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-rw-r--r--sound/aoa/codecs/Kconfig24
-rw-r--r--sound/aoa/codecs/Makefile7
-rw-r--r--sound/aoa/codecs/onyx.c1060
-rw-r--r--sound/aoa/codecs/onyx.h75
-rw-r--r--sound/aoa/codecs/tas-basstreble.h134
-rw-r--r--sound/aoa/codecs/tas-gain-table.h209
-rw-r--r--sound/aoa/codecs/tas.c949
-rw-r--r--sound/aoa/codecs/tas.h55
-rw-r--r--sound/aoa/codecs/toonie.c151
9 files changed, 2664 insertions, 0 deletions
diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig
new file mode 100644
index 000000000..0c68e3283
--- /dev/null
+++ b/sound/aoa/codecs/Kconfig
@@ -0,0 +1,24 @@
+config SND_AOA_ONYX
+ tristate "support Onyx chip"
+ select I2C
+ select I2C_POWERMAC
+ ---help---
+ This option enables support for the Onyx (pcm3052)
+ codec chip found in the latest Apple machines
+ (most of those with digital audio output).
+
+config SND_AOA_TAS
+ tristate "support TAS chips"
+ select I2C
+ select I2C_POWERMAC
+ ---help---
+ This option enables support for the tas chips
+ found in a lot of Apple Machines, especially
+ iBooks and PowerBooks without digital.
+
+config SND_AOA_TOONIE
+ tristate "support Toonie chip"
+ ---help---
+ This option enables support for the toonie codec
+ found in the Mac Mini. If you have a Mac Mini and
+ want to hear sound, select this option.
diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile
new file mode 100644
index 000000000..c3ee77fc4
--- /dev/null
+++ b/sound/aoa/codecs/Makefile
@@ -0,0 +1,7 @@
+snd-aoa-codec-onyx-objs := onyx.o
+snd-aoa-codec-tas-objs := tas.o
+snd-aoa-codec-toonie-objs := toonie.o
+
+obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
+obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
+obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o
diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
new file mode 100644
index 000000000..23c371ecf
--- /dev/null
+++ b/sound/aoa/codecs/onyx.c
@@ -0,0 +1,1060 @@
+/*
+ * Apple Onboard Audio driver for Onyx codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ *
+ * This is a driver for the pcm3052 codec chip (codenamed Onyx)
+ * that is present in newer Apple hardware (with digital output).
+ *
+ * The Onyx codec has the following connections (listed by the bit
+ * to be used in aoa_codec.connected):
+ * 0: analog output
+ * 1: digital output
+ * 2: line input
+ * 3: microphone input
+ * Note that even though I know of no machine that has for example
+ * the digital output connected but not the analog, I have handled
+ * all the different cases in the code so that this driver may serve
+ * as a good example of what to do.
+ *
+ * NOTE: This driver assumes that there's at most one chip to be
+ * used with one alsa card, in form of creating all kinds
+ * of mixer elements without regard for their existence.
+ * But snd-aoa assumes that there's at most one card, so
+ * this means you can only have one onyx on a system. This
+ * should probably be fixed by changing the assumption of
+ * having just a single card on a system, and making the
+ * 'card' pointer accessible to anyone who needs it instead
+ * of hiding it in the aoa_snd_* functions...
+ *
+ */
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa");
+
+#include "onyx.h"
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-onyx: "
+
+struct onyx {
+ /* cache registers 65 to 80, they are write-only! */
+ u8 cache[16];
+ struct i2c_client *i2c;
+ struct aoa_codec codec;
+ u32 initialised:1,
+ spdif_locked:1,
+ analog_locked:1,
+ original_mute:2;
+ int open_count;
+ struct codec_info *codec_info;
+
+ /* mutex serializes concurrent access to the device
+ * and this structure.
+ */
+ struct mutex mutex;
+};
+#define codec_to_onyx(c) container_of(c, struct onyx, codec)
+
+/* both return 0 if all ok, else on error */
+static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value)
+{
+ s32 v;
+
+ if (reg != ONYX_REG_CONTROL) {
+ *value = onyx->cache[reg-FIRSTREGISTER];
+ return 0;
+ }
+ v = i2c_smbus_read_byte_data(onyx->i2c, reg);
+ if (v < 0)
+ return -1;
+ *value = (u8)v;
+ onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value;
+ return 0;
+}
+
+static int onyx_write_register(struct onyx *onyx, u8 reg, u8 value)
+{
+ int result;
+
+ result = i2c_smbus_write_byte_data(onyx->i2c, reg, value);
+ if (!result)
+ onyx->cache[reg-FIRSTREGISTER] = value;
+ return result;
+}
+
+/* alsa stuff */
+
+static int onyx_dev_register(struct snd_device *dev)
+{
+ return 0;
+}
+
+static struct snd_device_ops ops = {
+ .dev_register = onyx_dev_register,
+};
+
+/* this is necessary because most alsa mixer programs
+ * can't properly handle the negative range */
+#define VOLUME_RANGE_SHIFT 128
+
+static int onyx_snd_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = -128 + VOLUME_RANGE_SHIFT;
+ uinfo->value.integer.max = -1 + VOLUME_RANGE_SHIFT;
+ return 0;
+}
+
+static int onyx_snd_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ s8 l, r;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] = l + VOLUME_RANGE_SHIFT;
+ ucontrol->value.integer.value[1] = r + VOLUME_RANGE_SHIFT;
+
+ return 0;
+}
+
+static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ s8 l, r;
+
+ if (ucontrol->value.integer.value[0] < -128 + VOLUME_RANGE_SHIFT ||
+ ucontrol->value.integer.value[0] > -1 + VOLUME_RANGE_SHIFT)
+ return -EINVAL;
+ if (ucontrol->value.integer.value[1] < -128 + VOLUME_RANGE_SHIFT ||
+ ucontrol->value.integer.value[1] > -1 + VOLUME_RANGE_SHIFT)
+ return -EINVAL;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
+
+ if (l + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[0] &&
+ r + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[1]) {
+ mutex_unlock(&onyx->mutex);
+ return 0;
+ }
+
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT,
+ ucontrol->value.integer.value[0]
+ - VOLUME_RANGE_SHIFT);
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT,
+ ucontrol->value.integer.value[1]
+ - VOLUME_RANGE_SHIFT);
+ mutex_unlock(&onyx->mutex);
+
+ return 1;
+}
+
+static struct snd_kcontrol_new volume_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_vol_info,
+ .get = onyx_snd_vol_get,
+ .put = onyx_snd_vol_put,
+};
+
+/* like above, this is necessary because a lot
+ * of alsa mixer programs don't handle ranges
+ * that don't start at 0 properly.
+ * even alsamixer is one of them... */
+#define INPUTGAIN_RANGE_SHIFT (-3)
+
+static int onyx_snd_inputgain_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 3 + INPUTGAIN_RANGE_SHIFT;
+ uinfo->value.integer.max = 28 + INPUTGAIN_RANGE_SHIFT;
+ return 0;
+}
+
+static int onyx_snd_inputgain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 ig;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &ig);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] =
+ (ig & ONYX_ADC_PGA_GAIN_MASK) + INPUTGAIN_RANGE_SHIFT;
+
+ return 0;
+}
+
+static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v, n;
+
+ if (ucontrol->value.integer.value[0] < 3 + INPUTGAIN_RANGE_SHIFT ||
+ ucontrol->value.integer.value[0] > 28 + INPUTGAIN_RANGE_SHIFT)
+ return -EINVAL;
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+ n = v;
+ n &= ~ONYX_ADC_PGA_GAIN_MASK;
+ n |= (ucontrol->value.integer.value[0] - INPUTGAIN_RANGE_SHIFT)
+ & ONYX_ADC_PGA_GAIN_MASK;
+ onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, n);
+ mutex_unlock(&onyx->mutex);
+
+ return n != v;
+}
+
+static struct snd_kcontrol_new inputgain_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Capture Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_inputgain_info,
+ .get = onyx_snd_inputgain_get,
+ .put = onyx_snd_inputgain_put,
+};
+
+static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = { "Line-In", "Microphone" };
+
+ return snd_ctl_enum_info(uinfo, 1, 2, texts);
+}
+
+static int onyx_snd_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ s8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.enumerated.item[0] = !!(v&ONYX_ADC_INPUT_MIC);
+
+ return 0;
+}
+
+static void onyx_set_capture_source(struct onyx *onyx, int mic)
+{
+ s8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+ v &= ~ONYX_ADC_INPUT_MIC;
+ if (mic)
+ v |= ONYX_ADC_INPUT_MIC;
+ onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, v);
+ mutex_unlock(&onyx->mutex);
+}
+
+static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ onyx_set_capture_source(snd_kcontrol_chip(kcontrol),
+ ucontrol->value.enumerated.item[0]);
+ return 1;
+}
+
+static struct snd_kcontrol_new capture_source_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* If we name this 'Input Source', it properly shows up in
+ * alsamixer as a selection, * but it's shown under the
+ * 'Playback' category.
+ * If I name it 'Capture Source', it shows up in strange
+ * ways (two bools of which one can be selected at a
+ * time) but at least it's shown in the 'Capture'
+ * category.
+ * I was told that this was due to backward compatibility,
+ * but I don't understand then why the mangling is *not*
+ * done when I name it "Input Source".....
