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>
23 #include <plat/regs-iis.h>
24 #include <mach/h1940-latch.h>
25 #include <asm/mach-types.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 runtime
->hw
.rate_min
= hw_rates
.list
[0];
70 runtime
->hw
.rate_max
= hw_rates
.list
[hw_rates
.count
- 1];
71 runtime
->hw
.rates
= SNDRV_PCM_RATE_KNOT
;
73 return snd_pcm_hw_constraint_list(runtime
, 0,
74 SNDRV_PCM_HW_PARAM_RATE
,
78 static int h1940_hw_params(struct snd_pcm_substream
*substream
,
79 struct snd_pcm_hw_params
*params
)
81 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
82 struct snd_soc_dai
*cpu_dai
= rtd
->cpu_dai
;
83 struct snd_soc_dai
*codec_dai
= rtd
->codec_dai
;
86 unsigned int rate
= params_rate(params
);
92 div
= s3c24xx_i2s_get_clockrate() / (384 * rate
);
93 if (s3c24xx_i2s_get_clockrate() % (384 * rate
) > (192 * rate
))
97 dev_err(&rtd
->dev
, "%s: rate %d is not supported\n",
102 /* set codec DAI configuration */
103 ret
= snd_soc_dai_set_fmt(codec_dai
, SND_SOC_DAIFMT_I2S
|
104 SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS
);
108 /* set cpu DAI configuration */
109 ret
= snd_soc_dai_set_fmt(cpu_dai
, SND_SOC_DAIFMT_I2S
|
110 SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS
);
114 /* select clock source */
115 ret
= snd_soc_dai_set_sysclk(cpu_dai
, S3C24XX_CLKSRC_PCLK
, rate
,
120 /* set MCLK division for sample rate */
121 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_MCLK
,
122 S3C2410_IISMOD_384FS
);
126 /* set BCLK division for sample rate */
127 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_BCLK
,
128 S3C2410_IISMOD_32FS
);
132 /* set prescaler division for sample rate */
133 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_PRESCALER
,
134 S3C24XX_PRESCALE(div
, div
));
141 static struct snd_soc_ops h1940_ops
= {
142 .startup
= h1940_startup
,
143 .hw_params
= h1940_hw_params
,
146 static int h1940_spk_power(struct snd_soc_dapm_widget
*w
,
147 struct snd_kcontrol
*kcontrol
, int event
)
149 if (SND_SOC_DAPM_EVENT_ON(event
))
150 gpio_set_value(H1940_LATCH_AUDIO_POWER
, 1);
152 gpio_set_value(H1940_LATCH_AUDIO_POWER
, 0);
157 /* h1940 machine dapm widgets */
158 static const struct snd_soc_dapm_widget uda1380_dapm_widgets
[] = {
159 SND_SOC_DAPM_HP("Headphone Jack", NULL
),
160 SND_SOC_DAPM_MIC("Mic Jack", NULL
),
161 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power
),
164 /* h1940 machine audio_map */
165 static const struct snd_soc_dapm_route audio_map
[] = {
166 /* headphone connected to VOUTLHP, VOUTRHP */
167 {"Headphone Jack", NULL
, "VOUTLHP"},
168 {"Headphone Jack", NULL
, "VOUTRHP"},
170 /* ext speaker connected to VOUTL, VOUTR */
171 {"Speaker", NULL
, "VOUTL"},
172 {"Speaker", NULL
, "VOUTR"},
174 /* mic is connected to VINM */
175 {"VINM", NULL
, "Mic Jack"},
178 static struct platform_device
*s3c24xx_snd_device
;
180 static int h1940_uda1380_init(struct snd_soc_pcm_runtime
*rtd
)
182 struct snd_soc_codec
*codec
= rtd
->codec
;
183 struct snd_soc_dapm_context
*dapm
= &codec
->dapm
;
186 snd_soc_dapm_enable_pin(dapm
, "Headphone Jack");
187 snd_soc_dapm_enable_pin(dapm
, "Speaker");
188 snd_soc_dapm_enable_pin(dapm
, "Mic Jack");
190 snd_soc_jack_new(codec
, "Headphone Jack", SND_JACK_HEADPHONE
,
193 snd_soc_jack_add_pins(&hp_jack
, ARRAY_SIZE(hp_jack_pins
),
196 snd_soc_jack_add_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
202 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
203 static struct snd_soc_dai_link h1940_uda1380_dai
[] = {
206 .stream_name
= "UDA1380 Duplex",
207 .cpu_dai_name
= "s3c24xx-iis",
208 .codec_dai_name
= "uda1380-hifi",
209 .init
= h1940_uda1380_init
,
210 .platform_name
= "samsung-audio",
211 .codec_name
= "uda1380-codec.0-001a",
216 static struct snd_soc_card h1940_asoc
= {
218 .owner
= THIS_MODULE
,
219 .dai_link
= h1940_uda1380_dai
,
220 .num_links
= ARRAY_SIZE(h1940_uda1380_dai
),
222 .dapm_widgets
= uda1380_dapm_widgets
,
223 .num_dapm_widgets
= ARRAY_SIZE(uda1380_dapm_widgets
),
224 .dapm_routes
= audio_map
,
225 .num_dapm_routes
= ARRAY_SIZE(audio_map
),
228 static int __init
h1940_init(void)
232 if (!machine_is_h1940())
235 /* configure some gpios */
236 ret
= gpio_request(H1940_LATCH_AUDIO_POWER
, "speaker-power");
240 ret
= gpio_direction_output(H1940_LATCH_AUDIO_POWER
, 0);
244 s3c24xx_snd_device
= platform_device_alloc("soc-audio", -1);
245 if (!s3c24xx_snd_device
) {
250 platform_set_drvdata(s3c24xx_snd_device
, &h1940_asoc
);
251 ret
= platform_device_add(s3c24xx_snd_device
);
259 platform_device_put(s3c24xx_snd_device
);
261 gpio_free(H1940_LATCH_AUDIO_POWER
);
267 static void __exit
h1940_exit(void)
269 platform_device_unregister(s3c24xx_snd_device
);
270 snd_soc_jack_free_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
272 gpio_free(H1940_LATCH_AUDIO_POWER
);
275 module_init(h1940_init
);
276 module_exit(h1940_exit
);
278 /* Module information */
279 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
280 MODULE_DESCRIPTION("ALSA SoC H1940");
281 MODULE_LICENSE("GPL");