PWN-Hunter/android_kernel_xiaomi_sdm845
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

Browse Source 
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (297 commits)
  ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
  ALSA: asihpi - HPI version 4.08
  ALSA: asihpi - Add volume mute controls
  ALSA: asihpi - Control name updates
  ALSA: asihpi - Use size_t for sizeof result
  ALSA: asihpi - Explicitly include mutex.h
  ALSA: asihpi - Add new node and message defines
  ALSA: asihpi - Make local function static
  ALSA: asihpi - Fix minor typos and spelling
  ALSA: asihpi - Remove unused structures, macros and functions
  ALSA: asihpi - Remove spurious adapter index check
  ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
  ALSA: asihpi - DSP code loader API now independent of OS
  ALSA: asihpi - Remove controlex structs and associated special data transfer code
  ALSA: asihpi - Increase request and response buffer sizes
  ALSA: asihpi - Give more meaningful name to hpi request message type
  ALSA: usb-audio - Add quirk for  Roland / BOSS BR-800
  ALSA: hda - Remove a superfluous argument of via_auto_init_output()
  ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
  ALSA: hda - Add documentation for codec-specific mixer controls
  ...
This commit is contained in:
Linus Torvalds 2011-07-23 10:59:37 -07:00
commit e498037105
225 changed files with 30897 additions and 25095 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
Documentation/DocBook/writing-an-alsa-driver.tmpl
View File

@ -1164,7 +1164,7 @@
}
chip->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_mychip_interrupt,
IRQF_SHARED, "My Chip", chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
snd_mychip_free(chip);
return -EBUSY;
@ -1197,7 +1197,7 @@
/* pci_driver definition */
static struct pci_driver driver = {
.name = "My Own Chip",
.name = KBUILD_MODNAME,
.id_table = snd_mychip_ids,
.probe = snd_mychip_probe,
.remove = __devexit_p(snd_mychip_remove),
@ -1340,7 +1340,7 @@
<programlisting>
<![CDATA[
if (request_irq(pci->irq, snd_mychip_interrupt,
IRQF_SHARED, "My Chip", chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
snd_mychip_free(chip);
return -EBUSY;
@ -1616,7 +1616,7 @@
<programlisting>
<![CDATA[
static struct pci_driver driver = {
.name = "My Own Chip",
.name = KBUILD_MODNAME,
.id_table = snd_mychip_ids,
.probe = snd_mychip_probe,
.remove = __devexit_p(snd_mychip_remove),
@ -5816,7 +5816,7 @@ struct _snd_pcm_runtime {
<programlisting>
<![CDATA[
static struct pci_driver driver = {
.name = "My Chip",
.name = KBUILD_MODNAME,
.id_table = snd_my_ids,
.probe = snd_my_probe,
.remove = __devexit_p(snd_my_remove),

100
Documentation/sound/alsa/HD-Audio-Controls.txt Normal file
View File

@ -0,0 +1,100 @@
This file explains the codec-specific mixer controls.
Realtek codecs
--------------
* Channel Mode
This is an enum control to change the surround-channel setup,
appears only when the surround channels are available.
It gives the number of channels to be used, "2ch", "4ch", "6ch",
and "8ch". According to the configuration, this also controls the
jack-retasking of multi-I/O jacks.
* Auto-Mute Mode
This is an enum control to change the auto-mute behavior of the
headphone and line-out jacks. If built-in speakers and headphone
and/or line-out jacks are available on a machine, this controls
appears.
When there are only either headphones or line-out jacks, it gives
"Disabled" and "Enabled" state. When enabled, the speaker is muted
automatically when a jack is plugged.
When both headphone and line-out jacks are present, it gives
"Disabled", "Speaker Only" and "Line-Out+Speaker". When
speaker-only is chosen, plugging into a headphone or a line-out jack
mutes the speakers, but not line-outs. When line-out+speaker is
selected, plugging to a headphone jack mutes both speakers and
line-outs.
IDT/Sigmatel codecs
-------------------
* Analog Loopback
This control enables/disables the analog-loopback circuit. This
appears only when "loopback" is set to true in a codec hint
(see HD-Audio.txt). Note that on some codecs the analog-loopback
and the normal PCM playback are exclusive, i.e. when this is on, you
won't hear any PCM stream.
* Swap Center/LFE
Swaps the center and LFE channel order. Normally, the left
corresponds to the center and the right to the LFE. When this is
ON, the left to the LFE and the right to the center.
* Headphone as Line Out
When this control is ON, treat the headphone jacks as line-out
jacks. That is, the headphone won't auto-mute the other line-outs,
and no HP-amp is set to the pins.
* Mic Jack Mode, Line Jack Mode, etc
These enum controls the direction and the bias of the input jack
pins. Depending on the jack type, it can set as "Mic In" and "Line
In", for determining the input bias, or it can be set to "Line Out"
when the pin is a multi-I/O jack for surround channels.
VIA codecs
----------
* Smart 5.1
An enum control to re-task the multi-I/O jacks for surround outputs.
When it's ON, the corresponding input jacks (usually a line-in and a
mic-in) are switched as the surround and the CLFE output jacks.
* Independent HP
When this enum control is enabled, the headphone output is routed
from an individual stream (the third PCM such as hw:0,2) instead of
the primary stream. In the case the headphone DAC is shared with a
side or a CLFE-channel DAC, the DAC is switched to the headphone
automatically.
* Loopback Mixing
An enum control to determine whether the analog-loopback route is
enabled or not. When it's enabled, the analog-loopback is mixed to
the front-channel. Also, the same route is used for the headphone
and speaker outputs. As a side-effect, when this mode is set, the
individual volume controls will be no longer available for
headphones and speakers because there is only one DAC connected to a
mixer widget.
* Dynamic Power-Control
This control determines whether the dynamic power-control per jack
detection is enabled or not. When enabled, the widgets power state
(D0/D3) are changed dynamically depending on the jack plugging
state for saving power consumptions. However, if your system
doesn't provide a proper jack-detection, this won't work; in such a
case, turn this control OFF.
* Jack Detect
This control is provided only for VT1708 codec which gives no proper
unsolicited event per jack plug. When this is on, the driver polls
the jack detection so that the headphone auto-mute can work, while
turning this off would reduce the power consumption.
Conexant codecs
---------------
* Auto-Mute Mode
See Reatek codecs.

2
MAINTAINERS
View File

@ -534,6 +534,8 @@ L: device-drivers-devel@blackfin.uclinux.org
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
W: http://wiki.analog.com/
S: Supported
F: sound/soc/codecs/adau*
F: sound/soc/codecs/adav*
F: sound/soc/codecs/ad1*
F: sound/soc/codecs/ssm*

1
include/linux/pci_ids.h
View File

@ -1308,6 +1308,7 @@
#define PCI_SUBDEVICE_ID_CREATIVE_SB08801 0x0041
#define PCI_SUBDEVICE_ID_CREATIVE_SB08802 0x0042
#define PCI_SUBDEVICE_ID_CREATIVE_SB08803 0x0043
#define PCI_SUBDEVICE_ID_CREATIVE_SB1270 0x0062
#define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX 0x6000
#define PCI_VENDOR_ID_ECTIVA 0x1102 /* duplicate: CREATIVE */

4
include/sound/rawmidi.h
View File

@ -27,6 +27,7 @@
#include <linux/spinlock.h>
#include <linux/wait.h>
#include <linux/mutex.h>
#include <linux/workqueue.h>
#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
#include "seq_device.h"
@ -63,6 +64,7 @@ struct snd_rawmidi_global_ops {
};
struct snd_rawmidi_runtime {
struct snd_rawmidi_substream *substream;
unsigned int drain: 1, /* drain stage */
oss: 1; /* OSS compatible mode */
/* midi stream buffer */
@ -79,7 +81,7 @@ struct snd_rawmidi_runtime {
/* event handler (new bytes, input only) */
void (*event)(struct snd_rawmidi_substream *substream);
/* defers calls to event [input] or ops->trigger [output] */
struct tasklet_struct tasklet;
struct work_struct event_work;
/* private data */
void *private_data;
void (*private_free)(struct snd_rawmidi_substream *substream);

4
include/sound/soc-dai.h
View File

@ -209,6 +209,10 @@ struct snd_soc_dai_driver {
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
unsigned int symmetric_rates:1;
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
};
/*

7
include/sound/soc-dapm.h
View File

@ -348,6 +348,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
@ -429,6 +431,7 @@ struct snd_soc_dapm_path {
/* status */
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
u32 weak:1; /* path ignored for power management */
int (*connected)(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink);
@ -444,6 +447,7 @@ struct snd_soc_dapm_widget {
char *name; /* widget name */
char *sname; /* stream name */
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct list_head list;
struct snd_soc_dapm_context *dapm;
@ -507,10 +511,11 @@ struct snd_soc_dapm_context {
struct device *dev; /* from parent - for debug */
struct snd_soc_codec *codec; /* parent codec */
struct snd_soc_platform *platform; /* parent platform */
struct snd_soc_card *card; /* parent card */
/* used during DAPM updates */
int dev_power;
enum snd_soc_bias_level target_bias_level;
struct list_head list;
#ifdef CONFIG_DEBUG_FS

59
include/sound/soc.h
View File

@ -202,6 +202,16 @@
#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
/*
* Component probe and remove ordering levels for components with runtime
* dependencies.
*/
#define SND_SOC_COMP_ORDER_FIRST -2
#define SND_SOC_COMP_ORDER_EARLY -1
#define SND_SOC_COMP_ORDER_NORMAL 0
#define SND_SOC_COMP_ORDER_LATE 1
#define SND_SOC_COMP_ORDER_LAST 2
/*
* Bias levels
*
@ -214,10 +224,10 @@
* @OFF: Power Off. No restrictions on transition times.
*/
enum snd_soc_bias_level {
SND_SOC_BIAS_OFF,
SND_SOC_BIAS_STANDBY,
SND_SOC_BIAS_PREPARE,
SND_SOC_BIAS_ON,
SND_SOC_BIAS_OFF = 0,
SND_SOC_BIAS_STANDBY = 1,
SND_SOC_BIAS_PREPARE = 2,
SND_SOC_BIAS_ON = 3,
};
struct snd_jack;
@ -258,6 +268,11 @@ enum snd_soc_compress_type {
SND_SOC_RBTREE_COMPRESSION
};
enum snd_soc_pcm_subclass {
SND_SOC_PCM_CLASS_PCM = 0,
SND_SOC_PCM_CLASS_BE = 1,
};
int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
unsigned int freq, int dir);
int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@ -297,6 +312,10 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec,
unsigned int reg);
int snd_soc_default_writable_register(struct snd_soc_codec *codec,
unsigned int reg);
int snd_soc_platform_read(struct snd_soc_platform *platform,
unsigned int reg);
int snd_soc_platform_write(struct snd_soc_platform *platform,
unsigned int reg, unsigned int val);
/* Utility functions to get clock rates from various things */
int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
@ -349,6 +368,8 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
const char *prefix);
int snd_soc_add_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@ -612,6 +633,10 @@ struct snd_soc_codec_driver {
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
};
/* SoC platform interface */
@ -623,10 +648,17 @@ struct snd_soc_platform_driver {
int (*resume)(struct snd_soc_dai *dai);
/* pcm creation and destruction */
int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
struct snd_pcm *);
int (*pcm_new)(struct snd_soc_pcm_runtime *);
void (*pcm_free)(struct snd_pcm *);
/* Default control and setup, added after probe() is run */
const struct snd_kcontrol_new *controls;
int num_controls;
const struct snd_soc_dapm_widget *dapm_widgets;
int num_dapm_widgets;
const struct snd_soc_dapm_route *dapm_routes;
int num_dapm_routes;
/*
* For platform caused delay reporting.
* Optional.
@ -636,6 +668,14 @@ struct snd_soc_platform_driver {
/* platform stream ops */
struct snd_pcm_ops *ops;
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
/* platform IO - used for platform DAPM */
unsigned int (*read)(struct snd_soc_platform *, unsigned int);
int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
};
struct snd_soc_platform {
@ -650,6 +690,8 @@ struct snd_soc_platform {
struct snd_soc_card *card;
struct list_head list;
struct list_head card_list;
struct snd_soc_dapm_context dapm;
};
struct snd_soc_dai_link {
@ -725,8 +767,10 @@ struct snd_soc_card {
/* callbacks */
int (*set_bias_level)(struct snd_soc_card *,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
int (*set_bias_level_post)(struct snd_soc_card *,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
long pmdown_time;
@ -789,6 +833,9 @@ struct snd_soc_pcm_runtime {
struct device dev;
struct snd_soc_card *card;
struct snd_soc_dai_link *dai_link;
struct mutex pcm_mutex;
enum snd_soc_pcm_subclass pcm_subclass;
struct snd_pcm_ops ops;
unsigned int complete:1;
unsigned int dev_registered:1;

45
include/trace/events/asoc.h
View File

@ -9,6 +9,7 @@
struct snd_soc_jack;
struct snd_soc_codec;
struct snd_soc_platform;
struct snd_soc_card;
struct snd_soc_dapm_widget;
@ -59,6 +60,50 @@ DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read,
);
DECLARE_EVENT_CLASS(snd_soc_preg,
TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
unsigned int val),
TP_ARGS(platform, reg, val),
TP_STRUCT__entry(
__string( name, platform->name )
__field( int, id )
__field( unsigned int, reg )
__field( unsigned int, val )
),
TP_fast_assign(
__assign_str(name, platform->name);
__entry->id = platform->id;
__entry->reg = reg;
__entry->val = val;
),
TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name),
(int)__entry->id, (unsigned int)__entry->reg,
(unsigned int)__entry->val)
);
DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write,
TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
unsigned int val),
TP_ARGS(platform, reg, val)
);
DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read,
TP_PROTO(struct snd_soc_platform *platform, unsigned int reg,
unsigned int val),
TP_ARGS(platform, reg, val)
);
DECLARE_EVENT_CLASS(snd_soc_card,
TP_PROTO(struct snd_soc_card *card, int val),

45
sound/core/rawmidi.c
View File

@ -92,16 +92,12 @@ static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substre
(!substream->append || runtime->avail >= count);
}
static void snd_rawmidi_input_event_tasklet(unsigned long data)
static void snd_rawmidi_input_event_work(struct work_struct *work)
{
struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
substream->runtime->event(substream);
}
static void snd_rawmidi_output_trigger_tasklet(unsigned long data)
{
struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
substream->ops->trigger(substream, 1);
struct snd_rawmidi_runtime *runtime =
container_of(work, struct snd_rawmidi_runtime, event_work);
if (runtime->event)
runtime->event(runtime->substream);
}
static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
@ -110,16 +106,10 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
if ((runtime = kzalloc(sizeof(*runtime), GFP_KERNEL)) == NULL)
return -ENOMEM;
runtime->substream = substream;
spin_lock_init(&runtime->lock);
init_waitqueue_head(&runtime->sleep);
if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT)
tasklet_init(&runtime->tasklet,
snd_rawmidi_input_event_tasklet,
(unsigned long)substream);
else
tasklet_init(&runtime->tasklet,
snd_rawmidi_output_trigger_tasklet,
(unsigned long)substream);
INIT_WORK(&runtime->event_work, snd_rawmidi_input_event_work);
runtime->event = NULL;
runtime->buffer_size = PAGE_SIZE;
runtime->avail_min = 1;
@ -150,12 +140,7 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs
{
if (!substream->opened)
return;
if (up) {
tasklet_schedule(&substream->runtime->tasklet);
} else {
tasklet_kill(&substream->runtime->tasklet);
substream->ops->trigger(substream, 0);
}
substream->ops->trigger(substream, up);
}
static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
@ -163,8 +148,8 @@ static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, i
if (!substream->opened)
return;
substream->ops->trigger(substream, up);
if (!up && substream->runtime->event)
tasklet_kill(&substream->runtime->tasklet);
if (!up)
cancel_work_sync(&substream->runtime->event_work);
}
int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream)
@ -641,10 +626,10 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
return -EINVAL;
}
if (params->buffer_size != runtime->buffer_size) {
newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
newbuf = krealloc(runtime->buffer, params->buffer_size,
GFP_KERNEL);
if (!newbuf)
return -ENOMEM;
kfree(runtime->buffer);
runtime->buffer = newbuf;
runtime->buffer_size = params->buffer_size;
runtime->avail = runtime->buffer_size;
@ -668,10 +653,10 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream,
return -EINVAL;
}
if (params->buffer_size != runtime->buffer_size) {
newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
newbuf = krealloc(runtime->buffer, params->buffer_size,
GFP_KERNEL);
if (!newbuf)
return -ENOMEM;
kfree(runtime->buffer);
runtime->buffer = newbuf;
runtime->buffer_size = params->buffer_size;
}
@ -926,7 +911,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
}
if (result > 0) {
if (runtime->event)
tasklet_schedule(&runtime->tasklet);
schedule_work(&runtime->event_work);
else if (snd_rawmidi_ready(substream))
wake_up(&runtime->sleep);
}

2
sound/firewire/speakers.c
View File

@ -171,7 +171,7 @@ static int fwspk_open(struct snd_pcm_substream *substream)
err = snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_PERIOD_TIME,
5000, 8192000);
5000, UINT_MAX);
if (err < 0)
return err;

4
sound/pci/ad1889.c
View File

@ -944,7 +944,7 @@ snd_ad1889_create(struct snd_card *card,
spin_lock_init(&chip->lock); /* only now can we call ad1889_free */
if (request_irq(pci->irq, snd_ad1889_interrupt,
IRQF_SHARED, card->driver, chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR PFX "cannot obtain IRQ %d\n", pci->irq);
snd_ad1889_free(chip);
return -EBUSY;
@ -1055,7 +1055,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = {
MODULE_DEVICE_TABLE(pci, snd_ad1889_ids);
static struct pci_driver ad1889_pci_driver = {
.name = "AD1889 Audio",
.name = KBUILD_MODNAME,
.id_table = snd_ad1889_ids,
.probe = snd_ad1889_probe,
.remove = __devexit_p(snd_ad1889_remove),

4
sound/pci/ali5451/ali5451.c
View File

@ -2090,7 +2090,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec)
codec->port = pci_resource_start(codec->pci, 0);
if (request_irq(codec->pci->irq, snd_ali_card_interrupt,
IRQF_SHARED, "ALI 5451", codec)) {
IRQF_SHARED, KBUILD_MODNAME, codec)) {
snd_printk(KERN_ERR "Unable to request irq.\n");
return -EBUSY;
}
@ -2295,7 +2295,7 @@ static void __devexit snd_ali_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "ALI 5451",
.name = KBUILD_MODNAME,
.id_table = snd_ali_ids,
.probe = snd_ali_probe,
.remove = __devexit_p(snd_ali_remove),

