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authorRichard Purdie <rpurdie@rpsys.net>2006-10-06 18:37:32 +0200
committerJaroslav Kysela <perex@suse.cz>2007-02-09 09:00:23 +0100
commit10c5cf30446fe91b7173436b75c4f00dfb4cd9f8 (patch)
treeca3dcc7fc72a1a635bf6f786dc5c5c005cfc7bbb /sound
parentabadfc928a27e1cf27c834e8e29e6b1f64ca2d55 (diff)
[ALSA] ASoC codecs: WM9712 support
This patch adds ASoC support for the WM9712 codec. Supported features:- o Capture/Playback/Sidetone/Bypass. o Aux DAC. o 8k - 48k sample rates. o DAPM. Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/codecs/wm9712.c778
-rw-r--r--sound/soc/codecs/wm9712.h14
2 files changed, 792 insertions, 0 deletions
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
new file mode 100644
index 00000000000..4850550e2e3
--- /dev/null
+++ b/sound/soc/codecs/wm9712.c
@@ -0,0 +1,778 @@
+/*
+ * wm9712.c -- ALSA Soc WM9712 codec support
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.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.
+ *
+ * Revision history
+ * 4th Feb 2006 Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#define WM9712_VERSION "0.4"
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val);
+
+#define AC97_DIR \
+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AC97_RATES \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+/* may need to expand this */
+static struct snd_soc_dai_mode ac97_modes[] = {
+ {0, 0, SNDRV_PCM_FMTBIT_S16_LE, AC97_RATES},
+ {0, 0, SNDRV_PCM_FMTBIT_S18_3LE, AC97_RATES},
+};
+
+/*
+ * WM9712 register cache
+ */
+static const u16 wm9712_reg[] = {
+ 0x6174, 0x8000, 0x8000, 0x8000, // 6
+ 0xf0f0, 0xaaa0, 0xc008, 0x6808, // e
+ 0xe808, 0xaaa0, 0xad00, 0x8000, // 16
+ 0xe808, 0x3000, 0x8000, 0x0000, // 1e
+ 0x0000, 0x0000, 0x0000, 0x000f, // 26
+ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
+ 0x0000, 0xbb80, 0x0000, 0x0000, // 36
+ 0x0000, 0x2000, 0x0000, 0x0000, // 3e
+ 0x0000, 0x0000, 0x0000, 0x0000, // 46
+ 0x0000, 0x0000, 0xf83e, 0xffff, // 4e
+ 0x0000, 0x0000, 0x0000, 0xf83e, // 56
+ 0x0008, 0x0000, 0x0000, 0x0000, // 5e
+ 0xb032, 0x3e00, 0x0000, 0x0000, // 66
+ 0x0000, 0x0000, 0x0000, 0x0000, // 6e
+ 0x0000, 0x0000, 0x0000, 0x0006, // 76
+ 0x0001, 0x0000, 0x574d, 0x4c12, // 7e
+ 0x0000, 0x0000 // virtual hp mixers
+};
+
+/* virtual HP mixers regs */
+#define HPL_MIXER 0x80
+#define HPR_MIXER 0x82
+
+static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
+static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right",
+ "Mono"};
+static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"};
+static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
+static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2",
+ "Stereo"};
+static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer",
+ "Line", "Headphone Mixer", "Phone Mixer", "Phone"};
+static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"};
+static const char *wm9712_diff_sel[] = {"Mic", "Line"};
+
+static const struct soc_enum wm9712_enum[] = {
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select),
+SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux),
+SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src),
+SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc),
+SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base),
+SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain),
+SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type),
+SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel),
+};
+
+static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1),
+
+SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
+SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0),
+SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0),
+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0),
+
+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+SOC_ENUM("ALC Function", wm9712_enum[0]),
+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+SOC_ENUM("ALC NG Type", wm9712_enum[10]),
+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
+
+SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1),
+SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
+
+SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
+SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1),
+SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
+
+SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
+SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1),
+SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1),
+
+SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1),
+SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1),
+SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1),
+
+SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 0),
+SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
+
+SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
+SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
+
+SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
+SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
+SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0),
+
+SOC_ENUM("Bass Control", wm9712_enum[5]),
+SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
+SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
+SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0),
+SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0),
+
+SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
+
+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
+};
+
+/* add non dapm controls */
+static int wm9712_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* We have to create a fake left and right HP mixers because
+ * the codec only has a single control that is shared by both channels.
