At Tue, 20 Nov 2007 09:25:43 +0000,
Mark Brown wrote:
>
> From: Liam Girdwood <liam@xxxxxxxxxxxxxxxxxxxxx>
>
> Signed-off-by: Philipp Zabel <philipp.zabel@xxxxxxxxx>
> Signed-off-by: Liam Girdwood <lg@xxxxxxxxxxxxxxxxxxxxxxxxxxx>
Hmm... Some patches seem to have inconsistent From: and the author
attribute. For exmaple, this module has
MODULE_AUTHOR("Philipp Zabel");
while Liam is in From header here (and it's a broken address :)
Otherwise it looks OK except for a few coding style issues.
To be sure, try checkpatch.pl.
Takashi
> ---
> sound/soc/pxa/Kconfig | 11 +
> sound/soc/pxa/Makefile | 2 +
> sound/soc/pxa/magician.c | 539 ++++++++++++++++++++++++++++++++++++++++++++++
> 3 files changed, 552 insertions(+), 0 deletions(-)
> create mode 100644 sound/soc/pxa/magician.c
>
> diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
> index bcb3aa0..3682f38 100644
> --- a/sound/soc/pxa/Kconfig
> +++ b/sound/soc/pxa/Kconfig
> @@ -58,6 +58,17 @@ config SND_PXA2XX_SOC_TOSA
> Say Y if you want to add support for SoC audio on Sharp
> Zaurus SL-C6000x models (Tosa).
>
> +config SND_PXA2XX_SOC_MAGICIAN
> + tristate "SoC Audio support for HTC Magician"
> + depends on SND_PXA2XX_SOC
> + select SND_PXA2XX_SOC_I2S
> + select SND_PXA2XX_SOC_SSP
> + select SND_SOC_UDA1380
> + help
> + Say Y if you want to add support for SoC audio on the
> + HTC Magician.
> +
> +
> config SND_PXA2XX_SOC_AMESOM_TLV320
> tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k"
> depends on SND_PXA2XX_SOC && MACH_AMESOM
> diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
> index 931bdc7..1faa751 100644
> --- a/sound/soc/pxa/Makefile
> +++ b/sound/soc/pxa/Makefile
> @@ -15,9 +15,11 @@ snd-soc-poodle-objs := poodle.o
> snd-soc-tosa-objs := tosa.o
> snd-soc-spitz-objs := spitz.o
> snd-soc-amesom-tlv320-objs := amesom_tlv320.o
> +snd-soc-magician-objs := magician.o
>
> obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
> obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
> obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
> obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
> obj-$(CONFIG_SND_PXA2XX_SOC_AMESOM_TLV320) += snd-soc-amesom-tlv320.o
> +obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
> \ No newline at end of file
> diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
> new file mode 100644
> index 0000000..7eb671c
> --- /dev/null
> +++ b/sound/soc/pxa/magician.c
> @@ -0,0 +1,539 @@
> +/*
> + * SoC audio for HTC Magician
> + *
> + * Copyright (c) 2006 Philipp Zabel <philipp.zabel@xxxxxxxxx>
> + *
> + * based on spitz.c,
> + * Authors: Liam Girdwood <liam.girdwood@xxxxxxxxxxxxxxxx>
> + * Richard Purdie <richard@xxxxxxxxxxxxxx>
> + *
> + * 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/timer.h>
> +#include <linux/interrupt.h>
> +#include <linux/platform_device.h>
> +#include <linux/delay.h>
> +#include <sound/driver.h>
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/soc.h>
> +#include <sound/soc-dapm.h>
> +
> +#include <asm/hardware/scoop.h>
> +#include <asm/arch/pxa-regs.h>
> +#include <asm/arch/hardware.h>
> +#include <asm/arch/magician.h>
> +#include <asm/arch/magician_cpld.h>
> +#include <asm/mach-types.h>
> +#include "../codecs/uda1380.h"
> +#include "pxa2xx-pcm.h"
> +#include "pxa2xx-i2s.h"
> +#include "pxa2xx-ssp.h"
> +
> +#define MAGICIAN_HP_ON 0
> +#define MAGICIAN_HP_OFF 1
> +
> +#define MAGICIAN_SPK_ON 0
> +#define MAGICIAN_SPK_OFF 1
> +
> +#define MAGICIAN_MIC 0
> +#define MAGICIAN_MIC_EXT 1
> +
> +/*
> + * SSP GPIO's
> + */
> +#define GPIO23_SSPSCLK_MD (23 | GPIO_ALT_FN_2_OUT)
> +#define GPIO24_SSPSFRM_MD (24 | GPIO_ALT_FN_2_OUT)
> +#define GPIO25_SSPTXD_MD (25 | GPIO_ALT_FN_2_OUT)
> +
