152 lines
3.8 KiB
C
152 lines
3.8 KiB
C
/*
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* afeb9260.c -- SoC audio for AFEB9260
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*
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* Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
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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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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/kernel.h>
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#include <linux/clk.h>
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#include <linux/platform_device.h>
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#include <linux/atmel-ssc.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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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 "../codecs/tlv320aic23.h"
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#include "atmel-pcm.h"
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#include "atmel_ssc_dai.h"
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#define CODEC_CLOCK 12000000
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static int afeb9260_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->codec_dai;
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int err;
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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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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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return err;
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}
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static struct snd_soc_ops afeb9260_ops = {
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.hw_params = afeb9260_hw_params,
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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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static const struct snd_soc_dapm_route afeb9260_audio_map[] = {
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{"Headphone Jack", NULL, "LHPOUT"},
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{"Headphone Jack", NULL, "RHPOUT"},
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{"LLINEIN", NULL, "Line In"},
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{"RLINEIN", NULL, "Line In"},
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{"MICIN", NULL, "Mic Jack"},
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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 afeb9260_dai = {
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.name = "TLV320AIC23",
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.stream_name = "AIC23",
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.cpu_dai_name = "atmel-ssc-dai.0",
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.codec_dai_name = "tlv320aic23-hifi",
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.platform_name = "atmel_pcm-audio",
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.codec_name = "tlv320aic23-codec.0-001a",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_CBM_CFM,
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.ops = &afeb9260_ops,
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};
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/* Audio machine driver */
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static struct snd_soc_card snd_soc_machine_afeb9260 = {
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.name = "AFEB9260",
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.owner = THIS_MODULE,
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.dai_link = &afeb9260_dai,
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.num_links = 1,
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.dapm_widgets = tlv320aic23_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
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.dapm_routes = afeb9260_audio_map,
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.num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
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};
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static struct platform_device *afeb9260_snd_device;
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static int __init afeb9260_soc_init(void)
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{
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int err;
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struct device *dev;
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if (!(machine_is_afeb9260()))
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return -ENODEV;
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afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
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if (!afeb9260_snd_device) {
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printk(KERN_ERR "ASoC: Platform device allocation failed\n");
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return -ENOMEM;
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}
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platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260);
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err = platform_device_add(afeb9260_snd_device);
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if (err)
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goto err1;
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dev = &afeb9260_snd_device->dev;
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return 0;
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err1:
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platform_device_put(afeb9260_snd_device);
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return err;
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}
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static void __exit afeb9260_soc_exit(void)
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{
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platform_device_unregister(afeb9260_snd_device);
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}
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module_init(afeb9260_soc_init);
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module_exit(afeb9260_soc_exit);
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MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
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MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
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MODULE_LICENSE("GPL");
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