a9b696c851
Merge series from Linus Walleij <linus.walleij@linaro.org>:
This cleans up and rewrites the GPIO usage in the TI
ASoC components to use GPIO descriptors exclusively.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
---
Linus Walleij (5):
ASoC: ti: Convert N810 ASoC to GPIO descriptors
ASoC: ti: Convert RX51 to use exclusively GPIO descriptors
ASoC: ti: Convert TWL4030 to use GPIO descriptors
ASoC: ti: Convert Pandora ASoC to GPIO descriptors
ASoC: ti: osk5912: Drop unused include
arch/arm/mach-omap2/board-n8x0.c | 10 +++++
arch/arm/mach-omap2/pdata-quirks.c | 10 +++++
include/linux/platform_data/omap-twl4030.h | 3 --
sound/soc/ti/n810.c | 31 ++++++++-------
sound/soc/ti/omap-twl4030.c | 20 ++++------
sound/soc/ti/omap3pandora.c | 63 +++++++++++-------------------
sound/soc/ti/osk5912.c | 1 -
sound/soc/ti/rx51.c | 19 ++-------
8 files changed, 72 insertions(+), 85 deletions(-)
---
base-commit: 0bb80ecc33
change-id: 20230922-descriptors-asoc-ti-a852eff479ed
Best regards,
--
Linus Walleij <linus.walleij@linaro.org>
372 lines
9.0 KiB
C
372 lines
9.0 KiB
C
// SPDX-License-Identifier: GPL-2.0-only
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/*
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* n810.c -- SoC audio for Nokia N810
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*
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* Copyright (C) 2008 Nokia Corporation
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*
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* Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
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*/
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#include <linux/clk.h>
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#include <linux/i2c.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 <asm/mach-types.h>
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#include <linux/gpio/consumer.h>
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#include <linux/module.h>
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#include <linux/platform_data/asoc-ti-mcbsp.h>
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#include "omap-mcbsp.h"
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static struct gpio_desc *n810_headset_amp;
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static struct gpio_desc *n810_speaker_amp;
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enum {
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N810_JACK_DISABLED,
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N810_JACK_HP,
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N810_JACK_HS,
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N810_JACK_MIC,
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};
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static struct clk *sys_clkout2;
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static struct clk *sys_clkout2_src;
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static struct clk *func96m_clk;
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static int n810_spk_func;
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static int n810_jack_func;
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static int n810_dmic_func;
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static void n810_ext_control(struct snd_soc_dapm_context *dapm)
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{
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int hp = 0, line1l = 0;
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switch (n810_jack_func) {
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case N810_JACK_HS:
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line1l = 1;
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fallthrough;
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case N810_JACK_HP:
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hp = 1;
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break;
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case N810_JACK_MIC:
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line1l = 1;
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break;
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}
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snd_soc_dapm_mutex_lock(dapm);
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if (n810_spk_func)
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snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
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if (hp)
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snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
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if (line1l)
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snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
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if (n810_dmic_func)
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snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
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snd_soc_dapm_sync_unlocked(dapm);
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snd_soc_dapm_mutex_unlock(dapm);
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}
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static int n810_startup(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
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snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
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n810_ext_control(&rtd->card->dapm);
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return clk_prepare_enable(sys_clkout2);
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}
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static void n810_shutdown(struct snd_pcm_substream *substream)
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{
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clk_disable_unprepare(sys_clkout2);
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}
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static int n810_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 = snd_soc_substream_to_rtd(substream);
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struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
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int err;
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/* Set the codec system clock for DAC and ADC */
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err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
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SND_SOC_CLOCK_IN);
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return err;
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}
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static const struct snd_soc_ops n810_ops = {
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.startup = n810_startup,
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.hw_params = n810_hw_params,
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.shutdown = n810_shutdown,
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};
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static int n810_get_spk(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.enumerated.item[0] = n810_spk_func;
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return 0;
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}
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static int n810_set_spk(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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if (n810_spk_func == ucontrol->value.enumerated.item[0])
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return 0;
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n810_spk_func = ucontrol->value.enumerated.item[0];
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n810_ext_control(&card->dapm);
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return 1;
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}
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static int n810_get_jack(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.enumerated.item[0] = n810_jack_func;
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return 0;
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}
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static int n810_set_jack(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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if (n810_jack_func == ucontrol->value.enumerated.item[0])
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return 0;
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n810_jack_func = ucontrol->value.enumerated.item[0];
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n810_ext_control(&card->dapm);
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return 1;
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}
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static int n810_get_input(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.enumerated.item[0] = n810_dmic_func;
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return 0;
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}
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static int n810_set_input(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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if (n810_dmic_func == ucontrol->value.enumerated.item[0])
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return 0;
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n810_dmic_func = ucontrol->value.enumerated.item[0];
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n810_ext_control(&card->dapm);
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return 1;
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}
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static int n810_spk_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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if (SND_SOC_DAPM_EVENT_ON(event))
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gpiod_set_value(n810_speaker_amp, 1);
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else
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gpiod_set_value(n810_speaker_amp, 0);
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return 0;
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}
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static int n810_jack_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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if (SND_SOC_DAPM_EVENT_ON(event))
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gpiod_set_value(n810_headset_amp, 1);
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else
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gpiod_set_value(n810_headset_amp, 0);
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return 0;
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}
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static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
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SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
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SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
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SND_SOC_DAPM_MIC("DMic", NULL),
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SND_SOC_DAPM_MIC("HS Mic", 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, "HPLOUT"},
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{"Headphone Jack", NULL, "HPROUT"},
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{"Ext Spk", NULL, "LLOUT"},
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{"Ext Spk", NULL, "RLOUT"},
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{"DMic Rate 64", NULL, "DMic"},
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{"DMic", NULL, "Mic Bias"},
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/*
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* Note that the mic bias is coming from Retu/Vilma and we don't have
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* control over it atm. The analog HS mic is not working. <- TODO
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*/
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{"LINE1L", NULL, "HS Mic"},
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};
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static const char *spk_function[] = {"Off", "On"};
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static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
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static const char *input_function[] = {"ADC", "Digital Mic"};
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static const struct soc_enum n810_enum[] = {
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SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
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SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
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SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
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};
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static const struct snd_kcontrol_new aic33_n810_controls[] = {
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SOC_ENUM_EXT("Speaker Function", n810_enum[0],
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n810_get_spk, n810_set_spk),
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SOC_ENUM_EXT("Jack Function", n810_enum[1],
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n810_get_jack, n810_set_jack),
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SOC_ENUM_EXT("Input Select", n810_enum[2],
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n810_get_input, n810_set_input),
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};
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/* Digital audio interface glue - connects codec <--> CPU */
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SND_SOC_DAILINK_DEFS(aic33,
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DAILINK_COMP_ARRAY(COMP_CPU("48076000.mcbsp")),
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DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
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"tlv320aic3x-hifi")),
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DAILINK_COMP_ARRAY(COMP_PLATFORM("48076000.mcbsp")));
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static struct snd_soc_dai_link n810_dai = {
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.name = "TLV320AIC33",
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.stream_name = "AIC33",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBM_CFM,
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.ops = &n810_ops,
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SND_SOC_DAILINK_REG(aic33),
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};
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/* Audio machine driver */
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static struct snd_soc_card snd_soc_n810 = {
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.name = "N810",
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.owner = THIS_MODULE,
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.dai_link = &n810_dai,
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.num_links = 1,
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.controls = aic33_n810_controls,
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.num_controls = ARRAY_SIZE(aic33_n810_controls),
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.dapm_widgets = aic33_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets),
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.dapm_routes = audio_map,
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.num_dapm_routes = ARRAY_SIZE(audio_map),
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.fully_routed = true,
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};
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static struct platform_device *n810_snd_device;
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static int __init n810_soc_init(void)
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{
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int err;
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struct device *dev;
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if (!of_have_populated_dt() ||
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(!of_machine_is_compatible("nokia,n810") &&
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!of_machine_is_compatible("nokia,n810-wimax")))
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return -ENODEV;
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n810_snd_device = platform_device_alloc("soc-audio", -1);
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if (!n810_snd_device)
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return -ENOMEM;
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platform_set_drvdata(n810_snd_device, &snd_soc_n810);
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err = platform_device_add(n810_snd_device);
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if (err)
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goto err1;
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dev = &n810_snd_device->dev;
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sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
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if (IS_ERR(sys_clkout2_src)) {
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dev_err(dev, "Could not get sys_clkout2_src clock\n");
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err = PTR_ERR(sys_clkout2_src);
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goto err2;
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}
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sys_clkout2 = clk_get(dev, "sys_clkout2");
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if (IS_ERR(sys_clkout2)) {
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dev_err(dev, "Could not get sys_clkout2\n");
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err = PTR_ERR(sys_clkout2);
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goto err3;
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}
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/*
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* Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
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* 96 MHz as its parent in order to get 12 MHz
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*/
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func96m_clk = clk_get(dev, "func_96m_ck");
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if (IS_ERR(func96m_clk)) {
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dev_err(dev, "Could not get func 96M clock\n");
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err = PTR_ERR(func96m_clk);
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goto err4;
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}
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clk_set_parent(sys_clkout2_src, func96m_clk);
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clk_set_rate(sys_clkout2, 12000000);
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n810_headset_amp = devm_gpiod_get(&n810_snd_device->dev,
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"headphone", GPIOD_OUT_LOW);
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if (IS_ERR(n810_headset_amp)) {
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err = PTR_ERR(n810_headset_amp);
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goto err4;
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}
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n810_speaker_amp = devm_gpiod_get(&n810_snd_device->dev,
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"speaker", GPIOD_OUT_LOW);
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if (IS_ERR(n810_speaker_amp)) {
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err = PTR_ERR(n810_speaker_amp);
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goto err4;
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}
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return 0;
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err4:
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clk_put(sys_clkout2);
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err3:
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clk_put(sys_clkout2_src);
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err2:
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platform_device_del(n810_snd_device);
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err1:
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platform_device_put(n810_snd_device);
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return err;
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}
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static void __exit n810_soc_exit(void)
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{
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clk_put(sys_clkout2_src);
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clk_put(sys_clkout2);
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clk_put(func96m_clk);
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platform_device_unregister(n810_snd_device);
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}
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module_init(n810_soc_init);
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module_exit(n810_soc_exit);
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MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
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MODULE_DESCRIPTION("ALSA SoC Nokia N810");
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MODULE_LICENSE("GPL");
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