Hi!
> Pavel Machek wrote:
> >
> > (I think this should go in through omap tree, because sx1 support in
> > mainline is nonexistent for now, but...)
>
> Yes, it's fine with me.
>
> > --- /dev/null
> > +++ b/sound/arm/omap/omap-alsa-sx1-mixer.c
> (snip)
> > +static int current_volume = 0; /* current volume, we cant read it */
> > +static int current_fm_volume = 0; /* current FM radio volume, we cant read it */
>
> Zero-initializations can be stripped. (And lines seem still too long
> :)
Fixed.
> > +int set_mixer_volume(int mixer_vol)
>
> Missing static? Also there are many global functions around here.
>
> > +{
> > + /* FIXME: Alsa has mixer_vol in 0-100 range, while SX1 needs 0-9 range */
>
> An incorrect information. ALSA native mixers can use any value ranges
> while OSS mixers require 0-100 range.
Aha, I had this fixed but not commited. Sorry.
> > +/* ---------------------------------------------------------------------------------------- */
> > +static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
>
> What is the reason to use __ prefix?
I guess Vladimir liked it, fixed.
> Don't use snd_xxx_t typedefs any more. They are obsolete (and will be
> removed in 2.6.20 tree).
Fixed.
> Also, the line is too long (ditto for other callbacks).
Yep, fixed.
> > +static int __pcm_playback_target_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
> > +{
> > + int ret_val = 0;
> > + int cur_val = ucontrol->value.integer.value[0];
> > +
> > + if ((cur_val >= 0) &&
> > + (cur_val < PLAYBACK_TARGET_COUNT) &&
> > + (cur_val != current_playback_target)) {
> > + if (cur_val == PLAYBACK_TARGET_LOUDSPEAKER) {
> > + set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
> > + set_loudspeaker_to_playback_target();
> > + }
> > + else if (cur_val == PLAYBACK_TARGET_HEADPHONE) {
>
> '}' ane else should be in the same line.
Ok.
> > +/*
> > + * Alsa mixer interface function for getting the volume read from the SX1 in a
> > + * 0-100 alsa mixer format.
> > + */
>
> I don't think this comment is correct...
Hmm, plus it is completely useless. Removed.
> > +static int __pcm_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
> > +{
> > + ucontrol->value.integer.value[0] = current_volume;
> > + return 0;
> > +}
> > +
> > +static int __pcm_playback_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
> > +{
> > + return set_mixer_volume(ucontrol->value.integer.value[0]);
>
> The put callback is supposed to return the following values:
>
> 0 = the value isn't changed
> 1 = the value is changed
> netagive value = error
>
> The return of 1 is not strictly needed in practice, but it's better to
> behave in a correct manner.
Hmm, that leads to some quite ugly code, but ok.
> > --- /dev/null
> > +++ b/sound/arm/omap/omap-alsa-sx1.c
> (snip)
> > +/* Send IPC message to sound server */
> > +extern int cn_sx1snd_send(unsigned int cmd, unsigned int arg1, unsigned int arg2)
> > +{
> > + struct cn_msg *m;
> > + unsigned short data[3];
> > + int err;
> > +
> > + m = kzalloc(sizeof(*m) + sizeof(data), GFP_ATOMIC);
>
> Better to use gfp_any() here, too?
> (Or is it really too big for stack?)
Let's go with gfp_any() for now.
> > +/*
> > + * Defines codec specific functions pointers that can be used from the
> > + * common omap-alsa base driver for all omap codecs.
> > + */
> > +void egold_configure(void);
> > +void egold_set_samplerate(long rate);
> > +void egold_clock_setup(void);
> > +int egold_clock_on(void);
> > +int egold_clock_off(void);
> > +int egold_get_default_samplerate(void);
>
> Do they need to be exported? I see them only in a single file as
> callbacks...
Apparently they do not. Fixed.
Thanks for all the comments!
---
From: Vladimir Ananiev <[email protected]>
Add mixer support for Siemens SX1. As SX1 is an cellphone, and this
allows voice calls, it is quite important.
