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.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12489 a1c6a512-1295-4272-9138-f99709370657
diff --git a/apps/dsp.c b/apps/dsp.c
index 0ffaaea..f306069 100644
--- a/apps/dsp.c
+++ b/apps/dsp.c
@@ -116,9 +116,10 @@
/* 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 @@
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_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 @@
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 @@
}
#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 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 @@
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 @@
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;
}
diff --git a/apps/dsp.h b/apps/dsp.h
index 63dc68c..03118e8 100644
--- a/apps/dsp.h
+++ b/apps/dsp.h
@@ -51,6 +51,12 @@
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_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);
diff --git a/apps/eq.c b/apps/eq.c
index 588c23f..1d74db7 100644
--- a/apps/eq.c
+++ b/apps/eq.c
@@ -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 @@
* @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 @@
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 @@
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 @@
* @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)
diff --git a/apps/eq.h b/apps/eq.h
index 095c8e8..83d2359 100644
--- a/apps/eq.h
+++ b/apps/eq.h
@@ -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 @@
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);
diff --git a/apps/menus/sound_menu.c b/apps/menus/sound_menu.c
index 9dc9579..287b3ec 100644
--- a/apps/menus/sound_menu.c
+++ b/apps/menus/sound_menu.c
@@ -55,12 +55,8 @@
#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 @@
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
diff --git a/apps/settings.c b/apps/settings.c
index dd5e7c5..cc5ab12 100644
--- a/apps/settings.c
+++ b/apps/settings.c
@@ -642,6 +642,9 @@
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 @@
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 */
@@ -975,7 +978,7 @@
return set_int(string, unit, talkunit, variable, &stereo_width_set,
steps, min, max, NULL );
else
-#endif
+#endif
return set_int(string, unit, talkunit, variable, sound_callback,
steps, min, max, NULL );
else
diff --git a/firmware/export/config-iaudiox5.h b/firmware/export/config-iaudiox5.h
index 6aeaab0..ba9e953 100644
--- a/firmware/export/config-iaudiox5.h
+++ b/firmware/export/config-iaudiox5.h
@@ -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 */
diff --git a/firmware/export/sound.h b/firmware/export/sound.h
index 2079a84..1923840 100644
--- a/firmware/export/sound.h
+++ b/firmware/export/sound.h
@@ -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_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);
diff --git a/firmware/sound.c b/firmware/sound.c
index c3679d4..a2b4e96 100644
--- a/firmware/sound.c
+++ b/firmware/sound.c
@@ -87,6 +87,11 @@
[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},
[SOUND_STEREO_WIDTH] = {"%", 0, 1, 0, 255, 100, sound_set_stereo_width},
@@ -166,6 +171,22 @@
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 @@
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 @@
* 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 @@
#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 @@
#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 @@
{
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)value;
#endif
}
-#endif /* HAVE_TLV320 */
void sound_set_channels(int value)
{