Hi!
These are cleanups for codingstyle in sound parts of siemens sx1. They
should not change any code. Please apply,
Pavel
diff --git a/sound/arm/omap/omap-alsa-sx1-mixer.c b/sound/arm/omap/omap-alsa-sx1-mixer.c
index 8ca4c95..b036b3b 100644
--- a/sound/arm/omap/omap-alsa-sx1-mixer.c
+++ b/sound/arm/omap/omap-alsa-sx1-mixer.c
@@ -29,41 +29,34 @@ #define M_DPRINTK(ARGS...) /* nop */
static int current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
static int current_rec_src = REC_SRC_SINGLE_ENDED_MICIN_HED;
-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
+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 */
-/* TODO
- * For selecting SX1 recourd source.
+/*
+ * Select SX1 recording source.
*/
static void set_record_source(int val)
{
- // TODO Recording is done on McBSP2 and Mic only
+ /* TODO Recording is done on McBSP2 and Mic only */
current_rec_src = val;
}
-/*
- * Converts the Alsa mixer volume (0 - 100) to SX1
- * (0 - 9) volume.
- */
-int set_mixer_volume(int mixerVol)
+int set_mixer_volume(int mixer_vol)
{
- int retVal;
+ /* FIXME: Alsa has mixer_vol in 0-100 range, while SX1 needs 0-9 range */
- if ((mixerVol < 0) ||
- (mixerVol > 9) ){
- printk(KERN_ERR "Trying a bad mixer volume (%d)!\n", mixerVol);
+ if ((mixer_vol < 0) || (mixer_vol > 9)) {
+ printk(KERN_ERR "Trying a bad mixer volume (%d)!\n", mixer_vol);
return -EPERM;
}
- current_volume = mixerVol; // set current volume, we cant read it
- M_DPRINTK("mixer volume = %d\n", mixerVol);
+ current_volume = mixer_vol; /* set current volume, we cant read it */
- retVal = cn_sx1snd_send(DAC_VOLUME_UPDATE, mixerVol, 0 );
- return retVal;
+ return cn_sx1snd_send(DAC_VOLUME_UPDATE, mixer_vol, 0);
}
void set_loudspeaker_to_playback_target(void)
{
- // TODO
+ /* TODO */
cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_SPEAKER, 0);
current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
@@ -71,7 +64,7 @@ void set_loudspeaker_to_playback_target(
void set_headphone_to_playback_target(void)
{
- // TODO
+ /* TODO */
cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_HEADPHONE, 0);
current_playback_target = PLAYBACK_TARGET_HEADPHONE;
@@ -79,7 +72,7 @@ void set_headphone_to_playback_target(vo
void set_telephone_to_playback_target(void)
{
- // TODO
+ /* TODO */
cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_PHONE, 0);
current_playback_target = PLAYBACK_TARGET_CELLPHONE;
@@ -93,26 +86,21 @@ static void set_telephone_to_record_sour
void init_playback_targets(void)
{
set_loudspeaker_to_playback_target();
-
set_mixer_volume(DEFAULT_OUTPUT_VOLUME);
}
/*
- * Initializes SX1 recourd source (to mic) and playback target (to loudspeaker)
+ * Initializes SX1 record source (to mic) and playback target (to loudspeaker)
*/
void snd_omap_init_mixer(void)
{
- FN_IN;
-
- /* Select headset to record source (MIC_INHED)*/
+ /* Select headset to record source */
set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
/* Init loudspeaker as a default playback target*/
init_playback_targets();
-
- FN_OUT(0);
}
-/*--------------------------------------------------------------------------------------------*/
+/* ---------------------------------------------------------------------------------------- */
static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
static char *texts[PLAYBACK_TARGET_COUNT] = {
@@ -125,8 +113,8 @@ static int __pcm_playback_target_info(sn
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]);
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
return 0;
}
@@ -138,29 +126,27 @@ static int __pcm_playback_target_get(snd
static int __pcm_playback_target_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- int retVal;
- int curVal;
+ int ret_val = 0;
+ int cur_val = ucontrol->value.integer.value[0];
- retVal = 0;
- curVal = ucontrol->value.integer.value[0];
- if ((curVal >= 0) &&
- (curVal < PLAYBACK_TARGET_COUNT) &&
- (curVal != current_playback_target)) {
- if (curVal == PLAYBACK_TARGET_LOUDSPEAKER) {
+ 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 (curVal == PLAYBACK_TARGET_HEADPHONE) {
+ else if (cur_val == PLAYBACK_TARGET_HEADPHONE) {
set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HND);
set_headphone_to_playback_target();
}
- else if (curVal == PLAYBACK_TARGET_CELLPHONE) {
+ else if (cur_val == PLAYBACK_TARGET_CELLPHONE) {
set_telephone_to_record_source();
set_telephone_to_playback_target();
}
- retVal = 1;
+ ret_val = 1;
}
- return retVal;
+ return ret_val;
}
/*--------------------------------------------------------------------------------------------*/
@@ -175,14 +161,11 @@ static int __pcm_playback_volume_info(sn
/*
* Alsa mixer interface function for getting the volume read from the SX1 in a
- * 0 -100 alsa mixer format.
