diff options
author | Arun KS <arunks@mistralsolutions.com> | 2008-10-02 15:02:45 +0530 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2008-10-13 02:16:53 +0200 |
commit | 17f9ecf34aaa0ade5c89aba603847309c849297c (patch) | |
tree | f96b445f107cc1f887c5a39a2842f6025c04538c /sound | |
parent | c1f27190a72e9310f1777261b33a05319ff2822c (diff) |
ALSA: ASoC: Add support for osk5912
Adding ASoC machine driver for osk5912
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/omap/Kconfig | 8 | ||||
-rw-r--r-- | sound/soc/omap/Makefile | 2 | ||||
-rw-r--r-- | sound/soc/omap/osk5912.c | 232 |
3 files changed, 242 insertions, 0 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index aea27e70043c..8b7766b998d7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on osk5912. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d8d8d58075e3..e09d1f297f64 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-osk5912-objs := osk5912.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c new file mode 100644 index 000000000000..0fe733796898 --- /dev/null +++ b/sound/soc/omap/osk5912.c @@ -0,0 +1,232 @@ +/* + * osk5912.c -- SoC audio for OSK 5912 + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks@mistralsolutions.com> + * + * 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/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static struct clk *tlv320aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(tlv320aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(tlv320aic23_mclk); +} + +static int osk_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_DSP_A | + 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_DSP_A | + 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 osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +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 osk_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 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 osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = osk_tlv320aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "OSK5912", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *osk_snd_device; + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + tlv320aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(tlv320aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(tlv320aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(tlv320aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + err = -ECANCELED; + goto err1; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, + clk_get_usecount(tlv320aic23_mclk)); + + return 0; +err1: + clk_put(tlv320aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC OSK 5912"); +MODULE_LICENSE("GPL"); |