+++ /dev/null
-/*
- * afeb9260.c -- SoC audio for AFEB9260
- *
- * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-
-#include <linux/atmel-ssc.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <linux/gpio.h>
-
-#include "../codecs/tlv320aic23.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-#define CODEC_CLOCK 12000000
-
-static int afeb9260_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int err;
-
- /* Set the codec system clock for DAC and ADC */
- err =
- snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
-
- if (err < 0) {
- printk(KERN_ERR "can't set codec system clock\n");
- return err;
- }
-
- return err;
-}
-
-static struct snd_soc_ops afeb9260_ops = {
- .hw_params = afeb9260_hw_params,
-};
-
-static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_LINE("Line In", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
-};
-
-static const struct snd_soc_dapm_route afeb9260_audio_map[] = {
- {"Headphone Jack", NULL, "LHPOUT"},
- {"Headphone Jack", NULL, "RHPOUT"},
-
- {"LLINEIN", NULL, "Line In"},
- {"RLINEIN", NULL, "Line In"},
-
- {"MICIN", NULL, "Mic Jack"},
-};
-
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link afeb9260_dai = {
- .name = "TLV320AIC23",
- .stream_name = "AIC23",
- .cpu_dai_name = "atmel-ssc-dai.0",
- .codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "atmel_pcm-audio",
- .codec_name = "tlv320aic23-codec.0-001a",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &afeb9260_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_machine_afeb9260 = {
- .name = "AFEB9260",
- .owner = THIS_MODULE,
- .dai_link = &afeb9260_dai,
- .num_links = 1,
-
- .dapm_widgets = tlv320aic23_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
- .dapm_routes = afeb9260_audio_map,
- .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
-};
-
-static struct platform_device *afeb9260_snd_device;
-
-static int __init afeb9260_soc_init(void)
-{
- int err;
- struct device *dev;
-
- if (!(machine_is_afeb9260()))
- return -ENODEV;
-
-
- afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
- if (!afeb9260_snd_device) {
- printk(KERN_ERR "ASoC: Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260);
- err = platform_device_add(afeb9260_snd_device);
- if (err)
- goto err1;
-
- dev = &afeb9260_snd_device->dev;
-
- return 0;
-err1:
- platform_device_put(afeb9260_snd_device);
- return err;
-}
-
-static void __exit afeb9260_soc_exit(void)
-{
- platform_device_unregister(afeb9260_snd_device);
-}
-
-module_init(afeb9260_soc_init);
-module_exit(afeb9260_soc_exit);
-
-MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
-MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
-MODULE_LICENSE("GPL");
-