume},
[SOUND_BASS] = {"dB", 0, 1, -15, 15, 7, sound_set_bass},
[SOUND_TREBLE] = {"dB", 0, 1, -15, 15, 7, sound_set_treble},
#endif
/* Override any other potentially existing treble/bass controllers if wanted */
#ifdef HAVE_SW_TONE_CONTROLS
[SOUND_BASS] = {"dB", 0, 1, -24, 24, 0, sound_set_bass},
[SOUND_TREBLE] = {"dB", 0, 1, -24, 24, 0, sound_set_treble},
#endif
[SOUND_BALANCE] = {"%", 0, 1,-100, 100, 0, sound_set_balance},
[SOUND_CHANNELS] = {"", 0, 1, 0, 5, 0, sound_set_channels},
@ -166,6 +171,22 @@ sound_set_type* sound_get_fn(int setting)
return NULL;
}
#ifdef HAVE_SW_TONE_CONTROLS
/* Copied from dsp.h, nasty nasty, but we don't want to include dsp.h */
enum {
DSP_CALLBACK_SET_PRESCALE = 0,
DSP_CALLBACK_SET_BASS,
DSP_CALLBACK_SET_TREBLE
};
static int (*dsp_callback)(int, intptr_t) = NULL;
void sound_set_dsp_callback(int (*func)(int, intptr_t))
{
dsp_callback = func;
}
#endif
#ifndef SIMULATOR
#if CONFIG_CODEC == MAS3507D /* volume/balance/treble/bass interdependency */
#define VOLUME_MIN -780
@ -293,10 +314,9 @@ int current_bass = 0; /* -150..+150 0..+240 */
static void set_prescaled_volume(void)
{
int prescale = 0;
int prescale;
int l, r;
#ifndef HAVE_TLV320
prescale = MAX(current_bass, current_treble);
if (prescale < 0)
prescale = 0; /* no need to prescale if we don't boost
@ -307,13 +327,12 @@ static void set_prescaled_volume(void)
* instead (might cause clipping). */
if (current_volume + prescale > VOLUME_MAX)
prescale = VOLUME_MAX - current_volume;
#endif
#if CONFIG_CODEC == MAS3507D
#if defined(HAVE_SW_TONE_CONTROLS)
dsp_callback(DSP_CALLBACK_SET_PRESCALE, prescale);
#elif CONFIG_CODEC == MAS3507D
mas_writereg(MAS_REG_KPRESCALE, prescale_table[prescale/10]);
#elif defined(HAVE_UDA1380)
audiohw_set_mixer_vol(tenthdb2mixer(-prescale), tenthdb2mixer(-prescale));
#elif defined(HAVE_WM8975) || defined(HAVE_WM8758) \
#elif defined(HAVE_UDA1380) || defined(HAVE_WM8975) || defined(HAVE_WM8758) \
|| defined(HAVE_WM8731) || defined(HAVE_WM8721) || defined(HAVE_WM8751)
audiohw_set_mixer_vol(tenthdb2mixer(-prescale), tenthdb2mixer(-prescale));
#endif
@ -338,9 +357,7 @@ static void set_prescaled_volume(void)
#if CONFIG_CODEC == MAS3507D
dac_volume(tenthdb2reg(l), tenthdb2reg(r), false);
#elif defined(HAVE_UDA1380)
audiohw_set_master_vol(tenthdb2master(l), tenthdb2master(r));
#elif defined(HAVE_WM8975) || defined(HAVE_WM8758) \
#elif defined(HAVE_UDA1380) || defined(HAVE_WM8975) || defined(HAVE_WM8758) \
|| defined(HAVE_WM8731) || defined(HAVE_WM8721) || defined(HAVE_WM8751)
audiohw_set_master_vol(tenthdb2master(l), tenthdb2master(r));
#if defined(HAVE_WM8975) || defined(HAVE_WM8758) || defined(HAVE_WM8751)
@ -484,12 +501,15 @@ void sound_set_balance(int value)
#endif
}
#ifndef HAVE_TLV320
void sound_set_bass(int value)
{
if(!audio_is_initialized)
return;
#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
#if defined(HAVE_SW_TONE_CONTROLS)
current_bass = value * 10;
dsp_callback(DSP_CALLBACK_SET_BASS, current_bass);
set_prescaled_volume();
#elif (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
unsigned tmp = ((unsigned)(value * 8) & 0xff) << 8;
mas_codec_writereg(0x14, tmp);
#elif CONFIG_CODEC == MAS3507D
@ -515,7 +535,11 @@ void sound_set_treble(int value)
{
if(!audio_is_initialized)
return;
#if (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
#if defined(HAVE_SW_TONE_CONTROLS)
current_treble = value * 10;
dsp_callback(DSP_CALLBACK_SET_TREBLE, current_treble);
set_prescaled_volume();
#elif (CONFIG_CODEC == MAS3587F) || (CONFIG_CODEC == MAS3539F)
unsigned tmp = ((unsigned)(value * 8) & 0xff) << 8;
mas_codec_writereg(0x15, tmp);
#elif CONFIG_CODEC == MAS3507D
@ -536,7 +560,6 @@ void sound_set_treble(int value)
(void)value;
#endif
}
#endif /* HAVE_TLV320 */
void sound_set_channels(int value)
{