Signed-off-by: Pavel Machek <[email protected]>
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 8f34673..8c035b0 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -63,6 +63,17 @@ config SND_OMAP_TSC2101
To compile this driver as a module, choose M here: the module
will be called snd-omap-tsc2101.
+config SND_SX1
+ tristate "Siemens SX1 Egold alsa driver"
+ depends on ARCH_OMAP && SND
+ select SND_PCM
+ select OMAP_MCBSP
+ help
+ Say Y here if you have a OMAP310 based Siemens SX1.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-omap-sx1.
+
config SND_OMAP_TSC2102
tristate "OMAP TSC2102 alsa driver"
depends on ARCH_OMAP && SND
diff --git a/sound/arm/omap/Makefile b/sound/arm/omap/Makefile
index 64e341f..c6bebac 100644
--- a/sound/arm/omap/Makefile
+++ b/sound/arm/omap/Makefile
@@ -10,3 +10,6 @@ snd-omap-alsa-tsc2101-objs := omap-alsa.
obj-$(CONFIG_SND_OMAP_TSC2102) += snd-omap-alsa-tsc2102.o
snd-omap-alsa-tsc2102-objs := omap-alsa.o omap-alsa-dma.o omap-alsa-tsc2102.o omap-alsa-tsc2102-mixer.o
+
+obj-$(CONFIG_SND_SX1) += snd-omap-alsa-sx1.o
+snd-omap-alsa-sx1-objs := omap-alsa.o omap-alsa-dma.o omap-alsa-sx1.o omap-alsa-sx1-mixer.o
diff --git a/sound/arm/omap/omap-alsa-sx1-mixer.c b/sound/arm/omap/omap-alsa-sx1-mixer.c
new file mode 100644
index 0000000..d826f26
--- /dev/null
+++ b/sound/arm/omap/omap-alsa-sx1-mixer.c
@@ -0,0 +1,470 @@
+/*
+ * sound/arm/omap/omap-alsa-sx1-mixer.c
+ *
+ * Alsa codec Driver for Siemens SX1 board.
+ * based on omap-alsa-tsc2101-mixer.c
+ *
+ * Copyright (C) 2006 Vladimir Ananiev (vovan888 at gmail com)
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include "omap-alsa-sx1.h"
+#include "omap-alsa-sx1-mixer.h"
+
+#include <linux/types.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+
+static int current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
+static int current_rec_src = REC_SRC_SINGLE_ENDED_MICIN_HED;
+static int current_volume; /* current volume, we cant read it */
+static int current_fm_volume; /* current FM radio volume, we cant read it */
+
+/*
+ * Select SX1 recording source.
+ */
+static void set_record_source(int val)
+{
+ /* TODO Recording is done on McBSP2 and Mic only */
+ current_rec_src = val;
+}
+
+static int set_mixer_volume(int mixer_vol)
+{
+ int ret, i;
+ if ((mixer_vol < 0) || (mixer_vol > 9)) {
+ printk(KERN_ERR "Trying a bad mixer volume (%d)!\n", mixer_vol);
+ return -EPERM;
+ }
+ ret = (current_volume != mixer_vol);
+ current_volume = mixer_vol; /* set current volume, we cant read it */
+
+ i = cn_sx1snd_send(DAC_VOLUME_UPDATE, mixer_vol, 0);
+ if (i)
+ return i;
+ return ret;
+}
+
+static void set_loudspeaker_to_playback_target(void)
+{
+ /* TODO */
+ cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_SPEAKER, 0);
+
+ current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
+}
+
+static void set_headphone_to_playback_target(void)
+{
+ /* TODO */
+ cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_HEADPHONE, 0);
+
+ current_playback_target = PLAYBACK_TARGET_HEADPHONE;
+}
+
+static void set_telephone_to_playback_target(void)
+{
+ /* TODO */
+ cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_PHONE, 0);
+
+ current_playback_target = PLAYBACK_TARGET_CELLPHONE;
+}
+
+static void set_telephone_to_record_source(void)
+{
+ cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_PHONE, 0);
+}
+
+static void init_playback_targets(void)
+{
+ set_loudspeaker_to_playback_target();
+ set_mixer_volume(DEFAULT_OUTPUT_VOLUME);
+}
+
+/*
+ * Initializes SX1 record source (to mic) and playback target (to loudspeaker)
+ */
+void snd_omap_init_mixer(void)
+{
+ /* Select headset to record source */
+ set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
+ /* Init loudspeaker as a default playback target*/
+ init_playback_targets();
+}
+
+/* -------------------------------------------------------------------------- */
+static int pcm_playback_target_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[PLAYBACK_TARGET_COUNT] = {
+ "Loudspeaker", "Headphone", "Cellphone"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = PLAYBACK_TARGET_COUNT;
+ if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1) {
+ uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
+ }
