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-rw-r--r--sound/soc/codecs/Kconfig3
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/uda1380.c852
-rw-r--r--sound/soc/codecs/uda1380.h89
4 files changed, 946 insertions, 0 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d4a5fe42f6e..7beefccfa82 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC
tristate
select SND_AC97_CODEC
+config SND_SOC_UDA1380
+ tristate
+
config SND_SOC_WM8731
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4e1314c9d3e..d5926a11707 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,4 +1,5 @@
snd-soc-ac97-objs := ac97.o
+snd-soc-uda1380-objs := uda1380.o
snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
@@ -8,6 +9,7 @@ snd-soc-cs4270-objs := cs4270.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
new file mode 100644
index 00000000000..cb50486201f
--- /dev/null
+++ b/sound/soc/codecs/uda1380.c
@@ -0,0 +1,852 @@
+/*
+ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
+ *
+ * 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.
+ *
+ * Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com>
+ * Improved support for DAPM and audio routing/mixing capabilities,
+ * added TLV support.
+ *
+ * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC
+ * codec model.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ * Copyright 2005 Openedhand Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/types.h>
+#include <linux/string.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/ioctl.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "uda1380.h"
+
+#define UDA1380_VERSION "0.6"
+#define AUDIO_NAME "uda1380"
+
+/*
+ * uda1380 register cache
+ */
+static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
+ 0x0502, 0x0000, 0x0000, 0x3f3f,
+ 0x0202, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0xff00, 0x0000, 0x4800,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x8000, 0x0002, 0x0000,
+};
+
+/*
+ * read uda1380 register cache
+ */
+static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == UDA1380_RESET)
+ return 0;
+ if (reg >= UDA1380_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write uda1380 register cache
+ */
+static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= UDA1380_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the UDA1380 register space
+ */
+static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+
+ /* data is
+ * data[0] is register offset
+ * data[1] is MS byte
+ * data[2] is LS byte
+ */
+ data[0] = reg;
+ data[1] = (value & 0xff00) >> 8;
+ data[2] = value & 0x00ff;
+
+ uda1380_write_reg_cache(codec, reg, value);
+
+ /* the interpolator & decimator regs must only be written when the
+ * codec DAI is active.
+ */
+ if (!codec->active && (reg >= UDA1380_MVOL))
+ return 0;
+ pr_debug("uda1380: hw write %x val %x\n", reg, value);
+ if (codec->hw_write(codec->control_data, data, 3) == 3) {
+ unsigned int val;
+ i2c_master_send(codec->control_data, data, 1);
+ i2c_master_recv(codec->control_data, data, 2);
+ val = (data[0]<<8) | data[1];
+ if (val != value) {
+ pr_debug("uda1380: READ BACK VAL %x\n",
+ (data[0]<<8) | data[1]);
+ return -EIO;
+ }
+ return 0;
+ } else
+ return -EIO;
+}
+
+#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
+
+/* declarations of ALSA reg_elem_REAL controls */
+static const char *uda1380_deemp[] = {
+ "None",
+ "32kHz",
+ "44.1kHz",
+ "48kHz",
+ "96kHz",
+};
+static const char *uda1380_input_sel[] = {
+ "Line",
+ "Mic + Line R",
+ "Line L",
+ "Mic",
+};
+static const char *uda1380_output_sel[] = {
+ "DAC",
+ "Analog Mixer",
+};
+static const char *uda1380_spf_mode[] = {
+ "Flat",
+ "Minimum1",
+ "Minimum2",
+ "Maximum"
+};
+static const char *uda1380_capture_sel[] = {
+ "ADC",
+ "Digital Mixer"
+};
+static const char *uda1380_sel_ns[] = {
+ "3rd-order",
+ "5th-order"
+};
+static const char *uda1380_mix_control[] = {
+ "off",
+ "PCM only",
+ "before sound processing",
+ "after sound processing"
+};
+static const char *uda1380_sdet_setting[] = {
+ "3200",
+ "4800",
+ "9600",
+ "19200"
+};
+static const char *uda1380_os_setting[] = {
+ "single-speed",
+ "double-speed (no mixing)",
+ "quad-speed (no mixing)"
+};
+
+static const struct soc_enum uda1380_deemp_enum[] = {
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
+};
+static const struct soc_enum uda1380_input_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
+static const struct soc_enum uda1380_output_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */
+static const struct soc_enum uda1380_spf_enum =
+ SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */
+static const struct soc_enum uda1380_capture_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */
+static const struct soc_enum uda1380_sel_ns_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */
+static const struct soc_enum uda1380_mix_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */
+static const struct soc_enum uda1380_sdet_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */
+static const struct soc_enum uda1380_os_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */
+
+/*
+ * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
+ */
+static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1);
+
+/*
+ * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored),
+ * from -66 dB in 0.5 dB steps (2 dB steps, really) and
+ * from -52 dB in 0.25 dB steps
+ */
+static const unsigned int mvol_tlv[] = {
+ TLV_DB_RANGE_HEAD(3),
+ 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1),
+ 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0),
+ 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0),