> +static int magician_hp_func = MAGICIAN_HP_OFF;
> +static int magician_spk_func = MAGICIAN_SPK_ON;
> +static int magician_in_sel = MAGICIAN_MIC;
> +
> +extern struct platform_device magician_cpld;
> +
> +static void magician_ext_control(struct snd_soc_codec *codec)
> +{
> + snd_soc_dapm_set_endpoint(codec, "Speaker",
> + (magician_spk_func == MAGICIAN_SPK_ON));
> +
> + snd_soc_dapm_set_endpoint(codec, "Headphone Jack",
> + (magician_hp_func == MAGICIAN_HP_ON));
> +
> + switch (magician_in_sel) {
> + case MAGICIAN_MIC:
> + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
> + snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
> + break;
> + case MAGICIAN_MIC_EXT:
> + snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
> + snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
> + break;
> + }
> + snd_soc_dapm_sync_endpoints(codec);
> +}
> +
> +static int magician_startup(struct snd_pcm_substream *substream)
> +{
> + struct snd_soc_pcm_runtime *rtd = substream->private_data;
> + struct snd_soc_codec *codec = rtd->socdev->codec;
> +
> + /* check the jack status at stream startup */
> + magician_ext_control(codec);
> +
> + return 0;
> +}
> +
> +/*
> + * Magician uses SSP port for playback.
> + */
> +static int magician_playback_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_codec_dai *codec_dai = rtd->dai->codec_dai;
> + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
> + unsigned int acps, acds, div4;
> + int ret = 0;
> +
> + /*
> + * Rate = SSPSCLK / (word size(16))
> + * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
> + */
> + switch (params_rate(params)) {
> + case 8000:
> + acps = 32842000;
> + acds = PXA2XX_SSP_CLK_AUDIO_DIV_32; /* wrong - 32 bits/sample */
> + div4 = PXA2XX_SSP_CLK_SCDB_4;
> + break;
> + case 11025:
> + acps = 5622000;
> + acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; /* 16 bits/sample, 1 slot */
> + div4 = PXA2XX_SSP_CLK_SCDB_4;
> + break;
> + case 22050:
> + acps = 5622000;
> + acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
> + div4 = PXA2XX_SSP_CLK_SCDB_4;
> + break;
> + case 44100:
> + acps = 11345000;
> + acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
> + div4 = PXA2XX_SSP_CLK_SCDB_4;
> + break;
> + case 48000:
> + acps = 12235000;
> + acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
> + div4 = PXA2XX_SSP_CLK_SCDB_4;
> + break;
> + }
> +
> + /* set codec DAI configuration */
> + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
> + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
> + if (ret < 0)
> + return ret;
> +
> + /* set cpu DAI configuration */
> + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |
> + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
> + if (ret < 0)
> + return ret;
> +
> + /* set audio clock as clock source */
> + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0,
> + SND_SOC_CLOCK_OUT);
> + if (ret < 0)
> + return ret;
> +
> + /* set the SSP audio system clock ACDS divider */
> + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
> + PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
> + if (ret < 0)
> + return ret;
> +
> + /* set the SSP audio system clock SCDB divider4 */
> + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
> + PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
> + if (ret < 0)
> + return ret;
> +
> + /* set SSP audio pll clock */
> + ret = cpu_dai->dai_ops.set_pll(cpu_dai, 0, 0, acps);
> + if (ret < 0)
> + return ret;
> +
> + return 0;
> +}
> +
> +/*
> + * We have to enable the SSP port early so the UDA1380 can flush
> + * it's register cache. The UDA1380 can only write it's interpolator and
> + * decimator registers when the link is running.