+ */
+ .name = "Capture Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_capture_source_info,
+ .get = onyx_snd_capture_source_get,
+ .put = onyx_snd_capture_source_put,
+};
+
+#define onyx_snd_mute_info snd_ctl_boolean_stereo_info
+
+static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 c;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] = !(c & ONYX_MUTE_LEFT);
+ ucontrol->value.integer.value[1] = !(c & ONYX_MUTE_RIGHT);
+
+ return 0;
+}
+
+static int onyx_snd_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v = 0, c = 0;
+ int err = -EBUSY;
+
+ mutex_lock(&onyx->mutex);
+ if (onyx->analog_locked)
+ goto out_unlock;
+
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+ c = v;
+ c &= ~(ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT);
+ if (!ucontrol->value.integer.value[0])
+ c |= ONYX_MUTE_LEFT;
+ if (!ucontrol->value.integer.value[1])
+ c |= ONYX_MUTE_RIGHT;
+ err = onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, c);
+
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? (v != c) : err;
+}
+
+static struct snd_kcontrol_new mute_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_mute_info,
+ .get = onyx_snd_mute_get,
+ .put = onyx_snd_mute_put,
+};
+
+
+#define onyx_snd_single_bit_info snd_ctl_boolean_mono_info
+
+#define FLAG_POLARITY_INVERT 1
+#define FLAG_SPDIFLOCK 2
+
+static int onyx_snd_single_bit_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 c;
+ long int pv = kcontrol->private_value;
+ u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
+ u8 address = (pv >> 8) & 0xff;
+ u8 mask = pv & 0xff;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, address, &c);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] = !!(c & mask) ^ polarity;
+
+ return 0;
+}
+
+static int onyx_snd_single_bit_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v = 0, c = 0;
+ int err;
+ long int pv = kcontrol->private_value;
+ u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
+ u8 spdiflock = (pv >> 16) & FLAG_SPDIFLOCK;
+ u8 address = (pv >> 8) & 0xff;
+ u8 mask = pv & 0xff;
+
+ mutex_lock(&onyx->mutex);
+ if (spdiflock && onyx->spdif_locked) {
+ /* even if alsamixer doesn't care.. */
+ err = -EBUSY;
+ goto out_unlock;
+ }
+ onyx_read_register(onyx, address, &v);
+ c = v;
+ c &= ~(mask);
+ if (!!ucontrol->value.integer.value[0] ^ polarity)
+ c |= mask;
+ err = onyx_write_register(onyx, address, c);
+
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? (v != c) : err;
+}
+
+#define SINGLE_BIT(n, type, description, address, mask, flags) \
+static struct snd_kcontrol_new n##_control = { \
+ .iface = SNDRV_CTL_ELEM_IFACE_##type, \
+ .name = description, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = onyx_snd_single_bit_info, \
+ .get = onyx_snd_single_bit_get, \
+ .put = onyx_snd_single_bit_put, \
+ .private_value = (flags << 16) | (address << 8) | mask \
+}
+
+SINGLE_BIT(spdif,
+ MIXER,
+ SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
+ ONYX_REG_DIG_INFO4,
+ ONYX_SPDIF_ENABLE,
+ FLAG_SPDIFLOCK);
+SINGLE_BIT(ovr1,
+ MIXER,
+ "Oversampling Rate",
+ ONYX_REG_DAC_CONTROL,
+ ONYX_OVR1,
+ 0);
+SINGLE_BIT(flt0,
+ MIXER,
+ "Fast Digital Filter Rolloff",
+ ONYX_REG_DAC_FILTER,
+ ONYX_ROLLOFF_FAST,
+ FLAG_POLARITY_INVERT);
+SINGLE_BIT(hpf,
+ MIXER,
+ "Highpass Filter",
+ ONYX_REG_ADC_HPF_BYPASS,
+ ONYX_HPF_DISABLE,
+ FLAG_POLARITY_INVERT);
+SINGLE_BIT(dm12,
+ MIXER,
+ "Digital De-Emphasis",
+ ONYX_REG_DAC_DEEMPH,
+ ONYX_DIGDEEMPH_CTRL,
+ 0);
+
+static int onyx_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ /* datasheet page 30, all others are 0 */
+ ucontrol->value.iec958.status[0] = 0x3e;
+ ucontrol->value.iec958.status[1] = 0xff;
+
+ ucontrol->value.iec958.status[3] = 0x3f;
+ ucontrol->value.iec958.status[4] = 0x0f;
+
+ return 0;
+}
+
+static struct snd_kcontrol_new onyx_spdif_mask = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = onyx_spdif_info,
+ .get = onyx_spdif_mask_get,
+};
+
+static int onyx_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
+ ucontrol->value.iec958.status[0] = v & 0x3e;
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO2, &v);
+ ucontrol->value.iec958.status[1] = v;
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
+ ucontrol->value.iec958.status[3] = v & 0x3f;
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ ucontrol->value.iec958.status[4] = v & 0x0f;
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+static int onyx_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
+ v = (v & ~0x3e) | (ucontrol->value.iec958.status[0] & 0x3e);
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO1, v);
+
+ v = ucontrol->value.iec958.status[1];
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO2, v);
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
+ v = (v & ~0x3f) | (ucontrol->value.iec958.status[3] & 0x3f);
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO3, v);
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ v = (v & ~0x0f) | (ucontrol->value.iec958.status[4] & 0x0f);
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
+ mutex_unlock(&onyx->mutex);
+
+ return 1;
+}
+
+static struct snd_kcontrol_new onyx_spdif_ctrl = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = onyx_spdif_info,
+ .get = onyx_spdif_get,
+ .put = onyx_spdif_put,
+};
+
+/* our registers */
+
+static u8 register_map[] = {
+ ONYX_REG_DAC_ATTEN_LEFT,
+ ONYX_REG_DAC_ATTEN_RIGHT,
+ ONYX_REG_CONTROL,
+ ONYX_REG_DAC_CONTROL,
+ ONYX_REG_DAC_DEEMPH,
+ ONYX_REG_DAC_FILTER,
+ ONYX_REG_DAC_OUTPHASE,
+ ONYX_REG_ADC_CONTROL,
+ ONYX_REG_ADC_HPF_BYPASS,
+ ONYX_REG_DIG_INFO1,
+ ONYX_REG_DIG_INFO2,
+ ONYX_REG_DIG_INFO3,
+ ONYX_REG_DIG_INFO4
+};
+
+static u8 initial_values[ARRAY_SIZE(register_map)] = {
+ 0x80, 0x80, /* muted */
+ ONYX_MRST | ONYX_SRST, /* but handled specially! */
+ ONYX_MUTE_LEFT | ONYX_MUTE_RIGHT,
+ 0, /* no deemphasis */
+ ONYX_DAC_FILTER_ALWAYS,
+ ONYX_OUTPHASE_INVERTED,
+ (-1 /*dB*/ + 8) & 0xF, /* line in selected, -1 dB gain*/
+ ONYX_ADC_HPF_ALWAYS,
+ (1<<2), /* pcm audio */
+ 2, /* category: pcm coder */
+ 0, /* sampling frequency 44.1 kHz, clock accuracy level II */
+ 1 /* 24 bit depth */
+};
+
+/* reset registers of chip, either to initial or to previous values */
+static int onyx_register_init(struct onyx *onyx)
+{
+ int i;
+ u8 val;
+ u8 regs[sizeof(initial_values)];
+
+ if (!onyx->initialised) {
+ memcpy(regs, initial_values, sizeof(initial_values));
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &val))
+ return -1;
+ val &= ~ONYX_SILICONVERSION;
+ val |= initial_values[3];
+ regs[3] = val;
+ } else {
+ for (i=0; i<sizeof(register_map); i++)
+ regs[i] = onyx->cache[register_map[i]-FIRSTREGISTER];
+ }
+
+ for (i=0; i<sizeof(register_map); i++) {
+ if (onyx_write_register(onyx, register_map[i], regs[i]))
+ return -1;
+ }
+ onyx->initialised = 1;
+ return 0;
+}
+
+static struct transfer_info onyx_transfers[] = {
+ /* this is first so we can skip it if no input is present...
