Re: [PATCH v2 1/1] ASoC: Add HP iPAQ H1940 support

[Jassi Brar <jassisinghbrar@gmail.com>]

On Sun, Sep 12, 2010 at 9:18 PM, Vasily Khoruzhick <anarsoul@gmail.com> wrote:
> Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
> Tested-by: Arnaud Patard <arnaud.patard@rtp-net.org>
> ---
>  sound/soc/s3c24xx/Kconfig         |    8 +
>  sound/soc/s3c24xx/Makefile        |    2 +
>  sound/soc/s3c24xx/h1940_uda1380.c |  297 +++++++++++++++++++++++++++++++++++++
>  3 files changed, 307 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..5dbc0ea
> --- /dev/null
> +++ b/sound/soc/s3c24xx/h1940_uda1380.c
> @@ -0,0 +1,297 @@
> +/*
> + * 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/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:
> +               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;
> +       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");
> +MODULE_ALIAS("platform:soc-audio");

Is platform:soc-audio specific enough in multi-platform builds of the OS?
Though it is unlikely there would ever be such a conflict, but still...
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel