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[ALSA] snd-aoa: add snd-aoa

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This large patch adds all of snd-aoa.
Consisting of many modules, it currently replaces snd-powermac
for all layout-id based machines and handles many more (for
example new powerbooks and powermacs with digital output that
previously couldn't be used at all).
It also has support for all layout-IDs that Apple has (judging
from their Info.plist file) but not all are tested.
The driver currently has 2 known regressions over snd-powermac:
 * it doesn't handle powermac 7,2 and 7,3
 * it doesn't have a DRC control on snapper-based machines
I will fix those during the 2.6.18 development cycle.

Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is contained in:
Johannes Berg
2006-06-21 15:42:43 +02:00
committed by Jaroslav Kysela
parent 41f0cd3a0c
commit f3d9478b2c
35 changed files with 7009 additions and 1 deletions
Download Patch File Download Diff File Expand all files Collapse all files

32
sound/aoa/codecs/Kconfig Normal file
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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.

3
sound/aoa/codecs/Makefile Normal file
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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

1113
sound/aoa/codecs/snd-aoa-codec-onyx.c Normal file
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76
sound/aoa/codecs/snd-aoa-codec-onyx.h Normal file
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/*
* 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 */

209
sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h Normal file
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/*
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 */
};

654
sound/aoa/codecs/snd-aoa-codec-tas.c Normal file
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/*
* 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);

47
sound/aoa/codecs/snd-aoa-codec-tas.h Normal file
View File

@@ -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 */

141
sound/aoa/codecs/snd-aoa-codec-toonie.c Normal file
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@@ -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);