+ * No hardware exists with that, but it's here as an example
+ * of what to do :) */
+ {
+ /* analog input */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .transfer_in = 1,
+ .must_be_clock_source = 0,
+ .tag = 0,
+ },
+ {
+ /* if analog and digital are currently off, anything should go,
+ * so this entry describes everything we can do... */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+ | SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+#endif
+ ,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .tag = 0,
+ },
+ {
+ /* analog output */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .transfer_in = 0,
+ .must_be_clock_source = 0,
+ .tag = 1,
+ },
+ {
+ /* digital pcm output, also possible for analog out */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .transfer_in = 0,
+ .must_be_clock_source = 0,
+ .tag = 2,
+ },
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+ /* Once alsa gets supports for this kind of thing we can add it... */
+ {
+ /* digital compressed output */
+ .formats = SNDRV_PCM_FMTBIT_COMPRESSED_16BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .tag = 2,
+ },
+#endif
+ {}
+};
+
+static int onyx_usable(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out)
+{
+ u8 v;
+ struct onyx *onyx = cii->codec_data;
+ int spdif_enabled, analog_enabled;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ spdif_enabled = !!(v & ONYX_SPDIF_ENABLE);
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+ analog_enabled =
+ (v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT))
+ != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT);
+ mutex_unlock(&onyx->mutex);
+
+ switch (ti->tag) {
+ case 0: return 1;
+ case 1: return analog_enabled;
+ case 2: return spdif_enabled;
+ }
+ return 1;
+}
+
+static int onyx_prepare(struct codec_info_item *cii,
+ struct bus_info *bi,
+ struct snd_pcm_substream *substream)
+{
+ u8 v;
+ struct onyx *onyx = cii->codec_data;
+ int err = -EBUSY;
+
+ mutex_lock(&onyx->mutex);
+
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+ if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) {
+ /* mute and lock analog output */
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+ if (onyx_write_register(onyx,
+ ONYX_REG_DAC_CONTROL,
+ v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT))
+ goto out_unlock;
+ onyx->analog_locked = 1;
+ err = 0;
+ goto out_unlock;
+ }
+#endif
+ switch (substream->runtime->rate) {
+ case 32000:
+ case 44100:
+ case 48000:
+ /* these rates are ok for all outputs */
+ /* FIXME: program spdif channel control bits here so that
+ * userspace doesn't have to if it only plays pcm! */
+ err = 0;
+ goto out_unlock;
+ default:
+ /* got some rate that the digital output can't do,
+ * so disable and lock it */
+ onyx_read_register(cii->codec_data, ONYX_REG_DIG_INFO4, &v);
+ if (onyx_write_register(onyx,
+ ONYX_REG_DIG_INFO4,
+ v & ~ONYX_SPDIF_ENABLE))
+ goto out_unlock;
+ onyx->spdif_locked = 1;
+ err = 0;
+ goto out_unlock;
+ }
+
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return err;
+}
+
+static int onyx_open(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream)
+{
+ struct onyx *onyx = cii->codec_data;
+
+ mutex_lock(&onyx->mutex);
+ onyx->open_count++;
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+static int onyx_close(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream)
+{
+ struct onyx *onyx = cii->codec_data;
+
+ mutex_lock(&onyx->mutex);
+ onyx->open_count--;
+ if (!onyx->open_count)
+ onyx->spdif_locked = onyx->analog_locked = 0;
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+static int onyx_switch_clock(struct codec_info_item *cii,
+ enum clock_switch what)
+{
+ struct onyx *onyx = cii->codec_data;
+
+ mutex_lock(&onyx->mutex);
+ /* this *MUST* be more elaborate later... */
+ switch (what) {
+ case CLOCK_SWITCH_PREPARE_SLAVE:
+ onyx->codec.gpio->methods->all_amps_off(onyx->codec.gpio);
+ break;
+ case CLOCK_SWITCH_SLAVE:
+ onyx->codec.gpio->methods->all_amps_restore(onyx->codec.gpio);
+ break;
+ default: /* silence warning */
+ break;
+ }
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+
+static int onyx_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+ struct onyx *onyx = cii->codec_data;
+ u8 v;
+ int err = -ENXIO;
+
+ mutex_lock(&onyx->mutex);
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
+ goto out_unlock;
+ onyx_write_register(onyx, ONYX_REG_CONTROL, v | ONYX_ADPSV | ONYX_DAPSV);
+ /* Apple does a sleep here but the datasheet says to do it on resume */
+ err = 0;
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return err;
+}
+
+static int onyx_resume(struct codec_info_item *cii)
+{
+ struct onyx *onyx = cii->codec_data;
+ u8 v;
+ int err = -ENXIO;
+
+ mutex_lock(&onyx->mutex);
+
+ /* reset codec */
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+
+ /* take codec out of suspend (if it still is after reset) */
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
+ goto out_unlock;
+ onyx_write_register(onyx, ONYX_REG_CONTROL, v & ~(ONYX_ADPSV | ONYX_DAPSV));
+ /* FIXME: should divide by sample rate, but 8k is the lowest we go */
+ msleep(2205000/8000);
+ /* reset all values */
+ onyx_register_init(onyx);
+ err = 0;
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return err;
+}
+
+#endif /* CONFIG_PM */
+
+static struct codec_info onyx_codec_info = {
+ .transfers = onyx_transfers,
+ .sysclock_factor = 256,
+ .bus_factor = 64,
+ .owner = THIS_MODULE,
+ .usable = onyx_usable,
+ .prepare = onyx_prepare,
+ .open = onyx_open,
+ .close = onyx_close,
+ .switch_clock = onyx_switch_clock,
+#ifdef CONFIG_PM
+ .suspend = onyx_suspend,
+ .resume = onyx_resume,
+#endif
+};
+
+static int onyx_init_codec(struct aoa_codec *codec)
+{
+ struct onyx *onyx = codec_to_onyx(codec);
+ struct snd_kcontrol *ctl;
+ struct codec_info *ci = &onyx_codec_info;
+ u8 v;
+ int err;
+
+ if (!onyx->codec.gpio || !onyx->codec.gpio->methods) {
+ printk(KERN_ERR PFX "gpios not assigned!!\n");
+ return -EINVAL;
+ }
+
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+
+ if (onyx_register_init(onyx)) {
+ printk(KERN_ERR PFX "failed to initialise onyx registers\n");
+ return -ENODEV;
+ }
+
+ if (aoa_snd_device_new(SNDRV_DEV_CODEC, onyx, &ops)) {
+ printk(KERN_ERR PFX "failed to create onyx snd device!\n");
+ return -ENODEV;
+ }
+
+ /* nothing connected? what a joke! */
+ if ((onyx->codec.connected & 0xF) == 0)
+ return -ENOTCONN;
+
+ /* if no inputs are present... */
+ if ((onyx->codec.connected & 0xC) == 0) {
+ if (!onyx->codec_info)
+ onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
+ if (!onyx->codec_info)
+ return -ENOMEM;
+ ci = onyx->codec_info;
+ *ci = onyx_codec_info;
+ ci->transfers++;
+ }
+
+ /* if no outputs are present... */
+ if ((onyx->codec.connected & 3) == 0) {
+ if (!onyx->codec_info)
+ onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
+ if (!onyx->codec_info)
+ return -ENOMEM;
+ ci = onyx->codec_info;
+ /* this is fine as there have to be inputs
+ * if we end up in this part of the code */
+ *ci = onyx_codec_info;
+ ci->transfers[1].formats = 0;
+ }
+
+ if (onyx->codec.soundbus_dev->attach_codec(onyx->codec.soundbus_dev,
+ aoa_get_card(),
+ ci, onyx)) {
+ printk(KERN_ERR PFX "error creating onyx pcm\n");
+ return -ENODEV;
+ }
+#define ADDCTL(n) \
+ do { \
+ ctl = snd_ctl_new1(&n, onyx); \
+ if (ctl) { \
+ ctl->id.device = \
+ onyx->codec.soundbus_dev->pcm->device; \
+ err = aoa_snd_ctl_add(ctl); \
+ if (err) \
+ goto error; \
+ } \
+ } while (0)
+
+ if (onyx->codec.soundbus_dev->pcm) {
+ /* give the user appropriate controls
+ * depending on what inputs are connected */
+ if ((onyx->codec.connected & 0xC) == 0xC)
+ ADDCTL(capture_source_control);
+ else if (onyx->codec.connected & 4)
+ onyx_set_capture_source(onyx, 0);
+ else
