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Diffstat (limited to 'ANDROID_3.4.5/sound/soc/atmel/snd-soc-afeb9260.c')
-rw-r--r--ANDROID_3.4.5/sound/soc/atmel/snd-soc-afeb9260.c163
1 files changed, 163 insertions, 0 deletions
diff --git a/ANDROID_3.4.5/sound/soc/atmel/snd-soc-afeb9260.c b/ANDROID_3.4.5/sound/soc/atmel/snd-soc-afeb9260.c
new file mode 100644
index 00000000..f65f08be
--- /dev/null
+++ b/ANDROID_3.4.5/sound/soc/atmel/snd-soc-afeb9260.c
@@ -0,0 +1,163 @@
+/*
+ * 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 <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->codec_dai;
+ int 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 afeb9260_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_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+ .name = "TLV320AIC23",
+ .stream_name = "AIC23",
+ .cpu_dai_name = "atmel-ssc-dai.0",
+ .codec_dai_name = "tlv320aic23-hifi",
+ .platform_name = "atmel_pcm-audio",
+ .codec_name = "tlv320aic23-codec.0-001a",
+ .init = afeb9260_tlv320aic23_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+ .name = "AFEB9260",
+ .owner = THIS_MODULE,
+ .dai_link = &afeb9260_dai,
+ .num_links = 1,
+
+ .dapm_widgets = tlv320aic23_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
+ .dapm_routes = afeb9260_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+ int err;
+ struct device *dev;
+
+ if (!(machine_is_afeb9260()))
+ return -ENODEV;
+
+
+ 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, &snd_soc_machine_afeb9260);
+ err = platform_device_add(afeb9260_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &afeb9260_snd_device->dev;
+
+ return 0;
+err1:
+ platform_device_put(afeb9260_snd_device);
+ 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");
+