[alsa-devel] [PATCH 1/1] ASoC codec: SSM2602 audio codec driver (v2)
Bryan Wu
cooloney at kernel.org
Thu Sep 4 08:59:17 CEST 2008
From: Cliff Cai <cliff.cai at analog.com>
v1-v2:
- coding style fixing
- use pr_xxx macros to replace printk(KERN_XXX...)
- use new-style i2c interface
- update to use latest ASoC API
Signed-off-by: Cliff Cai <cliff.cai at analog.com>
Signed-off-by: Bryan Wu <cooloney at kernel.org>
---
sound/soc/codecs/Kconfig | 3 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/ssm2602.c | 773 ++++++++++++++++++++++++++++++++++++++++++++
sound/soc/codecs/ssm2602.h | 131 ++++++++
4 files changed, 909 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/codecs/ssm2602.c
create mode 100644 sound/soc/codecs/ssm2602.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 5d77dc3..8ef5ce6 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -92,3 +92,6 @@ config SND_SOC_TLV320AIC26
config SND_SOC_TLV320AIC3X
tristate
depends on I2C
+
+config SND_SOC_SSM2602
+ tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 35daaa9..0cd55ee 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -15,6 +15,7 @@ snd-soc-wm9713-objs := wm9713.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-ssm2602-objs := ssm2602.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
@@ -33,3 +34,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
new file mode 100644
index 0000000..feb462e
--- /dev/null
+++ b/sound/soc/codecs/ssm2602.c
@@ -0,0 +1,773 @@
+/*
+ * File: sound/soc/codecs/ssm2602.c
+ * Author: Cliff Cai <Cliff.Cai at analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: Driver for ssm2602 sound chip built in ADSP-BF52xC
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "ssm2602.h"
+
+#define AUDIO_NAME "ssm2602"
+#define SSM2602_VERSION "0.1"
+
+struct snd_soc_codec_device soc_codec_dev_ssm2602;
+
+/* codec private data */
+struct ssm2602_priv {
+ unsigned int sysclk;
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+/*
+ * ssm2602 register cache
+ * We can't read the ssm2602 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
+ 0x0017, 0x0017, 0x0079, 0x0079,
+ 0x0000, 0x0000, 0x0000, 0x000a,
+ 0x0000, 0x0000
+};
+
+/*
+ * read ssm2602 register cache
+ */
+static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == SSM2602_RESET)
+ return 0;
+ if (reg >= SSM2602_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write ssm2602 register cache
+ */
+static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= SSM2602_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the ssm2602 register space
+ */
+static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 ssm2602 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ ssm2602_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define ssm2602_reset(c) ssm2602_write(c, SSM2602_RESET, 0)
+/*Appending several "None"s just for OSS mixer use*/
+static const char *ssm2602_input_select[] = {
+ "Line", "Mic", "None", "None", "None",
+ "None", "None", "None",
+};
+
+static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum ssm2602_enum[] = {
+ SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select),
+ SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph),
+};
+
+static const struct snd_kcontrol_new ssm2602_snd_controls[] = {
+
+SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V,
+ 0, 127, 0),
+SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V,
+ 7, 1, 0),
+
+SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
+SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
+
+SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
+SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
+
+SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1),
+
+SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1),
+SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0),
+
+SOC_ENUM("Capture Source", ssm2602_enum[0]),
+
+SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
+};
+
+/* add non dapm controls */
+static int ssm2602_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Output Mixer */
+static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new ssm2602_input_mux_controls =
+SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]);
+
+static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1,
+ &ssm2602_output_mixer_controls[0],
+ ARRAY_SIZE(ssm2602_output_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
+SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls),
+SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1),
