diff options
Diffstat (limited to '')
| -rw-r--r-- | sound/aoa/codecs/Kconfig | 32 | ||||
| -rw-r--r-- | sound/aoa/codecs/Makefile | 3 | ||||
| -rw-r--r-- | sound/aoa/codecs/snd-aoa-codec-onyx.c | 1113 | ||||
| -rw-r--r-- | sound/aoa/codecs/snd-aoa-codec-onyx.h | 76 | ||||
| -rw-r--r-- | sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h | 209 | ||||
| -rw-r--r-- | sound/aoa/codecs/snd-aoa-codec-tas.c | 654 | ||||
| -rw-r--r-- | sound/aoa/codecs/snd-aoa-codec-tas.h | 47 | ||||
| -rw-r--r-- | sound/aoa/codecs/snd-aoa-codec-toonie.c | 141 |
8 files changed, 2275 insertions, 0 deletions
diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig new file mode 100644 index 000000000000..90cf58f68630 --- /dev/null +++ b/sound/aoa/codecs/Kconfig @@ -0,0 +1,32 @@ +config SND_AOA_ONYX + tristate "support Onyx chip" + depends on SND_AOA + ---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_TOPAZ +# tristate "support Topaz chips" +# depends on SND_AOA +# ---help--- +# This option enables support for the Topaz (CS84xx) +# codec chips found in the latest Apple machines, +# these chips do the digital input and output on +# some PowerMacs. + +config SND_AOA_TAS + tristate "support TAS chips" + depends on SND_AOA + ---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" + depends on SND_AOA + ---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 000000000000..31cbe68fd42f --- /dev/null +++ b/sound/aoa/codecs/Makefile @@ -0,0 +1,3 @@ +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/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c new file mode 100644 index 000000000000..0b7650788f1f --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c @@ -0,0 +1,1113 @@ +/* + * 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> +MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); + +#include "snd-aoa-codec-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; + + 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; + + 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 char *texts[] = { "Line-In", "Microphone" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item > 1) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +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) +{ + 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, +}; + +static int onyx_snd_mute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +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, +}; + + +static int onyx_snd_single_bit_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +#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); + /* take codec out of suspend */ + 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_LOWLEVEL, 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 struct i2c_driver onyx_driver; + +static int onyx_create(struct i2c_adapter *adapter, + struct device_node *node, + int addr) +{ + struct onyx *onyx; + u8 dummy; + + onyx = kzalloc(sizeof(struct onyx), GFP_KERNEL); + + if (!onyx) + return -ENOMEM; + + mutex_init(&onyx->mutex); + onyx->i2c.driver = &onyx_driver; + onyx->i2c.adapter = adapter; + onyx->i2c.addr = addr & 0x7f; + strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE-1); + + if (i2c_attach_client(&onyx->i2c)) { + printk(KERN_ERR PFX "failed to attach to i2c\n"); + goto fail; + } + + /* 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) { + i2c_detach_client(&onyx->i2c); + printk(KERN_ERR PFX "failed to read control register\n"); + goto fail; + } + + strlcpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN-1); + 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)) { + i2c_detach_client(&onyx->i2c); + goto fail; + } + printk(KERN_DEBUG PFX "created and attached onyx instance\n"); + return 0; + fail: + kfree(onyx); + return -EINVAL; +} + +static int onyx_i2c_attach(struct i2c_adapter *adapter) +{ + struct device_node *busnode, *dev = NULL; + struct pmac_i2c_bus *bus; + + bus = pmac_i2c_adapter_to_bus(adapter); + if (bus == NULL) + return -ENODEV; + busnode = pmac_i2c_get_bus_node(bus); + + while ((dev = of_get_next_child(busnode, dev)) != NULL) { + if (device_is_compatible(dev, "pcm3052")) { + u32 *addr; + printk(KERN_DEBUG PFX "found pcm3052\n"); + addr = (u32 *) get_property(dev, "reg", NULL); + if (!addr) + return -ENODEV; + return onyx_create(adapter, dev, (*addr)>>1); + } + } + + /* if that didn't work, try desperate mode for older + * machines that have stuff missing from the device tree */ + + if (!device_is_compatible(busnode, "k2-i2c")) + return -ENODEV; + + printk(KERN_DEBUG PFX "found k2-i2c, checking if onyx chip is on it\n"); + /* probe both possible addresses for the onyx chip */ + if (onyx_create(adapter, NULL, 0x46) == 0) + return 0; + return onyx_create(adapter, NULL, 0x47); +} + +static int onyx_i2c_detach(struct i2c_client *client) +{ + struct onyx *onyx = container_of(client, struct onyx, i2c); + int err; + + if ((err = i2c_detach_client(client))) + return err; + aoa_codec_unregister(&onyx->codec); + of_node_put(onyx->codec.node); + if (onyx->codec_info) + kfree(onyx->codec_info); + kfree(onyx); + return 0; +} + +static struct i2c_driver onyx_driver = { + .driver = { + .name = "aoa_codec_onyx", + .owner = THIS_MODULE, + }, + .attach_adapter = onyx_i2c_attach, + .detach_client = onyx_i2c_detach, +}; + +static int __init onyx_init(void) +{ + return i2c_add_driver(&onyx_driver); +} + +static void __exit onyx_exit(void) +{ + i2c_del_driver(&onyx_driver); +} + +module_init(onyx_init); +module_exit(onyx_exit); diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.h b/sound/aoa/codecs/snd-aoa-codec-onyx.h new file mode 100644 index 000000000000..aeedda773699 --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.h @@ -0,0 +1,76 @@ +/* + * 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 <linux/i2c-dev.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/snd-aoa-codec-tas-gain-table.h b/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h new file mode 100644 index 000000000000..4cfa6757715e --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-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/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c new file mode 100644 index 000000000000..2e39ff6ee349 --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -0,0 +1,654 @@ +/* + * 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 <linux/i2c-dev.h> +#include <asm/pmac_low_i2c.h> +#include <asm/prom.h> +#include <linux/delay.h> +#include <linux/module.h> +MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("tas codec driver for snd-aoa"); + +#include "snd-aoa-codec-tas.h" +#include "snd-aoa-codec-tas-gain-table.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 muted_l:1, muted_r:1, + controls_created:1; + u8 cached_volume_l, cached_volume_r; + u8 mixer_l[3], mixer_r[3]; + u8 acr; +}; + +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 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->muted_l) left = 0; + if (tas->muted_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); + + ucontrol->value.integer.value[0] = tas->cached_volume_l; + ucontrol->value.integer.value[1] = tas->cached_volume_r; + 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 (tas->cached_volume_l == ucontrol->value.integer.value[0] + && tas->cached_volume_r == ucontrol->value.integer.value[1]) + return 0; + + tas->cached_volume_l = ucontrol->value.integer.value[0]; + tas->cached_volume_r = ucontrol->value.integer.value[1]; + tas_set_volume(tas); + 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, +}; + +static int tas_snd_mute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int tas_snd_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = !tas->muted_l; + ucontrol->value.integer.value[1] = !tas->muted_r; + 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); + + if (tas->muted_l == !ucontrol->value.integer.value[0] + && tas->muted_r == !ucontrol->value.integer.value[1]) + return 0; + + tas->muted_l = !ucontrol->value.integer.value[0]; + tas->muted_r = !ucontrol->value.integer.value[1]; + tas_set_volume(tas); + 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; + + ucontrol->value.integer.value[0] = tas->mixer_l[idx]; + ucontrol->value.integer.value[1] = tas->mixer_r[idx]; + + 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; + + if (tas->mixer_l[idx] == ucontrol->value.integer.value[0] + && tas->mixer_r[idx] == ucontrol->value.integer.value[1]) + return 0; + + tas->mixer_l[idx] = ucontrol->value.integer.value[0]; + tas->mixer_r[idx] = ucontrol->value.integer.value[1]; + + tas_set_mixer(tas); + 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, "PCM1", 0); +MIXER_CONTROL(monitor, "Monitor", 2); + +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { "Line-In", "Microphone" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item > 1) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = !!(tas->acr & TAS_ACR_INPUT_B); + 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 = tas->acr; + + tas->acr &= ~TAS_ACR_INPUT_B; + if (ucontrol->value.enumerated.item[0]) + tas->acr |= TAS_ACR_INPUT_B; + if (oldacr == tas->acr) + return 0; + tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr); + 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 struct transfer_info tas_transfers[] = { + { + /* input */ + .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_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_S16_BE | + SNDRV_PCM_FMTBIT_S24_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->set_hw_reset(tas->codec.gpio, 0); + msleep(1); + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 1); + msleep(1); + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0); + msleep(1); + + tas->acr &= ~TAS_ACR_ANALOG_PDOWN; + tas->acr |= TAS_ACR_B_MONAUREAL | TAS_ACR_B_MON_SEL_RIGHT; + if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr)) + return -ENODEV; + + tmp = TAS_MCS_SCLK64 | TAS_MCS_SPORT_MODE_I2S | TAS_MCS_SPORT_WL_24BIT; + if (tas_write_reg(tas, TAS_REG_MCS, 1, &tmp)) + return -ENODEV; + + tmp = 0; + if (tas_write_reg(tas, TAS_REG_MCS2, 1, &tmp)) + return -ENODEV; + + return 0; +} + +/* 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) +{ + tas->acr |= TAS_ACR_ANALOG_PDOWN; + tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr); + return 0; +} + +static int tas_resume(struct tas *tas) +{ + /* reset codec */ + tas_reset_init(tas); + tas_set_volume(tas); + tas_set_mixer(tas); + return 0; +} + +#ifdef CONFIG_PM +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); +} +#endif + +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, +#ifdef CONFIG_PM + .suspend = _tas_suspend, + .resume = _tas_resume, +#endif +}; + +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; + } + + if (tas_reset_init(tas)) { + printk(KERN_ERR PFX "tas failed to initialise\n"); + return -ENXIO; + } + + 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_LOWLEVEL, 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; + + 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 struct i2c_driver tas_driver; + +static int tas_create(struct i2c_adapter *adapter, + struct device_node *node, + int addr) +{ + struct tas *tas; + + tas = kzalloc(sizeof(struct tas), GFP_KERNEL); + + if (!tas) + return -ENOMEM; + + tas->i2c.driver = &tas_driver; + tas->i2c.adapter = adapter; + tas->i2c.addr = addr; + strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE-1); + + if (i2c_attach_client(&tas->i2c)) { + printk(KERN_ERR PFX "failed to attach to i2c\n"); + goto fail; + } + + strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN-1); + 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 detach; + } + printk(KERN_DEBUG "snd-aoa-codec-tas: created and attached tas instance\n"); + return 0; + detach: + i2c_detach_client(&tas->i2c); + fail: + kfree(tas); + return -EINVAL; +} + +static int tas_i2c_attach(struct i2c_adapter *adapter) +{ + struct device_node *busnode, *dev = NULL; + struct pmac_i2c_bus *bus; + + bus = pmac_i2c_adapter_to_bus(adapter); + if (bus == NULL) + return -ENODEV; + busnode = pmac_i2c_get_bus_node(bus); + + while ((dev = of_get_next_child(busnode, dev)) != NULL) { + if (device_is_compatible(dev, "tas3004")) { + u32 *addr; + printk(KERN_DEBUG PFX "found tas3004\n"); + addr = (u32 *) get_property(dev, "reg", NULL); + if (!addr) + continue; + return tas_create(adapter, dev, ((*addr) >> 1) & 0x7f); + } + /* older machines have no 'codec' node with a 'compatible' + * property that says 'tas3004', they just have a 'deq' + * node without any such property... */ + if (strcmp(dev->name, "deq") == 0) { + u32 *_addr, addr; + printk(KERN_DEBUG PFX "found 'deq' node\n"); + _addr = (u32 *) get_property(dev, "i2c-address", NULL); + if (!_addr) + continue; + addr = ((*_addr) >> 1) & 0x7f; + /* now, if the address doesn't match any of the two + * that a tas3004 can have, we cannot handle this. + * I doubt it ever happens but hey. */ + if (addr != 0x34 && addr != 0x35) + continue; + return tas_create(adapter, dev, addr); + } + } + return -ENODEV; +} + +static int tas_i2c_detach(struct i2c_client *client) +{ + struct tas *tas = container_of(client, struct tas, i2c); + int err; + u8 tmp = TAS_ACR_ANALOG_PDOWN; + + if ((err = i2c_detach_client(client))) + return err; + aoa_codec_unregister(&tas->codec); + of_node_put(tas->codec.node); + + /* power down codec chip */ + tas_write_reg(tas, TAS_REG_ACR, 1, &tmp); + + kfree(tas); + return 0; +} + +static struct i2c_driver tas_driver = { + .driver = { + .name = "aoa_codec_tas", + .owner = THIS_MODULE, + }, + .attach_adapter = tas_i2c_attach, + .detach_client = tas_i2c_detach, +}; + +static int __init tas_init(void) +{ + return i2c_add_driver(&tas_driver); +} + +static void __exit tas_exit(void) +{ + i2c_del_driver(&tas_driver); +} + +module_init(tas_init); +module_exit(tas_exit); diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.h b/sound/aoa/codecs/snd-aoa-codec-tas.h new file mode 100644 index 000000000000..daf81f45d83a --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-tas.h @@ -0,0 +1,47 @@ +/* + * 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 + +#endif /* __SND_AOA_CODECTASH */ diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/snd-aoa-codec-toonie.c new file mode 100644 index 000000000000..bcc555647e79 --- /dev/null +++ b/sound/aoa/codecs/snd-aoa-codec-toonie.c @@ -0,0 +1,141 @@ +/* + * 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> +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, + }, + {} +}; + +#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, +#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); + + if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, toonie, &ops)) { + printk(KERN_ERR PFX "failed to create toonie snd device!\n"); + return -ENODEV; + } + + /* nothing connected? what a joke! */ + if (toonie->codec.connected != 1) + return -ENOTCONN; + + 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"); + 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); |