> + */
> +static int magician_playback_prepare(struct snd_pcm_substream *substream)
> +{
> + /* enable SSP clock - is this needed ? */
> + SSCR0_P(1) |= SSCR0_SSE;
> +
> + /* FIXME: ENABLE I2S */
> + SACR0 |= SACR0_BCKD;
> + SACR0 |= SACR0_ENB;
> + pxa_set_cken(CKEN8_I2S, 1);
> +
> + return 0;
> +}
> +
> +static int magician_playback_hw_free(struct snd_pcm_substream *substream)
> +{
> + /* FIXME: DISABLE I2S */
> + SACR0 &= ~SACR0_ENB;
> + SACR0 &= ~SACR0_BCKD;
> + pxa_set_cken(CKEN8_I2S, 0);
> + return 0;
> +}
> +
> +/*
> + * Magician uses I2S for capture.
> + */
> +static int magician_capture_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_codec_dai *codec_dai = rtd->dai->codec_dai;
> + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
> + int ret = 0;
> +
> + /* set codec DAI configuration */
> + ret = codec_dai->dai_ops.set_fmt(codec_dai,
> + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
> + if (ret < 0)
> + return ret;
> +
> + /* set cpu DAI configuration */
> + ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
> + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
> + if (ret < 0)
> + return ret;
> +
> + /* set the I2S system clock as output */
> + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
> + SND_SOC_CLOCK_OUT);
> + if (ret < 0)
> + return ret;
> +
> + return 0;
> +}
> +
> +/*
> + * We have to enable the I2S port early so the UDA1380 can flush
> + * it's register cache. The UDA1380 can only write it's interpolator and
> + * decimator registers when the link is running.
> + */
> +static int magician_capture_prepare(struct snd_pcm_substream *substream)
> +{
> + SACR0 |= SACR0_ENB;
> + return 0;
> +}
> +
> +static struct snd_soc_ops magician_capture_ops = {
> + .startup = magician_startup,
> + .hw_params = magician_capture_hw_params,
> + .prepare = magician_capture_prepare,
> +};
> +
> +static struct snd_soc_ops magician_playback_ops = {
> + .startup = magician_startup,
> + .hw_params = magician_playback_hw_params,
> + .prepare = magician_playback_prepare,
> + .hw_free = magician_playback_hw_free,
> +};
> +
> +static int magician_get_jack(struct snd_kcontrol * kcontrol,
> + struct snd_ctl_elem_value * ucontrol)
> +{
> + ucontrol->value.integer.value[0] = magician_hp_func;
> + return 0;
> +}
> +
> +static int magician_set_hp(struct snd_kcontrol * kcontrol,
> + struct snd_ctl_elem_value * ucontrol)
> +{
> + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
> +
> + if (magician_hp_func == ucontrol->value.integer.value[0])
> + return 0;
> +
> + magician_hp_func = ucontrol->value.integer.value[0];
> + magician_ext_control(codec);
> + return 1;
> +}
> +
> +static int magician_get_spk(struct snd_kcontrol * kcontrol,
> + struct snd_ctl_elem_value * ucontrol)
> +{
> + ucontrol->value.integer.value[0] = magician_spk_func;
> + return 0;
> +}
> +
> +static int magician_set_spk(struct snd_kcontrol * kcontrol,
> + struct snd_ctl_elem_value * ucontrol)
> +{
> + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
> +
> + if (magician_spk_func == ucontrol->value.integer.value[0])
> + return 0;
> +
> + magician_spk_func = ucontrol->value.integer.value[0];
> + magician_ext_control(codec);
> + return 1;
> +}
> +
> +static int magician_get_input(struct snd_kcontrol * kcontrol,
> + struct snd_ctl_elem_value * ucontrol)
> +{
> + ucontrol->value.integer.value[0] = magician_in_sel;
> + return 0;
> +}
> +
> +static int magician_set_input(struct snd_kcontrol * kcontrol,
> + struct snd_ctl_elem_value * ucontrol)
> +{
> + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
> +