4
sound/pci/als300.c
View File

@ -722,7 +722,7 @@ static int __devinit snd_als300_create(struct snd_card *card,
irq_handler = snd_als300_interrupt;
if (request_irq(pci->irq, irq_handler, IRQF_SHARED,
card->shortname, chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_als300_free(chip);
return -EBUSY;
@ -846,7 +846,7 @@ static int __devinit snd_als300_probe(struct pci_dev *pci,
}
static struct pci_driver driver = {
.name = "ALS300",
.name = KBUILD_MODNAME,
.id_table = snd_als300_ids,
.probe = snd_als300_probe,
.remove = __devexit_p(snd_als300_remove),

2
sound/pci/als4000.c
View File

@ -1036,7 +1036,7 @@ static int snd_als4000_resume(struct pci_dev *pci)
static struct pci_driver driver = {
.name = "ALS4000",
.name = KBUILD_MODNAME,
.id_table = snd_als4000_ids,
.probe = snd_card_als4000_probe,
.remove = __devexit_p(snd_card_als4000_remove),

81
sound/pci/asihpi/asihpi.c
View File

@ -49,19 +49,21 @@ MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx");
#if defined CONFIG_SND_DEBUG
/* copied from pcm_lib.c, hope later patch will make that version public
and this copy can be removed */
static void pcm_debug_name(struct snd_pcm_substream *substream,
char *name, size_t len)
static inline void
snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
{
snprintf(name, len, "pcmC%dD%d%c:%d",
snprintf(buf, size, "pcmC%dD%d%c:%d",
substream->pcm->card->number,
substream->pcm->device,
substream->stream ? 'c' : 'p',
substream->number);
}
#define DEBUG_NAME(substream, name) char name[16]; pcm_debug_name(substream, name, sizeof(name))
#else
#define pcm_debug_name(s, n, l) do { } while (0)
#define DEBUG_NAME(name, substream) do { } while (0)
static inline void
snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
{
*buf = 0;
}
#endif
#if defined CONFIG_SND_DEBUG_VERBOSE
@ -304,7 +306,8 @@ static u16 handle_error(u16 err, int line, char *filename)
static void print_hwparams(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *p)
{
DEBUG_NAME(substream, name);
char name[16];
snd_pcm_debug_name(substream, name, sizeof(name));
snd_printd("%s HWPARAMS\n", name);
snd_printd(" samplerate %d Hz\n", params_rate(p));
snd_printd(" channels %d\n", params_channels(p));
@ -576,8 +579,9 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
struct snd_card_asihpi *card = snd_pcm_substream_chip(substream);
struct snd_pcm_substream *s;
u16 e;
DEBUG_NAME(substream, name);
char name[16];
snd_pcm_debug_name(substream, name, sizeof(name));
snd_printdd("%s trigger\n", name);
switch (cmd) {
@ -741,7 +745,9 @@ static void snd_card_asihpi_timer_function(unsigned long data)
int loops = 0;
u16 state;
u32 buffer_size, bytes_avail, samples_played, on_card_bytes;
DEBUG_NAME(substream, name);
char name[16];
snd_pcm_debug_name(substream, name, sizeof(name));
snd_printdd("%s snd_card_asihpi_timer_function\n", name);
@ -1323,10 +1329,12 @@ static const char * const asihpi_src_names[] = {
"RF",
"Clock",
"Bitstream",
"Microphone",
"Cobranet",
"Mic",
"Net",
"Analog",
"Adapter",
"RTP",
"GPI",
};
compile_time_assert(
@ -1341,8 +1349,10 @@ static const char * const asihpi_dst_names[] = {
"Digital",
"RF",
"Speaker",
"Cobranet Out",
"Analog"
"Net",
"Analog",
"RTP",
"GPO",
};
compile_time_assert(
@ -1476,11 +1486,40 @@ static int snd_asihpi_volume_put(struct snd_kcontrol *kcontrol,
static const DECLARE_TLV_DB_SCALE(db_scale_100, -10000, VOL_STEP_mB, 0);
#define snd_asihpi_volume_mute_info snd_ctl_boolean_mono_info
static int snd_asihpi_volume_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
u32 h_control = kcontrol->private_value;
u32 mute;
hpi_handle_error(hpi_volume_get_mute(h_control, &mute));
ucontrol->value.integer.value[0] = mute ? 0 : 1;
return 0;
}
static int snd_asihpi_volume_mute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
u32 h_control = kcontrol->private_value;
int change = 1;
/* HPI currently only supports all or none muting of multichannel volume
ALSA Switch element has opposite sense to HPI mute: on==unmuted, off=muted
*/
int mute = ucontrol->value.integer.value[0] ? 0 : HPI_BITMASK_ALL_CHANNELS;
hpi_handle_error(hpi_volume_set_mute(h_control, mute));
return change;
}
static int __devinit snd_asihpi_volume_add(struct snd_card_asihpi *asihpi,
struct hpi_control *hpi_ctl)
{
struct snd_card *card = asihpi->card;
struct snd_kcontrol_new snd_control;
int err;
u32 mute;
asihpi_ctl_init(&snd_control, hpi_ctl, "Volume");
snd_control.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
@ -1490,7 +1529,19 @@ static int __devinit snd_asihpi_volume_add(struct snd_card_asihpi *asihpi,
snd_control.put = snd_asihpi_volume_put;
snd_control.tlv.p = db_scale_100;
return ctl_add(card, &snd_control, asihpi);
err = ctl_add(card, &snd_control, asihpi);
if (err)
return err;
if (hpi_volume_get_mute(hpi_ctl->h_control, &mute) == 0) {
asihpi_ctl_init(&snd_control, hpi_ctl, "Switch");
snd_control.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
snd_control.info = snd_asihpi_volume_mute_info;
snd_control.get = snd_asihpi_volume_mute_get;
snd_control.put = snd_asihpi_volume_mute_put;
err = ctl_add(card, &snd_control, asihpi);
}
return err;
}
/*------------------------------------------------------------
@ -2923,7 +2974,7 @@ static DEFINE_PCI_DEVICE_TABLE(asihpi_pci_tbl) = {
MODULE_DEVICE_TABLE(pci, asihpi_pci_tbl);
static struct pci_driver driver = {
.name = "asihpi",
.name = KBUILD_MODNAME,
.id_table = asihpi_pci_tbl,
.probe = snd_asihpi_probe,
.remove = __devexit_p(snd_asihpi_remove),

24
sound/pci/asihpi/hpi.h
View File

@ -1,7 +1,7 @@
/******************************************************************************
AudioScience HPI driver
Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@ -42,12 +42,11 @@ i.e 3.05.02 is a development version
#define HPI_VER_MINOR(v) ((int)((v >> 8) & 0xFF))
#define HPI_VER_RELEASE(v) ((int)(v & 0xFF))
/* Use single digits for versions less that 10 to avoid octal. */
#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 6, 0)
#define HPI_VER_STRING "4.06.00"
#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 8, 0)
#define HPI_VER_STRING "4.08.00"
/* Library version as documented in hpi-api-versions.txt */
#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(9, 0, 0)
#define HPI_LIB_VER HPI_VERSION_CONSTRUCTOR(10, 0, 0)
#include <linux/types.h>
#define HPI_BUILD_EXCLUDE_DEPRECATED
@ -211,8 +210,12 @@ enum HPI_SOURCENODES {
HPI_SOURCENODE_COBRANET = 109,
HPI_SOURCENODE_ANALOG = 110, /**< analog input node. */
HPI_SOURCENODE_ADAPTER = 111, /**< adapter node. */
/** RTP stream input node - This node is a destination for
packets of RTP audio samples from other devices. */
HPI_SOURCENODE_RTP_DESTINATION = 112,
HPI_SOURCENODE_GP_IN = 113, /**< general purpose input. */
/* !!!Update this AND hpidebug.h if you add a new sourcenode type!!! */
HPI_SOURCENODE_LAST_INDEX = 111 /**< largest ID */
HPI_SOURCENODE_LAST_INDEX = 113 /**< largest ID */
/* AX6 max sourcenode types = 15 */
};
@ -228,7 +231,7 @@ enum HPI_DESTNODES {
HPI_DESTNODE_NONE = 200,
/** In Stream (Record) node. */
HPI_DESTNODE_ISTREAM = 201,
HPI_DESTNODE_LINEOUT = 202, /**< line out node. */
HPI_DESTNODE_LINEOUT = 202, /**< line out node. */
HPI_DESTNODE_AESEBU_OUT = 203, /**< AES/EBU output node. */
HPI_DESTNODE_RF = 204, /**< RF output node. */
HPI_DESTNODE_SPEAKER = 205, /**< speaker output node. */
@ -236,9 +239,12 @@ enum HPI_DESTNODES {
Audio samples from the device are sent out on the Cobranet network.*/
HPI_DESTNODE_COBRANET = 206,
HPI_DESTNODE_ANALOG = 207, /**< analog output node. */
/** RTP stream output node - This node is a source for
packets of RTP audio samples that are sent to other devices. */
HPI_DESTNODE_RTP_SOURCE = 208,
HPI_DESTNODE_GP_OUT = 209, /**< general purpose output node. */
/* !!!Update this AND hpidebug.h if you add a new destnode type!!! */
HPI_DESTNODE_LAST_INDEX = 207 /**< largest ID */
HPI_DESTNODE_LAST_INDEX = 209 /**< largest ID */
/* AX6 max destnode types = 15 */
};

11
sound/pci/asihpi/hpi6000.c
View File

@ -359,7 +359,7 @@ void HPI_6000(struct hpi_message *phm, struct hpi_response *phr)
HPI_ERROR_PROCESSING_MESSAGE);
switch (phm->type) {
case HPI_TYPE_MESSAGE:
case HPI_TYPE_REQUEST:
switch (phm->object) {
case HPI_OBJ_SUBSYSTEM:
subsys_message(phm, phr);
@ -538,7 +538,7 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao,
HPI_DEBUG_LOG(VERBOSE, "send ADAPTER_GET_INFO\n");
memset(&hm, 0, sizeof(hm));
hm.type = HPI_TYPE_MESSAGE;
hm.type = HPI_TYPE_REQUEST;
hm.size = sizeof(struct hpi_message);
hm.object = HPI_OBJ_ADAPTER;
hm.function = HPI_ADAPTER_GET_INFO;
@ -946,11 +946,8 @@ static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
}
/* write the DSP code down into the DSPs memory */
/*HpiDspCode_Open(nBootLoadFamily,&DspCode,pdwOsErrorCode); */
dsp_code.ps_dev = pao->pci.pci_dev;
error = hpi_dsp_code_open(boot_load_family, &dsp_code,
pos_error_code);
error = hpi_dsp_code_open(boot_load_family, pao->pci.pci_dev,
&dsp_code, pos_error_code);
if (error)
return error;

52
sound/pci/asihpi/hpi6205.c
View File

@ -373,6 +373,7 @@ static void instream_message(struct hpi_adapter_obj *pao,
/** Entry point to this HPI backend
* All calls to the HPI start here
*/
static
void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
struct hpi_response *phr)
{
@ -392,7 +393,7 @@ void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
HPI_DEBUG_LOG(VERBOSE, "start of switch\n");
switch (phm->type) {
case HPI_TYPE_MESSAGE:
case HPI_TYPE_REQUEST:
switch (phm->object) {
case HPI_OBJ_SUBSYSTEM:
subsys_message(pao, phm, phr);
@ -402,7 +403,6 @@ void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm,
adapter_message(pao, phm, phr);
break;
case HPI_OBJ_CONTROLEX:
case HPI_OBJ_CONTROL:
control_message(pao, phm, phr);
break;
@ -634,11 +634,12 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
HPI_DEBUG_LOG(VERBOSE, "init ADAPTER_GET_INFO\n");
memset(&hm, 0, sizeof(hm));
hm.type = HPI_TYPE_MESSAGE;
/* wAdapterIndex == version == 0 */
hm.type = HPI_TYPE_REQUEST;
hm.size = sizeof(hm);
hm.object = HPI_OBJ_ADAPTER;
hm.function = HPI_ADAPTER_GET_INFO;
hm.adapter_index = 0;
memset(&hr, 0, sizeof(hr));
hr.size = sizeof(hr);
@ -658,9 +659,6 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao,
hr.u.ax.info.num_outstreams +
hr.u.ax.info.num_instreams;
hpios_locked_mem_prepare((max_streams * 6) / 10, max_streams,
65536, pao->pci.pci_dev);
HPI_DEBUG_LOG(VERBOSE,
"got adapter info type %x index %d serial %d\n",
hr.u.ax.info.adapter_type, hr.u.ax.info.adapter_index,
@ -709,9 +707,6 @@ static void delete_adapter_obj(struct hpi_adapter_obj *pao)
[i]);
phw->outstream_host_buffer_size[i] = 0;
}
hpios_locked_mem_unprepare(pao->pci.pci_dev);
kfree(phw);
}
@ -1371,9 +1366,8 @@ static u16 adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
return err;
/* write the DSP code down into the DSPs memory */
dsp_code.ps_dev = pao->pci.pci_dev;
err = hpi_dsp_code_open(boot_code_id[dsp], &dsp_code,
pos_error_code);
err = hpi_dsp_code_open(boot_code_id[dsp], pao->pci.pci_dev,
&dsp_code, pos_error_code);
if (err)
return err;
@ -2084,13 +2078,13 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao,
u16 err = 0;
message_count++;
if (phm->size > sizeof(interface->u)) {
if (phm->size > sizeof(interface->u.message_buffer)) {
phr->error = HPI_ERROR_MESSAGE_BUFFER_TOO_SMALL;
phr->specific_error = sizeof(interface->u);
phr->specific_error = sizeof(interface->u.message_buffer);
phr->size = sizeof(struct hpi_response_header);
HPI_DEBUG_LOG(ERROR,
"message len %d too big for buffer %zd \n", phm->size,
sizeof(interface->u));
sizeof(interface->u.message_buffer));
return 0;
}
@ -2122,18 +2116,19 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao,
/* read the result */
if (time_out) {
if (interface->u.response_buffer.size <= phr->size)
if (interface->u.response_buffer.response.size <= phr->size)
memcpy(phr, &interface->u.response_buffer,
interface->u.response_buffer.size);
interface->u.response_buffer.response.size);
else {
HPI_DEBUG_LOG(ERROR,
"response len %d too big for buffer %d\n",
interface->u.response_buffer.size, phr->size);
interface->u.response_buffer.response.size,
phr->size);
memcpy(phr, &interface->u.response_buffer,
sizeof(struct hpi_response_header));
phr->error = HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
phr->specific_error =
interface->u.response_buffer.size;
interface->u.response_buffer.response.size;
phr->size = sizeof(struct hpi_response_header);
}
}
@ -2202,23 +2197,6 @@ static void hw_message(struct hpi_adapter_obj *pao, struct hpi_message *phm,
phm->u.d.u.data.data_size, H620_HIF_GET_DATA);
break;
case HPI_CONTROL_SET_STATE:
if (phm->object == HPI_OBJ_CONTROLEX
&& phm->u.cx.attribute == HPI_COBRANET_SET_DATA)
err = hpi6205_transfer_data(pao,
phm->u.cx.u.cobranet_bigdata.pb_data,
phm->u.cx.u.cobranet_bigdata.byte_count,
H620_HIF_SEND_DATA);
break;
case HPI_CONTROL_GET_STATE:
if (phm->object == HPI_OBJ_CONTROLEX
&& phm->u.cx.attribute == HPI_COBRANET_GET_DATA)
err = hpi6205_transfer_data(pao,
phm->u.cx.u.cobranet_bigdata.pb_data,
phr->u.cx.u.cobranet_data.byte_count,
H620_HIF_GET_DATA);
break;
}
phr->error = err;

25
sound/pci/asihpi/hpi6205.h
View File

@ -1,7 +1,7 @@
/*****************************************************************************
AudioScience HPI driver
Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@ -70,15 +70,28 @@ The Host located memory buffer that the 6205 will bus master
in and out of.
************************************************************/
#define HPI6205_SIZEOF_DATA (16*1024)
struct message_buffer_6205 {
struct hpi_message message;
char data[256];
};
struct response_buffer_6205 {
struct hpi_response response;
char data[256];
};
union buffer_6205 {
struct message_buffer_6205 message_buffer;
struct response_buffer_6205 response_buffer;
u8 b_data[HPI6205_SIZEOF_DATA];
};
struct bus_master_interface {
u32 host_cmd;
u32 dsp_ack;
u32 transfer_size_in_bytes;
union {
struct hpi_message_header message_buffer;
struct hpi_response_header response_buffer;
u8 b_data[HPI6205_SIZEOF_DATA];
} u;
union buffer_6205 u;
struct controlcache_6205 control_cache;
struct async_event_buffer_6205 async_buffer;
struct hpi_hostbuffer_status