+ onyx_set_capture_source(onyx, 1);
+ if (onyx->codec.connected & 0xC)
+ ADDCTL(inputgain_control);
+
+ /* depending on what output is connected,
+ * give the user appropriate controls */
+ if (onyx->codec.connected & 1) {
+ ADDCTL(volume_control);
+ ADDCTL(mute_control);
+ ADDCTL(ovr1_control);
+ ADDCTL(flt0_control);
+ ADDCTL(hpf_control);
+ ADDCTL(dm12_control);
+ /* spdif control defaults to off */
+ }
+ if (onyx->codec.connected & 2) {
+ ADDCTL(onyx_spdif_mask);
+ ADDCTL(onyx_spdif_ctrl);
+ }
+ if ((onyx->codec.connected & 3) == 3)
+ ADDCTL(spdif_control);
+ /* if only S/PDIF is connected, enable it unconditionally */
+ if ((onyx->codec.connected & 3) == 2) {
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ v |= ONYX_SPDIF_ENABLE;
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
+ }
+ }
+#undef ADDCTL
+ printk(KERN_INFO PFX "attached to onyx codec via i2c\n");
+
+ return 0;
+ error:
+ onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
+ snd_device_free(aoa_get_card(), onyx);
+ return err;
+}
+
+static void onyx_exit_codec(struct aoa_codec *codec)
+{
+ struct onyx *onyx = codec_to_onyx(codec);
+
+ if (!onyx->codec.soundbus_dev) {
+ printk(KERN_ERR PFX "onyx_exit_codec called without soundbus_dev!\n");
+ return;
+ }
+ onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
+}
+
+static int onyx_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct device_node *node = client->dev.of_node;
+ struct onyx *onyx;
+ u8 dummy;
+
+ onyx = kzalloc(sizeof(struct onyx), GFP_KERNEL);
+
+ if (!onyx)
+ return -ENOMEM;
+
+ mutex_init(&onyx->mutex);
+ onyx->i2c = client;
+ i2c_set_clientdata(client, onyx);
+
+ /* we try to read from register ONYX_REG_CONTROL
+ * to check if the codec is present */
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) {
+ printk(KERN_ERR PFX "failed to read control register\n");
+ goto fail;
+ }
+
+ strlcpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN);
+ onyx->codec.owner = THIS_MODULE;
+ onyx->codec.init = onyx_init_codec;
+ onyx->codec.exit = onyx_exit_codec;
+ onyx->codec.node = of_node_get(node);
+
+ if (aoa_codec_register(&onyx->codec)) {
+ goto fail;
+ }
+ printk(KERN_DEBUG PFX "created and attached onyx instance\n");
+ return 0;
+ fail:
+ kfree(onyx);
+ return -ENODEV;
+}
+
+static int onyx_i2c_remove(struct i2c_client *client)
+{
+ struct onyx *onyx = i2c_get_clientdata(client);
+
+ aoa_codec_unregister(&onyx->codec);
+ of_node_put(onyx->codec.node);
+ kfree(onyx->codec_info);
+ kfree(onyx);
+ return 0;
+}
+
+static const struct i2c_device_id onyx_i2c_id[] = {
+ { "MAC,pcm3052", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c,onyx_i2c_id);
+
+static struct i2c_driver onyx_driver = {
+ .driver = {
+ .name = "aoa_codec_onyx",
+ .owner = THIS_MODULE,
+ },
+ .probe = onyx_i2c_probe,
+ .remove = onyx_i2c_remove,
+ .id_table = onyx_i2c_id,
+};
+
+module_i2c_driver(onyx_driver);
diff --git a/sound/aoa/codecs/onyx.h b/sound/aoa/codecs/onyx.h
new file mode 100644
index 000000000..ffd20254f
--- /dev/null
+++ b/sound/aoa/codecs/onyx.h
@@ -0,0 +1,75 @@
+/*
+ * Apple Onboard Audio driver for Onyx codec (header)
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SND_AOA_CODEC_ONYX_H
+#define __SND_AOA_CODEC_ONYX_H
+#include <stddef.h>
+#include <linux/i2c.h>
+#include <asm/pmac_low_i2c.h>
+#include <asm/prom.h>
+
+/* PCM3052 register definitions */
+
+/* the attenuation registers take values from
+ * -1 (0dB) to -127 (-63.0 dB) or others (muted) */
+#define ONYX_REG_DAC_ATTEN_LEFT 65
+#define FIRSTREGISTER ONYX_REG_DAC_ATTEN_LEFT
+#define ONYX_REG_DAC_ATTEN_RIGHT 66
+
+#define ONYX_REG_CONTROL 67
+# define ONYX_MRST (1<<7)
+# define ONYX_SRST (1<<6)
+# define ONYX_ADPSV (1<<5)
+# define ONYX_DAPSV (1<<4)
+# define ONYX_SILICONVERSION (1<<0)
+/* all others reserved */
+
+#define ONYX_REG_DAC_CONTROL 68
+# define ONYX_OVR1 (1<<6)
+# define ONYX_MUTE_RIGHT (1<<1)
+# define ONYX_MUTE_LEFT (1<<0)
+
+#define ONYX_REG_DAC_DEEMPH 69
+# define ONYX_DIGDEEMPH_SHIFT 5
+# define ONYX_DIGDEEMPH_MASK (3<<ONYX_DIGDEEMPH_SHIFT)
+# define ONYX_DIGDEEMPH_CTRL (1<<4)
+
+#define ONYX_REG_DAC_FILTER 70
+# define ONYX_ROLLOFF_FAST (1<<5)
+# define ONYX_DAC_FILTER_ALWAYS (1<<2)
+
+#define ONYX_REG_DAC_OUTPHASE 71
+# define ONYX_OUTPHASE_INVERTED (1<<0)
+
+#define ONYX_REG_ADC_CONTROL 72
+# define ONYX_ADC_INPUT_MIC (1<<5)
+/* 8 + input gain in dB, valid range for input gain is -4 .. 20 dB */
+# define ONYX_ADC_PGA_GAIN_MASK 0x1f
+
+#define ONYX_REG_ADC_HPF_BYPASS 75
+# define ONYX_HPF_DISABLE (1<<3)
+# define ONYX_ADC_HPF_ALWAYS (1<<2)
+
+#define ONYX_REG_DIG_INFO1 77
+# define ONYX_MASK_DIN_TO_BPZ (1<<7)
+/* bits 1-5 control channel bits 1-5 */
+# define ONYX_DIGOUT_DISABLE (1<<0)
+
+#define ONYX_REG_DIG_INFO2 78
+/* controls channel bits 8-15 */
+
+#define ONYX_REG_DIG_INFO3 79
+/* control channel bits 24-29, high 2 bits reserved */
+
+#define ONYX_REG_DIG_INFO4 80
+# define ONYX_VALIDL (1<<7)
+# define ONYX_VALIDR (1<<6)
+# define ONYX_SPDIF_ENABLE (1<<5)
+/* lower 4 bits control bits 32-35 of channel control and word length */
+# define ONYX_WORDLEN_MASK (0xF)
+
+#endif /* __SND_AOA_CODEC_ONYX_H */
diff --git a/sound/aoa/codecs/tas-basstreble.h b/sound/aoa/codecs/tas-basstreble.h
new file mode 100644
index 000000000..69b61136f
--- /dev/null
+++ b/sound/aoa/codecs/tas-basstreble.h
@@ -0,0 +1,134 @@
+/*
+ * This file is only included exactly once!
+ *
+ * The tables here are derived from the tas3004 datasheet,
+ * modulo typo corrections and some smoothing...
+ */
+
+#define TAS3004_TREBLE_MIN 0
+#define TAS3004_TREBLE_MAX 72
+#define TAS3004_BASS_MIN 0
+#define TAS3004_BASS_MAX 72
+#define TAS3004_TREBLE_ZERO 36
+#define TAS3004_BASS_ZERO 36
+
+static u8 tas3004_treble_table[] = {
+ 150, /* -18 dB */
+ 149,
+ 148,
+ 147,
+ 146,
+ 145,
+ 144,
+ 143,
+ 142,
+ 141,
+ 140,
+ 139,
+ 138,
+ 137,
+ 136,
+ 135,
+ 134,
+ 133,
+ 132,
+ 131,
+ 130,
+ 129,
+ 128,
+ 127,
+ 126,
+ 125,
+ 124,
+ 123,
+ 122,
+ 121,
+ 120,
+ 119,
+ 118,
+ 117,
+ 116,
+ 115,
+ 114, /* 0 dB */
+ 113,
+ 112,
+ 111,
+ 109,
+ 108,
+ 107,
+ 105,
+ 104,
+ 103,
+ 101,
+ 99,
+ 98,
+ 96,
+ 93,
+ 91,
+ 89,
+ 86,
+ 83,
+ 81,
+ 77,
+ 74,
+ 71,
+ 67,
+ 63,
+ 59,
+ 54,
+ 49,
+ 44,
+ 38,
+ 32,
+ 26,
+ 19,
+ 10,
+ 4,
+ 2,
+ 1, /* +18 dB */
+};
+
+static inline u8 tas3004_treble(int idx)
+{
+ return tas3004_treble_table[idx];
+}
+
+/* I only save the difference here to the treble table
+ * so that the binary is smaller...
+ * I have also ignored completely differences of
+ * +/- 1
+ */
+static s8 tas3004_bass_diff_to_treble[] = {
+ 2, /* 7 dB, offset 50 */
+ 2,
+ 2,
+ 2,
+ 2,
+ 1,
+ 2,
+ 2,
+ 2,
+ 3,
+ 4,
+ 4,
+ 5,
+ 6,
+ 7,
+ 8,
+ 9,
+ 10,
+ 11,
+ 14,
+ 13,
+ 8,
+ 1, /* 18 dB */
+};
+
+static inline u8 tas3004_bass(int idx)
+{
+ u8 result = tas3004_treble_table[idx];
+
+ if (idx >= 50)
+ result += tas3004_bass_diff_to_treble[idx-50];
+ return result;
+}
diff --git a/sound/aoa/codecs/tas-gain-table.h b/sound/aoa/codecs/tas-gain-table.h
new file mode 100644
index 000000000..4cfa67577
--- /dev/null
+++ b/sound/aoa/codecs/tas-gain-table.h
@@ -0,0 +1,209 @@
+/*
+ This is the program used to generate below table.
+
+#include <stdio.h>
+#include <math.h>
+int main() {
+ int dB2;
+ printf("/" "* This file is only included exactly once!\n");
+ printf(" *\n");
+ printf(" * If they'd only tell us that generating this table was\n");
+ printf(" * as easy as calculating\n");
+ printf(" * hwvalue = 1048576.0*exp(0.057564628*dB*2)\n");
+ printf(" * :) *" "/\n");
+ printf("static int tas_gaintable[] = {\n");
+ printf(" 0x000000, /" "* -infinity dB *" "/\n");
+ for (dB2=-140;dB2<=36;dB2++)
+ printf(" 0x%.6x, /" "* %-02.1f dB *" "/\n", (int)(1048576.0*exp(0.057564628*dB2)), dB2/2.0);
+ printf("};\n\n");
+}
+
+*/
+
+/* This file is only included exactly once!