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int pcm_playback_target_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = current_playback_target;
+ return 0;
+}
+
+static int pcm_playback_target_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int ret_val = 0;
+ int cur_val = ucontrol->value.integer.value[0];
+
+ if ((cur_val >= 0) &&
+ (cur_val < PLAYBACK_TARGET_COUNT) &&
+ (cur_val != current_playback_target)) {
+ if (cur_val == PLAYBACK_TARGET_LOUDSPEAKER) {
+ set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
+ set_loudspeaker_to_playback_target();
+ } else if (cur_val == PLAYBACK_TARGET_HEADPHONE) {
+ set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HND);
+ set_headphone_to_playback_target();
+ } else if (cur_val == PLAYBACK_TARGET_CELLPHONE) {
+ set_telephone_to_record_source();
+ set_telephone_to_playback_target();
+ }
+ ret_val = 1;
+ }
+ return ret_val;
+}
+
+/*-----------------------------------------------------------*/
+static int pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 9;
+ return 0;
+}
+
+static int pcm_playback_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = current_volume;
+ return 0;
+}
+
+static int pcm_playback_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return set_mixer_volume(ucontrol->value.integer.value[0]);
+}
+
+static int pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int pcm_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 1;
+ return 0;
+}
+
+static int pcm_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return 0;
+}
+
+/* ----------------------------------------------------------- */
+
+static int headset_playback_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 9;
+ return 0;
+}
+
+static int headset_playback_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = current_volume;
+ return 0;
+}
+
+static int headset_playback_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return set_mixer_volume(ucontrol->value.integer.value[0]);
+}
+
+static int headset_playback_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int headset_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 1;
+ return 0;
+}
+
+static int headset_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ /* mute/unmute headset */
+#if 0
+ return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+ TSC2101_HEADSET_GAIN_CTRL,
+ 15);
+#endif
+ return 0;
+}
+/* ----------------------------------------------------------- */
+static int fmradio_playback_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 9;
+ return 0;
+}
+
+static int fmradio_playback_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = current_fm_volume;
+ return 0;
+}
+
+static int fmradio_playback_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int ret = current_fm_volume != ucontrol->value.integer.value[0];
+ int i;
+ current_fm_volume = ucontrol->value.integer.value[0];
+ i = cn_sx1snd_send(DAC_FMRADIO_OPEN, current_fm_volume, 0);
+ if (i)
+ return i;
+ return ret;
+}
+
+static int fmradio_playback_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int fmradio_playback_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 1;
+ return 0;
+}
+
+static int fmradio_playback_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ /* mute/unmute FM radio */
+ if (ucontrol->value.integer.value[0])
+ cn_sx1snd_send(DAC_FMRADIO_OPEN, current_fm_volume, 0);
+ else
+ cn_sx1snd_send(DAC_FMRADIO_CLOSE, 0, 0);
+
+ return 0;
+}
+/* ----------------------------------------------------------- */
+static int cellphone_input_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int cellphone_input_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 1;
+ return 0;
+}
+
+static int cellphone_input_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+#if 0
+ return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+ TSC2101_BUZZER_GAIN_CTRL, 15);
+#endif
+ return 0;
+}
+/* ----------------------------------------------------------- */
+
+static int buzzer_input_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int buzzer_input_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 1;
+ return 0;
+}
+
+static int buzzer_input_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+#if 0
+ return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+ TSC2101_BUZZER_GAIN_CTRL, 6);
