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authorLinus Torvalds <torvalds@linux-foundation.org>2008-04-29 09:38:52 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2008-04-29 09:38:52 -0700
commit25a025863e024f6b86b48137b10b4960c50351b0 (patch)
tree72d2521585f61d904769d28cf1d7687b949a61a6 /sound/soc/omap/n810.c
parent1f43c5393033de90bac4410352b1d2a69dcbe7ef (diff)
parent7e48bf653c37eb32c2ba4c13f15aa154aa807e61 (diff)
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: [ALSA] soc - wm9712 - checkpatch fixes [ALSA] pcsp - Fix more dependency [ALSA] hda - Add support of Medion RIM 2150 [ALSA] ASoC: Add drivers for the Texas Instruments OMAP processors [ALSA] ice1724 - Enable watermarks [ALSA] Add MPU401_INFO_NO_ACK bitflag
Diffstat (limited to 'sound/soc/omap/n810.c')
-rw-r--r--sound/soc/omap/n810.c336
1 files changed, 336 insertions, 0 deletions
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
new file mode 100644
index 000000000000..83b1eb4e40f3
--- /dev/null
+++ b/sound/soc/omap/n810.c
@@ -0,0 +1,336 @@
+/*
+ * n810.c -- SoC audio for Nokia N810
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * 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/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/gpio.h>
+#include <asm/arch/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic3x.h"
+
+#define RX44_HEADSET_AMP_GPIO 10
+#define RX44_SPEAKER_AMP_GPIO 101
+
+static struct clk *sys_clkout2;
+static struct clk *sys_clkout2_src;
+static struct clk *func96m_clk;
+
+static int n810_spk_func;
+static int n810_jack_func;
+
+static void n810_ext_control(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func);
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func);
+
+ snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int n810_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ n810_ext_control(codec);
+ return clk_enable(sys_clkout2);
+}
+
+static void n810_shutdown(struct snd_pcm_substream *substream)
+{
+ clk_disable(sys_clkout2);
+}
+
+static int n810_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 err;
+
+ /* Set codec DAI configuration */
+ err = codec_dai->dai_ops.set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0)
+ return err;
+
+ /* Set cpu DAI configuration */
+ err = cpu_dai->dai_ops.set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0)
+ return err;
+
+ /* Set the codec system clock for DAC and ADC */
+ err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000,
+ SND_SOC_CLOCK_IN);
+
+ return err;
+}
+
+static struct snd_soc_ops n810_ops = {
+ .startup = n810_startup,
+ .hw_params = n810_hw_params,
+ .shutdown = n810_shutdown,
+};
+
+static int n810_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = n810_spk_func;
+
+ return 0;
+}
+
+static int n810_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (n810_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ n810_spk_func = ucontrol->value.integer.value[0];
+ n810_ext_control(codec);
+
+ return 1;
+}
+
+static int n810_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = n810_jack_func;
+
+ return 0;
+}
+
+static int n810_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (n810_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ n810_jack_func = ucontrol->value.integer.value[0];
+ n810_ext_control(codec);
+
+ return 1;
+}
+
+static int n810_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1);
+ else
+ omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0);
+
+ return 0;
+}
+
+static int n810_jack_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1);
+ else
+ omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
+ SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
+};
+
+static const char *audio_map[][3] = {
+ {"Headphone Jack", NULL, "HPLOUT"},
+ {"Headphone Jack", NULL, "HPROUT"},
+
+ {"Ext Spk", NULL, "LLOUT"},
+ {"Ext Spk", NULL, "RLOUT"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *jack_function[] = {"Off", "Headphone"};
+static const struct soc_enum n810_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(3, jack_function),
+};
+
+static const struct snd_kcontrol_new aic33_n810_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", n810_enum[0],
+ n810_get_spk, n810_set_spk),
+ SOC_ENUM_EXT("Jack Function", n810_enum[1],
+ n810_get_jack, n810_set_jack),
+};
+
+static int n810_aic33_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ /* Not connected */
+ snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
+ snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
+ snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+
+ /* Add N810 specific controls */
+ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&aic33_n810_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add N810 specific widgets */
+ for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]);
+
+ /* Set up N810 specific audio path audio_map */
+ for (i = 0; i < ARRAY_SIZE(audio_map); 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;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link n810_dai = {
+ .name = "TLV320AIC33",
+ .stream_name = "AIC33",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &aic3x_dai,
+ .init = n810_aic33_init,
+ .ops = &n810_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_n810 = {
+ .name = "N810",
+ .dai_link = &n810_dai,
+ .num_links = 1,
+};
+
+/* Audio private data */
+static struct aic3x_setup_data n810_aic33_setup = {
+ .i2c_address = 0x18,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device n810_snd_devdata = {
+ .machine = &snd_soc_machine_n810,
+ .platform = &omap_soc_platform,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &n810_aic33_setup,
+};
+
+static struct platform_device *n810_snd_device;
+
+static int __init n810_soc_init(void)
+{
+ int err;
+ struct device *dev;
+
+ if (!machine_is_nokia_n810())
+ return -ENODEV;
+
+ n810_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!n810_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(n810_snd_device, &n810_snd_devdata);
+ n810_snd_devdata.dev = &n810_snd_device->dev;
+ *(unsigned int *)n810_dai.cpu_dai->private_data = 1; /* McBSP2 */
+ err = platform_device_add(n810_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &n810_snd_device->dev;
+
+ sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
+ if (IS_ERR(sys_clkout2_src)) {
+ dev_err(dev, "Could not get sys_clkout2_src clock\n");
+ return -ENODEV;
+ }
+ sys_clkout2 = clk_get(dev, "sys_clkout2");
+ if (IS_ERR(sys_clkout2)) {
+ dev_err(dev, "Could not get sys_clkout2\n");
+ goto err1;
+ }
+ /*
+ * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
+ * 96 MHz as its parent in order to get 12 MHz
+ */
+ func96m_clk = clk_get(dev, "func_96m_ck");
+ if (IS_ERR(func96m_clk)) {
+ dev_err(dev, "Could not get func 96M clock\n");
+ goto err2;
+ }
+ clk_set_parent(sys_clkout2_src, func96m_clk);
+ clk_set_rate(sys_clkout2, 12000000);
+
+ if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0)
+ BUG();
+ if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0)
+ BUG();
+ omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0);
+ omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0);
+
+ return 0;
+err2:
+ clk_put(sys_clkout2);
+ platform_device_del(n810_snd_device);
+err1:
+ platform_device_put(n810_snd_device);
+
+ return err;
+
+}
+
+static void __exit n810_soc_exit(void)
+{
+ platform_device_unregister(n810_snd_device);
+}
+
+module_init(n810_soc_init);
+module_exit(n810_soc_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("ALSA SoC Nokia N810");
+MODULE_LICENSE("GPL");