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authorSergey Lapin <slapin@ossfans.org>2009-05-08 19:19:41 +0400
committerMark Brown <broonie@opensource.wolfsonmicro.com>2009-05-08 19:14:34 +0100
commitb4df0a6c9d88cfff77c73d33873cd60f9ab909b6 (patch)
treef408863cf0e4de7de4b9e273bea97585e9b2e71b /sound/soc/atmel
parentbec4c99e8637b5b8bd4b0513eacb51da25885e3b (diff)
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>
Diffstat (limited to 'sound/soc/atmel')
-rw-r--r--sound/soc/atmel/Kconfig8
-rw-r--r--sound/soc/atmel/Makefile1
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c203
3 files changed, 212 insertions, 0 deletions
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index a608d7009db..e720d5e6f04 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 f54a7cc68e6..e7ea56bd5f8 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 00000000000..23349de2731
--- /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");
+