+ * 0-100 alsa mixer format.
*/
static int __pcm_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
-
- ucontrol->value.integer.value[0] = current_volume;
-
- M_DPRINTK("mixer volume = %ld\n", current_volume);
+ ucontrol->value.integer.value[0] = current_volume;
return 0;
}
@@ -202,17 +185,17 @@ static int __pcm_playback_switch_info(sn
static int __pcm_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- ucontrol->value.integer.value[0] = 1;
+ ucontrol->value.integer.value[0] = 1;
return 0;
}
static int __pcm_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
-// return dac_gain_control_unmute(ucontrol->value.integer.value[0],
-// ucontrol->value.integer.value[1]);
return 0;
}
-/*--------------------------------------------------------------------------------------------*/
+
+/* -------------------------------------------------------------------------------------------- */
+
static int __headset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -225,8 +208,6 @@ static int __headset_playback_volume_inf
static int __headset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
ucontrol->value.integer.value[0] = current_volume;
-
- M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
return 0;
}
@@ -246,19 +227,21 @@ static int __headset_playback_switch_inf
static int __headset_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- ucontrol->value.integer.value[0] = 1;
+ ucontrol->value.integer.value[0] = 1;
return 0;
}
static int __headset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- // mute/unmute headset
-// return adc_pga_unmute_control(ucontrol->value.integer.value[0],
-// TSC2101_HEADSET_GAIN_CTRL,
-// 15);
+ /* 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -270,9 +253,7 @@ static int __fmradio_playback_volume_inf
static int __fmradio_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- ucontrol->value.integer.value[0] = current_fm_volume;
-
- M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
+ ucontrol->value.integer.value[0] = current_fm_volume;
return 0;
}
@@ -294,21 +275,21 @@ static int __fmradio_playback_switch_inf
static int __fmradio_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- ucontrol->value.integer.value[0] = 1;
+ ucontrol->value.integer.value[0] = 1;
return 0;
}
static int __fmradio_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- // mute/unmute FM radio
- if(ucontrol->value.integer.value[0])
+ /* 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
@@ -320,18 +301,19 @@ static int __cellphone_input_switch_info
static int __cellphone_input_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- ucontrol->value.integer.value[0] = 1;
+ ucontrol->value.integer.value[0] = 1;
return 0;
}
static int __cellphone_input_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
-// return adc_pga_unmute_control(ucontrol->value.integer.value[0],
-// TSC2101_BUZZER_GAIN_CTRL,
-// 15);
+#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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
@@ -344,15 +326,16 @@ static int __buzzer_input_switch_info(sn
static int __buzzer_input_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
- ucontrol->value.integer.value[0] = 1;
+ ucontrol->value.integer.value[0] = 1;
return 0;
}
static int __buzzer_input_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
-// return adc_pga_unmute_control(ucontrol->value.integer.value[0],
-// TSC2101_BUZZER_GAIN_CTRL,
-// 6);
+#if 0
+ return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+ TSC2101_BUZZER_GAIN_CTRL, 6);
+#endif
return 0;
}
/*--------------------------------------------------------------------------------------------*/
@@ -434,7 +417,6 @@ static snd_kcontrol_new_t egold_control[
};
#ifdef CONFIG_PM
-
void snd_omap_suspend_mixer(void)
{
}
@@ -447,18 +429,16 @@ #endif
int snd_omap_mixer(struct snd_card_omap_codec *egold)
{
- int i=0;
- int err=0;
+ int i = 0;
+ int err = 0;
- if (!egold) {
+ if (!egold)
return -EINVAL;
- }
+
for (i=0; i < ARRAY_SIZE(egold_control); i++) {
- if ((err = snd_ctl_add(egold->card,
- snd_ctl_new1(&egold_control[i],
- egold->card))) < 0) {
+ 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
index e67f48a..02b8b6a 100644
--- a/sound/arm/omap/omap-alsa-sx1-mixer.h
+++ b/sound/arm/omap/omap-alsa-sx1-mixer.h
@@ -30,8 +30,8 @@ #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_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
@@ -39,7 +39,7 @@ #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
+#define DEFAULT_OUTPUT_VOLUME 5 /* default output volume to dac dgc */
+#define DEFAULT_INPUT_VOLUME 2 /* default record volume */
-#endif /*OMAPALSATSC2101MIXER_H_*/
+#endif
diff --git a/sound/arm/omap/omap-alsa-sx1.c b/sound/arm/omap/omap-alsa-sx1.c
index 0edaf95..64c09dc 100644
--- a/sound/arm/omap/omap-alsa-sx1.c
+++ b/sound/arm/omap/omap-alsa-sx1.c
@@ -20,9 +20,7 @@ #include <asm/io.h>
#include <asm/arch/mcbsp.h>
#include <linux/slab.h>
-#ifdef CONFIG_PM
#include <linux/pm.h>
-#endif
#include <asm/mach-types.h>
#include <asm/arch/dma.h>
#include <asm/arch/clock.h>
@@ -31,18 +29,11 @@ #include <asm/arch/gpio.h>
#include <asm/arch/omap-alsa.h>
#include "omap-alsa-sx1.h"
-//#include <linux/skbuff.h>
#include <linux/connector.h>
-//#define M_DPRINTK(ARGS...) printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
-#define M_DPRINTK(ARGS...) /* nop */
-
-//static struct clk *egold_mclk = 0;
-
/* Connector implementation */
static struct cb_id cn_sx1snd_id = { CN_IDX_SX1SND, CN_VAL_SX1SND };
static char cn_sx1snd_name[] = "cn_sx1snd";
-//static struct sock *nls;
void cn_sx1snd_callback(void *data)
{
@@ -52,39 +43,39 @@ void cn_sx1snd_callback(void *data)
__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;
+ int err;
- m = kmalloc(sizeof(*m) + sizeof(data), GFP_ATOMIC);
- if (m) {
- memset(m, 0, sizeof(*m) + sizeof(data));
+ m = kzalloc(sizeof(*m) + sizeof(data), GFP_ATOMIC);
+ if (!m)
+ return -1;
- memcpy(&m->id, &cn_sx1snd_id, sizeof(m->id));
- m->seq = 1;//cn_test_timer_counter;
- m->len = sizeof(data);
+ 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;
+ data[0] = (unsigned short)cmd;
+ data[1] = (unsigned short)arg1;
+ data[2] = (unsigned short)arg2;
- memcpy(m + 1, data, m->len);
+ memcpy(m + 1, data, m->len);
- err = cn_netlink_send(m, CN_IDX_SX1SND, gfp_any());
- M_DPRINTK("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;
- }
- return -1; // some error
+ 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 */
-/*
+/* Hardware capabilities
+ *
* DAC USB-mode sampling rates (MCLK = 12 MHz)
* The rates and rate_reg_into MUST be in the same order
*/
@@ -142,7 +133,7 @@ static snd_pcm_hardware_t egold_snd_omap
.fifo_size = 0,
};
-static long current_rate = -1;// current rate in egold format 0..8
+static long current_rate = -1; /* current rate in egold format 0..8 */
/*
* ALSA operations according to board file
*/
@@ -154,14 +145,14 @@ void egold_set_samplerate(long sample_ra
{
int egold_rate = 0;
int clkgdv = 0;
-
u16 srgr1, srgr2;
- ADEBUG();
/* Set the sample rate */
-// clkgdv = CODEC_CLOCK / (sample_rate * (DEFAULT_BITPERSAMPLE * 2 - 1)); //fw15: 5005E490 - divs are different !!!
- switch(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;
@@ -171,7 +162,7 @@ void egold_set_samplerate(long sample_ra
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)));
@@ -179,7 +170,7 @@ void egold_set_samplerate(long sample_ra
OMAP_MCBSP_WRITE(OMAP1510_MCBSP1_BASE, SRGR2, srgr2);
OMAP_MCBSP_WRITE(OMAP1510_MCBSP1_BASE, SRGR1, srgr1);
current_rate = egold_rate;
- M_DPRINTK("set samplerate=%ld\n", sample_rate);
+ snd_printd("set samplerate=%ld\n", sample_rate);
}
@@ -201,8 +192,8 @@ void egold_configure(void)
void egold_clock_setup(void)
{
omap_request_gpio(OSC_EN);
- omap_set_gpio_direction(OSC_EN, 0); // output pin
- M_DPRINTK("\n");
+ omap_set_gpio_direction(OSC_EN, 0); /* output */
+ snd_printd("\n");
}
/*
@@ -211,10 +202,10 @@ void egold_clock_setup(void)
int egold_clock_on(void)
{
omap_set_gpio_dataout(OSC_EN, 1);
- egold_set_samplerate(44100);// TODO
+ egold_set_samplerate(44100); /* TODO */
cn_sx1snd_send(DAC_SETAUDIODEVICE, SX1_DEVICE_SPEAKER, 0);
cn_sx1snd_send(DAC_OPEN_DEFAULT, current_rate , 4);
- M_DPRINTK("\n");
+ snd_printd("\n");
return 0;
}
@@ -226,37 +217,37 @@ 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);
- M_DPRINTK("\n");
+ snd_printd("\n");
return 0;
}
int egold_get_default_samplerate(void)
{
- M_DPRINTK("\n");
+ 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;
+ int ret;
+ struct omap_alsa_codec_config *codec_cfg;
codec_cfg = pdev->dev.platform_data;
- if (codec_cfg != NULL) {
- 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);
- }
- else
- ret = -ENODEV;
- M_DPRINTK("\n");
+ 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;
}
@@ -273,7 +264,7 @@ static struct platform_driver omap_alsa_
static int __init omap_alsa_egold_init(void)
{
int retval;
- ADEBUG();
+
retval = cn_add_callback(&cn_sx1snd_id, cn_sx1snd_name, cn_sx1snd_callback);
if(retval)
printk(KERN_WARNING "cn_sx1snd failed to register\n");
@@ -282,7 +273,6 @@ static int __init omap_alsa_egold_init(v
static void __exit omap_alsa_egold_exit(void)
{
- ADEBUG();
cn_del_callback(&cn_sx1snd_id);
platform_driver_unregister(&omap_alsa_driver);
}
diff --git a/sound/arm/omap/omap-alsa-sx1.h b/sound/arm/omap/omap-alsa-sx1.h
index fdf55ee..34e26fc 100644
--- a/sound/arm/omap/omap-alsa-sx1.h
+++ b/sound/arm/omap/omap-alsa-sx1.h
@@ -27,14 +27,14 @@ #define DEFAULT_BITPERSAMPLE 16
#endif
#define DEFAULT_SAMPLE_RATE 44100
-// fw15: 18356000
+/* fw15: 18356000 */
#define CODEC_CLOCK 18359000
-// McBSP for playing music
+/* McBSP for playing music */
#define AUDIO_MCBSP OMAP_MCBSP1
-// McBSP for record/play audio from phone and mic
+/* 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
+/* gpio pin for enable/disable clock */
+#define OSC_EN 2
/*
* Defines codec specific functions pointers that can be used from the
@@ -49,25 +49,25 @@ int egold_get_default_samplerate(void);
/* 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
+/* 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_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)
+/* cmd for IPC_GROUP_PCM */
+#define PCM_PLAY (0+8)
#define PCM_RECORD (1+8)
-#define PCM_CLOSE (2+8)
+#define PCM_CLOSE (2+8)
-// for DAC_SETAUDIODEVICE
+/* for DAC_SETAUDIODEVICE */
#define SX1_DEVICE_SPEAKER 0
#define SX1_DEVICE_HEADPHONE 4
#define SX1_DEVICE_PHONE 3
-// frequencies for MdaDacOpenDefaultL, MdaDacOpenRingL
+/* frequencies for MdaDacOpenDefaultL, MdaDacOpenRingL */
#define FRQ_8000 0
#define FRQ_11025 1
#define FRQ_12000 2
@@ -78,6 +78,4 @@ #define FRQ_32000 6
#define FRQ_44100 7
#define FRQ_48000 8
- /* Netlink socket defs for connection with userspace MUX */
-
-#endif /*OMAP_ALSA_SX1_H_*/
+#endif
--
(english) http://www.livejournal.com/~pavelmachek
(cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html
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