ASoC: Add HP iPAQ H1940 support
authorVasily Khoruzhick <anarsoul@gmail.com>
Thu, 9 Dec 2010 19:17:56 +0000 (21:17 +0200)
committerMark Brown <broonie@opensource.wolfsonmicro.com>
Fri, 10 Dec 2010 17:40:15 +0000 (17:40 +0000)
Add glue driver to make s3c24xx-i2s and uda1380 produce some sound on
H1940.

Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/soc/samsung/Kconfig
sound/soc/samsung/Makefile
sound/soc/samsung/h1940_uda1380.c [new file with mode: 0644]

index eb45cf90d4e2b03fb47abd958baf4bd2649e2b3d..67cdad4ee535415ae532c221263a0e89ca72baf5 100644 (file)
@@ -116,6 +116,14 @@ config ASOC_SAMSUNG_SIMTEC_HERMES
        select SND_SOC_TLV320AIC3X
        select ASOC_SAMSUNG_SIMTEC
 
+config ASOC_SAMSUNG_H1940_UDA1380
+       tristate "Audio support for the HP iPAQ H1940"
+       depends on ASOC_SAMSUNG && ARCH_H1940
+       select SND_S3C24XX_I2S
+       select SND_SOC_UDA1380
+       help
+         This driver provides audio support for HP iPAQ h1940 PDA.
+
 config ASOC_SAMSUNG_RX1950_UDA1380
        tristate "Audio support for the HP iPAQ RX1950"
        depends on ASOC_SAMSUNG && MACH_RX1950
index 0d24f95c8b1c120c923abf0d269a8f98aa85c804..622e76eb97753c1ccefd9bf09ef7a06f7c857d21 100644 (file)
@@ -27,6 +27,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
 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-h1940-uda1380-objs := h1940_uda1380.o
 snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o
 snd-soc-smdk-wm8580-objs := smdk_wm8580.o
 snd-soc-smdk-wm9713-objs := smdk_wm9713.o
@@ -43,6 +44,7 @@ obj-$(CONFIG_ASOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
 obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC) += snd-soc-s3c24xx-simtec.o
 obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
 obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+obj-$(CONFIG_ASOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o
 obj-$(CONFIG_ASOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o
 obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o
 obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
new file mode 100644 (file)
index 0000000..c45f7ce
--- /dev/null
@@ -0,0 +1,296 @@
+/*
+ * h1940-uda1380.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
+ *
+ * Based on version from Arnaud Patard <arnaud.patard@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/uda1380.h>
+#include <sound/jack.h>
+
+#include <plat/regs-iis.h>
+
+#include <mach/h1940-latch.h>
+
+#include <asm/mach-types.h>
+
+#include "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:
+               dev_err(&rtd->dev, "%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;
+       struct snd_soc_dapm_context *dapm = &codec->dapm;
+       int err;
+
+       /* Add h1940 specific widgets */
+       err = snd_soc_dapm_new_controls(dapm, 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(dapm, audio_map,
+                                     ARRAY_SIZE(audio_map));
+       if (err)
+               return err;
+
+       snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+       snd_soc_dapm_enable_pin(dapm, "Speaker");
+       snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+
+       snd_soc_dapm_sync(dapm);
+
+       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  = "samsung-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");