155
sound/pci/asihpi/hpi_internal.h
View File

@ -1,7 +1,7 @@
/******************************************************************************
AudioScience HPI driver
Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@ -32,12 +32,6 @@ HPI internal definitions
#include "hpios.h"
/* physical memory allocation */
void hpios_locked_mem_init(void
);
void hpios_locked_mem_free_all(void
);
#define hpios_locked_mem_prepare(a, b, c, d);
#define hpios_locked_mem_unprepare(a)
/** Allocate and map an area of locked memory for bus master DMA operations.
@ -226,8 +220,8 @@ enum HPI_CONTROL_ATTRIBUTES {
HPI_COBRANET_SET = HPI_CTL_ATTR(COBRANET, 1),
HPI_COBRANET_GET = HPI_CTL_ATTR(COBRANET, 2),
HPI_COBRANET_SET_DATA = HPI_CTL_ATTR(COBRANET, 3),
HPI_COBRANET_GET_DATA = HPI_CTL_ATTR(COBRANET, 4),
/*HPI_COBRANET_SET_DATA = HPI_CTL_ATTR(COBRANET, 3), */
/*HPI_COBRANET_GET_DATA = HPI_CTL_ATTR(COBRANET, 4), */
HPI_COBRANET_GET_STATUS = HPI_CTL_ATTR(COBRANET, 5),
HPI_COBRANET_SEND_PACKET = HPI_CTL_ATTR(COBRANET, 6),
HPI_COBRANET_GET_PACKET = HPI_CTL_ATTR(COBRANET, 7),
@ -364,10 +358,12 @@ Used in DLL to indicate device not present
#define HPI_ADAPTER_ASI(f) (f)
enum HPI_MESSAGE_TYPES {
HPI_TYPE_MESSAGE = 1,
HPI_TYPE_REQUEST = 1,
HPI_TYPE_RESPONSE = 2,
HPI_TYPE_DATA = 3,
HPI_TYPE_SSX2BYPASS_MESSAGE = 4
HPI_TYPE_SSX2BYPASS_MESSAGE = 4,
HPI_TYPE_COMMAND = 5,
HPI_TYPE_NOTIFICATION = 6
};
enum HPI_OBJECT_TYPES {
@ -383,7 +379,7 @@ enum HPI_OBJECT_TYPES {
HPI_OBJ_WATCHDOG = 10,
HPI_OBJ_CLOCK = 11,
HPI_OBJ_PROFILE = 12,
HPI_OBJ_CONTROLEX = 13,
/* HPI_ OBJ_ CONTROLEX = 13, */
HPI_OBJ_ASYNCEVENT = 14
#define HPI_OBJ_MAXINDEX 14
};
@ -608,7 +604,7 @@ struct hpi_data_compat32 {
#endif
struct hpi_buffer {
/** placehoder for backward compatibility (see dwBufferSize) */
/** placeholder for backward compatibility (see dwBufferSize) */
struct hpi_msg_format reserved;
u32 command; /**< HPI_BUFFER_CMD_xxx*/
u32 pci_address; /**< PCI physical address of buffer for DSP DMA */
@ -912,95 +908,13 @@ union hpi_control_union_res {
u32 remaining_chars;
} chars8;
char c_data12[12];
};
/* HPI_CONTROLX_STRUCTURES */
/* Message */
/** Used for all HMI variables where max length <= 8 bytes
*/
struct hpi_controlx_msg_cobranet_data {
u32 hmi_address;
u32 byte_count;
u32 data[2];
};
/** Used for string data, and for packet bridge
*/
struct hpi_controlx_msg_cobranet_bigdata {
u32 hmi_address;
u32 byte_count;
u8 *pb_data;
#ifndef HPI64BIT
u32 padding;
#endif
};
/** Used for PADS control reading of string fields.
*/
struct hpi_controlx_msg_pad_data {
u32 field;
u32 byte_count;
u8 *pb_data;
#ifndef HPI64BIT
u32 padding;
#endif
};
/** Used for generic data
*/
struct hpi_controlx_msg_generic {
u32 param1;
u32 param2;
};
struct hpi_controlx_msg {
u16 attribute; /* control attribute or property */
u16 saved_index;
union {
struct hpi_controlx_msg_cobranet_data cobranet_data;
struct hpi_controlx_msg_cobranet_bigdata cobranet_bigdata;
struct hpi_controlx_msg_generic generic;
struct hpi_controlx_msg_pad_data pad_data;
/*struct param_value universal_value; */
/* nothing extra to send for status read */
} u;
};
/* Response */
/**
*/
struct hpi_controlx_res_cobranet_data {
u32 byte_count;
u32 data[2];
};
struct hpi_controlx_res_cobranet_bigdata {
u32 byte_count;
};
struct hpi_controlx_res_cobranet_status {
u32 status;
u32 readable_size;
u32 writeable_size;
};
struct hpi_controlx_res_generic {
u32 param1;
u32 param2;
};
struct hpi_controlx_res {
union {
struct hpi_controlx_res_cobranet_bigdata cobranet_bigdata;
struct hpi_controlx_res_cobranet_data cobranet_data;
struct hpi_controlx_res_cobranet_status cobranet_status;
struct hpi_controlx_res_generic generic;
/*struct param_info universal_info; */
/*struct param_value universal_value; */
} u;
struct {
u32 status;
u32 readable_size;
u32 writeable_size;
} status;
} cobranet;
};
struct hpi_nvmemory_msg {
@ -1126,7 +1040,6 @@ struct hpi_message {
/* identical to struct hpi_control_msg,
but field naming is improved */
struct hpi_control_union_msg cu;
struct hpi_controlx_msg cx; /* extended mixer control; */
struct hpi_nvmemory_msg n;
struct hpi_gpio_msg l; /* digital i/o */
struct hpi_watchdog_msg w;
@ -1151,7 +1064,7 @@ struct hpi_message {
sizeof(struct hpi_message_header) + sizeof(struct hpi_watchdog_msg),\
sizeof(struct hpi_message_header) + sizeof(struct hpi_clock_msg),\
sizeof(struct hpi_message_header) + sizeof(struct hpi_profile_msg),\
sizeof(struct hpi_message_header) + sizeof(struct hpi_controlx_msg),\
sizeof(struct hpi_message_header), /* controlx obj removed */ \
sizeof(struct hpi_message_header) + sizeof(struct hpi_async_msg) \
}
@ -1188,7 +1101,6 @@ struct hpi_response {
struct hpi_control_res c; /* mixer control; */
/* identical to hpi_control_res, but field naming is improved */
union hpi_control_union_res cu;
struct hpi_controlx_res cx; /* extended mixer control; */
struct hpi_nvmemory_res n;
struct hpi_gpio_res l; /* digital i/o */
struct hpi_watchdog_res w;
@ -1213,7 +1125,7 @@ struct hpi_response {
sizeof(struct hpi_response_header) + sizeof(struct hpi_watchdog_res),\
sizeof(struct hpi_response_header) + sizeof(struct hpi_clock_res),\
sizeof(struct hpi_response_header) + sizeof(struct hpi_profile_res),\
sizeof(struct hpi_response_header) + sizeof(struct hpi_controlx_res),\
sizeof(struct hpi_response_header), /* controlx obj removed */ \
sizeof(struct hpi_response_header) + sizeof(struct hpi_async_res) \
}
@ -1308,6 +1220,30 @@ struct hpi_res_adapter_debug_read {
u8 bytes[256];
};
struct hpi_msg_cobranet_hmi {
u16 attribute;
u16 padding;
u32 hmi_address;
u32 byte_count;
};
struct hpi_msg_cobranet_hmiwrite {
struct hpi_message_header h;
struct hpi_msg_cobranet_hmi p;
u8 bytes[256];
};
struct hpi_msg_cobranet_hmiread {
struct hpi_message_header h;
struct hpi_msg_cobranet_hmi p;
};
struct hpi_res_cobranet_hmiread {
struct hpi_response_header h;
u32 byte_count;
u8 bytes[256];
};
#if 1
#define hpi_message_header_v1 hpi_message_header
#define hpi_response_header_v1 hpi_response_header
@ -1338,7 +1274,6 @@ struct hpi_msg_payload_v0 {
union hpi_mixerx_msg mx;
struct hpi_control_msg c;
struct hpi_control_union_msg cu;
struct hpi_controlx_msg cx;
struct hpi_nvmemory_msg n;
struct hpi_gpio_msg l;
struct hpi_watchdog_msg w;
@ -1358,7 +1293,6 @@ struct hpi_res_payload_v0 {
union hpi_mixerx_res mx;
struct hpi_control_res c;
union hpi_control_union_res cu;
struct hpi_controlx_res cx;
struct hpi_nvmemory_res n;
struct hpi_gpio_res l;
struct hpi_watchdog_res w;
@ -1493,12 +1427,6 @@ struct hpi_control_cache_microphone {
char temp_padding[6];
};
struct hpi_control_cache_generic {
struct hpi_control_cache_info i;
u32 dw1;
u32 dw2;
};
struct hpi_control_cache_single {
union {
struct hpi_control_cache_info i;
@ -1514,7 +1442,6 @@ struct hpi_control_cache_single {
struct hpi_control_cache_silencedetector silence;
struct hpi_control_cache_sampleclock clk;
struct hpi_control_cache_microphone microphone;
struct hpi_control_cache_generic generic;
} u;
};

17
sound/pci/asihpi/hpicmn.c
View File

@ -57,7 +57,7 @@ u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr)
}
if (phr->function != phm->function) {
HPI_DEBUG_LOG(ERROR, "header type %d invalid\n",
HPI_DEBUG_LOG(ERROR, "header function %d invalid\n",
phr->function);
return HPI_ERROR_INVALID_RESPONSE;
}
@ -315,8 +315,7 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
short found = 1;
struct hpi_control_cache_info *pI;
struct hpi_control_cache_single *pC;
struct hpi_control_cache_pad *p_pad;
size_t response_size;
if (!find_control(phm->obj_index, p_cache, &pI)) {
HPI_DEBUG_LOG(VERBOSE,
"HPICMN find_control() failed for adap %d\n",
@ -326,11 +325,15 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
phr->error = 0;
/* set the default response size */
response_size =
sizeof(struct hpi_response_header) +
sizeof(struct hpi_control_res);
/* pC is the default cached control strucure. May be cast to
something else in the following switch statement.
*/
pC = (struct hpi_control_cache_single *)pI;
p_pad = (struct hpi_control_cache_pad *)pI;
switch (pI->control_type) {
@ -529,9 +532,7 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
pI->control_index, pI->control_type, phm->u.c.attribute);
if (found)
phr->size =
sizeof(struct hpi_response_header) +
sizeof(struct hpi_control_res);
phr->size = (u16)response_size;
return found;
}
@ -682,7 +683,7 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr)
void HPI_COMMON(struct hpi_message *phm, struct hpi_response *phr)
{
switch (phm->type) {
case HPI_TYPE_MESSAGE:
case HPI_TYPE_REQUEST:
switch (phm->object) {
case HPI_OBJ_SUBSYSTEM:
subsys_message(phm, phr);

136
sound/pci/asihpi/hpidspcd.c
View File

@ -1,8 +1,8 @@
/***********************************************************************/
/*!
/**
AudioScience HPI driver
Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@ -18,90 +18,59 @@
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
\file
Functions for reading DSP code to load into DSP
(Linux only:) If DSPCODE_FIRMWARE_LOADER is defined, code is read using
Functions for reading DSP code using
hotplug firmware loader from individual dsp code files
If neither of the above is defined, code is read from linked arrays.
DSPCODE_ARRAY is defined.
HPI_INCLUDE_**** must be defined
and the appropriate hzz?????.c or hex?????.c linked in
*/
*/
/***********************************************************************/
#define SOURCEFILE_NAME "hpidspcd.c"
#include "hpidspcd.h"
#include "hpidebug.h"
/**
Header structure for binary dsp code file (see asidsp.doc)
This structure must match that used in s2bin.c for generation of asidsp.bin
*/
#ifndef DISABLE_PRAGMA_PACK1
#pragma pack(push, 1)
#endif
struct code_header {
u32 size;
char type[4];
u32 adapter;
u32 version;
u32 crc;
struct dsp_code_private {
/** Firmware descriptor */
const struct firmware *firmware;
struct pci_dev *dev;
};
#ifndef DISABLE_PRAGMA_PACK1
#pragma pack(pop)
#endif
#define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \
HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER)))
/***********************************************************************/
#include <linux/pci.h>
/*-------------------------------------------------------------------*/
short hpi_dsp_code_open(u32 adapter, struct dsp_code *ps_dsp_code,
u32 *pos_error_code)
short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code,
u32 *os_error_code)
{
const struct firmware *ps_firmware = ps_dsp_code->ps_firmware;
const struct firmware *firmware;
struct pci_dev *dev = os_data;
struct code_header header;
char fw_name[20];
int err;
sprintf(fw_name, "asihpi/dsp%04x.bin", adapter);
err = request_firmware(&ps_firmware, fw_name,
&ps_dsp_code->ps_dev->dev);
err = request_firmware(&firmware, fw_name, &dev->dev);
if (err != 0) {
dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
if (err || !firmware) {
dev_printk(KERN_ERR, &dev->dev,
"%d, request_firmware failed for %s\n", err,
fw_name);
goto error1;
}
if (ps_firmware->size < sizeof(header)) {
dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
"Header size too small %s\n", fw_name);
if (firmware->size < sizeof(header)) {
dev_printk(KERN_ERR, &dev->dev, "Header size too small %s\n",
fw_name);
goto error2;
}
memcpy(&header, ps_firmware->data, sizeof(header));
if (header.adapter != adapter) {
dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
"Adapter type incorrect %4x != %4x\n", header.adapter,
adapter);
goto error2;
}
if (header.size != ps_firmware->size) {
dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
"Code size wrong %d != %ld\n", header.size,
(unsigned long)ps_firmware->size);
memcpy(&header, firmware->data, sizeof(header));
if ((header.type != 0x45444F43) || /* "CODE" */
(header.adapter != adapter)
|| (header.size != firmware->size)) {
dev_printk(KERN_ERR, &dev->dev, "Invalid firmware file\n");
goto error2;
}
if (header.version / 100 != HPI_VER_DECIMAL / 100) {
dev_printk(KERN_ERR, &ps_dsp_code->ps_dev->dev,
if ((header.version / 100 & ~1) != (HPI_VER_DECIMAL / 100 & ~1)) {
dev_printk(KERN_ERR, &dev->dev,
"Incompatible firmware version "
"DSP image %d != Driver %d\n", header.version,
HPI_VER_DECIMAL);
@ -109,67 +78,70 @@ short hpi_dsp_code_open(u32 adapter, struct dsp_code *ps_dsp_code,
}
if (header.version != HPI_VER_DECIMAL) {
dev_printk(KERN_WARNING, &ps_dsp_code->ps_dev->dev,
dev_printk(KERN_WARNING, &dev->dev,
"Firmware: release version mismatch DSP image %d != Driver %d\n",
header.version, HPI_VER_DECIMAL);
}
HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name);
ps_dsp_code->ps_firmware = ps_firmware;
ps_dsp_code->block_length = header.size / sizeof(u32);
ps_dsp_code->word_count = sizeof(header) / sizeof(u32);
ps_dsp_code->version = header.version;
ps_dsp_code->crc = header.crc;
dsp_code->pvt = kmalloc(sizeof(*dsp_code->pvt), GFP_KERNEL);
if (!dsp_code->pvt)
return HPI_ERROR_MEMORY_ALLOC;
dsp_code->pvt->dev = dev;
dsp_code->pvt->firmware = firmware;
dsp_code->header = header;
dsp_code->block_length = header.size / sizeof(u32);
dsp_code->word_count = sizeof(header) / sizeof(u32);
return 0;
error2:
release_firmware(ps_firmware);
release_firmware(firmware);
error1:
ps_dsp_code->ps_firmware = NULL;
ps_dsp_code->block_length = 0;
dsp_code->block_length = 0;
return HPI_ERROR_DSP_FILE_NOT_FOUND;
}
/*-------------------------------------------------------------------*/
void hpi_dsp_code_close(struct dsp_code *ps_dsp_code)
void hpi_dsp_code_close(struct dsp_code *dsp_code)
{
if (ps_dsp_code->ps_firmware != NULL) {
if (dsp_code->pvt->firmware) {
HPI_DEBUG_LOG(DEBUG, "dsp code closed\n");
release_firmware(ps_dsp_code->ps_firmware);
ps_dsp_code->ps_firmware = NULL;
release_firmware(dsp_code->pvt->firmware);
dsp_code->pvt->firmware = NULL;
}
kfree(dsp_code->pvt);
}
/*-------------------------------------------------------------------*/
void hpi_dsp_code_rewind(struct dsp_code *ps_dsp_code)
void hpi_dsp_code_rewind(struct dsp_code *dsp_code)
{
/* Go back to start of data, after header */
ps_dsp_code->word_count = sizeof(struct code_header) / sizeof(u32);
dsp_code->word_count = sizeof(struct code_header) / sizeof(u32);
}
/*-------------------------------------------------------------------*/
short hpi_dsp_code_read_word(struct dsp_code *ps_dsp_code, u32 *pword)
short hpi_dsp_code_read_word(struct dsp_code *dsp_code, u32 *pword)
{
if (ps_dsp_code->word_count + 1 > ps_dsp_code->block_length)
if (dsp_code->word_count + 1 > dsp_code->block_length)
return HPI_ERROR_DSP_FILE_FORMAT;
*pword = ((u32 *)(ps_dsp_code->ps_firmware->data))[ps_dsp_code->
*pword = ((u32 *)(dsp_code->pvt->firmware->data))[dsp_code->
word_count];
ps_dsp_code->word_count++;
dsp_code->word_count++;
return 0;
}
/*-------------------------------------------------------------------*/
short hpi_dsp_code_read_block(size_t words_requested,
struct dsp_code *ps_dsp_code, u32 **ppblock)
struct dsp_code *dsp_code, u32 **ppblock)
{
if (ps_dsp_code->word_count + words_requested >
ps_dsp_code->block_length)
if (dsp_code->word_count + words_requested > dsp_code->block_length)
return HPI_ERROR_DSP_FILE_FORMAT;
*ppblock =
((u32 *)(ps_dsp_code->ps_firmware->data)) +
ps_dsp_code->word_count;
ps_dsp_code->word_count += words_requested;
((u32 *)(dsp_code->pvt->firmware->data)) +
dsp_code->word_count;
dsp_code->word_count += words_requested;
return 0;
}

72
sound/pci/asihpi/hpidspcd.h
View File

@ -2,7 +2,7 @@
/**
AudioScience HPI driver
Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@ -20,19 +20,6 @@
\file
Functions for reading DSP code to load into DSP
hpi_dspcode_defines HPI DSP code loading method
Define exactly one of these to select how the DSP code is supplied to
the adapter.
End users writing applications that use the HPI interface do not have to
use any of the below defines; they are only necessary for building drivers
HPI_DSPCODE_FILE:
DSP code is supplied as a file that is opened and read from by the driver.
HPI_DSPCODE_FIRMWARE:
DSP code is read using the hotplug firmware loader module.
Only valid when compiling the HPI kernel driver under Linux.
*/
/***********************************************************************/
#ifndef _HPIDSPCD_H_
@ -40,37 +27,56 @@ DSP code is read using the hotplug firmware loader module.
#include "hpi_internal.h"
#ifndef DISABLE_PRAGMA_PACK1
#pragma pack(push, 1)
#endif
/** Code header version is decimal encoded e.g. 4.06.10 is 40601 */
#define HPI_VER_DECIMAL ((int)(HPI_VER_MAJOR(HPI_VER) * 10000 + \
HPI_VER_MINOR(HPI_VER) * 100 + HPI_VER_RELEASE(HPI_VER)))
/** Header structure for dsp firmware file
This structure must match that used in s2bin.c for generation of asidsp.bin
*/
/*#ifndef DISABLE_PRAGMA_PACK1 */
/*#pragma pack(push, 1) */
/*#endif */
struct code_header {
/** Size in bytes including header */
u32 size;
/** File type tag "CODE" == 0x45444F43 */
u32 type;
/** Adapter model number */
u32 adapter;
/** Firmware version*/
u32 version;
/** Data checksum */
u32 checksum;
};
/*#ifndef DISABLE_PRAGMA_PACK1 */
/*#pragma pack(pop) */
/*#endif */
/*? Don't need the pragmas? */
compile_time_assert((sizeof(struct code_header) == 20), code_header_size);
/** Descriptor for dspcode from firmware loader */
struct dsp_code {
/** Firmware descriptor */
const struct firmware *ps_firmware;
struct pci_dev *ps_dev;
/** copy of file header */
struct code_header header;
/** Expected number of words in the whole dsp code,INCL header */
long int block_length;
u32 block_length;
/** Number of words read so far */
long int word_count;
/** Version read from dsp code file */
u32 version;
/** CRC read from dsp code file */
u32 crc;
u32 word_count;
/** internal state of DSP code reader */
struct dsp_code_private *pvt;
};
#ifndef DISABLE_PRAGMA_PACK1
#pragma pack(pop)
#endif
/** Prepare *psDspCode to refer to the requuested adapter.
Searches the file, or selects the appropriate linked array
/** Prepare *psDspCode to refer to the requested adapter's firmware.
Code file name is obtained from HpiOs_GetDspCodePath
\return 0 for success, or error code if requested code is not available
*/
short hpi_dsp_code_open(
/** Code identifier, usually adapter family */
u32 adapter,
u32 adapter, void *pci_dev,
/** Pointer to DSP code control structure */
struct dsp_code *ps_dsp_code,
/** Pointer to dword to receive OS specific error code */