+ * This makes it impossible to determine the audio path.
+ */
+static int mixer_event (struct snd_soc_dapm_widget *w, int event)
+{
+ u16 l, r, beep, line, phone, mic, pcm, aux;
+
+ l = ac97_read(w->codec, HPL_MIXER);
+ r = ac97_read(w->codec, HPR_MIXER);
+ beep = ac97_read(w->codec, AC97_PC_BEEP);
+ mic = ac97_read(w->codec, AC97_VIDEO);
+ phone = ac97_read(w->codec, AC97_PHONE);
+ line = ac97_read(w->codec, AC97_LINE);
+ pcm = ac97_read(w->codec, AC97_PCM);
+ aux = ac97_read(w->codec, AC97_CD);
+
+ if (l & 0x1 || r & 0x1)
+ ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
+
+ if (l & 0x2 || r & 0x2)
+ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+
+ if (l & 0x4 || r & 0x4)
+ ac97_write(w->codec, AC97_LINE, line & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_LINE, line | 0x8000);
+
+ if (l & 0x8 || r & 0x8)
+ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+
+ if (l & 0x10 || r & 0x10)
+ ac97_write(w->codec, AC97_CD, aux & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_CD, aux | 0x8000);
+
+ if (l & 0x20 || r & 0x20)
+ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+
+ return 0;
+}
+
+/* Left Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0),
+};
+
+/* Right Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0),
+};
+
+/* Speaker Mixer */
+static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1),
+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1),
+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1),
+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1),
+};
+
+/* Phone Mixer */
+static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1),
+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1),
+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1),
+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1),
+ SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1),
+ SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1),
+};
+
+/* ALC headphone mux */
+static const struct snd_kcontrol_new wm9712_alc_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[1]);
+
+/* out 3 mux */
+static const struct snd_kcontrol_new wm9712_out3_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[2]);
+
+/* spk mux */
+static const struct snd_kcontrol_new wm9712_spk_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[3]);
+
+/* Capture to Phone mux */
+static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[4]);
+
+/* Capture left select */
+static const struct snd_kcontrol_new wm9712_capture_selectl_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[8]);
+
+/* Capture right select */
+static const struct snd_kcontrol_new wm9712_capture_selectr_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[9]);
+
+/* Mic select */
+static const struct snd_kcontrol_new wm9712_mic_src_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+
+/* diff select */
+static const struct snd_kcontrol_new wm9712_diff_sel_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[11]);
+
+static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = {
+SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_alc_mux_controls),
+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_out3_mux_controls),
+SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_spk_mux_controls),
+SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_phone_mux_controls),
+SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_selectl_controls),
+SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_selectr_controls),
+SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_diff_sel_controls),
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
+ &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
+ &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
+ &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
+ &wm9712_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm9712_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1),
+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1),
+SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("LOUT2"),
+SND_SOC_DAPM_OUTPUT("ROUT2"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_INPUT("LINEINL"),
+SND_SOC_DAPM_INPUT("LINEINR"),
+SND_SOC_DAPM_INPUT("PHONE"),
+SND_SOC_DAPM_INPUT("PCBEEP"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+static const char *audio_map[][3] = {
+ /* virtual mixer - mixes left & right channels for spk and mono */
+ {"AC97 Mixer", NULL, "Left DAC"},
+ {"AC97 Mixer", NULL, "Right DAC"},
+
+ /* Left HP mixer */
+ {"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Left HP Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
+ {"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
+ {"Left HP Mixer", NULL, "ALC Sidetone Mux"},
+ //{"Right HP Mixer", NULL, "HP Mixer"},
+
+ /* Right HP mixer */
+ {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Right HP Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
+ {"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
+ {"Right HP Mixer", NULL, "ALC Sidetone Mux"},
+
+ /* speaker mixer */
+ {"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Speaker Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
+
+ /* Phone mixer */
+ {"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Phone Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Phone Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"},
+ {"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"},
+
+ /* inputs */
+ {"Line PGA", NULL, "LINEINL"},
+ {"Line PGA", NULL, "LINEINR"},
+ {"Phone PGA", NULL, "PHONE"},
+ {"Mic PGA", NULL, "MIC1"},
+ {"Mic PGA", NULL, "MIC2"},
+
+ /* left capture selector */
+ {"Left Capture Select", "Mic", "MIC1"},
+ {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
+ {"Left Capture Select", "Line", "LINEINL"},
+ {"Left Capture Select", "Headphone Mixer", "Left HP Mixer"},
+ {"Left Capture Select", "Phone Mixer", "Phone Mixer"},
+ {"Left Capture Select", "Phone", "PHONE"},
+
+ /* right capture selector */
+ {"Right Capture Select", "Mic", "MIC2"},
+ {"Right Capture Select", "Speaker Mixer", "Speaker Mixer"},
+ {"Right Capture Select", "Line", "LINEINR"},
+ {"Right Capture Select", "Headphone Mixer", "Right HP Mixer"},
+ {"Right Capture Select", "Phone Mixer", "Phone Mixer"},
+ {"Right Capture Select", "Phone", "PHONE"},
+
+ /* ALC Sidetone */
+ {"ALC Sidetone Mux", "Stereo", "Left Capture Select"},
+ {"ALC Sidetone Mux", "Stereo", "Right Capture Select"},
+ {"ALC Sidetone Mux", "Left", "Left Capture Select"},
+ {"ALC Sidetone Mux", "Right", "Right Capture Select"},
+
+ /* ADC's */
+ {"Left ADC", NULL, "Left Capture Select"},
+ {"Right ADC", NULL, "Right Capture Select"},
+
+ /* outputs */
+ {"MONOOUT", NULL, "Phone Mixer"},
+ {"HPOUTL", NULL, "Headphone PGA"},
+ {"Headphone PGA", NULL, "Left HP Mixer"},
+ {"HPOUTR", NULL, "Headphone PGA"},
+ {"Headphone PGA", NULL, "Right HP Mixer"},
+
+ /* mono hp mixer */
+ {"Mono HP Mixer", NULL, "Left HP Mixer"},
+ {"Mono HP Mixer", NULL, "Right HP Mixer"},
+
+ /* Out3 Mux */
+ {"Out3 Mux", "Left", "Left HP Mixer"},
+ {"Out3 Mux", "Mono", "Phone Mixer"},
+ {"Out3 Mux", "Left + Right", "Mono HP Mixer"},
+ {"Out 3 PGA", NULL, "Out3 Mux"},
+ {"OUT3", NULL, "Out 3 PGA"},
+
+ /* speaker Mux */
+ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
+ {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"},
+ {"Speaker PGA", NULL, "Speaker Mux"},
+ {"LOUT2", NULL, "Speaker PGA"},
+ {"ROUT2", NULL, "Speaker PGA"},
+
+ {NULL, NULL, NULL},
+};
+
+static int wm9712_add_widgets(struct snd_soc_codec *codec)
+{
+ int i;
+
+ for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
+ }
+
+ /* set up 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_new_widgets(codec);
+ return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
+ reg == AC97_REC_GAIN)
+ return soc_ac97_ops.read(codec->ac97, reg);
+ else {
+ reg = reg >> 1;
+
+ if (reg > (ARRAY_SIZE(wm9712_reg)))
+ return -EIO;
+
+ return cache[reg];
+ }
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ reg = reg >> 1;
+ if (reg <= (ARRAY_SIZE(wm9712_reg)))
+ cache[reg] = val;
+
+ return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg;
+ u16 vra;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 vra, xsle;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+ xsle = ac97_read(codec, AC97_PCI_SID);
+ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+
+ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
+}
+
+struct snd_soc_codec_dai wm9712_dai[] = {
+{
+ .name = "AC97 HiFi",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,},
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,},
+ .ops = {
+ .prepare = ac97_prepare,},
+ .caps = {
+ .num_modes = ARRAY_SIZE(ac97_modes),
+ .mode = ac97_modes,},
+ },
+ {
+ .name = "AC97 Aux",
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,},
+ .ops = {
+ .prepare = ac97_aux_prepare,},
+ .caps = {
+ .num_modes = ARRAY_SIZE(ac97_modes),
+ .mode = ac97_modes,},
+ },
+};
+EXPORT_SYMBOL_GPL(wm9712_dai);
+
+static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
+{
+ u16 reg;
+
+ switch (event) {
+ case SNDRV_CTL_POWER_D0: /* full On */
+ /* liam - maybe enable thermal shutdown */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ break;
+ case SNDRV_CTL_POWER_D1: /* partial On */
+ case SNDRV_CTL_POWER_D2: /* partial On */
+ break;
+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ /* enable master bias and vmid */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ /* disable everything including AC link */
+ ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+ ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->dapm_state = event;
+ return 0;
+}
+
+static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (!(ac97_read(codec, 0) & 0x8000))
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (ac97_read(codec, 0) & 0x8000)
+ goto err;
+ return 0;
+
+err:
+ printk(KERN_ERR "WM9712 AC97 reset failed\n");
+ return -EIO;
+}
+
+static int wm9712_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+ return 0;
+}
+
+static int wm9712_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9712_reset(codec, 1);
+ if (ret < 0){
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+ if (ret == 0) {
+ /* Sync reg_cache with the hardware after cold reset */
+ for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) {
+ if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
+ (i > 0x58 && i != 0x5c))
+ continue;
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+ }
+
+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0);
+
+ return ret;
+}
+
+static int wm9712_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache =
+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9712_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ kfree(codec->ac97);
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return -ENOMEM;
+ }
+ memcpy(codec->reg_cache, wm9712_reg, sizeof(u16) * ARRAY_SIZE(wm9712_reg));
+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9712_reg);
+ codec->reg_cache_step = 2;
+
+ codec->name = "WM9712";
+ codec->owner = THIS_MODULE;
+ codec->dai = wm9712_dai;
+ codec->num_dai = ARRAY_SIZE(wm9712_dai);
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ codec->dapm_event = wm9712_dapm_event;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0)
+ goto err;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ ret = wm9712_reset(codec, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "AC97 link error\n");
+ goto reset_err;
+ }
+
+ /* set alc mux to none */
+ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
+
+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+ wm9712_add_controls(codec);
+ wm9712_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0)
+ goto reset_err;
+
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+
+err:
+ kfree(socdev->codec->reg_cache);
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int wm9712_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_dapm_free(socdev);
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9712 = {
+ .probe = wm9712_soc_probe,
+ .remove = wm9712_soc_remove,
+ .suspend = wm9712_soc_suspend,
+ .resume = wm9712_soc_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
+
+MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
new file mode 100644
index 00000000000..719105d61e6
--- /dev/null
+++ b/sound/soc/codecs/wm9712.h
@@ -0,0 +1,14 @@
+/*
+ * wm9712.h -- WM9712 Soc Audio driver
+ */
+
+#ifndef _WM9712_H
+#define _WM9712_H
+
+#define WM9712_DAI_AC97_HIFI 0
+#define WM9712_DAI_AC97_AUX 1
+
+extern struct snd_soc_codec_dai wm9712_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9712;
+
+#endif