+};
+
+/*
+ * from -72 dB in 1.5 dB steps (6 dB steps really),
+ * from -66 dB in 0.75 dB steps (3 dB steps really),
+ * from -60 dB in 0.5 dB steps (2 dB steps really) and
+ * from -46 dB in 0.25 dB steps
+ */
+static const unsigned int vc_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0),
+ 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0),
+ 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0),
+};
+
+/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */
+static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0);
+
+/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts
+ * off at 18 dB max) */
+static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0);
+
+/* from -63 to 24 dB in 0.5 dB steps (-128...48) */
+static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1);
+
+/* from 0 to 24 dB in 3 dB steps */
+static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
+
+/* from 0 to 30 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0);
+
+static const struct snd_kcontrol_new uda1380_snd_controls[] = {
+ SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */
+ SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */
+ SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */
+ SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */
+ SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */
+ SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */
+ SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */
+/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */
+ SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */
+ SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */
+ SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */
+ SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */
+ SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */
+ SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */
+ SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */
+ SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */
+ SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */
+ SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */
+ SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */
+ SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */
+/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */
+ SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */
+ SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */
+ SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */
+ SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */
+ SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */
+ SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */
+ SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */
+ /* -5.5, -8, -11.5, -14 dBFS */
+ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
+};
+
+/* add non dapm controls */
+static int uda1380_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Input mux */
+static const struct snd_kcontrol_new uda1380_input_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
+
+/* Output mux */
+static const struct snd_kcontrol_new uda1380_output_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_output_sel_enum);
+
+/* Capture mux */
+static const struct snd_kcontrol_new uda1380_capture_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum);
+
+
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_input_mux_control),
+ SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_output_mux_control),
+ SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_capture_mux_control),
+ SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
+ SND_SOC_DAPM_INPUT("VINM"),
+ SND_SOC_DAPM_INPUT("VINL"),
+ SND_SOC_DAPM_INPUT("VINR"),
+ SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("VOUTLHP"),
+ SND_SOC_DAPM_OUTPUT("VOUTRHP"),
+ SND_SOC_DAPM_OUTPUT("VOUTL"),
+ SND_SOC_DAPM_OUTPUT("VOUTR"),
+ SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
+ SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* output mux */
+ {"HeadPhone Driver", NULL, "Output Mux"},
+ {"VOUTR", NULL, "Output Mux"},
+ {"VOUTL", NULL, "Output Mux"},
+
+ {"Analog Mixer", NULL, "VINR"},
+ {"Analog Mixer", NULL, "VINL"},
+ {"Analog Mixer", NULL, "DAC"},
+
+ {"Output Mux", "DAC", "DAC"},
+ {"Output Mux", "Analog Mixer", "Analog Mixer"},
+
+ /* {"DAC", "Digital Mixer", "I2S" } */
+
+ /* headphone driver */
+ {"VOUTLHP", NULL, "HeadPhone Driver"},
+ {"VOUTRHP", NULL, "HeadPhone Driver"},
+
+ /* input mux */
+ {"Left ADC", NULL, "Input Mux"},
+ {"Input Mux", "Mic", "Mic LNA"},
+ {"Input Mux", "Mic + Line R", "Mic LNA"},
+ {"Input Mux", "Line L", "Left PGA"},
+ {"Input Mux", "Line", "Left PGA"},
+
+ /* right input */
+ {"Right ADC", "Mic + Line R", "Right PGA"},
+ {"Right ADC", "Line", "Right PGA"},
+
+ /* inputs */
+ {"Mic LNA", NULL, "VINM"},
+ {"Left PGA", NULL, "VINL"},
+ {"Right PGA", NULL, "VINR"},
+};
+
+static int uda1380_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int iface;
+
+ /* set up DAI based upon fmt */
+ iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+ iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
+
+ /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= R01_SFORI_I2S | R01_SFORO_I2S;
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+ }
+
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+ iface |= R01_SIM;
+
+ uda1380_write(codec, UDA1380_IFACE, iface);
+
+ return 0;
+}
+
+/*
+ * Flush reg cache
+ * We can only write the interpolator and decimator registers
+ * when the DAI is being clocked by the CPU DAI. It's up to the
+ * machine and cpu DAI driver to do this before we are called.