88
sound/pci/asihpi/hpifunc.c
View File

@ -1663,68 +1663,64 @@ u16 hpi_channel_mode_get(u32 h_control, u16 *mode)
u16 hpi_cobranet_hmi_write(u32 h_control, u32 hmi_address, u32 byte_count,
u8 *pb_data)
{
struct hpi_message hm;
struct hpi_response hr;
struct hpi_msg_cobranet_hmiwrite hm;
struct hpi_response_header hr;
hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
HPI_CONTROL_SET_STATE);
if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
hpi_init_message_responseV1(&hm.h, sizeof(hm), &hr, sizeof(hr),
HPI_OBJ_CONTROL, HPI_CONTROL_SET_STATE);
if (hpi_handle_indexes(h_control, &hm.h.adapter_index,
&hm.h.obj_index))
return HPI_ERROR_INVALID_HANDLE;
hm.u.cx.u.cobranet_data.byte_count = byte_count;
hm.u.cx.u.cobranet_data.hmi_address = hmi_address;
if (byte_count > sizeof(hm.bytes))
return HPI_ERROR_MESSAGE_BUFFER_TOO_SMALL;
if (byte_count <= 8) {
memcpy(hm.u.cx.u.cobranet_data.data, pb_data, byte_count);
hm.u.cx.attribute = HPI_COBRANET_SET;
} else {
hm.u.cx.u.cobranet_bigdata.pb_data = pb_data;
hm.u.cx.attribute = HPI_COBRANET_SET_DATA;
}
hpi_send_recv(&hm, &hr);
hm.p.attribute = HPI_COBRANET_SET;
hm.p.byte_count = byte_count;
hm.p.hmi_address = hmi_address;
memcpy(hm.bytes, pb_data, byte_count);
hm.h.size = (u16)(sizeof(hm.h) + sizeof(hm.p) + byte_count);
hpi_send_recvV1(&hm.h, &hr);
return hr.error;
}
u16 hpi_cobranet_hmi_read(u32 h_control, u32 hmi_address, u32 max_byte_count,
u32 *pbyte_count, u8 *pb_data)
{
struct hpi_message hm;
struct hpi_response hr;
struct hpi_msg_cobranet_hmiread hm;
struct hpi_res_cobranet_hmiread hr;
hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
HPI_CONTROL_GET_STATE);
if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
hpi_init_message_responseV1(&hm.h, sizeof(hm), &hr.h, sizeof(hr),
HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE);
if (hpi_handle_indexes(h_control, &hm.h.adapter_index,
&hm.h.obj_index))
return HPI_ERROR_INVALID_HANDLE;
hm.u.cx.u.cobranet_data.byte_count = max_byte_count;
hm.u.cx.u.cobranet_data.hmi_address = hmi_address;
if (max_byte_count > sizeof(hr.bytes))
return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
if (max_byte_count <= 8) {
hm.u.cx.attribute = HPI_COBRANET_GET;
} else {
hm.u.cx.u.cobranet_bigdata.pb_data = pb_data;
hm.u.cx.attribute = HPI_COBRANET_GET_DATA;
}
hm.p.attribute = HPI_COBRANET_GET;
hm.p.byte_count = max_byte_count;
hm.p.hmi_address = hmi_address;
hpi_send_recv(&hm, &hr);
if (!hr.error && pb_data) {
hpi_send_recvV1(&hm.h, &hr.h);
*pbyte_count = hr.u.cx.u.cobranet_data.byte_count;
if (!hr.h.error && pb_data) {
if (hr.byte_count > sizeof(hr.bytes))
if (*pbyte_count < max_byte_count)
return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL;
*pbyte_count = hr.byte_count;
if (hr.byte_count < max_byte_count)
max_byte_count = *pbyte_count;
if (hm.u.cx.attribute == HPI_COBRANET_GET) {
memcpy(pb_data, hr.u.cx.u.cobranet_data.data,
max_byte_count);
} else {
}
memcpy(pb_data, hr.bytes, max_byte_count);
}
return hr.error;
return hr.h.error;
}
u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus,
@ -1733,23 +1729,23 @@ u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus,
struct hpi_message hm;
struct hpi_response hr;
hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROLEX,
hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL,
HPI_CONTROL_GET_STATE);
if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index))
return HPI_ERROR_INVALID_HANDLE;
hm.u.cx.attribute = HPI_COBRANET_GET_STATUS;
hm.u.c.attribute = HPI_COBRANET_GET_STATUS;
hpi_send_recv(&hm, &hr);
if (!hr.error) {
if (pstatus)
*pstatus = hr.u.cx.u.cobranet_status.status;
*pstatus = hr.u.cu.cobranet.status.status;
if (preadable_size)
*preadable_size =
hr.u.cx.u.cobranet_status.readable_size;
hr.u.cu.cobranet.status.readable_size;
if (pwriteable_size)
*pwriteable_size =
hr.u.cx.u.cobranet_status.writeable_size;
hr.u.cu.cobranet.status.writeable_size;
}
return hr.error;
}

4
sound/pci/asihpi/hpimsginit.c
View File

@ -46,7 +46,7 @@ static void hpi_init_message(struct hpi_message *phm, u16 object,
if (gwSSX2_bypass)
phm->type = HPI_TYPE_SSX2BYPASS_MESSAGE;
else
phm->type = HPI_TYPE_MESSAGE;
phm->type = HPI_TYPE_REQUEST;
phm->object = object;
phm->function = function;
phm->version = 0;
@ -89,7 +89,7 @@ static void hpi_init_messageV1(struct hpi_message_header *phm, u16 size,
memset(phm, 0, sizeof(*phm));
if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) {
phm->size = size;
phm->type = HPI_TYPE_MESSAGE;
phm->type = HPI_TYPE_REQUEST;
phm->object = object;
phm->function = function;
phm->version = 1;

6
sound/pci/asihpi/hpimsgx.c
View File

@ -16,7 +16,7 @@
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Extended Message Function With Response Cacheing
Extended Message Function With Response Caching
(C) Copyright AudioScience Inc. 2002
*****************************************************************************/
@ -186,7 +186,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr,
/* Initialize this module's internal state */
hpios_msgxlock_init(&msgx_lock);
memset(&hpi_entry_points, 0, sizeof(hpi_entry_points));
hpios_locked_mem_init();
/* Init subsys_findadapters response to no-adapters */
HPIMSGX__reset(HPIMSGX_ALLADAPTERS);
hpi_init_response(phr, HPI_OBJ_SUBSYSTEM,
@ -197,7 +196,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr,
case HPI_SUBSYS_DRIVER_UNLOAD:
HPI_COMMON(phm, phr);
HPIMSGX__cleanup(HPIMSGX_ALLADAPTERS, h_owner);
hpios_locked_mem_free_all();
hpi_init_response(phr, HPI_OBJ_SUBSYSTEM,
HPI_SUBSYS_DRIVER_UNLOAD, 0);
return;
@ -315,7 +313,7 @@ void hpi_send_recv_ex(struct hpi_message *phm, struct hpi_response *phr,
{
HPI_DEBUG_MESSAGE(DEBUG, phm);
if (phm->type != HPI_TYPE_MESSAGE) {
if (phm->type != HPI_TYPE_REQUEST) {
hpi_init_response(phr, phm->object, phm->function,
HPI_ERROR_INVALID_TYPE);
return;

10
sound/pci/asihpi/hpioctl.c
View File

@ -1,7 +1,7 @@
/*******************************************************************************
AudioScience HPI driver
Copyright (C) 1997-2010 AudioScience Inc. <support@audioscience.com>
Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License as
@ -157,11 +157,6 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
goto out;
}
if (hm->h.adapter_index >= HPI_MAX_ADAPTERS) {
err = -EINVAL;
goto out;
}
switch (hm->h.function) {
case HPI_SUBSYS_CREATE_ADAPTER:
case HPI_ADAPTER_DELETE:
@ -187,7 +182,6 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
/* -1=no data 0=read from user mem, 1=write to user mem */
int wrflag = -1;
u32 adapter = hm->h.adapter_index;
pa = &adapters[adapter];
if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) {
hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER,
@ -203,6 +197,8 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
goto out;
}
pa = &adapters[adapter];
if (mutex_lock_interruptible(&adapters[adapter].mutex)) {
err = -EINTR;
goto out;

8
sound/pci/asihpi/hpios.c
View File

@ -39,10 +39,6 @@ void hpios_delay_micro_seconds(u32 num_micro_sec)
}
void hpios_locked_mem_init(void)
{
}
/** Allocated an area of locked memory for bus master DMA operations.
On error, return -ENOMEM, and *pMemArea.size = 0
@ -85,7 +81,3 @@ u16 hpios_locked_mem_free(struct consistent_dma_area *p_mem_area)
return 1;
}
}
void hpios_locked_mem_free_all(void)
{
}

1
sound/pci/asihpi/hpios.h
View File

@ -38,6 +38,7 @@ HPI Operating System Specific macros for Linux Kernel driver
#include <linux/firmware.h>
#include <linux/interrupt.h>
#include <linux/pci.h>
#include <linux/mutex.h>
#define HPI_NO_OS_FILE_OPS

4
sound/pci/atiixp.c
View File

@ -1624,7 +1624,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
}
if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
card->shortname, chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_atiixp_free(chip);
return -EBUSY;
@ -1701,7 +1701,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "ATI IXP AC97 controller",
.name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
.remove = __devexit_p(snd_atiixp_remove),

4
sound/pci/atiixp_modem.c
View File

@ -1260,7 +1260,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card,
}
if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
card->shortname, chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_atiixp_free(chip);
return -EBUSY;
@ -1332,7 +1332,7 @@ static void __devexit snd_atiixp_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "ATI IXP MC97 controller",
.name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
.remove = __devexit_p(snd_atiixp_remove),

4
sound/pci/au88x0/au88x0.c
View File

@ -196,7 +196,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
}
if ((err = request_irq(pci->irq, vortex_interrupt,
IRQF_SHARED, CARD_NAME_SHORT,
IRQF_SHARED, KBUILD_MODNAME,
chip)) != 0) {
printk(KERN_ERR "cannot grab irq\n");
goto irq_out;
@ -375,7 +375,7 @@ static void __devexit snd_vortex_remove(struct pci_dev *pci)
// pci_driver definition
static struct pci_driver driver = {
.name = CARD_NAME_SHORT,
.name = KBUILD_MODNAME,
.id_table = snd_vortex_ids,
.probe = snd_vortex_probe,
.remove = __devexit_p(snd_vortex_remove),

4
sound/pci/aw2/aw2-alsa.c
View File

@ -171,7 +171,7 @@ MODULE_DEVICE_TABLE(pci, snd_aw2_ids);
/* pci_driver definition */
static struct pci_driver driver = {
.name = "Emagic Audiowerk 2",
.name = KBUILD_MODNAME,
.id_table = snd_aw2_ids,
.probe = snd_aw2_probe,
.remove = __devexit_p(snd_aw2_remove),
@ -317,7 +317,7 @@ static int __devinit snd_aw2_create(struct snd_card *card,
snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
IRQF_SHARED, "Audiowerk2", chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "aw2: Cannot grab irq %d\n", pci->irq);
iounmap(chip->iobase_virt);

4
sound/pci/azt3328.c
View File

@ -2559,7 +2559,7 @@ snd_azf3328_create(struct snd_card *card,
codec_setup->name = "I2S_OUT";
if (request_irq(pci->irq, snd_azf3328_interrupt,
IRQF_SHARED, card->shortname, chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
err = -EBUSY;
goto out_err;
@ -2860,7 +2860,7 @@ snd_azf3328_resume(struct pci_dev *pci)
static struct pci_driver driver = {
.name = "AZF3328",
.name = KBUILD_MODNAME,
.id_table = snd_azf3328_ids,
.probe = snd_azf3328_probe,
.remove = __devexit_p(snd_azf3328_remove),

4
sound/pci/bt87x.c
View File

@ -760,7 +760,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card,
snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS);
err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
"Bt87x audio", chip);
KBUILD_MODNAME, chip);
if (err < 0) {
snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
goto fail;
@ -965,7 +965,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = {
};
static struct pci_driver driver = {
.name = "Bt87x",
.name = KBUILD_MODNAME,
.id_table = snd_bt87x_ids,
.probe = snd_bt87x_probe,
.remove = __devexit_p(snd_bt87x_remove),

4
sound/pci/ca0106/ca0106_main.c
View File

@ -1666,7 +1666,7 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
}
if (request_irq(pci->irq, snd_ca0106_interrupt,
IRQF_SHARED, "snd_ca0106", chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
@ -1933,7 +1933,7 @@ MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
// pci_driver definition
static struct pci_driver driver = {
.name = "CA0106",
.name = KBUILD_MODNAME,
.id_table = snd_ca0106_ids,
.probe = snd_ca0106_probe,
.remove = __devexit_p(snd_ca0106_remove),

4
sound/pci/cmipci.c
View File

@ -3053,7 +3053,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
cm->iobase = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_cmipci_interrupt,
IRQF_SHARED, card->driver, cm)) {
IRQF_SHARED, KBUILD_MODNAME, cm)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_cmipci_free(cm);
return -EBUSY;
@ -3398,7 +3398,7 @@ static int snd_cmipci_resume(struct pci_dev *pci)
#endif /* CONFIG_PM */
static struct pci_driver driver = {
.name = "C-Media PCI",
.name = KBUILD_MODNAME,
.id_table = snd_cmipci_ids,
.probe = snd_cmipci_probe,
.remove = __devexit_p(snd_cmipci_remove),

4
sound/pci/cs4281.c
View File

@ -1382,7 +1382,7 @@ static int __devinit snd_cs4281_create(struct snd_card *card,
}
if (request_irq(pci->irq, snd_cs4281_interrupt, IRQF_SHARED,
"CS4281", chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_cs4281_free(chip);
return -ENOMEM;
@ -2085,7 +2085,7 @@ static int cs4281_resume(struct pci_dev *pci)
#endif /* CONFIG_PM */
static struct pci_driver driver = {
.name = "CS4281",
.name = KBUILD_MODNAME,
.id_table = snd_cs4281_ids,
.probe = snd_cs4281_probe,
.remove = __devexit_p(snd_cs4281_remove),

2
sound/pci/cs46xx/cs46xx.c
View File

@ -162,7 +162,7 @@ static void __devexit snd_card_cs46xx_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "Sound Fusion CS46xx",
.name = KBUILD_MODNAME,
.id_table = snd_cs46xx_ids,
.probe = snd_card_cs46xx_probe,
.remove = __devexit_p(snd_card_cs46xx_remove),

2
sound/pci/cs46xx/cs46xx_lib.c
View File

@ -3835,7 +3835,7 @@ int __devinit snd_cs46xx_create(struct snd_card *card,
}
if (request_irq(pci->irq, snd_cs46xx_interrupt, IRQF_SHARED,
"CS46XX", chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_cs46xx_free(chip);
return -EBUSY;

2
sound/pci/cs5530.c
View File

@ -285,7 +285,7 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci,
}
static struct pci_driver driver = {
.name = "CS5530_Audio",
.name = KBUILD_MODNAME,
.id_table = snd_cs5530_ids,
.probe = snd_cs5530_probe,
.remove = __devexit_p(snd_cs5530_remove),

4
sound/pci/cs5535audio/cs5535audio.c
View File

@ -311,7 +311,7 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card,
cs5535au->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_cs5535audio_interrupt,
IRQF_SHARED, "CS5535 Audio", cs5535au)) {
IRQF_SHARED, KBUILD_MODNAME, cs5535au)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
err = -EBUSY;
goto sndfail;
@ -395,7 +395,7 @@ static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = DRIVER_NAME,
.name = KBUILD_MODNAME,
.id_table = snd_cs5535audio_ids,
.probe = snd_cs5535audio_probe,
.remove = __devexit_p(snd_cs5535audio_remove),

1
sound/pci/ctxfi/ct20k2reg.h
View File

@ -55,6 +55,7 @@
/* GPIO Registers */
#define GPIO_DATA 0x1B7020
#define GPIO_CTRL 0x1B7024
#define GPIO_EXT_DATA 0x1B70A0
/* Virtual memory registers */
#define VMEM_PTPAL 0x1C6300 /* 0x1C6300 + (16 * Chn) */