+SND_SOC_DAPM_INPUT("MICIN"),
+SND_SOC_DAPM_INPUT("RLINEIN"),
+SND_SOC_DAPM_INPUT("LLINEIN"),
+};
+
+static const struct snd_soc_dapm_route audio_conn[] = {
+ /* output mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+ /* outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+
+ /* input mux */
+ {"Input Mux", "Line", "Line Input"},
+ {"Input Mux", "Mic", "Mic Bias"},
+ {"ADC", NULL, "Input Mux"},
+
+ /* inputs */
+ {"Line Input", NULL, "LLINEIN"},
+ {"Line Input", NULL, "RLINEIN"},
+ {"Mic Bias", NULL, "MICIN"},
+
+ /* terminator */
+ {NULL, NULL, NULL},
+};
+
+static int ssm2602_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets,
+ ARRAY_SIZE(ssm2602_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:4;
+ u8 bosr:1;
+ u8 usb:1;
+};
+
+/* codec mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0, 0x0},
+ {18432000, 48000, 384, 0x0, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x0, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0x6, 0x0, 0x0},
+ {18432000, 32000, 576, 0x6, 0x1, 0x0},
+ {12000000, 32000, 375, 0x6, 0x0, 0x1},
+
+ /* 8k */
+ {12288000, 8000, 1536, 0x3, 0x0, 0x0},
+ {18432000, 8000, 2304, 0x3, 0x1, 0x0},
+ {11289600, 8000, 1408, 0xb, 0x0, 0x0},
+ {16934400, 8000, 2112, 0xb, 0x1, 0x0},
+ {12000000, 8000, 1500, 0x3, 0x0, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0x7, 0x0, 0x0},
+ {18432000, 96000, 192, 0x7, 0x1, 0x0},
+ {12000000, 96000, 125, 0x7, 0x0, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x8, 0x0, 0x0},
+ {16934400, 44100, 384, 0x8, 0x1, 0x0},
+ {12000000, 44100, 272, 0x8, 0x1, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0xf, 0x0, 0x0},
+ {16934400, 88200, 192, 0xf, 0x1, 0x0},
+ {12000000, 88200, 136, 0xf, 0x1, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return 0;
+}
+
+static int ssm2602_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;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
+ u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
+ int i = get_coeff(ssm2602->sysclk, params_rate(params));
+ u16 srate = (coeff_div[i].sr << 2) |
+ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+ ssm2602_write(codec, SSM2602_SRATE, srate);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x000c;
+ break;
+ }
+ ssm2602_write(codec, SSM2602_IFACE, iface);
+ ssm2602_write(codec, SSM2602_ACTIVE, ACTIVATE_CODEC);
+ return 0;
+}
+
+static int ssm2602_startup(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;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
+ struct snd_pcm_runtime *master_runtime;
+
+ /* The DAI has shared clocks so if we already have a playback or
+ * capture going then constrain this substream to match it.
+ */
+ if (ssm2602->master_substream) {
+ master_runtime = ssm2602->master_substream->runtime;
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ ssm2602->slave_substream = substream;
+ } else
+ ssm2602->master_substream = substream;
+
+ return 0;
+}
+
+static int ssm2602_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;
+ /* set active */
+ ssm2602_write(codec, SSM2602_ACTIVE, ACTIVATE_CODEC);
+
+ return 0;
+}
+
+static void ssm2602_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;
+ /* deactivate */
+ if (!codec->active) {
+ udelay(50);
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+ }
+}
+
+static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & 0xfff7;
+ if (mute)
+ ssm2602_write(codec, SSM2602_APDIGI,
+ mute_reg | ENABLE_DAC_MUTE);
+ else
+ ssm2602_write(codec, SSM2602_APDIGI, mute_reg);
+ return 0;
+}
+
+static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ ssm2602->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ ssm2602_write(codec, SSM2602_IFACE, iface);
+ return 0;
+}
+
+static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* vref/mid, osc on, dac unmute */
+ ssm2602_write(codec, SSM2602_PWR, 0);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* everything off except vref/vmid, */
+ ssm2602_write(codec, SSM2602_PWR, reg | CLK_OUT_PDN);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+ ssm2602_write(codec, SSM2602_PWR, 0xffff);
+ break;
+
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define SSM2602_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 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+struct snd_soc_dai ssm2602_dai = {
+ .name = "SSM2602",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SSM2602_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SSM2602_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .ops = {