+#endif
+ return 0;
+}
+/*-----------------------------------------------------------*/
+
+static struct snd_kcontrol_new egold_control[] __devinitdata = {
+ {
+ .name = "Playback Playback Route",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = pcm_playback_target_info,
+ .get = pcm_playback_target_get,
+ .put = pcm_playback_target_put,
+ }, {
+ .name = "Master Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = pcm_playback_volume_info,
+ .get = pcm_playback_volume_get,
+ .put = pcm_playback_volume_put,
+ }, {
+ .name = "Master Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = pcm_playback_switch_info,
+ .get = pcm_playback_switch_get,
+ .put = pcm_playback_switch_put,
+ }, {
+ .name = "Headset Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 1,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = headset_playback_volume_info,
+ .get = headset_playback_volume_get,
+ .put = headset_playback_volume_put,
+ }, {
+ .name = "Headset Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 1,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = headset_playback_switch_info,
+ .get = headset_playback_switch_get,
+ .put = headset_playback_switch_put,
+ }, {
+ .name = "FM Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 2,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = fmradio_playback_volume_info,
+ .get = fmradio_playback_volume_get,
+ .put = fmradio_playback_volume_put,
+ }, {
+ .name = "FM Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 2,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = fmradio_playback_switch_info,
+ .get = fmradio_playback_switch_get,
+ .put = fmradio_playback_switch_put,
+ }, {
+ .name = "Cellphone Input Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = cellphone_input_switch_info,
+ .get = cellphone_input_switch_get,
+ .put = cellphone_input_switch_put,
+ }, {
+ .name = "Buzzer Input Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .index = 0,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = buzzer_input_switch_info,
+ .get = buzzer_input_switch_get,
+ .put = buzzer_input_switch_put,
+ }
+};
+
+#ifdef CONFIG_PM
+void snd_omap_suspend_mixer(void)
+{
+}
+
+void snd_omap_resume_mixer(void)
+{
+ snd_omap_init_mixer();
+}
+#endif
+
+int snd_omap_mixer(struct snd_card_omap_codec *egold)
+{
+ int i = 0;
+ int err = 0;
+
+ if (!egold)
+ return -EINVAL;
+
+ for (i=0; i < ARRAY_SIZE(egold_control); i++) {
+ err = snd_ctl_add(egold->card,
+ snd_ctl_new1(&egold_control[i], egold->card));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
diff --git a/sound/arm/omap/omap-alsa-sx1-mixer.h b/sound/arm/omap/omap-alsa-sx1-mixer.h
new file mode 100644
index 0000000..02b8b6a
--- /dev/null
+++ b/sound/arm/omap/omap-alsa-sx1-mixer.h
@@ -0,0 +1,45 @@
+/*
+ * sound/arm/omap/omap-alsa-sx1-mixer.h
+ *
+ * Alsa codec Driver for Siemens SX1 board.
+ * based on omap-alsa-tsc2101-mixer.c
+ *
+ * Copyright (C) 2006 Vladimir Ananiev (vovan888 at gmail com)
+ *
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#ifndef OMAPALSATSC2101MIXER_H_
+#define OMAPALSATSC2101MIXER_H_
+
+#include "omap-alsa-dma.h"
+
+#define PLAYBACK_TARGET_COUNT 0x03
+#define PLAYBACK_TARGET_LOUDSPEAKER 0x00
+#define PLAYBACK_TARGET_HEADPHONE 0x01
+#define PLAYBACK_TARGET_CELLPHONE 0x02
+
+/* following are used for register 03h Mixer PGA control bits D7-D5 for selecting record source */
+#define REC_SRC_TARGET_COUNT 0x08
+#define REC_SRC_SINGLE_ENDED_MICIN_HED 0x00 /* oss code referred to MIXER_LINE */
+#define REC_SRC_SINGLE_ENDED_MICIN_HND 0x01 /* oss code referred to MIXER_MIC */
+#define REC_SRC_SINGLE_ENDED_AUX1 0x02
+#define REC_SRC_SINGLE_ENDED_AUX2 0x03
+#define REC_SRC_MICIN_HED_AND_AUX1 0x04
+#define REC_SRC_MICIN_HED_AND_AUX2 0x05
+#define REC_SRC_MICIN_HND_AND_AUX1 0x06
+#define REC_SRC_MICIN_HND_AND_AUX2 0x07
+
+#define DEFAULT_OUTPUT_VOLUME 5 /* default output volume to dac dgc */
+#define DEFAULT_INPUT_VOLUME 2 /* default record volume */
+
+#endif
diff --git a/sound/arm/omap/omap-alsa-sx1.c b/sound/arm/omap/omap-alsa-sx1.c
new file mode 100644
index 0000000..c05a65e
--- /dev/null
+++ b/sound/arm/omap/omap-alsa-sx1.c
@@ -0,0 +1,281 @@
+/*
+ * arch/arm/mach-omap1/omap-alsa-sx1.c
+ *
+ * Alsa codec Driver for Siemens SX1 board.