107
sound/pci/ctxfi/ctatc.c
View File

@ -18,7 +18,6 @@
#include "ctatc.h"
#include "ctpcm.h"
#include "ctmixer.h"
#include "cthardware.h"
#include "ctsrc.h"
#include "ctamixer.h"
#include "ctdaio.h"
@ -30,7 +29,6 @@
#include <sound/asoundef.h>
#define MONO_SUM_SCALE 0x19a8 /* 2^(-0.5) in 14-bit floating format */
#define DAIONUM 7
#define MAX_MULTI_CHN 8
#define IEC958_DEFAULT_CON ((IEC958_AES0_NONAUDIO \
@ -53,6 +51,8 @@ static struct snd_pci_quirk __devinitdata subsys_20k1_list[] = {
static struct snd_pci_quirk __devinitdata subsys_20k2_list[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB0760,
"SB0760", CTSB0760),
SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB1270,
"SB1270", CTSB1270),
SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08801,
"SB0880", CTSB0880),
SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08802,
@ -75,6 +75,7 @@ static const char *ct_subsys_name[NUM_CTCARDS] = {
[CTSB0760] = "SB076x",
[CTHENDRIX] = "Hendrix",
[CTSB0880] = "SB0880",
[CTSB1270] = "SB1270",
[CT20K2_UNKNOWN] = "Unknown",
};
@ -459,12 +460,12 @@ static void setup_src_node_conf(struct ct_atc *atc, struct ct_atc_pcm *apcm,
apcm->substream->runtime->rate);
*n_srcc = 0;
if (1 == atc->msr) {
if (1 == atc->msr) { /* FIXME: do we really need SRC here if pitch==1 */
*n_srcc = apcm->substream->runtime->channels;
conf[0].pitch = pitch;
conf[0].mix_msr = conf[0].imp_msr = conf[0].msr = 1;
conf[0].vo = 1;
} else if (2 == atc->msr) {
} else if (2 <= atc->msr) {
if (0x8000000 < pitch) {
/* Need two-stage SRCs, SRCIMPs and
* AMIXERs for converting format */
@ -970,11 +971,39 @@ static int atc_select_mic_in(struct ct_atc *atc)
return 0;
}
static int atc_have_digit_io_switch(struct ct_atc *atc)
static struct capabilities atc_capabilities(struct ct_atc *atc)
{
struct hw *hw = atc->hw;
return hw->have_digit_io_switch(hw);
return hw->capabilities(hw);
}
static int atc_output_switch_get(struct ct_atc *atc)
{
struct hw *hw = atc->hw;
return hw->output_switch_get(hw);
}
static int atc_output_switch_put(struct ct_atc *atc, int position)
{
struct hw *hw = atc->hw;
return hw->output_switch_put(hw, position);
}
static int atc_mic_source_switch_get(struct ct_atc *atc)
{
struct hw *hw = atc->hw;
return hw->mic_source_switch_get(hw);
}
static int atc_mic_source_switch_put(struct ct_atc *atc, int position)
{
struct hw *hw = atc->hw;
return hw->mic_source_switch_put(hw, position);
}
static int atc_select_digit_io(struct ct_atc *atc)
@ -1045,6 +1074,11 @@ static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
return atc_daio_unmute(atc, state, LINEIM);
}
static int atc_mic_unmute(struct ct_atc *atc, unsigned char state)
{
return atc_daio_unmute(atc, state, MIC);
}
static int atc_spdif_out_unmute(struct ct_atc *atc, unsigned char state)
{
return atc_daio_unmute(atc, state, SPDIFOO);
@ -1331,17 +1365,20 @@ static int atc_get_resources(struct ct_atc *atc)
struct srcimp_mgr *srcimp_mgr;
struct sum_desc sum_dsc = {0};
struct sum_mgr *sum_mgr;
int err, i;
int err, i, num_srcs, num_daios;
atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL);
num_daios = ((atc->model == CTSB1270) ? 8 : 7);
num_srcs = ((atc->model == CTSB1270) ? 6 : 4);
atc->daios = kzalloc(sizeof(void *)*num_daios, GFP_KERNEL);
if (!atc->daios)
return -ENOMEM;
atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
atc->srcs = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL);
if (!atc->srcs)
return -ENOMEM;
atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
atc->srcimps = kzalloc(sizeof(void *)*num_srcs, GFP_KERNEL);
if (!atc->srcimps)
return -ENOMEM;
@ -1351,8 +1388,9 @@ static int atc_get_resources(struct ct_atc *atc)
daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO];
da_desc.msr = atc->msr;
for (i = 0, atc->n_daio = 0; i < DAIONUM-1; i++) {
da_desc.type = i;
for (i = 0, atc->n_daio = 0; i < num_daios; i++) {
da_desc.type = (atc->model != CTSB073X) ? i :
((i == SPDIFIO) ? SPDIFI1 : i);
err = daio_mgr->get_daio(daio_mgr, &da_desc,
(struct daio **)&atc->daios[i]);
if (err) {
@ -1362,23 +1400,12 @@ static int atc_get_resources(struct ct_atc *atc)
}
atc->n_daio++;
}
if (atc->model == CTSB073X)
da_desc.type = SPDIFI1;
else
da_desc.type = SPDIFIO;
err = daio_mgr->get_daio(daio_mgr, &da_desc,
(struct daio **)&atc->daios[i]);
if (err) {
printk(KERN_ERR "ctxfi: Failed to get S/PDIF-in resource!!!\n");
return err;
}
atc->n_daio++;
src_mgr = atc->rsc_mgrs[SRC];
src_dsc.multi = 1;
src_dsc.msr = atc->msr;
src_dsc.mode = ARCRW;
for (i = 0, atc->n_src = 0; i < (2*2); i++) {
for (i = 0, atc->n_src = 0; i < num_srcs; i++) {
err = src_mgr->get_src(src_mgr, &src_dsc,
(struct src **)&atc->srcs[i]);
if (err)
@ -1388,8 +1415,8 @@ static int atc_get_resources(struct ct_atc *atc)
}
srcimp_mgr = atc->rsc_mgrs[SRCIMP];
srcimp_dsc.msr = 8; /* SRCIMPs for S/PDIFIn SRT */
for (i = 0, atc->n_srcimp = 0; i < (2*1); i++) {
srcimp_dsc.msr = 8;
for (i = 0, atc->n_srcimp = 0; i < num_srcs; i++) {
err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
(struct srcimp **)&atc->srcimps[i]);
if (err)
@ -1397,15 +1424,6 @@ static int atc_get_resources(struct ct_atc *atc)
atc->n_srcimp++;
}
srcimp_dsc.msr = 8; /* SRCIMPs for LINE/MICIn SRT */
for (i = 0; i < (2*1); i++) {
err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
(struct srcimp **)&atc->srcimps[2*1+i]);
if (err)
return err;
atc->n_srcimp++;
}
sum_mgr = atc->rsc_mgrs[SUM];
sum_dsc.msr = atc->msr;
@ -1488,6 +1506,18 @@ static void atc_connect_resources(struct ct_atc *atc)
src = atc->srcs[3];
mixer->set_input_right(mixer, MIX_LINE_IN, &src->rsc);
if (atc->model == CTSB1270) {
/* Titanium HD has a dedicated ADC for the Mic. */
dai = container_of(atc->daios[MIC], struct dai, daio);
atc_connect_dai(atc->rsc_mgrs[SRC], dai,
(struct src **)&atc->srcs[4],
(struct srcimp **)&atc->srcimps[4]);
src = atc->srcs[4];
mixer->set_input_left(mixer, MIX_MIC_IN, &src->rsc);
src = atc->srcs[5];
mixer->set_input_right(mixer, MIX_MIC_IN, &src->rsc);
}
dai = container_of(atc->daios[SPDIFIO], struct dai, daio);
atc_connect_dai(atc->rsc_mgrs[SRC], dai,
(struct src **)&atc->srcs[0],
@ -1606,12 +1636,17 @@ static struct ct_atc atc_preset __devinitdata = {
.line_clfe_unmute = atc_line_clfe_unmute,
.line_rear_unmute = atc_line_rear_unmute,
.line_in_unmute = atc_line_in_unmute,
.mic_unmute = atc_mic_unmute,
.spdif_out_unmute = atc_spdif_out_unmute,
.spdif_in_unmute = atc_spdif_in_unmute,
.spdif_out_get_status = atc_spdif_out_get_status,
.spdif_out_set_status = atc_spdif_out_set_status,
.spdif_out_passthru = atc_spdif_out_passthru,
.have_digit_io_switch = atc_have_digit_io_switch,
.capabilities = atc_capabilities,
.output_switch_get = atc_output_switch_get,
.output_switch_put = atc_output_switch_put,
.mic_source_switch_get = atc_mic_source_switch_get,
.mic_source_switch_put = atc_mic_source_switch_put,
#ifdef CONFIG_PM
.suspend = atc_suspend,
.resume = atc_resume,

8
sound/pci/ctxfi/ctatc.h
View File

@ -25,6 +25,7 @@
#include <sound/core.h>
#include "ctvmem.h"
#include "cthardware.h"
#include "ctresource.h"
enum CTALSADEVS { /* Types of alsa devices */
@ -115,12 +116,17 @@ struct ct_atc {
int (*line_clfe_unmute)(struct ct_atc *atc, unsigned char state);
int (*line_rear_unmute)(struct ct_atc *atc, unsigned char state);
int (*line_in_unmute)(struct ct_atc *atc, unsigned char state);
int (*mic_unmute)(struct ct_atc *atc, unsigned char state);
int (*spdif_out_unmute)(struct ct_atc *atc, unsigned char state);
int (*spdif_in_unmute)(struct ct_atc *atc, unsigned char state);
int (*spdif_out_get_status)(struct ct_atc *atc, unsigned int *status);
int (*spdif_out_set_status)(struct ct_atc *atc, unsigned int status);
int (*spdif_out_passthru)(struct ct_atc *atc, unsigned char state);
int (*have_digit_io_switch)(struct ct_atc *atc);
struct capabilities (*capabilities)(struct ct_atc *atc);
int (*output_switch_get)(struct ct_atc *atc);
int (*output_switch_put)(struct ct_atc *atc, int position);
int (*mic_source_switch_get)(struct ct_atc *atc);
int (*mic_source_switch_put)(struct ct_atc *atc, int position);
/* Don't touch! Used for internal object. */
void *rsc_mgrs[NUM_RSCTYP]; /* chip resource managers */

23
sound/pci/ctxfi/ctdaio.c
View File

@ -22,20 +22,9 @@
#include <linux/slab.h>
#include <linux/kernel.h>
#define DAIO_RESOURCE_NUM NUM_DAIOTYP
#define DAIO_OUT_MAX SPDIFOO
union daio_usage {
struct {
unsigned short lineo1:1;
unsigned short lineo2:1;
unsigned short lineo3:1;
unsigned short lineo4:1;
unsigned short spdifoo:1;
unsigned short lineim:1;
unsigned short spdifio:1;
unsigned short spdifi1:1;
} bf;
struct daio_usage {
unsigned short data;
};
@ -61,6 +50,7 @@ struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
[LINEO3] = {.left = 0x50, .right = 0x51},
[LINEO4] = {.left = 0x70, .right = 0x71},
[LINEIM] = {.left = 0x45, .right = 0xc5},
[MIC] = {.left = 0x55, .right = 0xd5},
[SPDIFOO] = {.left = 0x00, .right = 0x01},
[SPDIFIO] = {.left = 0x05, .right = 0x85},
};
@ -138,6 +128,7 @@ static unsigned int daio_device_index(enum DAIOTYP type, struct hw *hw)
case LINEO3: return 5;
case LINEO4: return 6;
case LINEIM: return 4;
case MIC: return 5;
default: return -EINVAL;
}
default:
@ -519,17 +510,17 @@ static int dai_rsc_uninit(struct dai *dai)
static int daio_mgr_get_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
{
if (((union daio_usage *)mgr->rscs)->data & (0x1 << type))
if (((struct daio_usage *)mgr->rscs)->data & (0x1 << type))
return -ENOENT;
((union daio_usage *)mgr->rscs)->data |= (0x1 << type);
((struct daio_usage *)mgr->rscs)->data |= (0x1 << type);
return 0;
}
static int daio_mgr_put_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
{
((union daio_usage *)mgr->rscs)->data &= ~(0x1 << type);
((struct daio_usage *)mgr->rscs)->data &= ~(0x1 << type);
return 0;
}
@ -712,7 +703,7 @@ int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr)
if (!daio_mgr)
return -ENOMEM;
err = rsc_mgr_init(&daio_mgr->mgr, DAIO, DAIO_RESOURCE_NUM, hw);
err = rsc_mgr_init(&daio_mgr->mgr, DAIO, NUM_DAIOTYP, hw);
if (err)
goto error1;

1
sound/pci/ctxfi/ctdaio.h
View File

@ -33,6 +33,7 @@ enum DAIOTYP {
SPDIFOO, /* S/PDIF Out (Flexijack/Optical) */
LINEIM,
SPDIFIO, /* S/PDIF In (Flexijack/Optical) on the card */
MIC, /* Dedicated mic on Titanium HD */
SPDIFI1, /* S/PDIF In on internal Drive Bay */
NUM_DAIOTYP
};

14
sound/pci/ctxfi/cthardware.h
View File

@ -39,6 +39,7 @@ enum CTCARDS {
CT20K2_MODEL_FIRST = CTSB0760,
CTHENDRIX,
CTSB0880,
CTSB1270,
CT20K2_UNKNOWN,
NUM_CTCARDS /* This should always be the last */
};
@ -60,6 +61,13 @@ struct card_conf {
unsigned int msr; /* master sample rate in rsrs */
};
struct capabilities {
unsigned int digit_io_switch:1;
unsigned int dedicated_mic:1;
unsigned int output_switch:1;
unsigned int mic_source_switch:1;
};
struct hw {
int (*card_init)(struct hw *hw, struct card_conf *info);
int (*card_stop)(struct hw *hw);
@ -70,7 +78,11 @@ struct hw {
#endif
int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source);
int (*select_adc_source)(struct hw *hw, enum ADCSRC source);
int (*have_digit_io_switch)(struct hw *hw);
struct capabilities (*capabilities)(struct hw *hw);
int (*output_switch_get)(struct hw *hw);
int (*output_switch_put)(struct hw *hw, int position);
int (*mic_source_switch_get)(struct hw *hw);
int (*mic_source_switch_put)(struct hw *hw, int position);
/* SRC operations */
int (*src_rsc_get_ctrl_blk)(void **rblk);

15
sound/pci/ctxfi/cthw20k1.c
View File

@ -1777,10 +1777,17 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
return adc_init_SBx(hw, info->input, info->mic20db);
}
static int hw_have_digit_io_switch(struct hw *hw)
static struct capabilities hw_capabilities(struct hw *hw)
{
struct capabilities cap;
/* SB073x and Vista compatible cards have no digit IO switch */
return !(hw->model == CTSB073X || hw->model == CTUAA);
cap.digit_io_switch = !(hw->model == CTSB073X || hw->model == CTUAA);
cap.dedicated_mic = 0;
cap.output_switch = 0;
cap.mic_source_switch = 0;
return cap;
}
#define CTLBITS(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
@ -1933,7 +1940,7 @@ static int hw_card_start(struct hw *hw)
if (hw->irq < 0) {
err = request_irq(pci->irq, ct_20k1_interrupt, IRQF_SHARED,
"ctxfi", hw);
KBUILD_MODNAME, hw);
if (err < 0) {
printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
goto error2;
@ -2172,7 +2179,7 @@ static struct hw ct20k1_preset __devinitdata = {
.pll_init = hw_pll_init,
.is_adc_source_selected = hw_is_adc_input_selected,
.select_adc_source = hw_adc_input_select,
.have_digit_io_switch = hw_have_digit_io_switch,
.capabilities = hw_capabilities,
#ifdef CONFIG_PM
.suspend = hw_suspend,
.resume = hw_resume,