+ *
+ * If they'd only tell us that generating this table was
+ * as easy as calculating
+ * hwvalue = 1048576.0*exp(0.057564628*dB*2)
+ * :) */
+static int tas_gaintable[] = {
+ 0x000000, /* -infinity dB */
+ 0x00014b, /* -70.0 dB */
+ 0x00015f, /* -69.5 dB */
+ 0x000174, /* -69.0 dB */
+ 0x00018a, /* -68.5 dB */
+ 0x0001a1, /* -68.0 dB */
+ 0x0001ba, /* -67.5 dB */
+ 0x0001d4, /* -67.0 dB */
+ 0x0001f0, /* -66.5 dB */
+ 0x00020d, /* -66.0 dB */
+ 0x00022c, /* -65.5 dB */
+ 0x00024d, /* -65.0 dB */
+ 0x000270, /* -64.5 dB */
+ 0x000295, /* -64.0 dB */
+ 0x0002bc, /* -63.5 dB */
+ 0x0002e6, /* -63.0 dB */
+ 0x000312, /* -62.5 dB */
+ 0x000340, /* -62.0 dB */
+ 0x000372, /* -61.5 dB */
+ 0x0003a6, /* -61.0 dB */
+ 0x0003dd, /* -60.5 dB */
+ 0x000418, /* -60.0 dB */
+ 0x000456, /* -59.5 dB */
+ 0x000498, /* -59.0 dB */
+ 0x0004de, /* -58.5 dB */
+ 0x000528, /* -58.0 dB */
+ 0x000576, /* -57.5 dB */
+ 0x0005c9, /* -57.0 dB */
+ 0x000620, /* -56.5 dB */
+ 0x00067d, /* -56.0 dB */
+ 0x0006e0, /* -55.5 dB */
+ 0x000748, /* -55.0 dB */
+ 0x0007b7, /* -54.5 dB */
+ 0x00082c, /* -54.0 dB */
+ 0x0008a8, /* -53.5 dB */
+ 0x00092b, /* -53.0 dB */
+ 0x0009b6, /* -52.5 dB */
+ 0x000a49, /* -52.0 dB */
+ 0x000ae5, /* -51.5 dB */
+ 0x000b8b, /* -51.0 dB */
+ 0x000c3a, /* -50.5 dB */
+ 0x000cf3, /* -50.0 dB */
+ 0x000db8, /* -49.5 dB */
+ 0x000e88, /* -49.0 dB */
+ 0x000f64, /* -48.5 dB */
+ 0x00104e, /* -48.0 dB */
+ 0x001145, /* -47.5 dB */
+ 0x00124b, /* -47.0 dB */
+ 0x001361, /* -46.5 dB */
+ 0x001487, /* -46.0 dB */
+ 0x0015be, /* -45.5 dB */
+ 0x001708, /* -45.0 dB */
+ 0x001865, /* -44.5 dB */
+ 0x0019d8, /* -44.0 dB */
+ 0x001b60, /* -43.5 dB */
+ 0x001cff, /* -43.0 dB */
+ 0x001eb7, /* -42.5 dB */
+ 0x002089, /* -42.0 dB */
+ 0x002276, /* -41.5 dB */
+ 0x002481, /* -41.0 dB */
+ 0x0026ab, /* -40.5 dB */
+ 0x0028f5, /* -40.0 dB */
+ 0x002b63, /* -39.5 dB */
+ 0x002df5, /* -39.0 dB */
+ 0x0030ae, /* -38.5 dB */
+ 0x003390, /* -38.0 dB */
+ 0x00369e, /* -37.5 dB */
+ 0x0039db, /* -37.0 dB */
+ 0x003d49, /* -36.5 dB */
+ 0x0040ea, /* -36.0 dB */
+ 0x0044c3, /* -35.5 dB */
+ 0x0048d6, /* -35.0 dB */
+ 0x004d27, /* -34.5 dB */
+ 0x0051b9, /* -34.0 dB */
+ 0x005691, /* -33.5 dB */
+ 0x005bb2, /* -33.0 dB */
+ 0x006121, /* -32.5 dB */
+ 0x0066e3, /* -32.0 dB */
+ 0x006cfb, /* -31.5 dB */
+ 0x007370, /* -31.0 dB */
+ 0x007a48, /* -30.5 dB */
+ 0x008186, /* -30.0 dB */
+ 0x008933, /* -29.5 dB */
+ 0x009154, /* -29.0 dB */
+ 0x0099f1, /* -28.5 dB */
+ 0x00a310, /* -28.0 dB */
+ 0x00acba, /* -27.5 dB */
+ 0x00b6f6, /* -27.0 dB */
+ 0x00c1cd, /* -26.5 dB */
+ 0x00cd49, /* -26.0 dB */
+ 0x00d973, /* -25.5 dB */
+ 0x00e655, /* -25.0 dB */
+ 0x00f3fb, /* -24.5 dB */
+ 0x010270, /* -24.0 dB */
+ 0x0111c0, /* -23.5 dB */
+ 0x0121f9, /* -23.0 dB */
+ 0x013328, /* -22.5 dB */
+ 0x01455b, /* -22.0 dB */
+ 0x0158a2, /* -21.5 dB */
+ 0x016d0e, /* -21.0 dB */
+ 0x0182af, /* -20.5 dB */
+ 0x019999, /* -20.0 dB */
+ 0x01b1de, /* -19.5 dB */
+ 0x01cb94, /* -19.0 dB */
+ 0x01e6cf, /* -18.5 dB */
+ 0x0203a7, /* -18.0 dB */
+ 0x022235, /* -17.5 dB */
+ 0x024293, /* -17.0 dB */
+ 0x0264db, /* -16.5 dB */
+ 0x02892c, /* -16.0 dB */
+ 0x02afa3, /* -15.5 dB */
+ 0x02d862, /* -15.0 dB */
+ 0x03038a, /* -14.5 dB */
+ 0x033142, /* -14.0 dB */
+ 0x0361af, /* -13.5 dB */
+ 0x0394fa, /* -13.0 dB */
+ 0x03cb50, /* -12.5 dB */
+ 0x0404de, /* -12.0 dB */
+ 0x0441d5, /* -11.5 dB */
+ 0x048268, /* -11.0 dB */
+ 0x04c6d0, /* -10.5 dB */
+ 0x050f44, /* -10.0 dB */
+ 0x055c04, /* -9.5 dB */
+ 0x05ad50, /* -9.0 dB */
+ 0x06036e, /* -8.5 dB */
+ 0x065ea5, /* -8.0 dB */
+ 0x06bf44, /* -7.5 dB */
+ 0x07259d, /* -7.0 dB */
+ 0x079207, /* -6.5 dB */
+ 0x0804dc, /* -6.0 dB */
+ 0x087e80, /* -5.5 dB */
+ 0x08ff59, /* -5.0 dB */
+ 0x0987d5, /* -4.5 dB */
+ 0x0a1866, /* -4.0 dB */
+ 0x0ab189, /* -3.5 dB */
+ 0x0b53be, /* -3.0 dB */
+ 0x0bff91, /* -2.5 dB */
+ 0x0cb591, /* -2.0 dB */
+ 0x0d765a, /* -1.5 dB */
+ 0x0e4290, /* -1.0 dB */
+ 0x0f1adf, /* -0.5 dB */
+ 0x100000, /* 0.0 dB */
+ 0x10f2b4, /* 0.5 dB */
+ 0x11f3c9, /* 1.0 dB */
+ 0x13041a, /* 1.5 dB */
+ 0x14248e, /* 2.0 dB */
+ 0x15561a, /* 2.5 dB */
+ 0x1699c0, /* 3.0 dB */
+ 0x17f094, /* 3.5 dB */
+ 0x195bb8, /* 4.0 dB */
+ 0x1adc61, /* 4.5 dB */
+ 0x1c73d5, /* 5.0 dB */
+ 0x1e236d, /* 5.5 dB */
+ 0x1fec98, /* 6.0 dB */
+ 0x21d0d9, /* 6.5 dB */
+ 0x23d1cd, /* 7.0 dB */
+ 0x25f125, /* 7.5 dB */
+ 0x2830af, /* 8.0 dB */
+ 0x2a9254, /* 8.5 dB */
+ 0x2d1818, /* 9.0 dB */
+ 0x2fc420, /* 9.5 dB */
+ 0x3298b0, /* 10.0 dB */
+ 0x35982f, /* 10.5 dB */
+ 0x38c528, /* 11.0 dB */
+ 0x3c224c, /* 11.5 dB */
+ 0x3fb278, /* 12.0 dB */
+ 0x4378b0, /* 12.5 dB */
+ 0x477829, /* 13.0 dB */
+ 0x4bb446, /* 13.5 dB */
+ 0x5030a1, /* 14.0 dB */
+ 0x54f106, /* 14.5 dB */
+ 0x59f980, /* 15.0 dB */
+ 0x5f4e52, /* 15.5 dB */
+ 0x64f403, /* 16.0 dB */
+ 0x6aef5e, /* 16.5 dB */
+ 0x714575, /* 17.0 dB */
+ 0x77fbaa, /* 17.5 dB */
+ 0x7f17af, /* 18.0 dB */
+};
+
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
new file mode 100644
index 000000000..364c7c441
--- /dev/null
+++ b/sound/aoa/codecs/tas.c
@@ -0,0 +1,949 @@
+/*
+ * Apple Onboard Audio driver for tas codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ * Open questions:
+ * - How to distinguish between 3004 and versions?