+ .startup = ssm2602_startup,
+ .prepare = ssm2602_pcm_prepare,
+ .hw_params = ssm2602_hw_params,
+ .shutdown = ssm2602_shutdown,
+ },
+ .dai_ops = {
+ .digital_mute = ssm2602_mute,
+ .set_sysclk = ssm2602_set_dai_sysclk,
+ .set_fmt = ssm2602_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(ssm2602_dai);
+
+static int ssm2602_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;
+
+ ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int ssm2602_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(ssm2602_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ ssm2602_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the ssm2602 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int ssm2602_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "SSM2602";
+ codec->owner = THIS_MODULE;
+ codec->read = ssm2602_read_reg_cache;
+ codec->write = ssm2602_write;
+ codec->set_bias_level = ssm2602_set_bias_level;
+ codec->dai = &ssm2602_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = sizeof(ssm2602_reg);
+ codec->reg_cache = kmemdup(ssm2602_reg, sizeof(ssm2602_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ ssm2602_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ pr_err("ssm2602: failed to create pcms\n");
+ goto pcm_err;
+ }
+ /*power on device*/
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+ /* set the update bits */
+ reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL);
+ ssm2602_write(codec, SSM2602_LINVOL, reg | LRIN_BOTH);
+ reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL);
+ ssm2602_write(codec, SSM2602_RINVOL, reg | RLIN_BOTH);
+ reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V);
+ ssm2602_write(codec, SSM2602_LOUT1V, reg | LRHP_BOTH);
+ reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V);
+ ssm2602_write(codec, SSM2602_ROUT1V, reg | RLHP_BOTH);
+ /*select Line in as default input*/
+ ssm2602_write(codec, SSM2602_APANA,
+ ENABLE_MIC_BOOST2 | SELECT_DAC | ENABLE_MIC_BOOST);
+ ssm2602_write(codec, SSM2602_PWR, 0);
+
+ ssm2602_add_controls(codec);
+ ssm2602_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ pr_err("ssm2602: 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 *ssm2602_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * ssm2602 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static int ssm2602_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = ssm2602_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = ssm2602_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise SSM2602\n");
+
+ return ret;
+}
+
+static int ssm2602_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id ssm2602_i2c_id[] = {
+ { "ssm2602", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
+/* corgi i2c codec control layer */
+static struct i2c_driver ssm2602_i2c_driver = {
+ .driver = {
+ .name = "SSM2602 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ssm2602_i2c_probe,
+ .remove = ssm2602_i2c_remove,
+ .id_table = ssm2602_i2c_id,
+};
+
+static int ssm2602_add_i2c_device(struct platform_device *pdev,
+ const struct ssm2602_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&ssm2602_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "ssm2602", I2C_NAME_SIZE);
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+ return 0;
+err_driver:
+ i2c_del_driver(&ssm2602_i2c_driver);
+ return -ENODEV;
+}
+
+#endif
+
+static int ssm2602_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct ssm2602_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct ssm2602_priv *ssm2602;
+ int ret = 0;
+
+ pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL);
+ if (ssm2602 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = ssm2602;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ssm2602_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = ssm2602_add_i2c_device(pdev, setup);
+ }
+#else
+ /* other interfaces */
+#endif
+ return ret;
+}
+
+/* remove everything here */
+static int ssm2602_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)
+ ssm2602_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_unregister_device(codec->control_data);
+ i2c_del_driver(&ssm2602_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
+ .probe = ssm2602_probe,
+ .remove = ssm2602_remove,
+ .suspend = ssm2602_suspend,
+ .resume = ssm2602_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
+
+MODULE_DESCRIPTION("ASoC ssm2602 driver");
+MODULE_AUTHOR("Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h