+ * based on omap-alsa-tsc2101.c and cn_test.c example by Evgeniy Polyakov
+ *
+ * Copyright (C) 2006 Vladimir Ananiev (vovan888 at gmail com)
+ *
+ * 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.
+ */
+
+#include <linux/delay.h>
+#include <linux/soundcard.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <asm/io.h>
+#include <asm/arch/mcbsp.h>
+
+#include <linux/slab.h>
+#include <linux/pm.h>
+#include <asm/mach-types.h>
+#include <asm/arch/dma.h>
+#include <asm/arch/clock.h>
+#include <asm/arch/gpio.h>
+
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-sx1.h"
+
+#include <linux/connector.h>
+
+/* Connector implementation */
+static struct cb_id cn_sx1snd_id = { CN_IDX_SX1SND, CN_VAL_SX1SND };
+static char cn_sx1snd_name[] = "cn_sx1snd";
+
+void cn_sx1snd_callback(void *data)
+{
+ struct cn_msg *msg = (struct cn_msg *)data;
+
+ printk("%s: %lu: idx=%x, val=%x, seq=%u, ack=%u, len=%d: %s.\n",
+ __func__, jiffies, msg->id.idx, msg->id.val,
+ msg->seq, msg->ack, msg->len, (char *)msg->data);
+}
+
+/* Send IPC message to sound server */
+extern int cn_sx1snd_send(unsigned int cmd, unsigned int arg1, unsigned int arg2)
+{
+ struct cn_msg *m;
+ unsigned short data[3];
+ int err;
+
+ m = kzalloc(sizeof(*m) + sizeof(data), gfp_any());
+ if (!m)
+ return -1;
+
+ memcpy(&m->id, &cn_sx1snd_id, sizeof(m->id));
+ m->seq = 1;
+ m->len = sizeof(data);
+
+ data[0] = (unsigned short)cmd;
+ data[1] = (unsigned short)arg1;
+ data[2] = (unsigned short)arg2;
+
+ memcpy(m + 1, data, m->len);
+
+ err = cn_netlink_send(m, CN_IDX_SX1SND, gfp_any());
+ snd_printd("sent= %02X %02X %02X, err=%d\n", cmd,arg1,arg2,err);
+ kfree(m);
+
+ if (err == -ESRCH)
+ return -1; /* there are no listeners on socket */
+ return 0;
+}
+
+/* Hardware capabilities
+ *
+ * DAC USB-mode sampling rates (MCLK = 12 MHz)
+ * The rates and rate_reg_into MUST be in the same order
+ */
+static unsigned int rates[] = {
+ 8000, 11025, 12000,
+ 16000, 22050, 24000,
+ 32000, 44100, 48000,
+};
+
+static snd_pcm_hw_constraint_list_t egold_hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static snd_pcm_hardware_t egold_snd_omap_alsa_playback = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static snd_pcm_hardware_t egold_snd_omap_alsa_capture = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static long current_rate = -1; /* current rate in egold format 0..8 */
+/*
+ * ALSA operations according to board file
+ */
+
+/*
+ * Sample rate changing
+ */
+void egold_set_samplerate(long sample_rate)
+{
+ int egold_rate = 0;
+ int clkgdv = 0;
+ u16 srgr1, srgr2;
+
+ /* Set the sample rate */
+#if 0
+ /* fw15: 5005E490 - divs are different !!! */
+ clkgdv = CODEC_CLOCK / (sample_rate * (DEFAULT_BITPERSAMPLE * 2 - 1));
+#endif
+ switch (sample_rate) {
+ case 8000: clkgdv = 71; egold_rate = FRQ_8000; break;
