Re: [PATCH] Add Amesom ASoC machine support.

["Vladimir A. Barinov" <vbarinov@ru.mvista.com>]

Mark,

I suggest to do the same naming changes in the machine part like I 've 
suggested for tlv32024K codec driver: rename all tlv320 places to 
tlv320aic2x and "aic2x_"

Also I suggest to name all static functions in the machine ASoC part to 
begin from ameson_ and not the codec one.

Regards,
Vladimir

Mark Brown wrote:
> From: Liam Girdwood <liam@localhost.localdomain>
>
> Signed-off-by: Nicola Perrino <nicola.perrino@atlab.it>
> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
> ---
>  sound/soc/pxa/Kconfig         |   10 ++
>  sound/soc/pxa/Makefile        |    3 +-
>  sound/soc/pxa/amesom_tlv320.c |  211 +++++++++++++++++++++++++++++++++++++++++
>  3 files changed, 223 insertions(+), 1 deletions(-)
>  create mode 100644 sound/soc/pxa/amesom_tlv320.c
>
> diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
> index ca825e6..bcb3aa0 100644
> --- a/sound/soc/pxa/Kconfig
> +++ b/sound/soc/pxa/Kconfig
> @@ -57,3 +57,13 @@ config SND_PXA2XX_SOC_TOSA
>  	help
>  	  Say Y if you want to add support for SoC audio on Sharp
>  	  Zaurus SL-C6000x models (Tosa).
> +
> +config SND_PXA2XX_SOC_AMESOM_TLV320
> +	tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k"
> +	depends on SND_PXA2XX_SOC && MACH_AMESOM
> +	select SND_PXA2XX_SOC_I2S
> +	select SND_PXA2XX_SOC_SSP
> +	select SND_SOC_TLV320
> +	help
> +	  Say Y if you want to add support for SoC audio on Amesom
> +	  with the tlv320.
> diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
> index c631651..931bdc7 100644
> --- a/sound/soc/pxa/Makefile
> +++ b/sound/soc/pxa/Makefile
> @@ -14,9 +14,10 @@ snd-soc-corgi-objs := corgi.o
>  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
>  
>  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
> diff --git a/sound/soc/pxa/amesom_tlv320.c b/sound/soc/pxa/amesom_tlv320.c
> new file mode 100644
> index 0000000..6aa1c2c
> --- /dev/null
> +++ b/sound/soc/pxa/amesom_tlv320.c
> @@ -0,0 +1,211 @@
> +/*
> + * amesom_tlv320.c  --  SoC audio for Amesom
> + *
> + * Copyright 2005 Wolfson Microelectronics PLC.
> + * Copyright 2006 Atlab srl.
> + *
> + * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
> + *          Nicola Perrino <nicola.perrino@atlab.it>
> + *
> + *  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.
> + *
> + *  Revision history
> + *    5th Dec 2006   Initial version.
> + *
> + */
> +
> +#include <linux/module.h>
> +#include <linux/moduleparam.h>
> +#include <linux/device.h>
> +#include <linux/i2c.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.h>
> +#include <asm/arch/pxa-regs.h>
> +#include <asm/arch/audio.h>
> +
> +#include "../codecs/tlv320.h"
> +#include "pxa2xx-pcm.h"
> +#include "pxa2xx-i2s.h"
> +#include "pxa2xx-ssp.h"
> +
> +
> +/*
> + * SSP2 GPIO's
> + */
> +
> +#define GPIO11_SSP2RX_MD	(11 | GPIO_ALT_FN_2_IN)
> +#define GPIO13_SSP2TX_MD	(13 | GPIO_ALT_FN_1_OUT)
> +#define GPIO50_SSP2CLKS_MD	(50 | GPIO_ALT_FN_3_IN)
> +#define GPIO14_SSP2FRMS_MD	(14 | GPIO_ALT_FN_2_IN)
> +#define GPIO50_SSP2CLKM_MD	(50 | GPIO_ALT_FN_3_OUT)
> +#define GPIO14_SSP2FRMM_MD	(14 | GPIO_ALT_FN_2_OUT)
> +
> +
> +static struct snd_soc_machine amesom;
> +
> +
> +static int amesom_probe(struct platform_device *pdev)
> +{
> +	return 0;
> +}
> +
> +static int amesom_remove(struct platform_device *pdev)
> +{
> +	return 0;
> +}
> +
