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/gpio.h>
18 #include <sound/soc.h>
19 #include <sound/jack.h>
21 #include <plat/regs-iis.h>
22 #include <mach/h1940-latch.h>
23 #include <asm/mach-types.h>
25 #include "s3c24xx-i2s.h"
27 static unsigned int rates
[] = {
33 static struct snd_pcm_hw_constraint_list hw_rates
= {
34 .count
= ARRAY_SIZE(rates
),
39 static struct snd_soc_jack hp_jack
;
41 static struct snd_soc_jack_pin hp_jack_pins
[] = {
43 .pin
= "Headphone Jack",
44 .mask
= SND_JACK_HEADPHONE
,
48 .mask
= SND_JACK_HEADPHONE
,
53 static struct snd_soc_jack_gpio hp_jack_gpios
[] = {
55 .gpio
= S3C2410_GPG(4),
57 .report
= SND_JACK_HEADPHONE
,
63 static int h1940_startup(struct snd_pcm_substream
*substream
)
65 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
67 runtime
->hw
.rate_min
= hw_rates
.list
[0];
68 runtime
->hw
.rate_max
= hw_rates
.list
[hw_rates
.count
- 1];
69 runtime
->hw
.rates
= SNDRV_PCM_RATE_KNOT
;
71 return snd_pcm_hw_constraint_list(runtime
, 0,
72 SNDRV_PCM_HW_PARAM_RATE
,
76 static int h1940_hw_params(struct snd_pcm_substream
*substream
,
77 struct snd_pcm_hw_params
*params
)
79 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
80 struct snd_soc_dai
*cpu_dai
= rtd
->cpu_dai
;
81 struct snd_soc_dai
*codec_dai
= rtd
->codec_dai
;
84 unsigned int rate
= params_rate(params
);
90 div
= s3c24xx_i2s_get_clockrate() / (384 * rate
);
91 if (s3c24xx_i2s_get_clockrate() % (384 * rate
) > (192 * rate
))
95 dev_err(&rtd
->dev
, "%s: rate %d is not supported\n",
100 /* set codec DAI configuration */
101 ret
= snd_soc_dai_set_fmt(codec_dai
, SND_SOC_DAIFMT_I2S
|
102 SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS
);
106 /* set cpu DAI configuration */
107 ret
= snd_soc_dai_set_fmt(cpu_dai
, SND_SOC_DAIFMT_I2S
|
108 SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS
);
112 /* select clock source */
113 ret
= snd_soc_dai_set_sysclk(cpu_dai
, S3C24XX_CLKSRC_PCLK
, rate
,
118 /* set MCLK division for sample rate */
119 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_MCLK
,
120 S3C2410_IISMOD_384FS
);
124 /* set BCLK division for sample rate */
125 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_BCLK
,
126 S3C2410_IISMOD_32FS
);
130 /* set prescaler division for sample rate */
131 ret
= snd_soc_dai_set_clkdiv(cpu_dai
, S3C24XX_DIV_PRESCALER
,
132 S3C24XX_PRESCALE(div
, div
));
139 static struct snd_soc_ops h1940_ops
= {
140 .startup
= h1940_startup
,
141 .hw_params
= h1940_hw_params
,
144 static int h1940_spk_power(struct snd_soc_dapm_widget
*w
,
145 struct snd_kcontrol
*kcontrol
, int event
)
147 if (SND_SOC_DAPM_EVENT_ON(event
))
148 gpio_set_value(H1940_LATCH_AUDIO_POWER
, 1);
150 gpio_set_value(H1940_LATCH_AUDIO_POWER
, 0);
155 /* h1940 machine dapm widgets */
156 static const struct snd_soc_dapm_widget uda1380_dapm_widgets
[] = {
157 SND_SOC_DAPM_HP("Headphone Jack", NULL
),
158 SND_SOC_DAPM_MIC("Mic Jack", NULL
),
159 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power
),
162 /* h1940 machine audio_map */
163 static const struct snd_soc_dapm_route audio_map
[] = {
164 /* headphone connected to VOUTLHP, VOUTRHP */
165 {"Headphone Jack", NULL
, "VOUTLHP"},
166 {"Headphone Jack", NULL
, "VOUTRHP"},
168 /* ext speaker connected to VOUTL, VOUTR */
169 {"Speaker", NULL
, "VOUTL"},
170 {"Speaker", NULL
, "VOUTR"},
172 /* mic is connected to VINM */
173 {"VINM", NULL
, "Mic Jack"},
176 static struct platform_device
*s3c24xx_snd_device
;
178 static int h1940_uda1380_init(struct snd_soc_pcm_runtime
*rtd
)
180 struct snd_soc_codec
*codec
= rtd
->codec
;
181 struct snd_soc_dapm_context
*dapm
= &codec
->dapm
;
184 /* Add h1940 specific widgets */
185 err
= snd_soc_dapm_new_controls(dapm
, uda1380_dapm_widgets
,
186 ARRAY_SIZE(uda1380_dapm_widgets
));
190 /* Set up h1940 specific audio path audio_mapnects */
191 err
= snd_soc_dapm_add_routes(dapm
, audio_map
,
192 ARRAY_SIZE(audio_map
));
196 snd_soc_dapm_enable_pin(dapm
, "Headphone Jack");
197 snd_soc_dapm_enable_pin(dapm
, "Speaker");
198 snd_soc_dapm_enable_pin(dapm
, "Mic Jack");
200 snd_soc_dapm_sync(dapm
);
202 snd_soc_jack_new(codec
, "Headphone Jack", SND_JACK_HEADPHONE
,
205 snd_soc_jack_add_pins(&hp_jack
, ARRAY_SIZE(hp_jack_pins
),
208 snd_soc_jack_add_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
214 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
215 static struct snd_soc_dai_link h1940_uda1380_dai
[] = {
218 .stream_name
= "UDA1380 Duplex",
219 .cpu_dai_name
= "s3c24xx-iis",
220 .codec_dai_name
= "uda1380-hifi",
221 .init
= h1940_uda1380_init
,
222 .platform_name
= "samsung-audio",
223 .codec_name
= "uda1380-codec.0-001a",
228 static struct snd_soc_card h1940_asoc
= {
230 .dai_link
= h1940_uda1380_dai
,
231 .num_links
= ARRAY_SIZE(h1940_uda1380_dai
),
234 static int __init
h1940_init(void)
238 if (!machine_is_h1940())
241 /* configure some gpios */
242 ret
= gpio_request(H1940_LATCH_AUDIO_POWER
, "speaker-power");
246 ret
= gpio_direction_output(H1940_LATCH_AUDIO_POWER
, 0);
250 s3c24xx_snd_device
= platform_device_alloc("soc-audio", -1);
251 if (!s3c24xx_snd_device
) {
256 platform_set_drvdata(s3c24xx_snd_device
, &h1940_asoc
);
257 ret
= platform_device_add(s3c24xx_snd_device
);
265 platform_device_put(s3c24xx_snd_device
);
267 gpio_free(H1940_LATCH_AUDIO_POWER
);
273 static void __exit
h1940_exit(void)
275 platform_device_unregister(s3c24xx_snd_device
);
276 snd_soc_jack_free_gpios(&hp_jack
, ARRAY_SIZE(hp_jack_gpios
),
278 gpio_free(H1940_LATCH_AUDIO_POWER
);
281 module_init(h1940_init
);
282 module_exit(h1940_exit
);
284 /* Module information */
285 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
286 MODULE_DESCRIPTION("ALSA SoC H1940");
287 MODULE_LICENSE("GPL");