+ */
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ 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, reg_start, reg_end, clk;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ reg_start = UDA1380_MVOL;
+ reg_end = UDA1380_MIXER;
+ } else {
+ reg_start = UDA1380_DEC;
+ reg_end = UDA1380_AGC;
+ }
+
+ /* FIXME disable DAC_CLK */
+ clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+ uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
+
+ for (reg = reg_start; reg <= reg_end; reg++) {
+ pr_debug("uda1380: flush reg %x val %x:", reg,
+ uda1380_read_reg_cache(codec, reg));
+ uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+ }
+
+ /* FIXME enable DAC_CLK */
+ uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
+
+ return 0;
+}
+
+static int uda1380_pcm_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_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+ /* set WSPLL power and divider if running from this clock */
+ if (clk & R00_DAC_CLK) {
+ int rate = params_rate(params);
+ u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+ clk &= ~0x3; /* clear SEL_LOOP_DIV */
+ switch (rate) {
+ case 6250 ... 12500:
+ clk |= 0x0;
+ break;
+ case 12501 ... 25000:
+ clk |= 0x1;
+ break;
+ case 25001 ... 50000:
+ clk |= 0x2;
+ break;
+ case 50001 ... 100000:
+ clk |= 0x3;
+ break;
+ }
+ uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ clk |= R00_EN_DAC | R00_EN_INT;
+ else
+ clk |= R00_EN_ADC | R00_EN_DEC;
+
+ uda1380_write(codec, UDA1380_CLK, clk);
+ return 0;
+}
+
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+ /* shut down WSPLL power if running from this clock */
+ if (clk & R00_DAC_CLK) {
+ u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+ uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ clk &= ~(R00_EN_DAC | R00_EN_INT);
+ else
+ clk &= ~(R00_EN_ADC | R00_EN_DEC);
+
+ uda1380_write(codec, UDA1380_CLK, clk);
+}
+
+static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
+
+ /* FIXME: mute(codec,0) is called when the magician clock is already
+ * set to WSPLL, but for some unknown reason writing to interpolator
+ * registers works only when clocked by SYSCLK */
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+ uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
+ if (mute)
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
+ else
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg);
+ uda1380_write(codec, UDA1380_CLK, clk);
+ return 0;
+}
+
+static int uda1380_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
+ break;
+ case SND_SOC_BIAS_OFF:
+ uda1380_write(codec, UDA1380_PM, 0x0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_codec_dai uda1380_dai[] = {
+{
+ .name = "UDA1380",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .digital_mute = uda1380_mute,
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+{ /* playback only - dual interface */
+ .name = "UDA1380",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .digital_mute = uda1380_mute,
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+{ /* capture only - dual interface*/
+ .name = "UDA1380",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+};
+EXPORT_SYMBOL_GPL(uda1380_dai);
+
+static int uda1380_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;
+
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int uda1380_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda1380_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the UDA1380 driver
+ * register mixer and dsp interfaces with the kernel
+ */
+static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "UDA1380";
+ codec->owner = THIS_MODULE;
+ codec->read = uda1380_read_reg_cache;
+ codec->write = uda1380_write;
+ codec->set_bias_level = uda1380_set_bias_level;
+ codec->dai = uda1380_dai;
+ codec->num_dai = ARRAY_SIZE(uda1380_dai);
+ codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+ codec->reg_cache_size = sizeof(uda1380_reg);
+ codec->reg_cache_step = 2;
+ uda1380_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ pr_err("uda1380: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ /* set clock input */
+ switch (dac_clk) {
+ case UDA1380_DAC_CLK_SYSCLK:
+ uda1380_write(codec, UDA1380_CLK, 0);
+ break;
+ case UDA1380_DAC_CLK_WSPLL:
+ uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
+ break;
+ }
+
+ /* uda1380 init */