> +static int tlv320_voice_startup(struct snd_pcm_substream *substream)
> +{
> +	return 0;
> +}
> +
> +static void tlv320_voice_shutdown(struct snd_pcm_substream *substream)
> +{
> +	return;
> +}
> +
> +/*
> + * Tlv320 uses SSP port for playback.
> + */
> +static int tlv320_voice_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;
> +
> +	//printk("tlv320_voice_hw_params enter\n");
> +	switch(params_rate(params)) {
> +	case 8000:
> +		//printk("tlv320_voice_hw_params 8000\n");
> +		break;
> +	case 16000:
> +		//printk("tlv320_voice_hw_params 16000\n");
> +		break;
> +	default:
> +		break;
> +	}
> +
> +	// CODEC MASTER, SSP SLAVE
> +
> +	/* 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_CBM_CFM);
> +	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_CBM_CFM);
> +	if (ret < 0)
> +		return ret;
> +
> +	/* set the SSP system clock as input (unused) */
> +	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_NET_PLL, 0,
> +		SND_SOC_CLOCK_IN);
> +	if (ret < 0)
> +		return ret;
> +
> +	/* set SSP slots */
> +	//ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x1, slots);
> +	ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x3, 1);
> +	if (ret < 0)
> +		return ret;
> +
> +	return 0;
> +}
> +
> +static int tlv320_voice_hw_free(struct snd_pcm_substream *substream)
> +{
> +	return 0;
> +}
> +
> +static struct snd_soc_ops tlv320_voice_ops = {
> +	.startup = tlv320_voice_startup,
> +	.shutdown = tlv320_voice_shutdown,
> +	.hw_params = tlv320_voice_hw_params,
> +	.hw_free = tlv320_voice_hw_free,
> +};
> +
> +
> +static struct snd_soc_dai_link amesom_dai[] = {
> +{
> +	.name = "TLV320",
> +	.stream_name = "TLV320 Voice",
> +	.cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2],
> +	.codec_dai = &tlv320_dai[TLV320_DAI_MODE1_VOICE],
> +	.ops = &tlv320_voice_ops,
> +},
> +};
> +
> +static struct snd_soc_machine amesom = {
> +	.name = "Amesom",
> +	.probe = amesom_probe,
> +	.remove = amesom_remove,
> +	.dai_link = amesom_dai,
> +	.num_links = ARRAY_SIZE(amesom_dai),
> +};
> +
> +static struct tlv320_setup_data amesom_tlv320_setup = {
> +#ifdef TLV320AIC24K //codec2
> +	.i2c_address = 0x41,
> +#else // TLV320AIC14k
> +	.i2c_address = 0x40,
> +#endif
> +};
> +
> +static struct snd_soc_device amesom_snd_devdata = {
> +	.machine = &amesom,
> +	.platform = &pxa2xx_soc_platform,
> +	.codec_dev = &soc_codec_dev_tlv320,
> +	.codec_data = &amesom_tlv320_setup,
> +};
> +
> +static struct platform_device *amesom_snd_device;
> +
> +static int __init amesom_init(void)
> +{
> +	int ret;
> +
> +	amesom_snd_device = platform_device_alloc("soc-audio", -1);
> +	if (!amesom_snd_device)
> +		return -ENOMEM;
> +
> +	platform_set_drvdata(amesom_snd_device, &amesom_snd_devdata);
> +	amesom_snd_devdata.dev = &amesom_snd_device->dev;
> +	ret = platform_device_add(amesom_snd_device);
> +
> +	if (ret)
> +		platform_device_put(amesom_snd_device);
> +
> +
> +	/* SSP port 2 slave */
> +	pxa_gpio_mode(GPIO11_SSP2RX_MD);
> +	pxa_gpio_mode(GPIO13_SSP2TX_MD);
> +	pxa_gpio_mode(GPIO50_SSP2CLKS_MD);
> +	pxa_gpio_mode(GPIO14_SSP2FRMS_MD);
> +
> +	return ret;
> +}
> +
> +static void __exit amesom_exit(void)
> +{
> +	platform_device_unregister(amesom_snd_device);
> +}
> +
> +module_init(amesom_init);
> +module_exit(amesom_exit);
> +
> +/* Module information */
> +MODULE_AUTHOR("Nicola Perrino");
> +MODULE_DESCRIPTION("ALSA SoC TLV320 Amesom");
> +MODULE_LICENSE("GPL");
>