ASoC: AFEB9260 driver

ASoC driver for AT91SAM9260-based AFEB9260 board

Signed-off-by: Sergey Lapin <slapin@ossfans.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

3 files changed, 212 insertions, 0 deletions

diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index a608d7009dbd..e720d5e6f04c 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
           and FRAME signals on the PlayPaq.  Unless you want to play
           with the AT32 as the SSC master, you probably want to say N here,
           as this will give you better sound quality.
+
+config SND_AT91_SOC_AFEB9260
+        tristate "SoC Audio support for AFEB9260 board"
+        depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+        select SND_ATMEL_SOC_SSC
+        select SND_SOC_TLV320AIC23
+        help
+          Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index f54a7cc68e66..e7ea56bd5f82 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
+obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
new file mode 100644
index 000000000000..23349de27313
--- /dev/null
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -0,0 +1,203 @@
+/*
+ * afeb9260.c  --  SoC audio for AFEB9260
+ *
+ * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+#define CODEC_CLOCK     12000000
+
+static int afeb9260_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 *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int err;
+
+        /* Set codec DAI configuration */
+        err = snd_soc_dai_set_fmt(codec_dai,
+                                  SND_SOC_DAIFMT_I2S|
+                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (err < 0) {
+                printk(KERN_ERR "can't set codec DAI configuration\n");
+                return err;
+        }
+
+        /* Set cpu DAI configuration */
+        err = snd_soc_dai_set_fmt(cpu_dai,
+                                  SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (err < 0) {
+                printk(KERN_ERR "can't set cpu DAI configuration\n");
+                return err;
+        }
+
+        /* Set the codec system clock for DAC and ADC */
+        err =
+            snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+        if (err < 0) {
+                printk(KERN_ERR "can't set codec system clock\n");
+                return err;
+        }
+
+        return err;
+}
+
+static struct snd_soc_ops afeb9260_ops = {
+        .hw_params = afeb9260_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+        SND_SOC_DAPM_HP("Headphone Jack", NULL),
+        SND_SOC_DAPM_LINE("Line In", NULL),
+        SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+        {"Headphone Jack", NULL, "LHPOUT"},
+        {"Headphone Jack", NULL, "RHPOUT"},
+
+        {"LLINEIN", NULL, "Line In"},
+        {"RLINEIN", NULL, "Line In"},
+
+        {"MICIN", NULL, "Mic Jack"},
+};
+
+static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+        /* Add afeb9260 specific widgets */
+        snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+                                  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+        /* Set up afeb9260 specific audio path audio_map */
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+        snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+        snd_soc_dapm_enable_pin(codec, "Line In");
+        snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+        snd_soc_dapm_sync(codec);
+
+        return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+        .name = "TLV320AIC23",
+        .stream_name = "AIC23",
+        .cpu_dai = &atmel_ssc_dai[0],
+        .codec_dai = &tlv320aic23_dai,
+        .init = afeb9260_tlv320aic23_init,
+        .ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+        .name = "AFEB9260",
+        .platform = &atmel_soc_platform,
+        .dai_link = &afeb9260_dai,
+        .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device afeb9260_snd_devdata = {
+        .card = &snd_soc_machine_afeb9260,
+        .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+        int err;
+        struct device *dev;
+        struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
+        struct ssc_device *ssc = NULL;
+
+        if (!(machine_is_afeb9260()))
+                return -ENODEV;
+
+        ssc = ssc_request(0);
+        if (IS_ERR(ssc)) {
+                printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
+                err = PTR_ERR(ssc);
+                ssc = NULL;
+                goto err_ssc;
+        }
+        ssc_p->ssc = ssc;
+
+        afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!afeb9260_snd_device) {
+                printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+                return -ENOMEM;
+        }
+
+        platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
+        afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
+        err = platform_device_add(afeb9260_snd_device);
+        if (err)
+                goto err1;
+
+        dev = &afeb9260_snd_device->dev;
+
+        return 0;
+err1:
+        platform_device_del(afeb9260_snd_device);
+        platform_device_put(afeb9260_snd_device);
+err_ssc:
+        return err;
+
+}
+
+static void __exit afeb9260_soc_exit(void)
+{
+        platform_device_unregister(afeb9260_snd_device);
+}
+
+module_init(afeb9260_soc_init);
+module_exit(afeb9260_soc_exit);
+
+MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
+MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
+MODULE_LICENSE("GPL");
+