+ uda1380_add_controls(codec);
+ uda1380_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ pr_err("uda1380: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *uda1380_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver uda1380_i2c_driver;
+static struct i2c_client client_template;
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+
+static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = uda1380_socdev;
+ struct uda1380_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ pr_err("uda1380: failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = uda1380_init(socdev, setup->dac_clk);
+ if (ret < 0) {
+ pr_err("uda1380: failed to initialise UDA1380\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int uda1380_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int uda1380_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, uda1380_codec_probe);
+}
+
+static struct i2c_driver uda1380_i2c_driver = {
+ .driver = {
+ .name = "UDA1380 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_UDA1380,
+ .attach_adapter = uda1380_i2c_attach,
+ .detach_client = uda1380_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "UDA1380",
+ .driver = &uda1380_i2c_driver,
+};
+#endif
+
+static int uda1380_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct uda1380_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ uda1380_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&uda1380_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int uda1380_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&uda1380_i2c_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_uda1380 = {
+ .probe = uda1380_probe,
+ .remove = uda1380_remove,
+ .suspend = uda1380_suspend,
+ .resume = uda1380_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
+
+MODULE_AUTHOR("Giorgio Padrin");
+MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h
new file mode 100644
index 00000000000..f9d885c8bf0
--- /dev/null
+++ b/sound/soc/codecs/uda1380.h
@@ -0,0 +1,89 @@
+/*
+ * Audio support for Philips UDA1380
+ *
+ * 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.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ */
+
+#ifndef _UDA1380_H
+#define _UDA1380_H
+
+#define UDA1380_CLK 0x00
+#define UDA1380_IFACE 0x01
+#define UDA1380_PM 0x02
+#define UDA1380_AMIX 0x03
+#define UDA1380_HP 0x04
+#define UDA1380_MVOL 0x10
+#define UDA1380_MIXVOL 0x11
+#define UDA1380_MODE 0x12
+#define UDA1380_DEEMP 0x13
+#define UDA1380_MIXER 0x14
+#define UDA1380_INTSTAT 0x18
+#define UDA1380_DEC 0x20
+#define UDA1380_PGA 0x21
+#define UDA1380_ADC 0x22
+#define UDA1380_AGC 0x23
+#define UDA1380_DECSTAT 0x28
+#define UDA1380_RESET 0x7f
+
+#define UDA1380_CACHEREGNUM 0x24
+
+/* Register flags */
+#define R00_EN_ADC 0x0800
+#define R00_EN_DEC 0x0400
+#define R00_EN_DAC 0x0200
+#define R00_EN_INT 0x0100
+#define R00_DAC_CLK 0x0010
+#define R01_SFORI_I2S 0x0000
+#define R01_SFORI_LSB16 0x0100
+#define R01_SFORI_LSB18 0x0200
+#define R01_SFORI_LSB20 0x0300
+#define R01_SFORI_MSB 0x0500
+#define R01_SFORI_MASK 0x0700
+#define R01_SFORO_I2S 0x0000
+#define R01_SFORO_LSB16 0x0001
+#define R01_SFORO_LSB18 0x0002
+#define R01_SFORO_LSB20 0x0003
+#define R01_SFORO_LSB24 0x0004
+#define R01_SFORO_MSB 0x0005
+#define R01_SFORO_MASK 0x0007
+#define R01_SEL_SOURCE 0x0040
+#define R01_SIM 0x0010
+#define R02_PON_PLL 0x8000
+#define R02_PON_HP 0x2000
+#define R02_PON_DAC 0x0400
+#define R02_PON_BIAS 0x0100
+#define R02_EN_AVC 0x0080
+#define R02_PON_AVC 0x0040
+#define R02_PON_LNA 0x0010
+#define R02_PON_PGAL 0x0008
+#define R02_PON_ADCL 0x0004
+#define R02_PON_PGAR 0x0002
+#define R02_PON_ADCR 0x0001
+#define R13_MTM 0x4000
+#define R14_SILENCE 0x0080
+#define R14_SDET_ON 0x0040
+#define R21_MT_ADC 0x8000
+#define R22_SEL_LNA 0x0008
+#define R22_SEL_MIC 0x0004
+#define R22_SKIP_DCFIL 0x0002
+#define R23_AGC_EN 0x0001
+
+struct uda1380_setup_data {
+ unsigned short i2c_address;
+ int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL 1
+};
+
+#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */
+#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */
+#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
+
+extern struct snd_soc_codec_dai uda1380_dai[3];
+extern struct snd_soc_codec_device soc_codec_dev_uda1380;
+
+#endif /* _UDA1380_H */
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-19 16:49:39 +0000