+ *
+ * FIXMEs:
+ * - This codec driver doesn't honour the 'connected'
+ * property of the aoa_codec struct, hence if
+ * it is used in machines where not everything is
+ * connected it will display wrong mixer elements.
+ * - Driver assumes that the microphone is always
+ * monaureal and connected to the right channel of
+ * the input. This should also be a codec-dependent
+ * flag, maybe the codec should have 3 different
+ * bits for the three different possibilities how
+ * it can be hooked up...
+ * But as long as I don't see any hardware hooked
+ * up that way...
+ * - As Apple notes in their code, the tas3004 seems
+ * to delay the right channel by one sample. You can
+ * see this when for example recording stereo in
+ * audacity, or recording the tas output via cable
+ * on another machine (use a sinus generator or so).
+ * I tried programming the BiQuads but couldn't
+ * make the delay work, maybe someone can read the
+ * datasheet and fix it. The relevant Apple comment
+ * is in AppleTAS3004Audio.cpp lines 1637 ff. Note
+ * that their comment describing how they program
+ * the filters sucks...
+ *
+ * Other things:
+ * - this should actually register *two* aoa_codec
+ * structs since it has two inputs. Then it must
+ * use the prepare callback to forbid running the
+ * secondary output on a different clock.
+ * Also, whatever bus knows how to do this must
+ * provide two soundbus_dev devices and the fabric
+ * must be able to link them correctly.
+ *
+ * I don't even know if Apple ever uses the second
+ * port on the tas3004 though, I don't think their
+ * i2s controllers can even do it. OTOH, they all
+ * derive the clocks from common clocks, so it
+ * might just be possible. The framework allows the
+ * codec to refine the transfer_info items in the
+ * usable callback, so we can simply remove the
+ * rates the second instance is not using when it
+ * actually is in use.
+ * Maybe we'll need to make the sound busses have
+ * a 'clock group id' value so the codec can
+ * determine if the two outputs can be driven at
+ * the same time. But that is likely overkill, up
+ * to the fabric to not link them up incorrectly,
+ * and up to the hardware designer to not wire
+ * them up in some weird unusable way.
+ */
+#include <stddef.h>
+#include <linux/i2c.h>
+#include <asm/pmac_low_i2c.h>
+#include <asm/prom.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/mutex.h>
+#include <linux/slab.h>
+
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("tas codec driver for snd-aoa");
+
+#include "tas.h"
+#include "tas-gain-table.h"
+#include "tas-basstreble.h"
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+#define PFX "snd-aoa-codec-tas: "
+
+
+struct tas {
+ struct aoa_codec codec;
+ struct i2c_client *i2c;
+ u32 mute_l:1, mute_r:1 ,
+ controls_created:1 ,
+ drc_enabled:1,
+ hw_enabled:1;
+ u8 cached_volume_l, cached_volume_r;
+ u8 mixer_l[3], mixer_r[3];
+ u8 bass, treble;
+ u8 acr;
+ int drc_range;
+ /* protects hardware access against concurrency from
+ * userspace when hitting controls and during
+ * codec init/suspend/resume */
+ struct mutex mtx;
+};
+
+static int tas_reset_init(struct tas *tas);
+
+static struct tas *codec_to_tas(struct aoa_codec *codec)
+{
+ return container_of(codec, struct tas, codec);
+}
+
+static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data)
+{
+ if (len == 1)
+ return i2c_smbus_write_byte_data(tas->i2c, reg, *data);
+ else
+ return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data);
+}
+
+static void tas3004_set_drc(struct tas *tas)
+{
+ unsigned char val[6];
+
+ if (tas->drc_enabled)
+ val[0] = 0x50; /* 3:1 above threshold */
+ else
+ val[0] = 0x51; /* disabled */
+ val[1] = 0x02; /* 1:1 below threshold */
+ if (tas->drc_range > 0xef)
+ val[2] = 0xef;
+ else if (tas->drc_range < 0)
+ val[2] = 0x00;
+ else
+ val[2] = tas->drc_range;
+ val[3] = 0xb0;
+ val[4] = 0x60;
+ val[5] = 0xa0;
+
+ tas_write_reg(tas, TAS_REG_DRC, 6, val);
+}
+
+static void tas_set_treble(struct tas *tas)
+{
+ u8 tmp;
+
+ tmp = tas3004_treble(tas->treble);
+ tas_write_reg(tas, TAS_REG_TREBLE, 1, &tmp);
+}
+
+static void tas_set_bass(struct tas *tas)
+{
+ u8 tmp;
+
+ tmp = tas3004_bass(tas->bass);
+ tas_write_reg(tas, TAS_REG_BASS, 1, &tmp);
+}
+
+static void tas_set_volume(struct tas *tas)
+{
+ u8 block[6];
+ int tmp;
+ u8 left, right;
+
+ left = tas->cached_volume_l;
+ right = tas->cached_volume_r;
+
+ if (left > 177) left = 177;
+ if (right > 177) right = 177;
+
+ if (tas->mute_l) left = 0;
+ if (tas->mute_r) right = 0;
+
+ /* analysing the volume and mixer tables shows
+ * that they are similar enough when we shift
+ * the mixer table down by 4 bits. The error
+ * is miniscule, in just one item the error
+ * is 1, at a value of 0x07f17b (mixer table
+ * value is 0x07f17a) */
+ tmp = tas_gaintable[left];
+ block[0] = tmp>>20;
+ block[1] = tmp>>12;
+ block[2] = tmp>>4;
+ tmp = tas_gaintable[right];
+ block[3] = tmp>>20;
+ block[4] = tmp>>12;
+ block[5] = tmp>>4;
+ tas_write_reg(tas, TAS_REG_VOL, 6, block);
+}
+
+static void tas_set_mixer(struct tas *tas)
+{
+ u8 block[9];
+ int tmp, i;
+ u8 val;
+
+ for (i=0;i<3;i++) {
+ val = tas->mixer_l[i];
+ if (val > 177) val = 177;
+ tmp = tas_gaintable[val];
+ block[3*i+0] = tmp>>16;
+ block[3*i+1] = tmp>>8;
+ block[3*i+2] = tmp;
+ }
+ tas_write_reg(tas, TAS_REG_LMIX, 9, block);
+
+ for (i=0;i<3;i++) {
+ val = tas->mixer_r[i];
+ if (val > 177) val = 177;
+ tmp = tas_gaintable[val];
+ block[3*i+0] = tmp>>16;
+ block[3*i+1] = tmp>>8;
+ block[3*i+2] = tmp;
+ }
+ tas_write_reg(tas, TAS_REG_RMIX, 9, block);
+}
+
+/* alsa stuff */
+
+static int tas_dev_register(struct snd_device *dev)
+{
+ return 0;
+}
+
+static struct snd_device_ops ops = {
+ .dev_register = tas_dev_register,
+};
+
+static int tas_snd_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 177;
+ return 0;
+}
+
+static int tas_snd_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->cached_volume_l;
+ ucontrol->value.integer.value[1] = tas->cached_volume_r;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] > 177)
+ return -EINVAL;
+ if (ucontrol->value.integer.value[1] < 0 ||
+ ucontrol->value.integer.value[1] > 177)
+ return -EINVAL;
+
+ mutex_lock(&tas->mtx);
+ if (tas->cached_volume_l == ucontrol->value.integer.value[0]
+ && tas->cached_volume_r == ucontrol->value.integer.value[1]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->cached_volume_l = ucontrol->value.integer.value[0];
+ tas->cached_volume_r = ucontrol->value.integer.value[1];
+ if (tas->hw_enabled)
+ tas_set_volume(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new volume_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_vol_info,
+ .get = tas_snd_vol_get,
+ .put = tas_snd_vol_put,
+};
+
+#define tas_snd_mute_info snd_ctl_boolean_stereo_info
+
+static int tas_snd_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = !tas->mute_l;
+ ucontrol->value.integer.value[1] = !tas->mute_r;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ if (tas->mute_l == !ucontrol->value.integer.value[0]
+ && tas->mute_r == !ucontrol->value.integer.value[1]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->mute_l = !ucontrol->value.integer.value[0];
+ tas->mute_r = !ucontrol->value.integer.value[1];
+ if (tas->hw_enabled)
+ tas_set_volume(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new mute_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_mute_info,
+ .get = tas_snd_mute_get,
+ .put = tas_snd_mute_put,
+};
+
+static int tas_snd_mixer_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 177;
+ return 0;
+}
+
+static int tas_snd_mixer_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+ int idx = kcontrol->private_value;
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->mixer_l[idx];
+ ucontrol->value.integer.value[1] = tas->mixer_r[idx];
+ mutex_unlock(&tas->mtx);
+
+ return 0;
+}
+
+static int tas_snd_mixer_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+ int idx = kcontrol->private_value;
+
+ mutex_lock(&tas->mtx);
+ if (tas->mixer_l[idx] == ucontrol->value.integer.value[0]
+ && tas->mixer_r[idx] == ucontrol->value.integer.value[1]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->mixer_l[idx] = ucontrol->value.integer.value[0];
+ tas->mixer_r[idx] = ucontrol->value.integer.value[1];
+
+ if (tas->hw_enabled)
+ tas_set_mixer(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+#define MIXER_CONTROL(n,descr,idx) \
+static struct snd_kcontrol_new n##_control = { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = descr " Playback Volume", \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = tas_snd_mixer_info, \
+ .get = tas_snd_mixer_get, \
+ .put = tas_snd_mixer_put, \
+ .private_value = idx, \
+}
+
+MIXER_CONTROL(pcm1, "PCM", 0);
+MIXER_CONTROL(monitor, "Monitor", 2);
+
+static int tas_snd_drc_range_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = TAS3004_DRC_MAX;
+ return 0;
+}
+
+static int tas_snd_drc_range_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->drc_range;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] > TAS3004_DRC_MAX)
+ return -EINVAL;
+
+ mutex_lock(&tas->mtx);
+ if (tas->drc_range == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->drc_range = ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas3004_set_drc(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new drc_range_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DRC Range",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_drc_range_info,
+ .get = tas_snd_drc_range_get,
+ .put = tas_snd_drc_range_put,
+};
+
+#define tas_snd_drc_switch_info snd_ctl_boolean_mono_info
+
+static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->drc_enabled;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ if (tas->drc_enabled == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->drc_enabled = !!ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas3004_set_drc(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new drc_switch_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DRC Range Switch",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_drc_switch_info,
+ .get = tas_snd_drc_switch_get,
+ .put = tas_snd_drc_switch_put,
+};
+
+static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = { "Line-In", "Microphone" };
+
+ return snd_ctl_enum_info(uinfo, 1, 2, texts);
+}
+
+static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.enumerated.item[0] = !!(tas->acr & TAS_ACR_INPUT_B);
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+ int oldacr;
+
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ mutex_lock(&tas->mtx);
+ oldacr = tas->acr;
+
+ /*
+ * Despite what the data sheet says in one place, the
+ * TAS_ACR_B_MONAUREAL bit forces mono output even when
+ * input A (line in) is selected.
+ */
+ tas->acr &= ~(TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL);
+ if (ucontrol->value.enumerated.item[0])
+ tas->acr |= TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL |
+ TAS_ACR_B_MON_SEL_RIGHT;
+ if (oldacr == tas->acr) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+ if (tas->hw_enabled)
+ tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new capture_source_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* If we name this 'Input Source', it properly shows up in
+ * alsamixer as a selection, * but it's shown under the
+ * 'Playback' category.
+ * If I name it 'Capture Source', it shows up in strange
+ * ways (two bools of which one can be selected at a
+ * time) but at least it's shown in the 'Capture'
+ * category.
+ * I was told that this was due to backward compatibility,
+ * but I don't understand then why the mangling is *not*
+ * done when I name it "Input Source".....
+ */
+ .name = "Capture Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_capture_source_info,
+ .get = tas_snd_capture_source_get,
+ .put = tas_snd_capture_source_put,
+};
+
+static int tas_snd_treble_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = TAS3004_TREBLE_MIN;
+ uinfo->value.integer.max = TAS3004_TREBLE_MAX;
+ return 0;
+}
+
+static int tas_snd_treble_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->treble;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_treble_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < TAS3004_TREBLE_MIN ||
+ ucontrol->value.integer.value[0] > TAS3004_TREBLE_MAX)
+ return -EINVAL;
+ mutex_lock(&tas->mtx);
+ if (tas->treble == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->treble = ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas_set_treble(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new treble_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Treble",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_treble_info,
+ .get = tas_snd_treble_get,
+ .put = tas_snd_treble_put,
+};
+
+static int tas_snd_bass_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = TAS3004_BASS_MIN;
+ uinfo->value.integer.max = TAS3004_BASS_MAX;
+ return 0;
+}
+
+static int tas_snd_bass_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->bass;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_bass_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < TAS3004_BASS_MIN ||
+ ucontrol->value.integer.value[0] > TAS3004_BASS_MAX)
+ return -EINVAL;
+ mutex_lock(&tas->mtx);
+ if (tas->bass == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->bass = ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas_set_bass(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new bass_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Bass",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_bass_info,
+ .get = tas_snd_bass_get,
+ .put = tas_snd_bass_put,
+};
+
+static struct transfer_info tas_transfers[] = {
+ {
+ /* input */
+ .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .transfer_in = 1,
+ },
+ {
+ /* output */
+ .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .transfer_in = 0,
+ },
+ {}
+};
+
+static int tas_usable(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out)
+{
+ return 1;
+}
+
+static int tas_reset_init(struct tas *tas)
+{
+ u8 tmp;
+
+ tas->codec.gpio->methods->all_amps_off(tas->codec.gpio);
+ msleep(5);
+ tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
+ msleep(5);
+ tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 1);
+ msleep(20);
+ tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
+ msleep(10);
+ tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio);
+
+ tmp = TAS_MCS_SCLK64 | TAS_MCS_SPORT_MODE_I2S | TAS_MCS_SPORT_WL_24BIT;
+ if (tas_write_reg(tas, TAS_REG_MCS, 1, &tmp))
+ goto outerr;
+
+ tas->acr |= TAS_ACR_ANALOG_PDOWN;
+ if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr))
+ goto outerr;
+
+ tmp = 0;
+ if (tas_write_reg(tas, TAS_REG_MCS2, 1, &tmp))
+ goto outerr;
+
+ tas3004_set_drc(tas);
+
+ /* Set treble & bass to 0dB */
+ tas->treble = TAS3004_TREBLE_ZERO;
+ tas->bass = TAS3004_BASS_ZERO;
+ tas_set_treble(tas);
+ tas_set_bass(tas);
+
+ tas->acr &= ~TAS_ACR_ANALOG_PDOWN;
+ if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr))
+ goto outerr;
+
+ return 0;
+ outerr:
+ return -ENODEV;
+}
+
+static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock)
+{
+ struct tas *tas = cii->codec_data;
+
+ switch(clock) {
+ case CLOCK_SWITCH_PREPARE_SLAVE:
+ /* Clocks are going away, mute mute mute */
+ tas->codec.gpio->methods->all_amps_off(tas->codec.gpio);
+ tas->hw_enabled = 0;
+ break;
+ case CLOCK_SWITCH_SLAVE:
+ /* Clocks are back, re-init the codec */
+ mutex_lock(&tas->mtx);
+ tas_reset_init(tas);
+ tas_set_volume(tas);
+ tas_set_mixer(tas);
+ tas->hw_enabled = 1;
+ tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio);
+ mutex_unlock(&tas->mtx);
+ break;
+ default:
+ /* doesn't happen as of now */
+ return -EINVAL;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_PM
+/* we are controlled via i2c and assume that is always up
+ * If that wasn't the case, we'd have to suspend once
+ * our i2c device is suspended, and then take note of that! */
+static int tas_suspend(struct tas *tas)
+{
+ mutex_lock(&tas->mtx);
+ tas->hw_enabled = 0;
+ tas->acr |= TAS_ACR_ANALOG_PDOWN;
+ tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_resume(struct tas *tas)
+{
+ /* reset codec */
+ mutex_lock(&tas->mtx);
+ tas_reset_init(tas);
+ tas_set_volume(tas);
+ tas_set_mixer(tas);
+ tas->hw_enabled = 1;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int _tas_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+ return tas_suspend(cii->codec_data);
+}
+
+static int _tas_resume(struct codec_info_item *cii)
+{
+ return tas_resume(cii->codec_data);
+}
+#else /* CONFIG_PM */
+#define _tas_suspend NULL
+#define _tas_resume NULL
+#endif /* CONFIG_PM */
+
+static struct codec_info tas_codec_info = {
+ .transfers = tas_transfers,
+ /* in theory, we can drive it at 512 too...
