/* * osk5912.c -- SoC audio for OSK 5912 * * Copyright (C) 2008 Mistral Solutions * * Contact: Arun KS * * 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 #include #include #include #include #include #include #include #include #include "omap-mcbsp.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->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 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"}, }; /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link osk_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", .cpu_dai_name = "omap-mcbsp.1", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-mcbsp.1", .codec_name = "tlv320aic23-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .ops = &osk_ops, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", .owner = THIS_MODULE, .dai_link = &osk_dai, .num_links = 1, .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; 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, &snd_soc_card_osk); 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"); err = PTR_ERR(tlv320aic23_mclk); goto err2; } /* * 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 err3; } } printk(KERN_INFO "MCLK = %d [%d]\n", (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); return 0; err3: clk_put(tlv320aic23_mclk); err2: platform_device_del(osk_snd_device); err1: platform_device_put(osk_snd_device); return err; } static void __exit osk_soc_exit(void) { clk_put(tlv320aic23_mclk); platform_device_unregister(osk_snd_device); } module_init(osk_soc_init); module_exit(osk_soc_exit); MODULE_AUTHOR("Arun KS "); MODULE_DESCRIPTION("ALSA SoC OSK 5912"); MODULE_LICENSE("GPL"); ='ctrl'>mode:
authorLinus Torvalds <torvalds@linux-foundation.org>2017-01-18 11:13:41 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2017-01-18 11:13:41 -0800
commitca92e6c7e6329029d7188487a5c32e86ef471977 (patch)
tree704fb5c2ca533cdb569826522eed0dbbcf31f316 /sound/soc/codecs/adav803.c
parent0b75f821ec8be459dd4dec77be39595d989d77ac (diff)
parent4205e4786d0b9fc3b4fec7b1910cf645a0468307 (diff)
Merge branch 'smp-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip
Pull SMP hotplug update from Thomas Gleixner: "This contains a trivial typo fix and an extension to the core code for dynamically allocating states in the prepare stage. The extension is necessary right now because we need a proper way to unbreak LTTNG, which iscurrently non functional due to the removal of the notifiers. Surely it's out of tree, but it's widely used by distros. The simple solution would have been to reserve a state for LTTNG, but I'm not fond about unused crap in the kernel and the dynamic range, which we admittedly should have done right away, allows us to remove quite some of the hardcoded states, i.e. those which have no ordering requirements. So doing the right thing now is better than having an smaller intermediate solution which needs to be reworked anyway" * 'smp-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: cpu/hotplug: Provide dynamic range for prepare stage perf/x86/amd/ibs: Fix typo after cleanup state names in cpu/hotplug
Diffstat (limited to 'sound/soc/codecs/adav803.c')