341
sound/pci/ctxfi/cthw20k2.c
View File

@ -8,7 +8,7 @@
* @File cthw20k2.c
*
* @Brief
* This file contains the implementation of hardware access methord for 20k2.
* This file contains the implementation of hardware access method for 20k2.
*
* @Author Liu Chun
* @Date May 14 2008
@ -38,6 +38,8 @@ struct hw20k2 {
unsigned char dev_id;
unsigned char addr_size;
unsigned char data_size;
int mic_source;
};
static u32 hw_read_20kx(struct hw *hw, u32 reg);
@ -1163,7 +1165,12 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x01010101);
hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
} else if (2 == info->msr) {
hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111);
if (hw->model != CTSB1270) {
hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111);
} else {
/* PCM4220 on Titanium HD is different. */
hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11011111);
}
/* Specify all playing 96khz
* EA [0] - Enabled
* RTA [4:5] - 96kHz
@ -1175,6 +1182,10 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
* RTD [28:29] - 96kHz */
hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x11111111);
hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
} else if ((4 == info->msr) && (hw->model == CTSB1270)) {
hw_write_20kx(hw, AUDIO_IO_MCLK, 0x21011111);
hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x21212121);
hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
} else {
printk(KERN_ALERT "ctxfi: ERROR!!! Invalid sampling rate!!!\n");
return -EINVAL;
@ -1182,6 +1193,8 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
for (i = 0; i < 8; i++) {
if (i <= 3) {
/* This comment looks wrong since loop is over 4 */
/* channels and emu20k2 supports 4 spdif IOs. */
/* 1st 3 channels are SPDIFs (SB0960) */
if (i == 3)
data = 0x1001001;
@ -1206,12 +1219,16 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_H+(0x40*i), 0x0B);
} else {
/* Again, loop is over 4 channels not 5. */
/* Next 5 channels are I2S (SB0960) */
data = 0x11;
hw_write_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i), data);
if (2 == info->msr) {
/* Four channels per sample period */
data |= 0x1000;
} else if (4 == info->msr) {
/* FIXME: check this against the chip spec */
data |= 0x2000;
}
hw_write_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i), data);
}
@ -1299,21 +1316,18 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr)
pllenb = 0xB;
hw_write_20kx(hw, PLL_ENB, pllenb);
pllctl = 0x20D00000;
set_field(&pllctl, PLLCTL_FD, 16 - 4);
hw_write_20kx(hw, PLL_CTL, pllctl);
mdelay(40);
pllctl = hw_read_20kx(hw, PLL_CTL);
pllctl = 0x20C00000;
set_field(&pllctl, PLLCTL_B, 0);
if (48000 == rsr) {
set_field(&pllctl, PLLCTL_FD, 16 - 2);
set_field(&pllctl, PLLCTL_RD, 1 - 1); /* 3000*16/1 = 48000 */
} else { /* 44100 */
set_field(&pllctl, PLLCTL_FD, 147 - 2);
set_field(&pllctl, PLLCTL_RD, 10 - 1); /* 3000*147/10 = 44100 */
}
set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4);
set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1);
hw_write_20kx(hw, PLL_CTL, pllctl);
mdelay(40);
pllctl = hw_read_20kx(hw, PLL_CTL);
set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2);
hw_write_20kx(hw, PLL_CTL, pllctl);
mdelay(40);
for (i = 0; i < 1000; i++) {
pllstat = hw_read_20kx(hw, PLL_STAT);
if (get_field(pllstat, PLLSTAT_PD))
@ -1557,7 +1571,7 @@ static int hw20k2_i2c_write(struct hw *hw, u16 addr, u32 data)
hw_write_20kx(hw, I2C_IF_STATUS, i2c_status);
hw20k2_i2c_wait_data_ready(hw);
/* Dummy write to trigger the write oprtation */
/* Dummy write to trigger the write operation */
hw_write_20kx(hw, I2C_IF_WDATA, 0);
hw20k2_i2c_wait_data_ready(hw);
@ -1568,6 +1582,30 @@ static int hw20k2_i2c_write(struct hw *hw, u16 addr, u32 data)
return 0;
}
static void hw_dac_stop(struct hw *hw)
{
u32 data;
data = hw_read_20kx(hw, GPIO_DATA);
data &= 0xFFFFFFFD;
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(10);
}
static void hw_dac_start(struct hw *hw)
{
u32 data;
data = hw_read_20kx(hw, GPIO_DATA);
data |= 0x2;
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(50);
}
static void hw_dac_reset(struct hw *hw)
{
hw_dac_stop(hw);
hw_dac_start(hw);
}
static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
{
int err;
@ -1594,6 +1632,21 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
0x00000000 /* Vol Control B4 */
};
if (hw->model == CTSB1270) {
hw_dac_stop(hw);
data = hw_read_20kx(hw, GPIO_DATA);
data &= ~0x0600;
if (1 == info->msr)
data |= 0x0000; /* Single Speed Mode 0-50kHz */
else if (2 == info->msr)
data |= 0x0200; /* Double Speed Mode 50-100kHz */
else
data |= 0x0600; /* Quad Speed Mode 100-200kHz */
hw_write_20kx(hw, GPIO_DATA, data);
hw_dac_start(hw);
return 0;
}
/* Set DAC reset bit as output */
data = hw_read_20kx(hw, GPIO_CTRL);
data |= 0x02;
@ -1606,22 +1659,8 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
for (i = 0; i < 2; i++) {
/* Reset DAC twice just in-case the chip
* didn't initialized properly */
data = hw_read_20kx(hw, GPIO_DATA);
/* GPIO data bit 1 */
data &= 0xFFFFFFFD;
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(10);
data |= 0x2;
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(50);
/* Reset the 2nd time */
data &= 0xFFFFFFFD;
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(10);
data |= 0x2;
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(50);
hw_dac_reset(hw);
hw_dac_reset(hw);
if (hw20k2_i2c_read(hw, CS4382_MC1, &cs_read.mode_control_1))
continue;
@ -1725,7 +1764,11 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type)
{
u32 data;
if (hw->model == CTSB1270) {
/* Titanium HD has two ADC chips, one for line in and one */
/* for MIC. We don't need to switch the ADC input. */
return 1;
}
data = hw_read_20kx(hw, GPIO_DATA);
switch (type) {
case ADC_MICIN:
@ -1742,35 +1785,47 @@ static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type)
#define MIC_BOOST_0DB 0xCF
#define MIC_BOOST_STEPS_PER_DB 2
#define MIC_BOOST_20DB (MIC_BOOST_0DB + 20 * MIC_BOOST_STEPS_PER_DB)
static void hw_wm8775_input_select(struct hw *hw, u8 input, s8 gain_in_db)
{
u32 adcmc, gain;
if (input > 3)
input = 3;
adcmc = ((u32)1 << input) | 0x100; /* Link L+R gain... */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, adcmc),
MAKE_WM8775_DATA(adcmc));
if (gain_in_db < -103)
gain_in_db = -103;
if (gain_in_db > 24)
gain_in_db = 24;
gain = gain_in_db * MIC_BOOST_STEPS_PER_DB + MIC_BOOST_0DB;
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, gain),
MAKE_WM8775_DATA(gain));
/* ...so there should be no need for the following. */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, gain),
MAKE_WM8775_DATA(gain));
}
static int hw_adc_input_select(struct hw *hw, enum ADCSRC type)
{
u32 data;
data = hw_read_20kx(hw, GPIO_DATA);
switch (type) {
case ADC_MICIN:
data |= (0x1 << 14);
hw_write_20kx(hw, GPIO_DATA, data);
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
MAKE_WM8775_DATA(0x101)); /* Mic-in */
hw20k2_i2c_write(hw,
MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB),
MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
hw20k2_i2c_write(hw,
MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB),
MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
hw_wm8775_input_select(hw, 0, 20); /* Mic, 20dB */
break;
case ADC_LINEIN:
data &= ~(0x1 << 14);
hw_write_20kx(hw, GPIO_DATA, data);
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
MAKE_WM8775_DATA(0x102)); /* Line-in */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
MAKE_WM8775_DATA(0xCF)); /* No boost */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
MAKE_WM8775_DATA(0xCF)); /* No boost */
hw_wm8775_input_select(hw, 1, 0); /* Line-in, 0dB */
break;
default:
break;
@ -1782,7 +1837,7 @@ static int hw_adc_input_select(struct hw *hw, enum ADCSRC type)
static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
{
int err;
u32 mux = 2, data, ctl;
u32 data, ctl;
/* Set ADC reset bit as output */
data = hw_read_20kx(hw, GPIO_CTRL);
@ -1796,19 +1851,42 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
goto error;
}
/* Make ADC in normal operation */
/* Reset the ADC (reset is active low). */
data = hw_read_20kx(hw, GPIO_DATA);
data &= ~(0x1 << 15);
hw_write_20kx(hw, GPIO_DATA, data);
if (hw->model == CTSB1270) {
/* Set up the PCM4220 ADC on Titanium HD */
data &= ~0x0C;
if (1 == info->msr)
data |= 0x00; /* Single Speed Mode 32-50kHz */
else if (2 == info->msr)
data |= 0x08; /* Double Speed Mode 50-108kHz */
else
data |= 0x04; /* Quad Speed Mode 108kHz-216kHz */
hw_write_20kx(hw, GPIO_DATA, data);
}
mdelay(10);
/* Return the ADC to normal operation. */
data |= (0x1 << 15);
hw_write_20kx(hw, GPIO_DATA, data);
mdelay(50);
/* I2C write to register offset 0x0B to set ADC LRCLK polarity */
/* invert bit, interface format to I2S, word length to 24-bit, */
/* enable ADC high pass filter. Fixes bug 5323? */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_IC, 0x26),
MAKE_WM8775_DATA(0x26));
/* Set the master mode (256fs) */
if (1 == info->msr) {
/* slave mode, 128x oversampling 256fs */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x02),
MAKE_WM8775_DATA(0x02));
} else if (2 == info->msr) {
} else if ((2 == info->msr) || (4 == info->msr)) {
/* slave mode, 64x oversampling, 256fs */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A),
MAKE_WM8775_DATA(0x0A));
} else {
@ -1818,55 +1896,113 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
goto error;
}
/* Configure GPIO bit 14 change to line-in/mic-in */
ctl = hw_read_20kx(hw, GPIO_CTRL);
ctl |= 0x1 << 14;
hw_write_20kx(hw, GPIO_CTRL, ctl);
/* Check using Mic-in or Line-in */
data = hw_read_20kx(hw, GPIO_DATA);
if (mux == 1) {
/* Configures GPIO data to select Mic-in */
data |= 0x1 << 14;
hw_write_20kx(hw, GPIO_DATA, data);
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
MAKE_WM8775_DATA(0x101)); /* Mic-in */
hw20k2_i2c_write(hw,
MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB),
MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
hw20k2_i2c_write(hw,
MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB),
MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */
} else if (mux == 2) {
/* Configures GPIO data to select Line-in */
data &= ~(0x1 << 14);
hw_write_20kx(hw, GPIO_DATA, data);
/* Setup ADC */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
MAKE_WM8775_DATA(0x102)); /* Line-in */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
MAKE_WM8775_DATA(0xCF)); /* No boost */
hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
MAKE_WM8775_DATA(0xCF)); /* No boost */
if (hw->model != CTSB1270) {
/* Configure GPIO bit 14 change to line-in/mic-in */
ctl = hw_read_20kx(hw, GPIO_CTRL);
ctl |= 0x1 << 14;
hw_write_20kx(hw, GPIO_CTRL, ctl);
hw_adc_input_select(hw, ADC_LINEIN);
} else {
printk(KERN_ALERT "ctxfi: ERROR!!! Invalid input mux!!!\n");
err = -EINVAL;
goto error;
hw_wm8775_input_select(hw, 0, 0);
}
return 0;
error:
hw20k2_i2c_uninit(hw);
return err;
}
static int hw_have_digit_io_switch(struct hw *hw)
static struct capabilities hw_capabilities(struct hw *hw)
{
return 0;
struct capabilities cap;
cap.digit_io_switch = 0;
cap.dedicated_mic = hw->model == CTSB1270;
cap.output_switch = hw->model == CTSB1270;
cap.mic_source_switch = hw->model == CTSB1270;
return cap;
}
static int hw_output_switch_get(struct hw *hw)
{
u32 data = hw_read_20kx(hw, GPIO_EXT_DATA);
switch (data & 0x30) {
case 0x00:
return 0;
case 0x10:
return 1;
case 0x20:
return 2;
default:
return 3;
}
}
static int hw_output_switch_put(struct hw *hw, int position)
{
u32 data;
if (position == hw_output_switch_get(hw))
return 0;
/* Mute line and headphones (intended for anti-pop). */
data = hw_read_20kx(hw, GPIO_DATA);
data |= (0x03 << 11);
hw_write_20kx(hw, GPIO_DATA, data);
data = hw_read_20kx(hw, GPIO_EXT_DATA) & ~0x30;
switch (position) {
case 0:
break;
case 1:
data |= 0x10;
break;
default:
data |= 0x20;
}
hw_write_20kx(hw, GPIO_EXT_DATA, data);
/* Unmute line and headphones. */
data = hw_read_20kx(hw, GPIO_DATA);
data &= ~(0x03 << 11);
hw_write_20kx(hw, GPIO_DATA, data);
return 1;
}
static int hw_mic_source_switch_get(struct hw *hw)
{
struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
return hw20k2->mic_source;
}
static int hw_mic_source_switch_put(struct hw *hw, int position)
{
struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
if (position == hw20k2->mic_source)
return 0;
switch (position) {
case 0:
hw_wm8775_input_select(hw, 0, 0); /* Mic, 0dB */
break;
case 1:
hw_wm8775_input_select(hw, 1, 0); /* FP Mic, 0dB */
break;
case 2:
hw_wm8775_input_select(hw, 3, 0); /* Aux Ext, 0dB */
break;
default:
return 0;
}
hw20k2->mic_source = position;
return 1;
}
static irqreturn_t ct_20k2_interrupt(int irq, void *dev_id)
@ -1925,7 +2061,7 @@ static int hw_card_start(struct hw *hw)
if (hw->irq < 0) {
err = request_irq(pci->irq, ct_20k2_interrupt, IRQF_SHARED,
"ctxfi", hw);
KBUILD_MODNAME, hw);
if (err < 0) {
printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
goto error2;
@ -2023,13 +2159,16 @@ static int hw_card_init(struct hw *hw, struct card_conf *info)
/* Reset all SRC pending interrupts */
hw_write_20kx(hw, SRC_IP, 0);
/* TODO: detect the card ID and configure GPIO accordingly. */
/* Configures GPIO (0xD802 0x98028) */
/*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/
/* Configures GPIO (SB0880) */
/*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/
hw_write_20kx(hw, GPIO_CTRL, 0xD802);
if (hw->model != CTSB1270) {
/* TODO: detect the card ID and configure GPIO accordingly. */
/* Configures GPIO (0xD802 0x98028) */
/*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/
/* Configures GPIO (SB0880) */
/*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/
hw_write_20kx(hw, GPIO_CTRL, 0xD802);
} else {
hw_write_20kx(hw, GPIO_CTRL, 0x9E5F);
}
/* Enable audio ring */
hw_write_20kx(hw, MIXER_AR_ENABLE, 0x01);
@ -2106,7 +2245,11 @@ static struct hw ct20k2_preset __devinitdata = {
.pll_init = hw_pll_init,
.is_adc_source_selected = hw_is_adc_input_selected,
.select_adc_source = hw_adc_input_select,
.have_digit_io_switch = hw_have_digit_io_switch,
.capabilities = hw_capabilities,
.output_switch_get = hw_output_switch_get,
.output_switch_put = hw_output_switch_put,
.mic_source_switch_get = hw_mic_source_switch_get,
.mic_source_switch_put = hw_mic_source_switch_put,
#ifdef CONFIG_PM
.suspend = hw_suspend,
.resume = hw_resume,

145
sound/pci/ctxfi/ctmixer.c
View File

@ -86,9 +86,7 @@ enum CTALSA_MIXER_CTL {
MIXER_LINEIN_C_S,
MIXER_MIC_C_S,
MIXER_SPDIFI_C_S,
MIXER_LINEIN_P_S,
MIXER_SPDIFO_P_S,
MIXER_SPDIFI_P_S,
MIXER_WAVEF_P_S,
MIXER_WAVER_P_S,
MIXER_WAVEC_P_S,
@ -137,11 +135,11 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
},
[MIXER_LINEIN_P] = {
.ctl = 1,
.name = "Line-in Playback Volume",
.name = "Line Playback Volume",
},
[MIXER_LINEIN_C] = {
.ctl = 1,
.name = "Line-in Capture Volume",
.name = "Line Capture Volume",
},
[MIXER_MIC_P] = {
.ctl = 1,
@ -153,15 +151,15 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
},
[MIXER_SPDIFI_P] = {
.ctl = 1,
.name = "S/PDIF-in Playback Volume",
.name = "IEC958 Playback Volume",
},
[MIXER_SPDIFI_C] = {
.ctl = 1,
.name = "S/PDIF-in Capture Volume",
.name = "IEC958 Capture Volume",
},
[MIXER_SPDIFO_P] = {
.ctl = 1,
.name = "S/PDIF-out Playback Volume",
.name = "Digital Playback Volume",
},
[MIXER_WAVEF_P] = {
.ctl = 1,
@ -179,14 +177,13 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
.ctl = 1,
.name = "Surround Playback Volume",
},
[MIXER_PCM_C_S] = {
.ctl = 1,
.name = "PCM Capture Switch",
},
[MIXER_LINEIN_C_S] = {
.ctl = 1,
.name = "Line-in Capture Switch",
.name = "Line Capture Switch",
},
[MIXER_MIC_C_S] = {
.ctl = 1,
@ -194,19 +191,11 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
},
[MIXER_SPDIFI_C_S] = {
.ctl = 1,
.name = "S/PDIF-in Capture Switch",
},
[MIXER_LINEIN_P_S] = {
.ctl = 1,
.name = "Line-in Playback Switch",
.name = "IEC958 Capture Switch",
},
[MIXER_SPDIFO_P_S] = {
.ctl = 1,
.name = "S/PDIF-out Playback Switch",
},
[MIXER_SPDIFI_P_S] = {
.ctl = 1,
.name = "S/PDIF-in Playback Switch",
.name = "Digital Playback Switch",
},
[MIXER_WAVEF_P_S] = {
.ctl = 1,
@ -236,6 +225,8 @@ ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type);
static void
ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type);
/* FIXME: this static looks like it would fail if more than one card was */
/* installed. */
static struct snd_kcontrol *kctls[2] = {NULL};
static enum CT_AMIXER_CTL get_amixer_index(enum CTALSA_MIXER_CTL alsa_index)
@ -420,6 +411,77 @@ static struct snd_kcontrol_new vol_ctl = {
.tlv = { .p = ct_vol_db_scale },
};
static int output_switch_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *info)
{
static const char *const names[3] = {
"FP Headphones", "Headphones", "Speakers"
};
return snd_ctl_enum_info(info, 1, 3, names);
}
static int output_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = atc->output_switch_get(atc);
return 0;
}
static int output_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
if (ucontrol->value.enumerated.item[0] > 2)
return -EINVAL;
return atc->output_switch_put(atc, ucontrol->value.enumerated.item[0]);
}
static struct snd_kcontrol_new output_ctl = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Analog Output Playback Enum",
.info = output_switch_info,
.get = output_switch_get,
.put = output_switch_put,
};
static int mic_source_switch_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *info)
{
static const char *const names[3] = {
"Mic", "FP Mic", "Aux"
};
return snd_ctl_enum_info(info, 1, 3, names);
}
static int mic_source_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = atc->mic_source_switch_get(atc);
return 0;
}
static int mic_source_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
if (ucontrol->value.enumerated.item[0] > 2)
return -EINVAL;
return atc->mic_source_switch_put(atc,
ucontrol->value.enumerated.item[0]);
}
static struct snd_kcontrol_new mic_source_ctl = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Mic Source Capture Enum",
.info = mic_source_switch_info,
.get = mic_source_switch_get,
.put = mic_source_switch_put,
};
static void
do_line_mic_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type)
{
@ -465,6 +527,7 @@ do_digit_io_switch(struct ct_atc *atc, int state)
static void do_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type, int state)
{
struct ct_mixer *mixer = atc->mixer;
struct capabilities cap = atc->capabilities(atc);
/* Do changes in mixer. */
if ((SWH_CAPTURE_START <= type) && (SWH_CAPTURE_END >= type)) {
@ -477,8 +540,17 @@ static void do_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type, int state)
}
}
/* Do changes out of mixer. */
if (state && (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type))
do_line_mic_switch(atc, type);
if (!cap.dedicated_mic &&
(MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type)) {
if (state)
do_line_mic_switch(atc, type);
atc->line_in_unmute(atc, state);
} else if (cap.dedicated_mic && (MIXER_LINEIN_C_S == type))
atc->line_in_unmute(atc, state);
else if (cap.dedicated_mic && (MIXER_MIC_C_S == type))
atc->mic_unmute(atc, state);
else if (MIXER_SPDIFI_C_S == type)
atc->spdif_in_unmute(atc, state);
else if (MIXER_WAVEF_P_S == type)
atc->line_front_unmute(atc, state);
else if (MIXER_WAVES_P_S == type)
@ -487,12 +559,8 @@ static void do_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type, int state)
atc->line_clfe_unmute(atc, state);
else if (MIXER_WAVER_P_S == type)
atc->line_rear_unmute(atc, state);
else if (MIXER_LINEIN_P_S == type)
atc->line_in_unmute(atc, state);
else if (MIXER_SPDIFO_P_S == type)
atc->spdif_out_unmute(atc, state);
else if (MIXER_SPDIFI_P_S == type)
atc->spdif_in_unmute(atc, state);
else if (MIXER_DIGITAL_IO_S == type)
do_digit_io_switch(atc, state);
@ -671,6 +739,7 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
{
enum CTALSA_MIXER_CTL type;
struct ct_atc *atc = mixer->atc;
struct capabilities cap = atc->capabilities(atc);
int err;
/* Create snd kcontrol instances on demand */
@ -684,8 +753,8 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
}
}
ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl =
atc->have_digit_io_switch(atc);
ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl = cap.digit_io_switch;
for (type = SWH_MIXER_START; type <= SWH_MIXER_END; type++) {
if (ct_kcontrol_init_table[type].ctl) {
swh_ctl.name = ct_kcontrol_init_table[type].name;
@ -708,6 +777,17 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
if (err)
return err;
if (cap.output_switch) {
err = ct_mixer_kcontrol_new(mixer, &output_ctl);
if (err)
return err;
}
if (cap.mic_source_switch) {
err = ct_mixer_kcontrol_new(mixer, &mic_source_ctl);
if (err)
return err;
}
atc->line_front_unmute(atc, 1);
set_switch_state(mixer, MIXER_WAVEF_P_S, 1);
atc->line_surround_unmute(atc, 0);
@ -719,13 +799,12 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
atc->spdif_out_unmute(atc, 0);
set_switch_state(mixer, MIXER_SPDIFO_P_S, 0);
atc->line_in_unmute(atc, 0);
set_switch_state(mixer, MIXER_LINEIN_P_S, 0);
if (cap.dedicated_mic)
atc->mic_unmute(atc, 0);
atc->spdif_in_unmute(atc, 0);
set_switch_state(mixer, MIXER_SPDIFI_P_S, 0);
set_switch_state(mixer, MIXER_PCM_C_S, 1);
set_switch_state(mixer, MIXER_LINEIN_C_S, 1);
set_switch_state(mixer, MIXER_SPDIFI_C_S, 1);
set_switch_state(mixer, MIXER_PCM_C_S, 0);
set_switch_state(mixer, MIXER_LINEIN_C_S, 0);
set_switch_state(mixer, MIXER_SPDIFI_C_S, 0);
return 0;
}

6
sound/pci/ctxfi/xfi.c
View File

@ -80,11 +80,11 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
"are 48000 and 44100, Value 48000 is assumed.\n");
reference_rate = 48000;
}
if ((multiple != 1) && (multiple != 2)) {
if ((multiple != 1) && (multiple != 2) && (multiple != 4)) {
printk(KERN_ERR "ctxfi: Invalid multiple value %u!!!\n",
multiple);
printk(KERN_ERR "ctxfi: The valid values for multiple are "
"1 and 2, Value 2 is assumed.\n");
"1, 2 and 4, Value 2 is assumed.\n");
multiple = 2;
}
err = ct_atc_create(card, pci, reference_rate, multiple,
@ -143,7 +143,7 @@ static int ct_card_resume(struct pci_dev *pci)
#endif
static struct pci_driver ct_driver = {
.name = "SB-XFi",
.name = KBUILD_MODNAME,
.id_table = ct_pci_dev_ids,
.probe = ct_card_probe,
.remove = __devexit_p(ct_card_remove),