+ case 11025: clkgdv = 51; egold_rate = FRQ_11025; break;
+ case 12000: clkgdv = 47; egold_rate = FRQ_12000; break;
+ case 16000: clkgdv = 35; egold_rate = FRQ_16000; break;
+ case 22050: clkgdv = 25; egold_rate = FRQ_22050; break;
+ case 24000: clkgdv = 23; egold_rate = FRQ_24000; break;
+ case 32000: clkgdv = 17; egold_rate = FRQ_32000; break;
+ case 44100: clkgdv = 12; egold_rate = FRQ_44100; break;
+ case 48000: clkgdv = 11; egold_rate = FRQ_48000; break;
+ }
+
+ srgr1 = (FWID(DEFAULT_BITPERSAMPLE - 1) | CLKGDV(clkgdv));
+ srgr2 = ((FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1)));
+
+ OMAP_MCBSP_WRITE(OMAP1510_MCBSP1_BASE, SRGR2, srgr2);
+ OMAP_MCBSP_WRITE(OMAP1510_MCBSP1_BASE, SRGR1, srgr1);
+ current_rate = egold_rate;
+ snd_printd("set samplerate=%ld\n", sample_rate);
+
+}
+
+void egold_configure(void)
+{
+}
+
+/*
+ * Omap MCBSP clock and Power Management configuration
+ *
+ * Here we have some functions that allows clock to be enabled and
+ * disabled only when needed. Besides doing clock configuration
+ * it allows turn on/turn off audio when necessary.
+ */
+
+/*
+ * Do clock framework mclk search
+ */
+void egold_clock_setup(void)
+{
+ omap_request_gpio(OSC_EN);
+ omap_set_gpio_direction(OSC_EN, 0); /* output */
+ snd_printd("\n");
+}
+
+/*
+ * Do some sanity check, set clock rate, starts it and turn codec audio on
+ */
+int egold_clock_on(void)
+{
+ omap_set_gpio_dataout(OSC_EN, 1);
+ egold_set_samplerate(44100); /* TODO */
+ cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_SPEAKER, 0);
+ cn_sx1snd_send(DAC_OPEN_DEFAULT, current_rate , 4);
+ snd_printd("\n");
+ return 0;
+}
+
+/*
+ * Do some sanity check, turn clock off and then turn codec audio off
+ */
+int egold_clock_off(void)
+{
+ cn_sx1snd_send(DAC_CLOSE, 0 , 0);
+ cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_PHONE, 0);
+ omap_set_gpio_dataout(OSC_EN, 0);
+ snd_printd("\n");
+ return 0;
+}
+
+int egold_get_default_samplerate(void)
+{
+ snd_printd("\n");
+ return DEFAULT_SAMPLE_RATE;
+}
+
+static int __init snd_omap_alsa_egold_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct omap_alsa_codec_config *codec_cfg;
+
+ codec_cfg = pdev->dev.platform_data;
+ if (!codec_cfg)
+ return -ENODEV;
+
+ codec_cfg->hw_constraints_rates = &egold_hw_constraints_rates;
+ codec_cfg->snd_omap_alsa_playback = &egold_snd_omap_alsa_playback;
+ codec_cfg->snd_omap_alsa_capture = &egold_snd_omap_alsa_capture;
+ codec_cfg->codec_configure_dev = egold_configure;
+ codec_cfg->codec_set_samplerate = egold_set_samplerate;
+ codec_cfg->codec_clock_setup = egold_clock_setup;
+ codec_cfg->codec_clock_on = egold_clock_on;
+ codec_cfg->codec_clock_off = egold_clock_off;
+ codec_cfg->get_default_samplerate = egold_get_default_samplerate;
