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f2055e145f
Create new directory to contain all Texas Instruments specific DAI, platform and machine drivers instead of scattering them under davinci and omap directories. There is already inter dependency between the two directories becasue of McASP (on dra7x it is serviced by sDMA, not EDMA). With the upcoming AM654 we will need to introduce new platform driver for UDMA and it does not fit under davinci, nor under omap. With the move I have restructured the Kconfig to be more usable in the era of simple-sound-card: CPU DAIs can be selected individually and they will select the platform driver they can be served with. To avoid breakage, I have moved over deprecated Kconfig options so defconfig builds will work without regression. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> For sound/soc/{omap => ti}: Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@kernel.org>
354 lines
9.3 KiB
C
354 lines
9.3 KiB
C
/*
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* omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and
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* twl6040 codec
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*
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* Author: Misael Lopez Cruz <misael.lopez@ti.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 <linux/mfd/twl6040.h>
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#include <linux/module.h>
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#include <linux/of.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/jack.h>
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#include "omap-dmic.h"
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#include "omap-mcpdm.h"
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#include "../codecs/twl6040.h"
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struct abe_twl6040 {
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struct snd_soc_card card;
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struct snd_soc_dai_link dai_links[2];
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int jack_detection; /* board can detect jack events */
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int mclk_freq; /* MCLK frequency speed for twl6040 */
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};
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static struct platform_device *dmic_codec_dev;
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static int omap_abe_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->codec_dai;
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struct snd_soc_card *card = rtd->card;
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struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
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int clk_id, freq;
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int ret;
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clk_id = twl6040_get_clk_id(codec_dai->component);
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if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
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freq = priv->mclk_freq;
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else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
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freq = 32768;
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else
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return -EINVAL;
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/* set the codec mclk */
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ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
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SND_SOC_CLOCK_IN);
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if (ret) {
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printk(KERN_ERR "can't set codec system clock\n");
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return ret;
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}
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return ret;
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}
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static const struct snd_soc_ops omap_abe_ops = {
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.hw_params = omap_abe_hw_params,
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};
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static int omap_abe_dmic_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 *cpu_dai = rtd->cpu_dai;
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int ret = 0;
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ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
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19200000, SND_SOC_CLOCK_IN);
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if (ret < 0) {
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printk(KERN_ERR "can't set DMIC cpu system clock\n");
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return ret;
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}
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ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
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SND_SOC_CLOCK_OUT);
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if (ret < 0) {
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printk(KERN_ERR "can't set DMIC output clock\n");
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return ret;
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}
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return 0;
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}
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static struct snd_soc_ops omap_abe_dmic_ops = {
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.hw_params = omap_abe_dmic_hw_params,
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};
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/* Headset jack */
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static struct snd_soc_jack hs_jack;
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/*Headset jack detection DAPM pins */
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static struct snd_soc_jack_pin hs_jack_pins[] = {
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{
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.pin = "Headset Mic",
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.mask = SND_JACK_MICROPHONE,
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},
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{
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.pin = "Headset Stereophone",
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.mask = SND_JACK_HEADPHONE,
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},
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};
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/* SDP4430 machine DAPM */
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static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
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/* Outputs */
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SND_SOC_DAPM_HP("Headset Stereophone", NULL),
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SND_SOC_DAPM_SPK("Earphone Spk", NULL),
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SND_SOC_DAPM_SPK("Ext Spk", NULL),
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SND_SOC_DAPM_LINE("Line Out", NULL),
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SND_SOC_DAPM_SPK("Vibrator", NULL),
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/* Inputs */
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
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SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
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SND_SOC_DAPM_LINE("Line In", NULL),
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/* Digital microphones */
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SND_SOC_DAPM_MIC("Digital Mic", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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/* Routings for outputs */
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{"Headset Stereophone", NULL, "HSOL"},
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{"Headset Stereophone", NULL, "HSOR"},
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{"Earphone Spk", NULL, "EP"},
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{"Ext Spk", NULL, "HFL"},
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{"Ext Spk", NULL, "HFR"},
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{"Line Out", NULL, "AUXL"},
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{"Line Out", NULL, "AUXR"},
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{"Vibrator", NULL, "VIBRAL"},
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{"Vibrator", NULL, "VIBRAR"},
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/* Routings for inputs */
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{"HSMIC", NULL, "Headset Mic"},
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{"Headset Mic", NULL, "Headset Mic Bias"},
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{"MAINMIC", NULL, "Main Handset Mic"},
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{"Main Handset Mic", NULL, "Main Mic Bias"},
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{"SUBMIC", NULL, "Sub Handset Mic"},
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{"Sub Handset Mic", NULL, "Main Mic Bias"},
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{"AFML", NULL, "Line In"},
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{"AFMR", NULL, "Line In"},
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};
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static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_component *component = rtd->codec_dai->component;
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struct snd_soc_card *card = rtd->card;
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struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
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int hs_trim;
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int ret = 0;
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/*
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* Configure McPDM offset cancellation based on the HSOTRIM value from
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* twl6040.