new file mode 100644
index 0000000..ec5e604
--- /dev/null
+++ b/sound/soc/codecs/ssm2602.h
@@ -0,0 +1,131 @@
+/*
+ * File: sound/soc/codecs/ssm2602.h
+ * Author: Cliff Cai <Cliff.Cai at analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef _SSM2602_H
+#define _SSM2602_H
+
+/* SSM2602 Codec Register definitions */
+
+#define SSM2602_LINVOL 0x00
+#define SSM2602_RINVOL 0x01
+#define SSM2602_LOUT1V 0x02
+#define SSM2602_ROUT1V 0x03
+#define SSM2602_APANA 0x04
+#define SSM2602_APDIGI 0x05
+#define SSM2602_PWR 0x06
+#define SSM2602_IFACE 0x07
+#define SSM2602_SRATE 0x08
+#define SSM2602_ACTIVE 0x09
+#define SSM2602_RESET 0x0f
+
+/*SSM2602 Codec Register Field definitions
+ *(Mask value to extract the corresponding Register field)
+ */
+
+/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/
+#define LIN_VOL 0x01F /* Left Channel PGA Volume control */
+#define LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */
+#define LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */
+
+/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/
+#define RIN_VOL 0x01F /* Right Channel PGA Volume control */
+#define RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */
+#define RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */
+
+/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/
+#define LHP_VOL 0x07F /* Left Channel Headphone volume control */
+#define ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */
+#define LRHP_BOTH 0x100 /* Left Channel Headphone volume update */
+
+/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/
+#define RHP_VOL 0x07F /* Right Channel Headphone volume control */
+#define ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */
+#define RLHP_BOTH 0x100 /* Right Channel Headphone volume update */
+
+/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/
+#define ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */
+#define ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */
+#define ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */
+#define ENABLE_BYPASS 0x008 /* Line input bypass to line output */
+#define SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */
+#define ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */
+#define SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */
+#define ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */
+
+/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/
+#define ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */
+#define DE_EMPHASIS 0x006 /* De-Emphasis Control */
+#define ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */
+#define STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */
+
+/*Power Down Control (SSM2602_REG_POWER)
+ *(1=Enable PowerDown, 0=Disable PowerDown)
+ */
+#define LINE_IN_PDN 0x001 /* Line Input Power Down */
+#define MIC_PDN 0x002 /* Microphone Input & Bias Power Down */
+#define ADC_PDN 0x004 /* ADC Power Down */
+#define DAC_PDN 0x008 /* DAC Power Down */
+#define OUT_PDN 0x010 /* Outputs Power Down */
+#define OSC_PDN 0x020 /* Oscillator Power Down */
+#define CLK_OUT_PDN 0x040 /* CLKOUT Power Down */
+#define POWER_OFF 0x080 /* POWEROFF Mode */
+
+/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/
+#define IFACE_FORMAT 0x003 /* Digital Audio input format control */
+#define AUDIO_DATA_LEN 0x00C /* Audio Data word length control */
+#define DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */
+#define DAC_LR_SWAP 0x020 /* Swap DAC data control */
+#define ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */
+#define BCLK_INVERT 0x080 /* Bit Clock Inversion control */
+
+/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/
+#define ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */
+#define BOS_RATE 0x002 /* Base Over-Sampling rate */
+#define SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */
+#define CORECLK_DIV2 0x040 /* Core Clock divider select */
+#define CLKOUT_DIV2 0x080 /* Clock Out divider select */
+
+/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/
+#define ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */
+
+/*********************************************************************/
+
+#define SSM2602_CACHEREGNUM 10
+
+#define SSM2602_SYSCLK 0
+#define SSM2602_DAI 0
+
+struct ssm2602_setup_data {
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai ssm2602_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ssm2602;
+
+#endif
--
1.5.6
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