Merge branch 'fix/hda' into for-linus
[deliverable/linux.git] / sound / soc / omap / osk5912.c
1 /*
2 * osk5912.c -- SoC audio for OSK 5912
3 *
4 * Copyright (C) 2008 Mistral Solutions
5 *
6 * Contact: Arun KS <arunks@mistralsolutions.com>
7 *
8 * This program is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU General Public License
10 * version 2 as published by the Free Software Foundation.
11 *
12 * This program is distributed in the hope that it will be useful, but
13 * WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * General Public License for more details.
16 *
17 * You should have received a copy of the GNU General Public License
18 * along with this program; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
20 * 02110-1301 USA
21 *
22 */
23
24 #include <linux/clk.h>
25 #include <linux/platform_device.h>
26 #include <sound/core.h>
27 #include <sound/pcm.h>
28 #include <sound/soc.h>
29 #include <sound/soc-dapm.h>
30
31 #include <asm/mach-types.h>
32 #include <mach/hardware.h>
33 #include <linux/gpio.h>
34 #include <plat/mcbsp.h>
35
36 #include "omap-mcbsp.h"
37 #include "omap-pcm.h"
38 #include "../codecs/tlv320aic23.h"
39
40 #define CODEC_CLOCK 12000000
41
42 static struct clk *tlv320aic23_mclk;
43
44 static int osk_startup(struct snd_pcm_substream *substream)
45 {
46 return clk_enable(tlv320aic23_mclk);
47 }
48
49 static void osk_shutdown(struct snd_pcm_substream *substream)
50 {
51 clk_disable(tlv320aic23_mclk);
52 }
53
54 static int osk_hw_params(struct snd_pcm_substream *substream,
55 struct snd_pcm_hw_params *params)
56 {
57 struct snd_soc_pcm_runtime *rtd = substream->private_data;
58 struct snd_soc_dai *codec_dai = rtd->codec_dai;
59 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
60 int err;
61
62 /* Set codec DAI configuration */
63 err = snd_soc_dai_set_fmt(codec_dai,
64 SND_SOC_DAIFMT_DSP_B |
65 SND_SOC_DAIFMT_NB_NF |
66 SND_SOC_DAIFMT_CBM_CFM);
67 if (err < 0) {
68 printk(KERN_ERR "can't set codec DAI configuration\n");
69 return err;
70 }
71
72 /* Set cpu DAI configuration */
73 err = snd_soc_dai_set_fmt(cpu_dai,
74 SND_SOC_DAIFMT_DSP_B |
75 SND_SOC_DAIFMT_NB_NF |
76 SND_SOC_DAIFMT_CBM_CFM);
77 if (err < 0) {
78 printk(KERN_ERR "can't set cpu DAI configuration\n");
79 return err;
80 }
81
82 /* Set the codec system clock for DAC and ADC */
83 err =
84 snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
85
86 if (err < 0) {
87 printk(KERN_ERR "can't set codec system clock\n");
88 return err;
89 }
90
91 return err;
92 }
93
94 static struct snd_soc_ops osk_ops = {
95 .startup = osk_startup,
96 .hw_params = osk_hw_params,
97 .shutdown = osk_shutdown,
98 };
99
100 static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
101 SND_SOC_DAPM_HP("Headphone Jack", NULL),
102 SND_SOC_DAPM_LINE("Line In", NULL),
103 SND_SOC_DAPM_MIC("Mic Jack", NULL),
104 };
105
106 static const struct snd_soc_dapm_route audio_map[] = {
107 {"Headphone Jack", NULL, "LHPOUT"},
108 {"Headphone Jack", NULL, "RHPOUT"},
109
110 {"LLINEIN", NULL, "Line In"},
111 {"RLINEIN", NULL, "Line In"},
112
113 {"MICIN", NULL, "Mic Jack"},
114 };
115
116 static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
117 {
118 struct snd_soc_codec *codec = rtd->codec;
119
120 /* Add osk5912 specific widgets */
121 snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
122 ARRAY_SIZE(tlv320aic23_dapm_widgets));
123
124 /* Set up osk5912 specific audio path audio_map */
125 snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
126
127 snd_soc_dapm_enable_pin(codec, "Headphone Jack");
128 snd_soc_dapm_enable_pin(codec, "Line In");
129 snd_soc_dapm_enable_pin(codec, "Mic Jack");
130
131 snd_soc_dapm_sync(codec);
132
133 return 0;
134 }
135
136 /* Digital audio interface glue - connects codec <--> CPU */
137 static struct snd_soc_dai_link osk_dai = {
138 .name = "TLV320AIC23",
139 .stream_name = "AIC23",
140 .cpu_dai_name = "omap-mcbsp-dai.0",
141 .codec_dai_name = "tlv320aic23-hifi",
142 .platform_name = "omap-pcm-audio",
143 .codec_name = "tlv320aic23-codec",
144 .init = osk_tlv320aic23_init,
145 .ops = &osk_ops,
146 };
147
148 /* Audio machine driver */
149 static struct snd_soc_card snd_soc_card_osk = {
150 .name = "OSK5912",
151 .dai_link = &osk_dai,
152 .num_links = 1,
153 };
154
155 static struct platform_device *osk_snd_device;
156
157 static int __init osk_soc_init(void)
158 {
159 int err;
160 u32 curRate;
161 struct device *dev;
162
163 if (!(machine_is_omap_osk()))
164 return -ENODEV;
165
166 osk_snd_device = platform_device_alloc("soc-audio", -1);
167 if (!osk_snd_device)
168 return -ENOMEM;
169
170 platform_set_drvdata(osk_snd_device, &snd_soc_card_osk);
171 err = platform_device_add(osk_snd_device);
172 if (err)
173 goto err1;
174
175 dev = &osk_snd_device->dev;
176
177 tlv320aic23_mclk = clk_get(dev, "mclk");
178 if (IS_ERR(tlv320aic23_mclk)) {
179 printk(KERN_ERR "Could not get mclk clock\n");
180 err = PTR_ERR(tlv320aic23_mclk);
181 goto err2;
182 }
183
184 /*
185 * Configure 12 MHz output on MCLK.
186 */
187 curRate = (uint) clk_get_rate(tlv320aic23_mclk);
188 if (curRate != CODEC_CLOCK) {
189 if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
190 printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
191 err = -ECANCELED;
192 goto err3;
193 }
194 }
195
196 printk(KERN_INFO "MCLK = %d [%d]\n",
197 (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
198
199 return 0;
200
201 err3:
202 clk_put(tlv320aic23_mclk);
203 err2:
204 platform_device_del(osk_snd_device);
205 err1:
206 platform_device_put(osk_snd_device);
207
208 return err;
209
210 }
211
212 static void __exit osk_soc_exit(void)
213 {
214 clk_put(tlv320aic23_mclk);
215 platform_device_unregister(osk_snd_device);
216 }
217
218 module_init(osk_soc_init);
219 module_exit(osk_soc_exit);
220
221 MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
222 MODULE_DESCRIPTION("ALSA SoC OSK 5912");
223 MODULE_LICENSE("GPL");
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