225 lines
		
	
	
		
			5.4 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			225 lines
		
	
	
		
			5.4 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * osk5912.c  --  SoC audio for OSK 5912
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|  *
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|  * Copyright (C) 2008 Mistral Solutions
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|  *
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|  * Contact: Arun KS  <arunks@mistralsolutions.com>
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|  *
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|  * This program is free software; you can redistribute it and/or
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|  * modify it under the terms of the GNU General Public License
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|  * version 2 as published by the Free Software Foundation.
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|  *
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|  * This program is distributed in the hope that it will be useful, but
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|  * WITHOUT ANY WARRANTY; without even the implied warranty of
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|  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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|  * General Public License for more details.
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|  *
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|  * You should have received a copy of the GNU General Public License
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|  * along with this program; if not, write to the Free Software
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|  * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
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|  * 02110-1301 USA
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|  *
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|  */
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| 
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| #include <linux/clk.h>
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| #include <linux/platform_device.h>
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| #include <sound/core.h>
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| #include <sound/pcm.h>
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| #include <sound/soc.h>
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| #include <sound/soc-dapm.h>
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| 
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| #include <asm/mach-types.h>
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| #include <mach/hardware.h>
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| #include <linux/gpio.h>
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| #include <mach/mcbsp.h>
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| 
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| #include "omap-mcbsp.h"
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| #include "omap-pcm.h"
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| #include "../codecs/tlv320aic23.h"
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| 
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| #define CODEC_CLOCK 	12000000
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| 
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| static struct clk *tlv320aic23_mclk;
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| 
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| static int osk_startup(struct snd_pcm_substream *substream)
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| {
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| 	return clk_enable(tlv320aic23_mclk);
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| }
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| 
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| static void osk_shutdown(struct snd_pcm_substream *substream)
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| {
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| 	clk_disable(tlv320aic23_mclk);
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| }
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| 
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| static int osk_hw_params(struct snd_pcm_substream *substream,
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| 			 struct snd_pcm_hw_params *params)
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| {
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| 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
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| 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
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| 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
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| 	int err;
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| 
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| 	/* Set codec DAI configuration */
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| 	err = snd_soc_dai_set_fmt(codec_dai,
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| 				  SND_SOC_DAIFMT_DSP_B |
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| 				  SND_SOC_DAIFMT_NB_NF |
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| 				  SND_SOC_DAIFMT_CBM_CFM);
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| 	if (err < 0) {
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| 		printk(KERN_ERR "can't set codec DAI configuration\n");
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| 		return err;
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| 	}
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| 
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| 	/* Set cpu DAI configuration */
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| 	err = snd_soc_dai_set_fmt(cpu_dai,
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| 				  SND_SOC_DAIFMT_DSP_B |
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| 				  SND_SOC_DAIFMT_NB_NF |
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| 				  SND_SOC_DAIFMT_CBM_CFM);
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| 	if (err < 0) {
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| 		printk(KERN_ERR "can't set cpu DAI configuration\n");
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| 		return err;
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| 	}
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| 
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| 	/* Set the codec system clock for DAC and ADC */
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| 	err =
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| 	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
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| 
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| 	if (err < 0) {
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| 		printk(KERN_ERR "can't set codec system clock\n");
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| 		return err;
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| 	}
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| 
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| 	return err;
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| }
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| 
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| static struct snd_soc_ops osk_ops = {
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| 	.startup = osk_startup,
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| 	.hw_params = osk_hw_params,
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| 	.shutdown = osk_shutdown,
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| };
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| 
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| static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
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| 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
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| 	SND_SOC_DAPM_LINE("Line In", NULL),
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| 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
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| };
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| 
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| static const struct snd_soc_dapm_route audio_map[] = {
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| 	{"Headphone Jack", NULL, "LHPOUT"},
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| 	{"Headphone Jack", NULL, "RHPOUT"},
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| 
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| 	{"LLINEIN", NULL, "Line In"},
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| 	{"RLINEIN", NULL, "Line In"},
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| 
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| 	{"MICIN", NULL, "Mic Jack"},
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| };
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| 
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| static int osk_tlv320aic23_init(struct snd_soc_codec *codec)
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| {
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| 
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| 	/* Add osk5912 specific widgets */
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| 	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
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| 				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
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| 
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| 	/* Set up osk5912 specific audio path audio_map */
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| 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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| 
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| 	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
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| 	snd_soc_dapm_enable_pin(codec, "Line In");
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| 	snd_soc_dapm_enable_pin(codec, "Mic Jack");
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| 
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| 	snd_soc_dapm_sync(codec);
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| 
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| 	return 0;
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| }
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| 
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| /* Digital audio interface glue - connects codec <--> CPU */
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| static struct snd_soc_dai_link osk_dai = {
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| 	.name = "TLV320AIC23",
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| 	.stream_name = "AIC23",
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| 	.cpu_dai = &omap_mcbsp_dai[0],
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| 	.codec_dai = &tlv320aic23_dai,
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| 	.init = osk_tlv320aic23_init,
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| 	.ops = &osk_ops,
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| };
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| 
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| /* Audio machine driver */
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| static struct snd_soc_card snd_soc_card_osk = {
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| 	.name = "OSK5912",
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| 	.platform = &omap_soc_platform,
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| 	.dai_link = &osk_dai,
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| 	.num_links = 1,
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| };
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| 
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| /* Audio subsystem */
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| static struct snd_soc_device osk_snd_devdata = {
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| 	.card = &snd_soc_card_osk,
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| 	.codec_dev = &soc_codec_dev_tlv320aic23,
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| };
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| 
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| static struct platform_device *osk_snd_device;
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| 
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| static int __init osk_soc_init(void)
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| {
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| 	int err;
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| 	u32 curRate;
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| 	struct device *dev;
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| 
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| 	if (!(machine_is_omap_osk()))
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| 		return -ENODEV;
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| 
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| 	osk_snd_device = platform_device_alloc("soc-audio", -1);
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| 	if (!osk_snd_device)
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| 		return -ENOMEM;
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| 
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| 	platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
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| 	osk_snd_devdata.dev = &osk_snd_device->dev;
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| 	*(unsigned int *)osk_dai.cpu_dai->private_data = 0;	/* McBSP1 */
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| 	err = platform_device_add(osk_snd_device);
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| 	if (err)
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| 		goto err1;
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| 
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| 	dev = &osk_snd_device->dev;
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| 
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| 	tlv320aic23_mclk = clk_get(dev, "mclk");
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| 	if (IS_ERR(tlv320aic23_mclk)) {
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| 		printk(KERN_ERR "Could not get mclk clock\n");
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| 		return -ENODEV;
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| 	}
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| 
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| 	/*
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| 	 * Configure 12 MHz output on MCLK.
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| 	 */
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| 	curRate = (uint) clk_get_rate(tlv320aic23_mclk);
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| 	if (curRate != CODEC_CLOCK) {
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| 		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
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| 			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
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| 			err = -ECANCELED;
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| 			goto err1;
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| 		}
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| 	}
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| 
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| 	printk(KERN_INFO "MCLK = %d [%d]\n",
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| 	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
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| 
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| 	return 0;
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| err1:
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| 	clk_put(tlv320aic23_mclk);
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| 	platform_device_del(osk_snd_device);
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| 	platform_device_put(osk_snd_device);
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| 
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| 	return err;
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| 
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| }
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| 
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| static void __exit osk_soc_exit(void)
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| {
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| 	platform_device_unregister(osk_snd_device);
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| }
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| 
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| module_init(osk_soc_init);
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| module_exit(osk_soc_exit);
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| 
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| MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
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| MODULE_DESCRIPTION("ALSA SoC OSK 5912");
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| MODULE_LICENSE("GPL");
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