Initial import

1 files changed, 407 insertions, 0 deletions

diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
new file mode 100644
index 000000000..6464caf72
--- /dev/null
+++ b/sound/soc/codecs/stac9766.c
@@ -0,0 +1,407 @@
+/*
+ * stac9766.c  --  ALSA SoC STAC9766 codec support
+ *
+ * Copyright 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ *  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.
+ *
+ *  Features:-
+ *
+ *   o Support for AC97 Codec, S/PDIF
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "stac9766.h"
+
+/*
+ * STAC9766 register cache
+ */
+static const u16 stac9766_reg[] = {
+	0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
+	0x0000, 0x0000, 0x8008, 0x8008, /* e */
+	0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
+	0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+	0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+	0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
+	0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
+	0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+	0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
+	0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+	0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
+};
+
+static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
+			"Line", "Stereo Mix", "Mono Mix", "Phone"};
+static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
+static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
+static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
+static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
+static const char *stac9766_record_all_mux[] = {"All analog",
+	"Analog plus DAC"};
+static const char *stac9766_boost1[] = {"0dB", "10dB"};
+static const char *stac9766_boost2[] = {"0dB", "20dB"};
+static const char *stac9766_stereo_mic[] = {"Off", "On"};
+
+static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum,
+			    AC97_REC_SEL, 8, 0, stac9766_record_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum,
+			    AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum,
+			    AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum,
+			    AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum,
+			    AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum,
+			    AC97_STAC_ANALOG_SPECIAL, 12,
+			    stac9766_record_all_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum,
+			    AC97_MIC, 6, stac9766_boost1); /* 0/10dB */
+static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum,
+			    AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */
+static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum,
+			    AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic);
+
+static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
+static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
+static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
+static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
+
+static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
+	SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
+	SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
+		       master_tlv),
+	SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
+	SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
+		       master_tlv),
+	SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+	SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
+	SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
+
+
+	SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
+	SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
+	SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
+	SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
+
+	SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
+	SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
+	SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+	SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
+
+	SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
+	SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
+	SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
+	SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
+
+	SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
+	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
+	SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
+	SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+
+	SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
+	SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
+	SOC_ENUM("Record All Mux", stac9766_record_all_enum),
+	SOC_ENUM("Record Mux", stac9766_record_enum),
+	SOC_ENUM("Mono Mux", stac9766_mono_enum),
+	SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
+};
+
+static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+			       unsigned int val)
+{
+	struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+	u16 *cache = codec->reg_cache;
+
+	if (reg > AC97_STAC_PAGE0) {
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+		soc_ac97_ops->write(ac97, reg, val);
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+		return 0;
+	}
+	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
+		return -EIO;
+
+	soc_ac97_ops->write(ac97, reg, val);
+	cache[reg / 2] = val;
+	return 0;
+}
+
+static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
+				       unsigned int reg)
+{
+	struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+	u16 val = 0, *cache = codec->reg_cache;
+