6
sound/pci/echoaudio/echoaudio.c
View File

@ -1995,7 +1995,7 @@ static __devinit int snd_echo_create(struct snd_card *card,
ioremap_nocache(chip->dsp_registers_phys, sz);
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
ECHOCARD_NAME, chip)) {
KBUILD_MODNAME, chip)) {
snd_echo_free(chip);
snd_printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
@ -2286,7 +2286,7 @@ static int snd_echo_resume(struct pci_dev *pci)
kfree(commpage_bak);
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
ECHOCARD_NAME, chip)) {
KBUILD_MODNAME, chip)) {
snd_echo_free(chip);
snd_printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
@ -2327,7 +2327,7 @@ static void __devexit snd_echo_remove(struct pci_dev *pci)
/* pci_driver definition */
static struct pci_driver driver = {
.name = "Echoaudio " ECHOCARD_NAME,
.name = KBUILD_MODNAME,
.id_table = snd_echo_ids,
.probe = snd_echo_probe,
.remove = __devexit_p(snd_echo_remove),

2
sound/pci/emu10k1/emu10k1.c
View File

@ -264,7 +264,7 @@ static int snd_emu10k1_resume(struct pci_dev *pci)
#endif
static struct pci_driver driver = {
.name = "EMU10K1_Audigy",
.name = KBUILD_MODNAME,
.id_table = snd_emu10k1_ids,
.probe = snd_card_emu10k1_probe,
.remove = __devexit_p(snd_card_emu10k1_remove),

2
sound/pci/emu10k1/emu10k1_main.c
View File

@ -1912,7 +1912,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
/* irq handler must be registered after I/O ports are activated */
if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
"EMU10K1", emu)) {
KBUILD_MODNAME, emu)) {
err = -EBUSY;
goto error;
}

4
sound/pci/emu10k1/emu10k1x.c
View File

@ -925,7 +925,7 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card,
}
if (request_irq(pci->irq, snd_emu10k1x_interrupt,
IRQF_SHARED, "EMU10K1X", chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "emu10k1x: cannot grab irq %d\n", pci->irq);
snd_emu10k1x_free(chip);
return -EBUSY;
@ -1613,7 +1613,7 @@ MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids);
// pci_driver definition
static struct pci_driver driver = {
.name = "EMU10K1X",
.name = KBUILD_MODNAME,
.id_table = snd_emu10k1x_ids,
.probe = snd_emu10k1x_probe,
.remove = __devexit_p(snd_emu10k1x_remove),

4
sound/pci/ens1370.c
View File

@ -2120,7 +2120,7 @@ static int __devinit snd_ensoniq_create(struct snd_card *card,
}
ensoniq->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_audiopci_interrupt, IRQF_SHARED,
"Ensoniq AudioPCI", ensoniq)) {
KBUILD_MODNAME, ensoniq)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_ensoniq_free(ensoniq);
return -EBUSY;
@ -2489,7 +2489,7 @@ static void __devexit snd_audiopci_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = DRIVER_NAME,
.name = KBUILD_MODNAME,
.id_table = snd_audiopci_ids,
.probe = snd_audiopci_probe,
.remove = __devexit_p(snd_audiopci_remove),

6
sound/pci/es1938.c
View File

@ -1514,7 +1514,7 @@ static int es1938_resume(struct pci_dev *pci)
}
if (request_irq(pci->irq, snd_es1938_interrupt,
IRQF_SHARED, "ES1938", chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "es1938: unable to grab IRQ %d, "
"disabling device\n", pci->irq);
snd_card_disconnect(card);
@ -1636,7 +1636,7 @@ static int __devinit snd_es1938_create(struct snd_card *card,
chip->mpu_port = pci_resource_start(pci, 3);
chip->game_port = pci_resource_start(pci, 4);
if (request_irq(pci->irq, snd_es1938_interrupt, IRQF_SHARED,
"ES1938", chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_es1938_free(chip);
return -EBUSY;
@ -1882,7 +1882,7 @@ static void __devexit snd_es1938_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "ESS ES1938 (Solo-1)",
.name = KBUILD_MODNAME,
.id_table = snd_es1938_ids,
.probe = snd_es1938_probe,
.remove = __devexit_p(snd_es1938_remove),

68
sound/pci/es1968.c
View File

@ -554,9 +554,8 @@ struct es1968 {
#else
struct snd_kcontrol *master_switch; /* for h/w volume control */
struct snd_kcontrol *master_volume;
spinlock_t ac97_lock;
struct tasklet_struct hwvol_tq;
#endif
struct work_struct hwvol_work;
#ifdef CONFIG_SND_ES1968_RADIO
struct snd_tea575x tea;
@ -646,38 +645,23 @@ static int snd_es1968_ac97_wait_poll(struct es1968 *chip)
static void snd_es1968_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
{
struct es1968 *chip = ac97->private_data;
#ifndef CONFIG_SND_ES1968_INPUT
unsigned long flags;
#endif
snd_es1968_ac97_wait(chip);
/* Write the bus */
#ifndef CONFIG_SND_ES1968_INPUT
spin_lock_irqsave(&chip->ac97_lock, flags);
#endif
outw(val, chip->io_port + ESM_AC97_DATA);
/*msleep(1);*/
outb(reg, chip->io_port + ESM_AC97_INDEX);
/*msleep(1);*/
#ifndef CONFIG_SND_ES1968_INPUT
spin_unlock_irqrestore(&chip->ac97_lock, flags);
#endif
}
static unsigned short snd_es1968_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
{
u16 data = 0;
struct es1968 *chip = ac97->private_data;
#ifndef CONFIG_SND_ES1968_INPUT
unsigned long flags;
#endif
snd_es1968_ac97_wait(chip);
#ifndef CONFIG_SND_ES1968_INPUT
spin_lock_irqsave(&chip->ac97_lock, flags);
#endif
outb(reg | 0x80, chip->io_port + ESM_AC97_INDEX);
/*msleep(1);*/
@ -685,9 +669,6 @@ static unsigned short snd_es1968_ac97_read(struct snd_ac97 *ac97, unsigned short
data = inw(chip->io_port + ESM_AC97_DATA);
/*msleep(1);*/
}
#ifndef CONFIG_SND_ES1968_INPUT
spin_unlock_irqrestore(&chip->ac97_lock, flags);
#endif
return data;
}
@ -1904,13 +1885,10 @@ static void snd_es1968_update_pcm(struct es1968 *chip, struct esschan *es)
(without wrap around) in response to volume button presses and then
generating an interrupt. The pair of counters is stored in bits 1-3 and 5-7
of a byte wide register. The meaning of bits 0 and 4 is unknown. */
static void es1968_update_hw_volume(unsigned long private_data)
static void es1968_update_hw_volume(struct work_struct *work)
{
struct es1968 *chip = (struct es1968 *) private_data;
struct es1968 *chip = container_of(work, struct es1968, hwvol_work);
int x, val;
#ifndef CONFIG_SND_ES1968_INPUT
unsigned long flags;
#endif
/* Figure out which volume control button was pushed,
based on differences from the default register
@ -1929,18 +1907,11 @@ static void es1968_update_hw_volume(unsigned long private_data)
if (! chip->master_switch || ! chip->master_volume)
return;
/* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
spin_lock_irqsave(&chip->ac97_lock, flags);
val = chip->ac97->regs[AC97_MASTER];
val = snd_ac97_read(chip->ac97, AC97_MASTER);
switch (x) {
case 0x88:
/* mute */
val ^= 0x8000;
chip->ac97->regs[AC97_MASTER] = val;
outw(val, chip->io_port + ESM_AC97_DATA);
outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
&chip->master_switch->id);
break;
case 0xaa:
/* volume up */
@ -1948,11 +1919,6 @@ static void es1968_update_hw_volume(unsigned long private_data)
val--;
if ((val & 0x7f00) > 0)
val -= 0x0100;
chip->ac97->regs[AC97_MASTER] = val;
outw(val, chip->io_port + ESM_AC97_DATA);
outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
&chip->master_volume->id);
break;
case 0x66:
/* volume down */
@ -1960,14 +1926,11 @@ static void es1968_update_hw_volume(unsigned long private_data)
val++;
if ((val & 0x7f00) < 0x1f00)
val += 0x0100;
chip->ac97->regs[AC97_MASTER] = val;
outw(val, chip->io_port + ESM_AC97_DATA);
outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
&chip->master_volume->id);
break;
}
spin_unlock_irqrestore(&chip->ac97_lock, flags);
if (snd_ac97_update(chip->ac97, AC97_MASTER, val))
snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
&chip->master_volume->id);
#else
if (!chip->input_dev)
return;
@ -2013,11 +1976,7 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
outw(inw(chip->io_port + 4) & 1, chip->io_port + 4);
if (event & ESM_HWVOL_IRQ)
#ifdef CONFIG_SND_ES1968_INPUT
es1968_update_hw_volume((unsigned long)chip);
#else
tasklet_schedule(&chip->hwvol_tq); /* we'll do this later */
#endif
schedule_work(&chip->hwvol_work);
/* else ack 'em all, i imagine */
outb(0xFF, chip->io_port + 0x1A);
@ -2426,6 +2385,7 @@ static int es1968_suspend(struct pci_dev *pci, pm_message_t state)
return 0;
chip->in_suspend = 1;
cancel_work_sync(&chip->hwvol_work);
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
snd_pcm_suspend_all(chip->pcm);
snd_ac97_suspend(chip->ac97);
@ -2638,6 +2598,7 @@ static struct snd_tea575x_ops snd_es1968_tea_ops = {
static int snd_es1968_free(struct es1968 *chip)
{
cancel_work_sync(&chip->hwvol_work);
#ifdef CONFIG_SND_ES1968_INPUT
if (chip->input_dev)
input_unregister_device(chip->input_dev);
@ -2728,10 +2689,7 @@ static int __devinit snd_es1968_create(struct snd_card *card,
INIT_LIST_HEAD(&chip->buf_list);
INIT_LIST_HEAD(&chip->substream_list);
mutex_init(&chip->memory_mutex);
#ifndef CONFIG_SND_ES1968_INPUT
spin_lock_init(&chip->ac97_lock);
tasklet_init(&chip->hwvol_tq, es1968_update_hw_volume, (unsigned long)chip);
#endif
INIT_WORK(&chip->hwvol_work, es1968_update_hw_volume);
chip->card = card;
chip->pci = pci;
chip->irq = -1;
@ -2746,7 +2704,7 @@ static int __devinit snd_es1968_create(struct snd_card *card,
}
chip->io_port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_es1968_interrupt, IRQF_SHARED,
"ESS Maestro", chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_es1968_free(chip);
return -EBUSY;
@ -2925,7 +2883,7 @@ static void __devexit snd_es1968_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "ES1968 (ESS Maestro)",
.name = KBUILD_MODNAME,
.id_table = snd_es1968_ids,
.probe = snd_es1968_probe,
.remove = __devexit_p(snd_es1968_remove),

4
sound/pci/fm801.c
View File

@ -1199,7 +1199,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
chip->port = pci_resource_start(pci, 0);
if ((tea575x_tuner & TUNER_ONLY) == 0) {
if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED,
"FM801", chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
snd_fm801_free(chip);
return -EBUSY;
@ -1394,7 +1394,7 @@ static int snd_fm801_resume(struct pci_dev *pci)
#endif
static struct pci_driver driver = {
.name = "FM801",
.name = KBUILD_MODNAME,
.id_table = snd_fm801_ids,
.probe = snd_card_fm801_probe,
.remove = __devexit_p(snd_card_fm801_remove),

39
sound/pci/hda/Kconfig
View File

@ -14,6 +14,19 @@ menuconfig SND_HDA_INTEL
if SND_HDA_INTEL
config SND_HDA_PREALLOC_SIZE
int "Pre-allocated buffer size for HD-audio driver"
range 0 32768
default 64
help
Specifies the default pre-allocated buffer-size in kB for the
HD-audio driver. A larger buffer (e.g. 2048) is preferred
for systems using PulseAudio. The default 64 is chosen just
for compatibility reasons.
Note that the pre-allocation size can be changed dynamically
via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too.
config SND_HDA_HWDEP
bool "Build hwdep interface for HD-audio driver"
select SND_HWDEP
@ -83,6 +96,19 @@ config SND_HDA_CODEC_REALTEK
snd-hda-codec-realtek.
This module is automatically loaded at probing.
config SND_HDA_ENABLE_REALTEK_QUIRKS
bool "Build static quirks for Realtek codecs"
depends on SND_HDA_CODEC_REALTEK
default y
help
Say Y here to build the static quirks codes for Realtek codecs.
If you need the "model" preset that the default BIOS auto-parser
can't handle, turn this option on.
If your device works with model=auto option, basically you don't
need the quirk code. By turning this off, you can reduce the
module size quite a lot.
config SND_HDA_CODEC_ANALOG
bool "Build Analog Device HD-audio codec support"
default y
@ -171,6 +197,19 @@ config SND_HDA_CODEC_CA0110
snd-hda-codec-ca0110.
This module is automatically loaded at probing.
config SND_HDA_CODEC_CA0132
bool "Build Creative CA0132 codec support"
depends on SND_HDA_INTEL
default y
help
Say Y here to include Creative CA0132 codec support in
snd-hda-intel driver.
When the HD-audio driver is built as a module, the codec
support code is also built as another module,
snd-hda-codec-ca0132.
This module is automatically loaded at probing.
config SND_HDA_CODEC_CMEDIA
bool "Build C-Media HD-audio codec support"
default y

4
sound/pci/hda/Makefile
View File

@ -13,6 +13,7 @@ snd-hda-codec-idt-objs := patch_sigmatel.o
snd-hda-codec-si3054-objs := patch_si3054.o
snd-hda-codec-cirrus-objs := patch_cirrus.o
snd-hda-codec-ca0110-objs := patch_ca0110.o
snd-hda-codec-ca0132-objs := patch_ca0132.o
snd-hda-codec-conexant-objs := patch_conexant.o
snd-hda-codec-via-objs := patch_via.o
snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o
@ -42,6 +43,9 @@ endif
ifdef CONFIG_SND_HDA_CODEC_CA0110
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
endif
ifdef CONFIG_SND_HDA_CODEC_CA0132
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o
endif
ifdef CONFIG_SND_HDA_CODEC_CONEXANT
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
endif

1272
sound/pci/hda/alc260_quirks.c Normal file
View File

File diff suppressed because it is too large Load Diff

1353
sound/pci/hda/alc262_quirks.c Normal file
View File

File diff suppressed because it is too large Load Diff

636
sound/pci/hda/alc268_quirks.c Normal file
View File

@ -0,0 +1,636 @@
/*
* ALC267/ALC268 quirk models
* included by patch_realtek.c
*/
/* ALC268 models */
enum {
ALC268_AUTO,
ALC267_QUANTA_IL1,
ALC268_3ST,
ALC268_TOSHIBA,
ALC268_ACER,
ALC268_ACER_DMIC,
ALC268_ACER_ASPIRE_ONE,
ALC268_DELL,
ALC268_ZEPTO,
#ifdef CONFIG_SND_DEBUG
ALC268_TEST,
#endif
ALC268_MODEL_LAST /* last tag */
};
/*
* ALC268 channel source setting (2 channel)
*/
#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID
#define alc268_modes alc260_modes
static const hda_nid_t alc268_dac_nids[2] = {
/* front, hp */
0x02, 0x03
};
static const hda_nid_t alc268_adc_nids[2] = {
/* ADC0-1 */
0x08, 0x07
};
static const hda_nid_t alc268_adc_nids_alt[1] = {
/* ADC0 */
0x08
};
static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
static const struct snd_kcontrol_new alc268_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc268_toshiba_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc268_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
/* Toshiba specific */
static const struct hda_verb alc268_toshiba_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{ } /* end */
};
/* Acer specific */
/* bind volumes of both NID 0x02 and 0x03 */
static const struct hda_bind_ctls alc268_acer_bind_master_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
0
},
};
static void alc268_acer_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
#define alc268_acer_master_sw_get alc262_hp_master_sw_get
#define alc268_acer_master_sw_put alc262_hp_master_sw_put
static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_NID_FLAG | 0x15,
.info = snd_ctl_boolean_mono_info,
.get = alc268_acer_master_sw_get,
.put = alc268_acer_master_sw_put,
},
HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc268_acer_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
.info = snd_ctl_boolean_mono_info,
.get = alc268_acer_master_sw_get,
.put = alc268_acer_master_sw_put,
},
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_NID_FLAG | 0x14,
.info = snd_ctl_boolean_mono_info,
.get = alc268_acer_master_sw_get,
.put = alc268_acer_master_sw_put,
},
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc268_acer_aspire_one_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
{ }
};
static const struct hda_verb alc268_acer_verbs[] = {
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{ }
};
/* unsolicited event for HP jack sensing */
#define alc268_toshiba_setup alc262_hippo_setup
static void alc268_acer_lc_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
}
static const struct snd_kcontrol_new alc268_dell_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc268_dell_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{ }
};
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc268_dell_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_PIN;
}
static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct hda_verb alc267_quanta_il1_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{ }
};
static void alc267_quanta_il1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_PIN;
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc268_base_init_verbs[] = {
/* Unmute DAC0-1 and set vol = 0 */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* set PCBEEP vol = 0, mute connections */
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* Unmute Selector 23h,24h and set the default input to mic-in */
{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{ }
};
/* only for model=test */
#ifdef CONFIG_SND_DEBUG
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc268_volume_init_verbs[] = {
/* set output DAC */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
#endif /* CONFIG_SND_DEBUG */
static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
_DEFINE_CAPSRC(1),
{ } /* end */
};
static const struct snd_kcontrol_new alc268_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
_DEFINE_CAPSRC(2),
{ } /* end */
};
static const struct hda_input_mux alc268_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x3 },
},
};
static const struct hda_input_mux alc268_acer_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
{ "Line", 0x2 },
},
};
static const struct hda_input_mux alc268_acer_dmic_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x6 },
{ "Line", 0x2 },
},
};
#ifdef CONFIG_SND_DEBUG
static const struct snd_kcontrol_new alc268_test_mixer[] = {
/* Volume widgets */
HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
/* The below appears problematic on some hardwares */
/*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
/* Modes for retasking pin widgets */
ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
/* Controls for GPIO pins, assuming they are configured as outputs */
ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
/* Switches to allow the digital SPDIF output pin to be enabled.
* The ALC268 does not have an SPDIF input.
*/
ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
/* A switch allowing EAPD to be enabled. Some laptops seem to use
* this output to turn on an external amplifier.
*/
ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
{ } /* end */
};
#endif
/*
* configuration and preset
*/
static const char * const alc268_models[ALC268_MODEL_LAST] = {
[ALC267_QUANTA_IL1] = "quanta-il1",
[ALC268_3ST] = "3stack",
[ALC268_TOSHIBA] = "toshiba",
[ALC268_ACER] = "acer",
[ALC268_ACER_DMIC] = "acer-dmic",
[ALC268_ACER_ASPIRE_ONE] = "acer-aspire",
[ALC268_DELL] = "dell",
[ALC268_ZEPTO] = "zepto",
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = "test",
#endif
[ALC268_AUTO] = "auto",
};
static const struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
/* almost compatible with toshiba but with optional digital outs;
* auto-probing seems working fine
*/
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
ALC268_AUTO),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
{}
};
/* Toshiba laptops have no unique PCI SSID but only codec SSID */
static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO),
SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO),
SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
ALC268_TOSHIBA),
{}
};
static const struct alc_config_preset alc268_presets[] = {
[ALC267_QUANTA_IL1] = {
.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc267_quanta_il1_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc267_quanta_il1_setup,
.init_hook = alc_inithook,
},
[ALC268_3ST] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_toshiba_setup,
.init_hook = alc_inithook,
},
[ALC268_ACER] = {
.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_acer_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_acer_setup,
.init_hook = alc_inithook,
},
[ALC268_ACER_DMIC] = {
.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_acer_dmic_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_acer_setup,
.init_hook = alc_inithook,
},
[ALC268_ACER_ASPIRE_ONE] = {
.mixers = { alc268_acer_aspire_one_mixer,
alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_aspire_one_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_acer_lc_setup,
.init_hook = alc_inithook,
},
[ALC268_DELL] = {
.mixers = { alc268_dell_mixer, alc268_beep_mixer,
alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_dell_setup,
.init_hook = alc_inithook,
},
[ALC268_ZEPTO] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc268_toshiba_setup,
.init_hook = alc_inithook,
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
.mixers = { alc268_test_mixer, alc268_capture_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_volume_init_verbs,
alc268_beep_init_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
.adc_nids = alc268_adc_nids_alt,
.capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x03,
.dig_out_nid = ALC268_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
},
#endif
};

