[PATCH 6/8] ASoC pxa2xx Tosa machine support

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This patch adds Alsa audio support to the Sharp Zaurus SL-C6000 (Tosa)
machine. 


Signed-off-by: Dirk Opfer <Dirk@xxxxxxxxxxxxxxx>
Signed-off-by: Richard Purdie <rpurdie@xxxxxxxxx>
Signed-off-by: Liam Girdwood <liam.girdwood@xxxxxxxxxxxxxxxx>
Index: alsa-kernel/soc/pxa/tosa.c
===================================================================
--- /dev/null
+++ alsa-kernel/soc/pxa/tosa.c
@@ -0,0 +1,279 @@
+/*
+ * tosa.c  --  SoC audio for Tosa
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@xxxxxxxxxxxxxxxx>
+ *          Richard Purdie <richard@xxxxxxxxxxxxxx>
+ *
+ *  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
+ *    30th Nov 2005   Initial version.
+ *
+ * GPIO's
+ *  1 - Jack Insertion
+ *  5 - Hookswitch (headset answer/hang up switch)
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/driver.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/hardware/tmio.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/audio.h>
+#include <asm/arch/tosa.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+
+static struct snd_soc_machine tosa;
+
+#define TOSA_HP        0
+#define TOSA_MIC_INT   1
+#define TOSA_HEADSET   2
+#define TOSA_HP_OFF    3
+#define TOSA_SPK_ON    0
+#define TOSA_SPK_OFF   1
+
+static int tosa_jack_func;
+static int tosa_spk_func;
+
+static void tosa_ext_control(struct snd_soc_codec *codec)
+{
+	int spk = 0, mic_int = 0, hp = 0, hs = 0;
+
+	/* set up jack connection */
+	switch (tosa_jack_func) {
+	case TOSA_HP:
+		hp = 1;
+		break;
+	case TOSA_MIC_INT:
+		mic_int = 1;
+		break;
+	case TOSA_HEADSET:
+		hs = 1;
+		break;
+	}
+
+	if (tosa_spk_func == TOSA_SPK_ON)
+		spk = 1;
+
+	snd_soc_dapm_set_endpoint(codec, "Speaker", spk);
+	snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int);
+	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
+	snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int tosa_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	tosa_ext_control(codec);
+	return 0;
+}
+
+static struct snd_soc_ops tosa_ops = {
+	.startup = tosa_startup,
+};
+
+static int tosa_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = tosa_jack_func;
+	return 0;
+}
+
+static int tosa_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	tosa_jack_func = ucontrol->value.integer.value[0];
+	tosa_ext_control(codec);
+	return 0;
+}
+
+static int tosa_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = tosa_spk_func;
+	return 0;
+}
+
+static int tosa_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	tosa_spk_func = ucontrol->value.integer.value[0];
+	tosa_ext_control(codec);
+	return 0;
+}
+
+/* tosa dapm event handlers */
+static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
+	else
+		reset_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
+	return 0;
+}
+
+/* tosa machine dapm widgets */
+static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* tosa audio map */
+static const char *audio_map[][3] = {
+
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to LOUT2, ROUT2 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+
+	/* internal mic is connected to mic1, mic2 differential - with bias */
+	{"MIC1", NULL, "Mic Bias"},
+	{"MIC2", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic (Internal)"},
+
+	/* headset is connected to HPOUTR, and LINEINR with bias */
+	{"Headset Jack", NULL, "HPOUTR"},
+	{"LINEINR", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Headset Jack"},
+
+	{NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+	"Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum tosa_enum[] = {
+	SOC_ENUM_SINGLE_EXT(5, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new tosa_controls[] = {
+	SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
+		tosa_set_jack),
+	SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
+		tosa_set_spk),
+};
+
+static int tosa_ac97_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
+	snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0);
+
+	/* add tosa specific controls */
+	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&tosa_controls[i],codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* add tosa specific widgets */
+	for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]);
+	}
+
+	/* set up tosa specific audio path audio_mapnects */
+	for (i = 0; audio_map[i][0] != NULL; 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;
+}
+
+static struct snd_soc_dai_link tosa_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.init = tosa_ac97_init,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+},
+};
+
+static struct snd_soc_machine tosa = {
+	.name = "Tosa",
+	.dai_link = tosa_dai,
+	.num_links = ARRAY_SIZE(tosa_dai),
+	.ops = &tosa_ops,
+};
+
+static struct snd_soc_device tosa_snd_devdata = {
+	.machine = &tosa,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *tosa_snd_device;
+
+static int __init tosa_init(void)
+{
+	int ret;
+
+	if (!machine_is_tosa())
+		return -ENODEV;
+
+	tosa_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!tosa_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata);
+	tosa_snd_devdata.dev = &tosa_snd_device->dev;
+	ret = platform_device_add(tosa_snd_device);
+
+	if (ret)
+		platform_device_put(tosa_snd_device);
+
+	return ret;
+}
+
+static void __exit tosa_exit(void)
+{
+	platform_device_unregister(tosa_snd_device);
+}
+
+module_init(tosa_init);
+module_exit(tosa_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Tosa");
+MODULE_LICENSE("GPL");
-------------------------------------------------------------------------
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