> + if (magician_in_sel == ucontrol->value.integer.value[0])
> + return 0;
> +
> + magician_in_sel = ucontrol->value.integer.value[0];
> +
> + switch (magician_in_sel) {
> + case MAGICIAN_MIC:
> + magician_egpio_disable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_IN_SEL0);
> + magician_egpio_enable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_IN_SEL1);
> + break;
> + case MAGICIAN_MIC_EXT:
> + magician_egpio_disable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_IN_SEL0);
> + magician_egpio_disable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_IN_SEL1);
> + }
> +
> + return 1;
> +}
> +
> +static int magician_spk_power(struct snd_soc_dapm_widget *w, int event)
> +{
> + if (SND_SOC_DAPM_EVENT_ON(event))
> + magician_egpio_enable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_SPK_POWER);
> + else
> + magician_egpio_disable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_SPK_POWER);
> + return 0;
> +}
> +
> +static int magician_hp_power(struct snd_soc_dapm_widget *w, int event)
> +{
> + if (SND_SOC_DAPM_EVENT_ON(event))
> + magician_egpio_enable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_EP_POWER);
> + else
> + magician_egpio_disable(&magician_cpld,
> + EGPIO_NR_MAGICIAN_EP_POWER);
> + return 0;
> +}
> +
> +static int magician_mic_bias(struct snd_soc_dapm_widget *w, int event)
> +{
> +// if (SND_SOC_DAPM_EVENT_ON(event))
> +// magician_egpio_enable(&magician_cpld,
> +// EGPIO_NR_MAGICIAN_MIC_POWER);
> +// else
> +// magician_egpio_disable(&magician_cpld,
> +// EGPIO_NR_MAGICIAN_MIC_POWER);
> + return 0;
> +}
> +
> +/* magician machine dapm widgets */
> +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
> + SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
> + SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
> + SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
> + SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
> +};
> +
> +/* magician machine audio_map */
> +static const char *audio_map[][3] = {
> +
> + /* Headphone connected to VOUTL, VOUTR */
> + {"Headphone Jack", NULL, "VOUTL"},
> + {"Headphone Jack", NULL, "VOUTR"},
> +
> + /* Speaker connected to VOUTL, VOUTR */
> + {"Speaker", NULL, "VOUTL"},
> + {"Speaker", NULL, "VOUTR"},
> +
> + /* Mics are connected to VINM */
> + {"VINM", NULL, "Headset Mic"},
> + {"VINM", NULL, "Call Mic"},
> +
> + {NULL, NULL, NULL},
> +};
> +
> +static const char *hp_function[] = { "On", "Off" };
> +static const char *spk_function[] = { "On", "Off" };
> +static const char *input_select[] = { "Call Mic", "Headset Mic" };
> +static const struct soc_enum magician_enum[] = {
> + SOC_ENUM_SINGLE_EXT(4, hp_function),
> + SOC_ENUM_SINGLE_EXT(2, spk_function),
> + SOC_ENUM_SINGLE_EXT(2, input_select),
> +};
> +
> +static const struct snd_kcontrol_new uda1380_magician_controls[] = {
> + SOC_ENUM_EXT("Headphone Switch", magician_enum[0], magician_get_jack,
> + magician_set_hp),
> + SOC_ENUM_EXT("Speaker Switch", magician_enum[1], magician_get_spk,
> + magician_set_spk),
> + SOC_ENUM_EXT("Input Select", magician_enum[2], magician_get_input,
> + magician_set_input),
> +};
> +
> +/*
> + * Logic for a uda1380 as connected on a HTC Magician
> + */
> +static int magician_uda1380_init(struct snd_soc_codec *codec)
> +{
> + int i, err;
> +
> + /* NC codec pins */
> + snd_soc_dapm_set_endpoint(codec, "VOUTLHP", 0);
> + snd_soc_dapm_set_endpoint(codec, "VOUTRHP", 0);
> +
> + /* FIXME: is anything connected here? */
> + snd_soc_dapm_set_endpoint(codec, "VINL", 0);
> + snd_soc_dapm_set_endpoint(codec, "VINR", 0);
> +
> + /* Add magician specific controls */
