Return-Path: Received: (majordomo@vger.kernel.org) by vger.kernel.org via listexpand id S1756576AbWKSLuj (ORCPT ); Sun, 19 Nov 2006 06:50:39 -0500 Received: (majordomo@vger.kernel.org) by vger.kernel.org id S1756586AbWKSLui (ORCPT ); Sun, 19 Nov 2006 06:50:38 -0500 Received: from gprs189-60.eurotel.cz ([160.218.189.60]:40873 "EHLO amd.ucw.cz") by vger.kernel.org with ESMTP id S1756576AbWKSLuh (ORCPT ); Sun, 19 Nov 2006 06:50:37 -0500 Date: Sun, 19 Nov 2006 12:49:38 +0100 From: Pavel Machek To: kernel list , Vladimir Ananiev Subject: Siemens SX1: sound cleanups Message-ID: <20061119114938.GA22514@elf.ucw.cz> MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Disposition: inline X-Warning: Reading this can be dangerous to your mental health. User-Agent: Mutt/1.5.11+cvs20060126 Sender: linux-kernel-owner@vger.kernel.org X-Mailing-List: linux-kernel@vger.kernel.org Content-Length: 22547 Lines: 695 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 #include #include -#ifdef CONFIG_PM #include -#endif #include #include #include @@ -31,18 +29,11 @@ #include #include #include "omap-alsa-sx1.h" -//#include #include -//#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 - To unsubscribe from this list: send the line "unsubscribe linux-kernel" in the body of a message to majordomo@vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html Please read the FAQ at http://www.tux.org/lkml/