[alsa-devel] [PATCH] ASoC: Add HP iPAQ H1940 support
Vasily Khoruzhick
anarsoul at gmail.com
Sun Sep 12 12:55:09 CEST 2010
This patch depends on gpiolib support for h1940 latch access,
here's link to patch in ml archive:
http://www.spinics.net/lists/arm-kernel/msg98042.html
I hope Ben will find some time to merge it into his tree
before 2.6.37 merge window :)
Signed-off-by: Vasily Khoruzhick <anarsoul at gmail.com>
Tested-by: Arnaud Patard <arnaud.patard at rtp-net.org>
---
sound/soc/s3c24xx/Kconfig | 8 +
sound/soc/s3c24xx/Makefile | 2 +
sound/soc/s3c24xx/h1940_uda1380.c | 296 +++++++++++++++++++++++++++++++++++++
3 files changed, 306 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 7d8235d..6b50509 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES
select SND_SOC_TLV320AIC3X
select SND_S3C24XX_SOC_SIMTEC
+config SND_S3C24XX_SOC_H1940_UDA1380
+ tristate "Audio support for the HP iPAQ H1940"
+ depends on SND_S3C24XX_SOC && ARCH_H1940
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_UDA1380
+ help
+ This driver provides audio support for HP iPAQ h1940 PDA.
+
config SND_S3C24XX_SOC_RX1950_UDA1380
tristate "Audio support for the HP iPAQ RX1950"
depends on SND_S3C24XX_SOC && MACH_RX1950
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index dd412a9..33a7c68 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o
+snd-soc-h1940-uda1380-objs := h1940_uda1380.o
snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o
snd-soc-smdk-wm9713-objs := smdk_wm9713.o
snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o
@@ -44,6 +45,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o
obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o
+obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o
obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o
obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o
obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o
diff --git a/sound/soc/s3c24xx/h1940_uda1380.c b/sound/soc/s3c24xx/h1940_uda1380.c
new file mode 100644
index 0000000..77550fa
--- /dev/null
+++ b/sound/soc/s3c24xx/h1940_uda1380.c
@@ -0,0 +1,296 @@
+/*
+ * h1940-uda1380.c -- ALSA Soc Audio Layer
+ *
+ * Copyright (c) 2010 Arnaud Patard <arnaud.patard at rtp-net.org>
+ * Copyright (c) 2010 Vasily Khoruzhick <anarsoul at gmail.com>
+ *
+ * Based on version from Arnaud Patard <arnaud.patard at rtp-net.org>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/uda1380.h>
+#include <sound/jack.h>
+
+#include <plat/regs-iis.h>
+
+#include <mach/h1940-latch.h>
+
+#include <asm/mach-types.h>
+
+#include "s3c-dma.h"
+#include "s3c24xx-i2s.h"
+#include "../codecs/uda1380.h"
+
+static unsigned int rates[] = {
+ 11025,
+ 22050,
+ 44100,
+};
+
+static struct snd_pcm_hw_constraint_list hw_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static struct snd_soc_jack hp_jack;
+
+static struct snd_soc_jack_pin hp_jack_pins[] = {
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Speaker",
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1,
+ },
+};
+
+static struct snd_soc_jack_gpio hp_jack_gpios[] = {
+ {
+ .gpio = S3C2410_GPG(4),
+ .name = "hp-gpio",
+ .report = SND_JACK_HEADPHONE,
+ .invert = 1,
+ .debounce_time = 200,
+ },
+};
+
+static int h1940_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.rate_min = hw_rates.list[0];
+ runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
+ runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+
+ return snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &hw_rates);
+}
+
+static int h1940_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 *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int div;
+ int ret;
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 11025:
+ case 22050:
+ case 44100:
+ div = s3c24xx_i2s_get_clockrate() / (384 * rate);
+ if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
+ div++;
+ break;
+ default:
+ printk(KERN_ERR "%s: rate %d is not supported\n",
+ __func__, rate);
+ return -EINVAL;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* select clock source */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* set MCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ S3C2410_IISMOD_384FS);
+ if (ret < 0)
+ return ret;
+
+ /* set BCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+
+ /* set prescaler division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(div, div));
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops h1940_ops = {
+ .startup = h1940_startup,
+ .hw_params = h1940_hw_params,
+};
+
+static int h1940_spk_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
+ else
+ gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
+
+ return 0;
+}
+
+/* h1940 machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
+};
+
+/* h1940 machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* headphone connected to VOUTLHP, VOUTRHP */
+ {"Headphone Jack", NULL, "VOUTLHP"},
+ {"Headphone Jack", NULL, "VOUTRHP"},
+
+ /* ext speaker connected to VOUTL, VOUTR */
+ {"Speaker", NULL, "VOUTL"},
+ {"Speaker", NULL, "VOUTR"},
+
+ /* mic is connected to VINM */
+ {"VINM", NULL, "Mic Jack"},
+};
+
+static struct platform_device *s3c24xx_snd_device;
+
+static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ int err;
+
+ /* Add h1940 specific widgets */
+ err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+ if (err)
+ return err;
+
+ /* Set up h1940 specific audio path audio_mapnects */
+ err = snd_soc_dapm_add_routes(codec, audio_map,
+ ARRAY_SIZE(audio_map));
+ if (err)
+ return err;
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ snd_soc_dapm_sync(codec);
+
+ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
+ &hp_jack);
+
+ snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
+ hp_jack_pins);
+
+ snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
+ hp_jack_gpios);
+
+ return 0;
+}
+
+/* s3c24xx digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link h1940_uda1380_dai[] = {
+ {
+ .name = "uda1380",
+ .stream_name = "UDA1380 Duplex",
+ .cpu_dai_name = "s3c24xx-iis",
+ .codec_dai_name = "uda1380-hifi",
+ .init = h1940_uda1380_init,
+ .platform_name = "s3c24xx-pcm-audio",
+ .codec_name = "uda1380-codec.0-001a",
+ .ops = &h1940_ops,
+ },
+};
+
+static struct snd_soc_card h1940_asoc = {
+ .name = "h1940",
+ .dai_link = h1940_uda1380_dai,
+ .num_links = ARRAY_SIZE(h1940_uda1380_dai),
+};
+
+static int __init h1940_init(void)
+{
+ int ret;
+
+ if (!machine_is_h1940())
+ return -ENODEV;
+
+ /* configure some gpios */
+ ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
+ if (ret)
+ goto err_out;
+
+ ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
+ if (ret)
+ goto err_gpio;
+
+ s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!s3c24xx_snd_device) {
+ ret = -ENOMEM;
+ goto err_gpio;
+ }
+
+ platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
+ ret = platform_device_add(s3c24xx_snd_device);
+
+ if (ret)
+ goto err_plat;
+
+ return 0;
+
+err_plat:
+ platform_device_put(s3c24xx_snd_device);
+err_gpio:
+ gpio_free(H1940_LATCH_AUDIO_POWER);
+
+err_out:
+ return ret;
+}
+
+static void __exit h1940_exit(void)
+{
+ platform_device_unregister(s3c24xx_snd_device);
+ snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
+ hp_jack_gpios);
+ gpio_free(H1940_LATCH_AUDIO_POWER);
+}
+
+module_init(h1940_init);
+module_exit(h1940_exit);
+
+/* Module information */
+MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
+MODULE_DESCRIPTION("ALSA SoC H1940");
+MODULE_LICENSE("GPL");
--
1.7.2.2
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