> + for (i = 0; i < ARRAY_SIZE(uda1380_magician_controls); i++) {
> + if ((err = snd_ctl_add(codec->card,
> + snd_soc_cnew(&uda1380_magician_controls[i],
> + codec, NULL))) < 0)
> + return err;
> + }
> +
> + /* Add magician specific widgets */
> + for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
> + snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
> + }
> +
> + /* Set up magician specific audio path interconnects */
> + for (i = 0; audio_map[i][0] != NULL; i++) {
> + snd_soc_dapm_connect_input(codec, audio_map[i][0],
> + audio_map[i][1], audio_map[i][2]);
> + }
> +
> + snd_soc_dapm_sync_endpoints(codec);
> + return 0;
> +}
> +
> +/* magician digital audio interface glue - connects codec <--> CPU */
> +static struct snd_soc_dai_link magician_dai[] = {
> +{
> + .name = "uda1380",
> + .stream_name = "UDA1380 Playback",
> + .cpu_dai = &pxa_ssp_dai[0],
> + .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
> + .init = magician_uda1380_init,
> + .ops = &magician_playback_ops,
> +},
> +{
> + .name = "uda1380",
> + .stream_name = "UDA1380 Capture",
> + .cpu_dai = &pxa_i2s_dai,
> + .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
> + .ops = &magician_capture_ops,
> +}
> +};
> +
> +/* magician audio machine driver */
> +static struct snd_soc_machine snd_soc_machine_magician = {
> + .name = "Magician",
> + .dai_link = magician_dai,
> + .num_links = ARRAY_SIZE(magician_dai),
> +};
> +
> +/* magician audio private data */
> +static struct uda1380_setup_data magician_uda1380_setup = {
> + .i2c_address = 0x18,
> + .dac_clk = UDA1380_DAC_CLK_WSPLL,
> +};
> +
> +/* magician audio subsystem */
> +static struct snd_soc_device magician_snd_devdata = {
> + .machine = &snd_soc_machine_magician,
> + .platform = &pxa2xx_soc_platform,
> + .codec_dev = &soc_codec_dev_uda1380,
> + .codec_data = &magician_uda1380_setup,
> +};
> +
> +static struct platform_device *magician_snd_device;
> +
> +static int __init magician_init(void)
> +{
> + int ret;
> +
> + if (!machine_is_magician())
> + return -ENODEV;
> +
> + magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
> +
> + /* we may need to have the clock running here - pH5 */
> + magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
> + udelay(5);
> + magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
> +
> + /* correct place? we'll need it to talk to the uda1380 */
> + request_module("i2c-pxa");
> +
> + magician_snd_device = platform_device_alloc("soc-audio", -1);
> + if (!magician_snd_device)
> + return -ENOMEM;
> +
> + platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
> + magician_snd_devdata.dev = &magician_snd_device->dev;
> + ret = platform_device_add(magician_snd_device);
> +
> + if (ret)
> + platform_device_put(magician_snd_device);
> +
> + pxa_gpio_mode(GPIO23_SSPSCLK_MD);
> + pxa_gpio_mode(GPIO24_SSPSFRM_MD);
> + pxa_gpio_mode(GPIO25_SSPTXD_MD);
> +
> + return ret;
> +}
> +
> +static void __exit magician_exit(void)
> +{
> + platform_device_unregister(magician_snd_device);
> +
> + magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
> + magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
> + magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
> + magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
> +}
> +
> +module_init(magician_init);
> +module_exit(magician_exit);
> +
> +MODULE_AUTHOR("Philipp Zabel");
> +MODULE_DESCRIPTION("ALSA SoC Magician");
> +MODULE_LICENSE("GPL");
> --
> 1.5.3.5
>
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>
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