2 * h1940-uda1380.c -- ALSA Soc Audio Layer
4 * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
5 * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
7 * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
9 * This program is free software; you can redistribute it and/or modify it
10 * under the terms of the GNU General Public License as published by the
11 * Free Software Foundation; either version 2 of the License, or (at your
12 * option) any later version.
16 #include <linux/types.h>
17 #include <linux/gpio.h>
18 #include <linux/module.h>
20 #include <sound/soc.h>
21 #include <sound/jack.h>
24 #include <asm/mach-types.h>
26 #include <mach/gpio-samsung.h>
27 #include "s3c24xx-i2s.h"
29 static const unsigned int rates
[] = {
35 static const struct snd_pcm_hw_constraint_list hw_rates
= {
36 .count
= ARRAY_SIZE(rates
),
40 static struct snd_soc_jack hp_jack
;
42 static struct snd_soc_jack_pin hp_jack_pins
[] = {
44 .pin
= "Headphone Jack",
45 .mask
= SND_JACK_HEADPHONE
,
49 .mask
= SND_JACK_HEADPHONE
,
54 static struct snd_soc_jack_gpio hp_jack_gpios
[] = {
56 .gpio
= S3C2410_GPG(4),
58 .report
= SND_JACK_HEADPHONE
,
64 static int h1940_startup(struct snd_pcm_substream
*substream
)
66 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
68 return snd_pcm_hw_constraint_list(runtime
, 0,
69 SNDRV_PCM_HW_PARAM_RATE
,
73 static int h1940_hw_params(struct snd_pcm_substream
*substream
,
74 struct snd_pcm_hw_params
*params
)
76 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
77 struct snd_soc_dai
*cpu_dai
= rtd
->cpu_dai
;
80 unsigned int rate
= params_rate(params
);
86 div
= s3c24xx_i2s_get_clockrate() / (384 * rate
);
87 if (s3c24xx_i2s_get_clockrate() % (384 * rate
) > (192 * rate
))
91 dev_err(rtd
->dev
, "%s: rate %d is not supported\n",
96 /* select clock source */
97 ret
= snd_soc_dai_set_sysclk(cpu_dai
, S3C24XX_CLKSRC_PCLK
, rate
,
102 /* set MCLK division for sample rate */
103 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_MCLK
,
104 S3C2410_IISMOD_384FS
);
108 /* set BCLK division for sample rate */
109 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_BCLK
,
110 S3C2410_IISMOD_32FS
);
114 /* set prescaler division for sample rate */
115 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_PRESCALER
,
116 S3C24XX_PRESCALE(div
, div
));
123 static struct snd_soc_ops h1940_ops
= {
124 .startup
= h1940_startup
,
125 .hw_params
= h1940_hw_params
,
128 static int h1940_spk_power(struct snd_soc_dapm_widget
*w
,
129 struct snd_kcontrol
*kcontrol
, int event
)
131 if (SND_SOC_DAPM_EVENT_ON(event
))
132 gpio_set_value(S3C_GPIO_END
+ 9, 1);
134 gpio_set_value(S3C_GPIO_END
+ 9, 0);
139 /* h1940 machine dapm widgets */
140 static const struct snd_soc_dapm_widget uda1380_dapm_widgets
[] = {
141 SND_SOC_DAPM_HP("Headphone Jack", NULL
),
142 SND_SOC_DAPM_MIC("Mic Jack", NULL
),
143 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power
),
146 /* h1940 machine audio_map */
147 static const struct snd_soc_dapm_route audio_map
[] = {
148 /* headphone connected to VOUTLHP, VOUTRHP */
149 {"Headphone Jack", NULL
, "VOUTLHP"},
150 {"Headphone Jack", NULL
, "VOUTRHP"},
152 /* ext speaker connected to VOUTL, VOUTR */
153 {"Speaker", NULL
, "VOUTL"},
154 {"Speaker", NULL
, "VOUTR"},
156 /* mic is connected to VINM */
157 {"VINM", NULL
, "Mic Jack"},
160 static struct platform_device
*s3c24xx_snd_device
;
162 static int h1940_uda1380_init(struct snd_soc_pcm_runtime
*rtd
)
164 snd_soc_card_jack_new(rtd
->card
, "Headphone Jack", SND_JACK_HEADPHONE
,
165 &hp_jack
, hp_jack_pins
, ARRAY_SIZE(hp_jack_pins
));
167 snd_soc_jack_add_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
173 static int h1940_uda1380_card_remove(struct snd_soc_card
*card
)
175 snd_soc_jack_free_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
181 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
182 static struct snd_soc_dai_link h1940_uda1380_dai
[] = {
185 .stream_name
= "UDA1380 Duplex",
186 .cpu_dai_name
= "s3c24xx-iis",
187 .codec_dai_name
= "uda1380-hifi",
188 .init
= h1940_uda1380_init
,
189 .platform_name
= "s3c24xx-iis",
190 .codec_name
= "uda1380-codec.0-001a",
191 .dai_fmt
= SND_SOC_DAIFMT_I2S
| SND_SOC_DAIFMT_NB_NF
|
192 SND_SOC_DAIFMT_CBS_CFS
,
197 static struct snd_soc_card h1940_asoc
= {
199 .owner
= THIS_MODULE
,
200 .remove
= h1940_uda1380_card_remove
,
201 .dai_link
= h1940_uda1380_dai
,
202 .num_links
= ARRAY_SIZE(h1940_uda1380_dai
),
204 .dapm_widgets
= uda1380_dapm_widgets
,
205 .num_dapm_widgets
= ARRAY_SIZE(uda1380_dapm_widgets
),
206 .dapm_routes
= audio_map
,
207 .num_dapm_routes
= ARRAY_SIZE(audio_map
),
210 static int __init
h1940_init(void)
214 if (!machine_is_h1940())
217 /* configure some gpios */
218 ret
= gpio_request(S3C_GPIO_END
+ 9, "speaker-power");
222 ret
= gpio_direction_output(S3C_GPIO_END
+ 9, 0);
226 s3c24xx_snd_device
= platform_device_alloc("soc-audio", -1);
227 if (!s3c24xx_snd_device
) {
232 platform_set_drvdata(s3c24xx_snd_device
, &h1940_asoc
);
233 ret
= platform_device_add(s3c24xx_snd_device
);
241 platform_device_put(s3c24xx_snd_device
);
243 gpio_free(S3C_GPIO_END
+ 9);
249 static void __exit
h1940_exit(void)
251 platform_device_unregister(s3c24xx_snd_device
);
252 gpio_free(S3C_GPIO_END
+ 9);
255 module_init(h1940_init
);
256 module_exit(h1940_exit
);
258 /* Module information */
259 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
260 MODULE_DESCRIPTION("ALSA SoC H1940");
261 MODULE_LICENSE("GPL");