/* * 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_B | 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_B | 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_card snd_soc_card_osk = { .name = "OSK5912", .platform = &omap_soc_platform, .dai_link = &osk_dai, .num_links = 1, }; /* Audio subsystem */ static struct snd_soc_device osk_snd_devdata = { .card = &snd_soc_card_osk, .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");