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Diffstat (limited to 'sound/soc/codecs/stac9766.c')
-rw-r--r--sound/soc/codecs/stac9766.c407
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");