forked from luck/tmp_suning_uos_patched
875065491f
One of the issues with the ASoC v1 API which has been addressed in the ASoC v2 work that Liam Girdwood has done is that the ALSA card provided by ASoC is distributed around the ASoC structures. For example, machine wide data such as the struct snd_card are maintained as part of the CODEC data structure, preventing the use of multiple codecs. This has been addressed by refactoring the data structures so that all the data for the ALSA card is contained in a single structure snd_soc_card which replaces the existing snd_soc_machine and snd_soc_device. Begin the process of backporting this by renaming struct snd_soc_machine to struct snd_soc_card, better reflecting its function and bringing it closer to standard ALSA terminology. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
233 lines
5.7 KiB
C
233 lines
5.7 KiB
C
/*
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* osk5912.c -- SoC audio for OSK 5912
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*
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* Copyright (C) 2008 Mistral Solutions
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*
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* Contact: Arun KS <arunks@mistralsolutions.com>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* version 2 as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*
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*/
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#include <linux/clk.h>
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#include <linux/platform_device.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/soc.h>
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#include <sound/soc-dapm.h>
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#include <asm/mach-types.h>
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#include <mach/hardware.h>
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#include <linux/gpio.h>
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#include <mach/mcbsp.h>
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#include "omap-mcbsp.h"
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#include "omap-pcm.h"
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#include "../codecs/tlv320aic23.h"
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#define CODEC_CLOCK 12000000
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static struct clk *tlv320aic23_mclk;
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static int osk_startup(struct snd_pcm_substream *substream)
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{
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return clk_enable(tlv320aic23_mclk);
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}
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static void osk_shutdown(struct snd_pcm_substream *substream)
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{
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clk_disable(tlv320aic23_mclk);
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}
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static int osk_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
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struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
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int err;
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/* Set codec DAI configuration */
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err = snd_soc_dai_set_fmt(codec_dai,
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SND_SOC_DAIFMT_DSP_A |
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SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (err < 0) {
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printk(KERN_ERR "can't set codec DAI configuration\n");
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return err;
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}
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/* Set cpu DAI configuration */
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err = snd_soc_dai_set_fmt(cpu_dai,
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SND_SOC_DAIFMT_DSP_A |
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SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_CBM_CFM);
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if (err < 0) {
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printk(KERN_ERR "can't set cpu DAI configuration\n");
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return err;
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}
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/* Set the codec system clock for DAC and ADC */
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err =
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snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
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if (err < 0) {
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printk(KERN_ERR "can't set codec system clock\n");
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return err;
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}
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return err;
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}
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static struct snd_soc_ops osk_ops = {
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.startup = osk_startup,
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.hw_params = osk_hw_params,
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.shutdown = osk_shutdown,
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};
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static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_LINE("Line In", NULL),
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SND_SOC_DAPM_MIC("Mic Jack", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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{"Headphone Jack", NULL, "LHPOUT"},
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{"Headphone Jack", NULL, "RHPOUT"},
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{"LLINEIN", NULL, "Line In"},
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{"RLINEIN", NULL, "Line In"},
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{"MICIN", NULL, "Mic Jack"},
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};
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static int osk_tlv320aic23_init(struct snd_soc_codec *codec)
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{
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/* Add osk5912 specific widgets */
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snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
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ARRAY_SIZE(tlv320aic23_dapm_widgets));
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/* Set up osk5912 specific audio path audio_map */
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snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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snd_soc_dapm_enable_pin(codec, "Headphone Jack");
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snd_soc_dapm_enable_pin(codec, "Line In");
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snd_soc_dapm_enable_pin(codec, "Mic Jack");
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snd_soc_dapm_sync(codec);
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return 0;
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}
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/* Digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link osk_dai = {
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.name = "TLV320AIC23",
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.stream_name = "AIC23",
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.cpu_dai = &omap_mcbsp_dai[0],
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.codec_dai = &tlv320aic23_dai,
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.init = osk_tlv320aic23_init,
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.ops = &osk_ops,
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};
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/* Audio machine driver */
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static struct snd_soc_card snd_soc_card_osk = {
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.name = "OSK5912",
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.dai_link = &osk_dai,
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.num_links = 1,
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};
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/* Audio subsystem */
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static struct snd_soc_device osk_snd_devdata = {
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.card = &snd_soc_card_osk,
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.platform = &omap_soc_platform,
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.codec_dev = &soc_codec_dev_tlv320aic23,
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};
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static struct platform_device *osk_snd_device;
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static int __init osk_soc_init(void)
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{
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int err;
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u32 curRate;
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struct device *dev;
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if (!(machine_is_omap_osk()))
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return -ENODEV;
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osk_snd_device = platform_device_alloc("soc-audio", -1);
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if (!osk_snd_device)
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return -ENOMEM;
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platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
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osk_snd_devdata.dev = &osk_snd_device->dev;
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*(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */
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err = platform_device_add(osk_snd_device);
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if (err)
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goto err1;
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dev = &osk_snd_device->dev;
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tlv320aic23_mclk = clk_get(dev, "mclk");
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if (IS_ERR(tlv320aic23_mclk)) {
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printk(KERN_ERR "Could not get mclk clock\n");
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return -ENODEV;
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}
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if (clk_get_usecount(tlv320aic23_mclk) > 0) {
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/* MCLK is already in use */
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printk(KERN_WARNING
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"MCLK in use at %d Hz. We change it to %d Hz\n",
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(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
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}
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/*
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* Configure 12 MHz output on MCLK.
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*/
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curRate = (uint) clk_get_rate(tlv320aic23_mclk);
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if (curRate != CODEC_CLOCK) {
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if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
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printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
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err = -ECANCELED;
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goto err1;
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}
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}
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printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
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(uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
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clk_get_usecount(tlv320aic23_mclk));
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return 0;
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err1:
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clk_put(tlv320aic23_mclk);
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platform_device_del(osk_snd_device);
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platform_device_put(osk_snd_device);
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return err;
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}
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static void __exit osk_soc_exit(void)
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{
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platform_device_unregister(osk_snd_device);
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}
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module_init(osk_soc_init);
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module_exit(osk_soc_exit);
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MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
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MODULE_DESCRIPTION("ALSA SoC OSK 5912");
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MODULE_LICENSE("GPL");
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