+	if (reg > AC97_STAC_PAGE0) {
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+		val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0);
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+		return val;
+	}
+	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
+		return -EIO;
+
+	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+		reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
+		reg == AC97_VENDOR_ID2) {
+
+		val = soc_ac97_ops->read(ac97, reg);
+		return val;
+	}
+	return cache[reg / 2];
+}
+
+static int ac97_analog_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned short reg, vra;
+
+	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+
+	vra |= 0x1; /* enable variable rate audio */
+	vra &= ~0x4; /* disable SPDIF output */
+
+	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = AC97_PCM_FRONT_DAC_RATE;
+	else
+		reg = AC97_PCM_LR_ADC_RATE;
+
+	return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned short reg, vra;
+
+	stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
+
+	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+	vra |= 0x5; /* Enable VRA and SPDIF out */
+
+	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+	reg = AC97_PCM_FRONT_DAC_RATE;
+
+	return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int stac9766_set_bias_level(struct snd_soc_codec *codec,
+				   enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON: /* full On */
+	case SND_SOC_BIAS_PREPARE: /* partial On */
+	case SND_SOC_BIAS_STANDBY: /* Off, with power */
+		stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
+		break;
+	case SND_SOC_BIAS_OFF: /* Off, without power */
+		/* disable everything including AC link */
+		stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
+{
+	struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+	if (try_warm && soc_ac97_ops->warm_reset) {
+		soc_ac97_ops->warm_reset(ac97);
+		if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
+			return 1;
+	}
+
+	soc_ac97_ops->reset(ac97);
+	if (soc_ac97_ops->warm_reset)
+		soc_ac97_ops->warm_reset(ac97);
+	if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
+		return -EIO;
+	return 0;
+}
+
+static int stac9766_codec_resume(struct snd_soc_codec *codec)
+{
+	struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+	u16 id, reset;
+
+	reset = 0;
+	/* give the codec an AC97 warm reset to start the link */
+reset:
+	if (reset > 5) {
+		dev_err(codec->dev, "Failed to resume\n");
+		return -EIO;
+	}
+	ac97->bus->ops->warm_reset(ac97);
+	id = soc_ac97_ops->read(ac97, AC97_VENDOR_ID2);
+	if (id != 0x4c13) {
+		stac9766_reset(codec, 0);
+		reset++;
+		goto reset;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops stac9766_dai_ops_analog = {
+	.prepare = ac97_analog_prepare,
+};
+
+static const struct snd_soc_dai_ops stac9766_dai_ops_digital = {
+	.prepare = ac97_digital_prepare,
+};
+
+static struct snd_soc_dai_driver stac9766_dai[] = {
+{
+	.name = "stac9766-hifi-analog",
+
+	/* stream cababilities */
+	.playback = {
+		.stream_name = "stac9766 analog",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SND_SOC_STD_AC97_FMTS,
+	},
+	.capture = {
+		.stream_name = "stac9766 analog",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SND_SOC_STD_AC97_FMTS,
+	},
+	/* alsa ops */
+	.ops = &stac9766_dai_ops_analog,
+},
+{
+	.name = "stac9766-hifi-IEC958",
+
+	/* stream cababilities */
+	.playback = {
+		.stream_name = "stac9766 IEC958",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_32000 | \
+			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
+	},
+	/* alsa ops */
+	.ops = &stac9766_dai_ops_digital,
+}
+};
+
+static int stac9766_codec_probe(struct snd_soc_codec *codec)
+{
+	struct snd_ac97 *ac97;
+	int ret = 0;
+
+	ac97 = snd_soc_new_ac97_codec(codec);
+	if (IS_ERR(ac97))
+		return PTR_ERR(ac97);
+
+	snd_soc_codec_set_drvdata(codec, ac97);
+
+	/* do a cold reset for the controller and then try
+	 * a warm reset followed by an optional cold reset for codec */
+	stac9766_reset(codec, 0);
+	ret = stac9766_reset(codec, 1);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to reset: AC97 link error\n");
+		goto codec_err;
+	}
+
+	return 0;
+
+codec_err:
+	snd_soc_free_ac97_codec(ac97);
+	return ret;
+}
+
+static int stac9766_codec_remove(struct snd_soc_codec *codec)
+{
+	struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
+
+	snd_soc_free_ac97_codec(ac97);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
+	.controls = stac9766_snd_ac97_controls,
+	.num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls),
+	.write = stac9766_ac97_write,
+	.read = stac9766_ac97_read,
+	.set_bias_level = stac9766_set_bias_level,
+	.suspend_bias_off = true,
+	.probe = stac9766_codec_probe,
+	.remove = stac9766_codec_remove,
+	.resume = stac9766_codec_resume,
+	.reg_cache_size = ARRAY_SIZE(stac9766_reg),
+	.reg_word_size = sizeof(u16),
+	.reg_cache_step = 2,
+	.reg_cache_default = stac9766_reg,
+};
+
+static int stac9766_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev,
+			&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
+}
+
+static int stac9766_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver stac9766_codec_driver = {
+	.driver = {
+			.name = "stac9766-codec",
+	},
+
+	.probe = stac9766_probe,
+	.remove = stac9766_remove,
+};
+
+module_platform_driver(stac9766_codec_driver);
+
+MODULE_DESCRIPTION("ASoC stac9766 driver");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_LICENSE("GPL");