+ * but so far the framework doesn't allow
+ * for that and I don't see much point in it. */
+ .sysclock_factor = 256,
+ /* same here, could be 32 for just one 16 bit format */
+ .bus_factor = 64,
+ .owner = THIS_MODULE,
+ .usable = tas_usable,
+ .switch_clock = tas_switch_clock,
+ .suspend = _tas_suspend,
+ .resume = _tas_resume,
+};
+
+static int tas_init_codec(struct aoa_codec *codec)
+{
+ struct tas *tas = codec_to_tas(codec);
+ int err;
+
+ if (!tas->codec.gpio || !tas->codec.gpio->methods) {
+ printk(KERN_ERR PFX "gpios not assigned!!\n");
+ return -EINVAL;
+ }
+
+ mutex_lock(&tas->mtx);
+ if (tas_reset_init(tas)) {
+ printk(KERN_ERR PFX "tas failed to initialise\n");
+ mutex_unlock(&tas->mtx);
+ return -ENXIO;
+ }
+ tas->hw_enabled = 1;
+ mutex_unlock(&tas->mtx);
+
+ if (tas->codec.soundbus_dev->attach_codec(tas->codec.soundbus_dev,
+ aoa_get_card(),
+ &tas_codec_info, tas)) {
+ printk(KERN_ERR PFX "error attaching tas to soundbus\n");
+ return -ENODEV;
+ }
+
+ if (aoa_snd_device_new(SNDRV_DEV_CODEC, tas, &ops)) {
+ printk(KERN_ERR PFX "failed to create tas snd device!\n");
+ return -ENODEV;
+ }
+ err = aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&drc_range_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&drc_switch_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&treble_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&bass_control, tas));
+ if (err)
+ goto error;
+
+ return 0;
+ error:
+ tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
+ snd_device_free(aoa_get_card(), tas);
+ return err;
+}
+
+static void tas_exit_codec(struct aoa_codec *codec)
+{
+ struct tas *tas = codec_to_tas(codec);
+
+ if (!tas->codec.soundbus_dev)
+ return;
+ tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
+}
+
+
+static int tas_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct device_node *node = client->dev.of_node;
+ struct tas *tas;
+
+ tas = kzalloc(sizeof(struct tas), GFP_KERNEL);
+
+ if (!tas)
+ return -ENOMEM;
+
+ mutex_init(&tas->mtx);
+ tas->i2c = client;
+ i2c_set_clientdata(client, tas);
+
+ /* seems that half is a saner default */
+ tas->drc_range = TAS3004_DRC_MAX / 2;
+
+ strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN);
+ tas->codec.owner = THIS_MODULE;
+ tas->codec.init = tas_init_codec;
+ tas->codec.exit = tas_exit_codec;
+ tas->codec.node = of_node_get(node);
+
+ if (aoa_codec_register(&tas->codec)) {
+ goto fail;
+ }
+ printk(KERN_DEBUG
+ "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n",
+ (unsigned int)client->addr, node->full_name);
+ return 0;
+ fail:
+ mutex_destroy(&tas->mtx);
+ kfree(tas);
+ return -EINVAL;
+}
+
+static int tas_i2c_remove(struct i2c_client *client)
+{
+ struct tas *tas = i2c_get_clientdata(client);
+ u8 tmp = TAS_ACR_ANALOG_PDOWN;
+
+ aoa_codec_unregister(&tas->codec);
+ of_node_put(tas->codec.node);
+
+ /* power down codec chip */
+ tas_write_reg(tas, TAS_REG_ACR, 1, &tmp);
+
+ mutex_destroy(&tas->mtx);
+ kfree(tas);
+ return 0;
+}
+
+static const struct i2c_device_id tas_i2c_id[] = {
+ { "MAC,tas3004", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c,tas_i2c_id);
+
+static struct i2c_driver tas_driver = {
+ .driver = {
+ .name = "aoa_codec_tas",
+ .owner = THIS_MODULE,
+ },
+ .probe = tas_i2c_probe,
+ .remove = tas_i2c_remove,
+ .id_table = tas_i2c_id,
+};
+
+module_i2c_driver(tas_driver);
diff --git a/sound/aoa/codecs/tas.h b/sound/aoa/codecs/tas.h
new file mode 100644
index 000000000..ae177e346
--- /dev/null
+++ b/sound/aoa/codecs/tas.h
@@ -0,0 +1,55 @@
+/*
+ * Apple Onboard Audio driver for tas codec (header)
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SND_AOA_CODECTASH
+#define __SND_AOA_CODECTASH
+
+#define TAS_REG_MCS 0x01 /* main control */
+# define TAS_MCS_FASTLOAD (1<<7)
+# define TAS_MCS_SCLK64 (1<<6)
+# define TAS_MCS_SPORT_MODE_MASK (3<<4)
+# define TAS_MCS_SPORT_MODE_I2S (2<<4)
+# define TAS_MCS_SPORT_MODE_RJ (1<<4)
+# define TAS_MCS_SPORT_MODE_LJ (0<<4)
+# define TAS_MCS_SPORT_WL_MASK (3<<0)
+# define TAS_MCS_SPORT_WL_16BIT (0<<0)
+# define TAS_MCS_SPORT_WL_18BIT (1<<0)
+# define TAS_MCS_SPORT_WL_20BIT (2<<0)
+# define TAS_MCS_SPORT_WL_24BIT (3<<0)
+
+#define TAS_REG_DRC 0x02
+#define TAS_REG_VOL 0x04
+#define TAS_REG_TREBLE 0x05
+#define TAS_REG_BASS 0x06
+#define TAS_REG_LMIX 0x07
+#define TAS_REG_RMIX 0x08
+
+#define TAS_REG_ACR 0x40 /* analog control */
+# define TAS_ACR_B_MONAUREAL (1<<7)
+# define TAS_ACR_B_MON_SEL_RIGHT (1<<6)
+# define TAS_ACR_DEEMPH_MASK (3<<2)
+# define TAS_ACR_DEEMPH_OFF (0<<2)
+# define TAS_ACR_DEEMPH_48KHz (1<<2)
+# define TAS_ACR_DEEMPH_44KHz (2<<2)
+# define TAS_ACR_INPUT_B (1<<1)
+# define TAS_ACR_ANALOG_PDOWN (1<<0)
+
+#define TAS_REG_MCS2 0x43 /* main control 2 */
+# define TAS_MCS2_ALLPASS (1<<1)
+
+#define TAS_REG_LEFT_BIQUAD6 0x10
+#define TAS_REG_RIGHT_BIQUAD6 0x19
+
+#define TAS_REG_LEFT_LOUDNESS 0x21
+#define TAS_REG_RIGHT_LOUDNESS 0x22
+#define TAS_REG_LEFT_LOUDNESS_GAIN 0x23
+#define TAS_REG_RIGHT_LOUDNESS_GAIN 0x24
+
+#define TAS3001_DRC_MAX 0x5f
+#define TAS3004_DRC_MAX 0xef
+
+#endif /* __SND_AOA_CODECTASH */
diff --git a/sound/aoa/codecs/toonie.c b/sound/aoa/codecs/toonie.c
new file mode 100644
index 000000000..7e8c3417c
--- /dev/null
+++ b/sound/aoa/codecs/toonie.c
@@ -0,0 +1,151 @@
+/*
+ * Apple Onboard Audio driver for Toonie codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ *
+ * This is a driver for the toonie codec chip. This chip is present
+ * on the Mac Mini and is nothing but a DAC.
+ */
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("toonie codec driver for snd-aoa");
+
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-toonie: "
+
+struct toonie {
+ struct aoa_codec codec;
+};
+#define codec_to_toonie(c) container_of(c, struct toonie, codec)
+
+static int toonie_dev_register(struct snd_device *dev)
+{
+ return 0;
+}
+
+static struct snd_device_ops ops = {
+ .dev_register = toonie_dev_register,
+};
+
+static struct transfer_info toonie_transfers[] = {
+ /* This thing *only* has analog output,
+ * the rates are taken from Info.plist
+ * from Darwin. */
+ {
+ .formats = SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ },
+ {}
+};
+
+static int toonie_usable(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out)
+{
+ return 1;
+}
+
+#ifdef CONFIG_PM
+static int toonie_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+ /* can we turn it off somehow? */
+ return 0;
+}
+
+static int toonie_resume(struct codec_info_item *cii)
+{
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static struct codec_info toonie_codec_info = {
+ .transfers = toonie_transfers,
+ .sysclock_factor = 256,
+ .bus_factor = 64,
+ .owner = THIS_MODULE,
+ .usable = toonie_usable,
+#ifdef CONFIG_PM
+ .suspend = toonie_suspend,
+ .resume = toonie_resume,
+#endif
+};
+
+static int toonie_init_codec(struct aoa_codec *codec)
+{
+ struct toonie *toonie = codec_to_toonie(codec);
+
+ /* nothing connected? what a joke! */
+ if (toonie->codec.connected != 1)
+ return -ENOTCONN;
+
+ if (aoa_snd_device_new(SNDRV_DEV_CODEC, toonie, &ops)) {
+ printk(KERN_ERR PFX "failed to create toonie snd device!\n");
+ return -ENODEV;
+ }
+
+ if (toonie->codec.soundbus_dev->attach_codec(toonie->codec.soundbus_dev,
+ aoa_get_card(),
+ &toonie_codec_info, toonie)) {
+ printk(KERN_ERR PFX "error creating toonie pcm\n");
+ snd_device_free(aoa_get_card(), toonie);
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static void toonie_exit_codec(struct aoa_codec *codec)
+{
+ struct toonie *toonie = codec_to_toonie(codec);
+
+ if (!toonie->codec.soundbus_dev) {
+ printk(KERN_ERR PFX "toonie_exit_codec called without soundbus_dev!\n");
+ return;
+ }
+ toonie->codec.soundbus_dev->detach_codec(toonie->codec.soundbus_dev, toonie);
+}
+
+static struct toonie *toonie;
+
+static int __init toonie_init(void)
+{
+ toonie = kzalloc(sizeof(struct toonie), GFP_KERNEL);
+
+ if (!toonie)
+ return -ENOMEM;
+
+ strlcpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name));
+ toonie->codec.owner = THIS_MODULE;
+ toonie->codec.init = toonie_init_codec;
+ toonie->codec.exit = toonie_exit_codec;
+
+ if (aoa_codec_register(&toonie->codec)) {
+ kfree(toonie);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static void __exit toonie_exit(void)
+{
+ aoa_codec_unregister(&toonie->codec);
+ kfree(toonie);
+}
+
+module_init(toonie_init);
+module_exit(toonie_exit);
generated by cgit v1.2.3-54-g00ecf (git 2.45.2) at 2024-11-18 15:19:55 +0000