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*/
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hs_trim = twl6040_get_trim_value(component, TWL6040_TRIM_HSOTRIM);
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omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
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TWL6040_HSF_TRIM_RIGHT(hs_trim));
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/* Headset jack detection only if it is supported */
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if (priv->jack_detection) {
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ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
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SND_JACK_HEADSET, &hs_jack,
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hs_jack_pins,
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ARRAY_SIZE(hs_jack_pins));
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if (ret)
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return ret;
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twl6040_hs_jack_detect(component, &hs_jack, SND_JACK_HEADSET);
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}
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return 0;
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}
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static const struct snd_soc_dapm_route dmic_audio_map[] = {
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{"DMic", NULL, "Digital Mic"},
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{"Digital Mic", NULL, "Digital Mic1 Bias"},
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};
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static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
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return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
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ARRAY_SIZE(dmic_audio_map));
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}
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static int omap_abe_probe(struct platform_device *pdev)
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{
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struct device_node *node = pdev->dev.of_node;
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struct snd_soc_card *card;
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struct device_node *dai_node;
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struct abe_twl6040 *priv;
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int num_links = 0;
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int ret = 0;
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if (!node) {
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dev_err(&pdev->dev, "of node is missing.\n");
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return -ENODEV;
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}
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priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
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if (priv == NULL)
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return -ENOMEM;
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card = &priv->card;
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card->dev = &pdev->dev;
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card->owner = THIS_MODULE;
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card->dapm_widgets = twl6040_dapm_widgets;
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card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
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card->dapm_routes = audio_map;
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card->num_dapm_routes = ARRAY_SIZE(audio_map);
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if (snd_soc_of_parse_card_name(card, "ti,model")) {
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dev_err(&pdev->dev, "Card name is not provided\n");
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return -ENODEV;
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}
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ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
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if (ret) {
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dev_err(&pdev->dev, "Error while parsing DAPM routing\n");
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return ret;
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}
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dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
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if (!dai_node) {
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dev_err(&pdev->dev, "McPDM node is not provided\n");
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return -EINVAL;
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}
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priv->dai_links[0].name = "DMIC";
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priv->dai_links[0].stream_name = "TWL6040";
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priv->dai_links[0].cpu_of_node = dai_node;
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priv->dai_links[0].platform_of_node = dai_node;
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priv->dai_links[0].codec_dai_name = "twl6040-legacy";
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priv->dai_links[0].codec_name = "twl6040-codec";
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priv->dai_links[0].init = omap_abe_twl6040_init;
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priv->dai_links[0].ops = &omap_abe_ops;
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dai_node = of_parse_phandle(node, "ti,dmic", 0);
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if (dai_node) {
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num_links = 2;
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priv->dai_links[1].name = "TWL6040";
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priv->dai_links[1].stream_name = "DMIC Capture";
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priv->dai_links[1].cpu_of_node = dai_node;
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priv->dai_links[1].platform_of_node = dai_node;
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priv->dai_links[1].codec_dai_name = "dmic-hifi";
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priv->dai_links[1].codec_name = "dmic-codec";
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priv->dai_links[1].init = omap_abe_dmic_init;
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priv->dai_links[1].ops = &omap_abe_dmic_ops;
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} else {
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num_links = 1;
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}
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priv->jack_detection = of_property_read_bool(node, "ti,jack-detection");
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of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq);
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if (!priv->mclk_freq) {
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dev_err(&pdev->dev, "MCLK frequency not provided\n");
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return -EINVAL;
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}
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card->fully_routed = 1;
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if (!priv->mclk_freq) {
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dev_err(&pdev->dev, "MCLK frequency missing\n");
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return -ENODEV;
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}
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card->dai_link = priv->dai_links;
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card->num_links = num_links;
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snd_soc_card_set_drvdata(card, priv);
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ret = devm_snd_soc_register_card(&pdev->dev, card);
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if (ret)
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dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n",
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ret);
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return ret;
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}
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static const struct of_device_id omap_abe_of_match[] = {
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{.compatible = "ti,abe-twl6040", },
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{ },
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};
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MODULE_DEVICE_TABLE(of, omap_abe_of_match);
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static struct platform_driver omap_abe_driver = {
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.driver = {
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.name = "omap-abe-twl6040",
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.pm = &snd_soc_pm_ops,
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.of_match_table = omap_abe_of_match,
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},
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.probe = omap_abe_probe,
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};
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static int __init omap_abe_init(void)
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{
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int ret;
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dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL,
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0);
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if (IS_ERR(dmic_codec_dev)) {
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pr_err("%s: dmic-codec device registration failed\n", __func__);
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return PTR_ERR(dmic_codec_dev);
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}
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ret = platform_driver_register(&omap_abe_driver);
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if (ret) {
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pr_err("%s: platform driver registration failed\n", __func__);
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platform_device_unregister(dmic_codec_dev);
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}
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return ret;
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}
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module_init(omap_abe_init);
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static void __exit omap_abe_exit(void)
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{
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platform_driver_unregister(&omap_abe_driver);
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platform_device_unregister(dmic_codec_dev);
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}
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module_exit(omap_abe_exit);
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MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
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MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
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MODULE_LICENSE("GPL");
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MODULE_ALIAS("platform:omap-abe-twl6040");
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