+ ret = snd_omap_alsa_post_probe(pdev, codec_cfg);
+
+ snd_printd("\n");
+ return ret;
+}
+
+static struct platform_driver omap_alsa_driver = {
+ .probe = snd_omap_alsa_egold_probe,
+ .remove = snd_omap_alsa_remove,
+ .suspend = snd_omap_alsa_suspend,
+ .resume = snd_omap_alsa_resume,
+ .driver = {
+ .name = "omap_alsa_mcbsp",
+ },
+};
+
+static int __init omap_alsa_egold_init(void)
+{
+ int retval;
+
+ retval = cn_add_callback(&cn_sx1snd_id, cn_sx1snd_name, cn_sx1snd_callback);
+ if(retval)
+ printk(KERN_WARNING "cn_sx1snd failed to register\n");
+ return platform_driver_register(&omap_alsa_driver);
+}
+
+static void __exit omap_alsa_egold_exit(void)
+{
+ cn_del_callback(&cn_sx1snd_id);
+ platform_driver_unregister(&omap_alsa_driver);
+}
+
+module_init(omap_alsa_egold_init);
+module_exit(omap_alsa_egold_exit);
diff --git a/sound/arm/omap/omap-alsa-sx1.h b/sound/arm/omap/omap-alsa-sx1.h
new file mode 100644
index 0000000..e7bec00
--- /dev/null
+++ b/sound/arm/omap/omap-alsa-sx1.h
@@ -0,0 +1,70 @@
+/*
+ * arch/arc/mach-omap1/omap-alsa-sx1.h
+ *
+ * based on omap-alsa-tsc2101.h
+ *
+ * Alsa Driver for Siemens SX1.
+ * Copyright (C) 2006 Vladimir Ananiev (vovan888 at gmail com)
+ *
+ * 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.
+ */
+
+#ifndef OMAP_ALSA_SX1_H_
+#define OMAP_ALSA_SX1_H_
+
+#include <linux/types.h>
+
+#define NUMBER_SAMPLE_RATES_SUPPORTED 9
+
+/*
+ * AUDIO related MACROS
+ */
+#ifndef DEFAULT_BITPERSAMPLE
+#define DEFAULT_BITPERSAMPLE 16
+#endif
+
+#define DEFAULT_SAMPLE_RATE 44100
+/* fw15: 18356000 */
+#define CODEC_CLOCK 18359000
+/* McBSP for playing music */
+#define AUDIO_MCBSP OMAP_MCBSP1
+/* McBSP for record/play audio from phone and mic */
+#define AUDIO_MCBSP_PCM OMAP_MCBSP2
+/* gpio pin for enable/disable clock */
+#define OSC_EN 2
+
+/* Send IPC message to sound server */
+extern int cn_sx1snd_send(unsigned int cmd, unsigned int arg1, unsigned int arg2);
+/* cmd for IPC_GROUP_DAC */
+#define DAC_VOLUME_UPDATE 0
+#define DAC_SETAUDIODEVICE 1
+#define DAC_OPEN_RING 2
+#define DAC_OPEN_DEFAULT 3
+#define DAC_CLOSE 4
+#define DAC_FMRADIO_OPEN 5
+#define DAC_FMRADIO_CLOSE 6
+#define DAC_PLAYTONE 7
+/* cmd for IPC_GROUP_PCM */
+#define PCM_PLAY (0+8)
+#define PCM_RECORD (1+8)
+#define PCM_CLOSE (2+8)
+
+/* for DAC_SETAUDIODEVICE */
+#define SX1_DEVICE_SPEAKER 0
+#define SX1_DEVICE_HEADPHONE 4
+#define SX1_DEVICE_PHONE 3
+/* frequencies for MdaDacOpenDefaultL, MdaDacOpenRingL */
+#define FRQ_8000 0
+#define FRQ_11025 1
+#define FRQ_12000 2
+#define FRQ_16000 3
+#define FRQ_22050 4
+#define FRQ_24000 5
+#define FRQ_32000 6
+#define FRQ_44100 7
+#define FRQ_48000 8
+
+#endif
--
(english) http://www.livejournal.com/~pavelmachek
(cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html
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