diff options
Diffstat (limited to 'sound/soc/atmel/snd-soc-afeb9260.c')
| -rw-r--r-- | sound/soc/atmel/snd-soc-afeb9260.c | 203 |
1 files changed, 203 insertions, 0 deletions
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c new file mode 100644 index 000000000000..23349de27313 --- /dev/null +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -0,0 +1,203 @@ +/* + * 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 <sound/soc-dapm.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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S| + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return 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 audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add afeb9260 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up afeb9260 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link afeb9260_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = afeb9260_tlv320aic23_init, + .ops = &afeb9260_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_machine_afeb9260 = { + .name = "AFEB9260", + .platform = &atmel_soc_platform, + .dai_link = &afeb9260_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device afeb9260_snd_devdata = { + .card = &snd_soc_machine_afeb9260, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *afeb9260_snd_device; + +static int __init afeb9260_soc_init(void) +{ + int err; + struct device *dev; + struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + if (!(machine_is_afeb9260())) + return -ENODEV; + + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); + err = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + 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, &afeb9260_snd_devdata); + afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev; + err = platform_device_add(afeb9260_snd_device); + if (err) + goto err1; + + dev = &afeb9260_snd_device->dev; + + return 0; +err1: + platform_device_del(afeb9260_snd_device); + platform_device_put(afeb9260_snd_device); +err_ssc: + 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"); + |