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 unsigned int rates
[] = {
35 static struct snd_pcm_hw_constraint_list hw_rates
= {
36 .count
= ARRAY_SIZE(rates
),
41 static struct snd_soc_jack hp_jack
;
43 static struct snd_soc_jack_pin hp_jack_pins
[] = {
45 .pin
= "Headphone Jack",
46 .mask
= SND_JACK_HEADPHONE
,
50 .mask
= SND_JACK_HEADPHONE
,
55 static struct snd_soc_jack_gpio hp_jack_gpios
[] = {
57 .gpio
= S3C2410_GPG(4),
59 .report
= SND_JACK_HEADPHONE
,
65 static int h1940_startup(struct snd_pcm_substream
*substream
)
67 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
69 return snd_pcm_hw_constraint_list(runtime
, 0,
70 SNDRV_PCM_HW_PARAM_RATE
,
74 static int h1940_hw_params(struct snd_pcm_substream
*substream
,
75 struct snd_pcm_hw_params
*params
)
77 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
78 struct snd_soc_dai
*cpu_dai
= rtd
->cpu_dai
;
81 unsigned int rate
= params_rate(params
);
87 div
= s3c24xx_i2s_get_clockrate() / (384 * rate
);
88 if (s3c24xx_i2s_get_clockrate() % (384 * rate
) > (192 * rate
))
92 dev_err(rtd
->dev
, "%s: rate %d is not supported\n",
97 /* select clock source */
98 ret
= snd_soc_dai_set_sysclk(cpu_dai
, S3C24XX_CLKSRC_PCLK
, rate
,
103 /* set MCLK division for sample rate */
104 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_MCLK
,
105 S3C2410_IISMOD_384FS
);
109 /* set BCLK division for sample rate */
110 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_BCLK
,
111 S3C2410_IISMOD_32FS
);
115 /* set prescaler division for sample rate */
116 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_PRESCALER
,
117 S3C24XX_PRESCALE(div
, div
));
124 static struct snd_soc_ops h1940_ops
= {
125 .startup
= h1940_startup
,
126 .hw_params
= h1940_hw_params
,
129 static int h1940_spk_power(struct snd_soc_dapm_widget
*w
,
130 struct snd_kcontrol
*kcontrol
, int event
)
132 if (SND_SOC_DAPM_EVENT_ON(event
))
133 gpio_set_value(S3C_GPIO_END
+ 9, 1);
135 gpio_set_value(S3C_GPIO_END
+ 9, 0);
140 /* h1940 machine dapm widgets */
141 static const struct snd_soc_dapm_widget uda1380_dapm_widgets
[] = {
142 SND_SOC_DAPM_HP("Headphone Jack", NULL
),
143 SND_SOC_DAPM_MIC("Mic Jack", NULL
),
144 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power
),
147 /* h1940 machine audio_map */
148 static const struct snd_soc_dapm_route audio_map
[] = {
149 /* headphone connected to VOUTLHP, VOUTRHP */
150 {"Headphone Jack", NULL
, "VOUTLHP"},
151 {"Headphone Jack", NULL
, "VOUTRHP"},
153 /* ext speaker connected to VOUTL, VOUTR */
154 {"Speaker", NULL
, "VOUTL"},
155 {"Speaker", NULL
, "VOUTR"},
157 /* mic is connected to VINM */
158 {"VINM", NULL
, "Mic Jack"},
161 static struct platform_device
*s3c24xx_snd_device
;
163 static int h1940_uda1380_init(struct snd_soc_pcm_runtime
*rtd
)
165 snd_soc_card_jack_new(rtd
->card
, "Headphone Jack", SND_JACK_HEADPHONE
,
166 &hp_jack
, hp_jack_pins
, ARRAY_SIZE(hp_jack_pins
));
168 snd_soc_jack_add_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
174 static int h1940_uda1380_card_remove(struct snd_soc_card
*card
)
176 snd_soc_jack_free_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
182 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
183 static struct snd_soc_dai_link h1940_uda1380_dai
[] = {
186 .stream_name
= "UDA1380 Duplex",
187 .cpu_dai_name
= "s3c24xx-iis",
188 .codec_dai_name
= "uda1380-hifi",
189 .init
= h1940_uda1380_init
,
190 .platform_name
= "s3c24xx-iis",
191 .codec_name
= "uda1380-codec.0-001a",
192 .dai_fmt
= SND_SOC_DAIFMT_I2S
| SND_SOC_DAIFMT_NB_NF
|
193 SND_SOC_DAIFMT_CBS_CFS
,
198 static struct snd_soc_card h1940_asoc
= {
200 .owner
= THIS_MODULE
,
201 .remove
= h1940_uda1380_card_remove
,
202 .dai_link
= h1940_uda1380_dai
,
203 .num_links
= ARRAY_SIZE(h1940_uda1380_dai
),
205 .dapm_widgets
= uda1380_dapm_widgets
,
206 .num_dapm_widgets
= ARRAY_SIZE(uda1380_dapm_widgets
),
207 .dapm_routes
= audio_map
,
208 .num_dapm_routes
= ARRAY_SIZE(audio_map
),
211 static int __init
h1940_init(void)
215 if (!machine_is_h1940())
218 /* configure some gpios */
219 ret
= gpio_request(S3C_GPIO_END
+ 9, "speaker-power");
223 ret
= gpio_direction_output(S3C_GPIO_END
+ 9, 0);
227 s3c24xx_snd_device
= platform_device_alloc("soc-audio", -1);
228 if (!s3c24xx_snd_device
) {
233 platform_set_drvdata(s3c24xx_snd_device
, &h1940_asoc
);
234 ret
= platform_device_add(s3c24xx_snd_device
);
242 platform_device_put(s3c24xx_snd_device
);
244 gpio_free(S3C_GPIO_END
+ 9);
250 static void __exit
h1940_exit(void)
252 platform_device_unregister(s3c24xx_snd_device
);
253 gpio_free(S3C_GPIO_END
+ 9);
256 module_init(h1940_init
);
257 module_exit(h1940_exit
);
259 /* Module information */
260 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
261 MODULE_DESCRIPTION("ALSA SoC H1940");
262 MODULE_LICENSE("GPL");