681
sound/pci/hda/alc269_quirks.c Normal file
View File

@ -0,0 +1,681 @@
/*
* ALC269/ALC270/ALC275/ALC276 quirk models
* included by patch_realtek.c
*/
/* ALC269 models */
enum {
ALC269_AUTO,
ALC269_BASIC,
ALC269_QUANTA_FL1,
ALC269_AMIC,
ALC269_DMIC,
ALC269VB_AMIC,
ALC269VB_DMIC,
ALC269_FUJITSU,
ALC269_LIFEBOOK,
ALC271_ACER,
ALC269_MODEL_LAST /* last tag */
};
/*
* ALC269 channel source setting (2 channel)
*/
#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID
#define alc269_dac_nids alc260_dac_nids
static const hda_nid_t alc269_adc_nids[1] = {
/* ADC1 */
0x08,
};
static const hda_nid_t alc269_capsrc_nids[1] = {
0x23,
};
static const hda_nid_t alc269vb_adc_nids[1] = {
/* ADC1 */
0x09,
};
static const hda_nid_t alc269vb_capsrc_nids[1] = {
0x22,
};
#define alc269_modes alc260_modes
#define alc269_capture_source alc880_lg_lw_capture_source
static const struct snd_kcontrol_new alc269_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_AMP_FLAG,
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
.put = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc269_lifebook_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.subdevice = HDA_SUBDEV_AMP_FLAG,
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
.put = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT),
{ }
};
static const struct snd_kcontrol_new alc269_laptop_mixer[] = {
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = {
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269_asus_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT),
{ } /* end */
};
/* capture mixer elements */
static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
{ } /* end */
};
/* FSC amilo */
#define alc269_fujitsu_mixer alc269_laptop_mixer
static const struct hda_verb alc269_quanta_fl1_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
static const struct hda_verb alc269_lifebook_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
{
alc_hp_automute(codec);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x680);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x480);
}
#define alc269_lifebook_speaker_automute \
alc269_quanta_fl1_speaker_automute
static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
{
unsigned int present_laptop;
unsigned int present_dock;
present_laptop = snd_hda_jack_detect(codec, 0x18);
present_dock = snd_hda_jack_detect(codec, 0x1b);
/* Laptop mic port overrides dock mic port, design decision */
if (present_dock)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x3);
if (present_laptop)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x0);
if (!present_dock && !present_laptop)
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, 0x1);
}
static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC_HP_EVENT:
alc269_quanta_fl1_speaker_automute(codec);
break;
case ALC_MIC_EVENT:
alc_mic_automute(codec);
break;
}
}
static void alc269_lifebook_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc269_lifebook_speaker_automute(codec);
if ((res >> 26) == ALC_MIC_EVENT)
alc269_lifebook_mic_autoswitch(codec);
}
static void alc269_quanta_fl1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
}
static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
{
alc269_quanta_fl1_speaker_automute(codec);
alc_mic_automute(codec);
}
static void alc269_lifebook_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.hp_pins[1] = 0x1a;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
}
static void alc269_lifebook_init_hook(struct hda_codec *codec)
{
alc269_lifebook_speaker_automute(codec);
alc269_lifebook_mic_autoswitch(codec);
}
static const struct hda_verb alc269_laptop_dmic_init_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc269_laptop_amic_init_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = {
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x22, AC_VERB_SET_CONNECT_SEL, 0x06},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = {
{0x21, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x22, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{}
};
static const struct hda_verb alc271_acer_dmic_verbs[] = {
{0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
{0x20, AC_VERB_SET_PROC_COEF, 0x4000},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{0x22, AC_VERB_SET_CONNECT_SEL, 6},
{ }
};
static void alc269_laptop_amic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
}
static void alc269_laptop_dmic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
}
static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x21;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
}
static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x21;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute_mixer_nid[0] = 0x0c;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_MIXER;
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc269_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/*
* Set up output mixers (0x02 - 0x03)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* FIXME: use Mux-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
/* set EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static const struct hda_verb alc269vb_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/*
* Set up output mixers (0x02 - 0x03)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* FIXME: use Mux-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x22, AC_VERB_SET_CONNECT_SEL, 0x00},
/* set EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
/*
* configuration and preset
*/
static const char * const alc269_models[ALC269_MODEL_LAST] = {
[ALC269_BASIC] = "basic",
[ALC269_QUANTA_FL1] = "quanta",
[ALC269_AMIC] = "laptop-amic",
[ALC269_DMIC] = "laptop-dmic",
[ALC269_FUJITSU] = "fujitsu",
[ALC269_LIFEBOOK] = "lifebook",
[ALC269_AUTO] = "auto",
};
static const struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER),
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269VB_AMIC),
SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_AMIC),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_AMIC),
SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_DMIC),
SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_DMIC),
{}
};
static const struct alc_config_preset alc269_presets[] = {
[ALC269_BASIC] = {
.mixers = { alc269_base_mixer },
.init_verbs = { alc269_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
},
[ALC269_QUANTA_FL1] = {
.mixers = { alc269_quanta_fl1_mixer },
.init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_quanta_fl1_unsol_event,
.setup = alc269_quanta_fl1_setup,
.init_hook = alc269_quanta_fl1_init_hook,
},
[ALC269_AMIC] = {
.mixers = { alc269_laptop_mixer },
.cap_mixer = alc269_laptop_analog_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_laptop_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269_laptop_amic_setup,
.init_hook = alc_inithook,
},
[ALC269_DMIC] = {
.mixers = { alc269_laptop_mixer },
.cap_mixer = alc269_laptop_digital_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_laptop_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269_laptop_dmic_setup,
.init_hook = alc_inithook,
},
[ALC269VB_AMIC] = {
.mixers = { alc269vb_laptop_mixer },
.cap_mixer = alc269vb_laptop_analog_capture_mixer,
.init_verbs = { alc269vb_init_verbs,
alc269vb_laptop_amic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269vb_laptop_amic_setup,
.init_hook = alc_inithook,
},
[ALC269VB_DMIC] = {
.mixers = { alc269vb_laptop_mixer },
.cap_mixer = alc269vb_laptop_digital_capture_mixer,
.init_verbs = { alc269vb_init_verbs,
alc269vb_laptop_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269vb_laptop_dmic_setup,
.init_hook = alc_inithook,
},
[ALC269_FUJITSU] = {
.mixers = { alc269_fujitsu_mixer },
.cap_mixer = alc269_laptop_digital_capture_mixer,
.init_verbs = { alc269_init_verbs,
alc269_laptop_dmic_init_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.unsol_event = alc_sku_unsol_event,
.setup = alc269_laptop_dmic_setup,
.init_hook = alc_inithook,
},
[ALC269_LIFEBOOK] = {
.mixers = { alc269_lifebook_mixer },
.init_verbs = { alc269_init_verbs, alc269_lifebook_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_lifebook_unsol_event,
.setup = alc269_lifebook_setup,
.init_hook = alc269_lifebook_init_hook,
},
[ALC271_ACER] = {
.mixers = { alc269_asus_mixer },
.cap_mixer = alc269vb_laptop_digital_capture_mixer,
.init_verbs = { alc269_init_verbs, alc271_acer_dmic_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.adc_nids = alc262_dmic_adc_nids,
.num_adc_nids = ARRAY_SIZE(alc262_dmic_adc_nids),
.capsrc_nids = alc262_dmic_capsrc_nids,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.dig_out_nid = ALC880_DIGOUT_NID,
.unsol_event = alc_sku_unsol_event,
.setup = alc269vb_laptop_dmic_setup,
.init_hook = alc_inithook,
},
};

1408
sound/pci/hda/alc662_quirks.c Normal file
View File

File diff suppressed because it is too large Load Diff

222
sound/pci/hda/alc680_quirks.c Normal file
View File

@ -0,0 +1,222 @@
/*
* ALC680 quirk models
* included by patch_realtek.c
*/
/* ALC680 models */
enum {
ALC680_AUTO,
ALC680_BASE,
ALC680_MODEL_LAST,
};
#define ALC680_DIGIN_NID ALC880_DIGIN_NID
#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID
#define alc680_modes alc260_modes
static const hda_nid_t alc680_dac_nids[3] = {
/* Lout1, Lout2, hp */
0x02, 0x03, 0x04
};
static const hda_nid_t alc680_adc_nids[3] = {
/* ADC0-2 */
/* DMIC, MIC, Line-in*/
0x07, 0x08, 0x09
};
/*
* Analog capture ADC cgange
*/
static hda_nid_t alc680_get_cur_adc(struct hda_codec *codec)
{
static hda_nid_t pins[] = {0x18, 0x19};
static hda_nid_t adcs[] = {0x08, 0x09};
int i;
for (i = 0; i < ARRAY_SIZE(pins); i++) {
if (!is_jack_detectable(codec, pins[i]))
continue;
if (snd_hda_jack_detect(codec, pins[i]))
return adcs[i];
}
return 0x07;
}
static void alc680_rec_autoswitch(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t nid = alc680_get_cur_adc(codec);
if (spec->cur_adc && nid != spec->cur_adc) {
__snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
spec->cur_adc = nid;
snd_hda_codec_setup_stream(codec, nid,
spec->cur_adc_stream_tag, 0,
spec->cur_adc_format);
}
}
static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
hda_nid_t nid = alc680_get_cur_adc(codec);
spec->cur_adc = nid;
spec->cur_adc_stream_tag = stream_tag;
spec->cur_adc_format = format;
snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
return 0;
}
static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct alc_spec *spec = codec->spec;
snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
spec->cur_adc = 0;
return 0;
}
static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
.substreams = 1, /* can be overridden */
.channels_min = 2,
.channels_max = 2,
/* NID is set in alc_build_pcms */
.ops = {
.prepare = alc680_capture_pcm_prepare,
.cleanup = alc680_capture_pcm_cleanup
},
};
static const struct snd_kcontrol_new alc680_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT),
{ }
};
static const struct hda_bind_ctls alc680_bind_cap_vol = {
.ops = &snd_hda_bind_vol,
.values = {
HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
0
},
};
static const struct hda_bind_ctls alc680_bind_cap_switch = {
.ops = &snd_hda_bind_sw,
.values = {
HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
0
},
};
static const struct snd_kcontrol_new alc680_master_capture_mixer[] = {
HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
{ } /* end */
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc680_init_verbs[] = {
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_MIC_EVENT | AC_USRSP_EN},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc680_base_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x16;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.num_inputs = 2;
spec->autocfg.inputs[0].pin = 0x18;
spec->autocfg.inputs[0].type = AUTO_PIN_MIC;
spec->autocfg.inputs[1].pin = 0x19;
spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc680_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc_hp_automute(codec);
if ((res >> 26) == ALC_MIC_EVENT)
alc680_rec_autoswitch(codec);
}
static void alc680_inithook(struct hda_codec *codec)
{
alc_hp_automute(codec);
alc680_rec_autoswitch(codec);
}
/*
* configuration and preset
*/
static const char * const alc680_models[ALC680_MODEL_LAST] = {
[ALC680_BASE] = "base",
[ALC680_AUTO] = "auto",
};
static const struct snd_pci_quirk alc680_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE),
{}
};
static const struct alc_config_preset alc680_presets[] = {
[ALC680_BASE] = {
.mixers = { alc680_base_mixer },
.cap_mixer = alc680_master_capture_mixer,
.init_verbs = { alc680_init_verbs },
.num_dacs = ARRAY_SIZE(alc680_dac_nids),
.dac_nids = alc680_dac_nids,
.dig_out_nid = ALC680_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc680_modes),
.channel_mode = alc680_modes,
.unsol_event = alc680_unsol_event,
.setup = alc680_base_setup,
.init_hook = alc680_inithook,
},
};

725
sound/pci/hda/alc861_quirks.c Normal file
View File

@ -0,0 +1,725 @@
/*
* ALC660/ALC861 quirk models
* included by patch_realtek.c
*/
/* ALC861 models */
enum {
ALC861_AUTO,
ALC861_3ST,
ALC660_3ST,
ALC861_3ST_DIG,
ALC861_6ST_DIG,
ALC861_UNIWILL_M31,
ALC861_TOSHIBA,
ALC861_ASUS,
ALC861_ASUS_LAPTOP,
ALC861_MODEL_LAST,
};
/*
* ALC861 channel source setting (2/6 channel selection for 3-stack)
*/
/*
* set the path ways for 2 channel output
* need to set the codec line out and mic 1 pin widgets to inputs
*/
static const struct hda_verb alc861_threestack_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1/2) for input, for mic also enable
* the vref
*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* end */
};
/*
* 6ch mode
* need to set the codec line out and mic 1 pin widgets to outputs
*/
static const struct hda_verb alc861_threestack_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static const struct hda_channel_mode alc861_threestack_modes[2] = {
{ 2, alc861_threestack_ch2_init },
{ 6, alc861_threestack_ch6_init },
};
/* Set mic1 as input and unmute the mixer */
static const struct hda_verb alc861_uniwill_m31_ch2_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
/* Set mic1 as output and mute mixer */
static const struct hda_verb alc861_uniwill_m31_ch4_init[] = {
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ } /* end */
};
static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
{ 2, alc861_uniwill_m31_ch2_init },
{ 4, alc861_uniwill_m31_ch4_init },
};
/* Set mic1 and line-in as input and unmute the mixer */
static const struct hda_verb alc861_asus_ch2_init[] = {
/* set pin widget 1Ah (line in) for input */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* set pin widget 18h (mic1/2) for input, for mic also enable
* the vref
*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
#endif
{ } /* end */
};
/* Set mic1 nad line-in as output and mute mixer */
static const struct hda_verb alc861_asus_ch6_init[] = {
/* set pin widget 1Ah (line in) for output (Back Surround)*/
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
/* set pin widget 18h (mic1) for output (CLFE)*/
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
{ 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
#if 0
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
#endif
{ } /* end */
};
static const struct hda_channel_mode alc861_asus_modes[2] = {
{ 2, alc861_asus_ch2_init },
{ 6, alc861_asus_ch6_init },
};
/* patch-ALC861 */
static const struct snd_kcontrol_new alc861_base_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
/*Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861_3ST_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
/* Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_threestack_modes),
},
{ } /* end */
};
static const struct snd_kcontrol_new alc861_toshiba_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
/* Input mixer control */
/* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
},
{ } /* end */
};
static const struct snd_kcontrol_new alc861_asus_mixer[] = {
/* output mixer control */
HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
/* Input mixer control */
HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
.private_value = ARRAY_SIZE(alc861_asus_modes),
},
{ }
};
/* additional mixer */
static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
{ }
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc861_base_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static const struct hda_verb alc861_threestack_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static const struct hda_verb alc861_uniwill_m31_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel) */
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
/* this has to be set to VREF80 */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
static const struct hda_verb alc861_asus_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
/* port-A for surround (rear panel)
* according to codec#0 this is the HP jack
*/
{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
/* route front PCM to HP */
{ 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* port-B for mic-in (rear panel) with vref */
{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-C for line-in (rear panel) */
{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* port-D for Front */
{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
{ 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-E for HP out (front panel) */
/* this has to be set to VREF80 */
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-F for mic-in (front panel) with vref */
{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
/* port-G for CLFE (rear panel) */
{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* port-H for side (rear panel) */
{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
/* CD-in */
{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
/* route front mic to ADC1*/
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute DAC0~3 & spdif out*/
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Unmute Mixer 14 (mic) 1c (Line in)*/
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Unmute Stereo Mixer 15 */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* hp used DAC 3 (Front) */
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{ }
};
/* additional init verbs for ASUS laptops */
static const struct hda_verb alc861_asus_laptop_init_verbs[] = {
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
{ }
};
static const struct hda_verb alc861_toshiba_init_verbs[] = {
{0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
static void alc861_toshiba_automute(struct hda_codec *codec)
{
unsigned int present = snd_hda_jack_detect(codec, 0x0f);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
}
static void alc861_toshiba_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC_HP_EVENT)
alc861_toshiba_automute(codec);
}
#define ALC861_DIGOUT_NID 0x07
static const struct hda_channel_mode alc861_8ch_modes[1] = {
{ 8, NULL }
};
static const hda_nid_t alc861_dac_nids[4] = {
/* front, surround, clfe, side */
0x03, 0x06, 0x05, 0x04
};
static const hda_nid_t alc660_dac_nids[3] = {
/* front, clfe, surround */
0x03, 0x05, 0x06
};
static const hda_nid_t alc861_adc_nids[1] = {
/* ADC0-2 */
0x08,
};
static const struct hda_input_mux alc861_capture_source = {
.num_items = 5,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x3 },
{ "Line", 0x1 },
{ "CD", 0x4 },
{ "Mixer", 0x5 },
},
};
/*
* configuration and preset
*/
static const char * const alc861_models[ALC861_MODEL_LAST] = {
[ALC861_3ST] = "3stack",
[ALC660_3ST] = "3stack-660",
[ALC861_3ST_DIG] = "3stack-dig",
[ALC861_6ST_DIG] = "6stack-dig",
[ALC861_UNIWILL_M31] = "uniwill-m31",
[ALC861_TOSHIBA] = "toshiba",
[ALC861_ASUS] = "asus",
[ALC861_ASUS_LAPTOP] = "asus-laptop",
[ALC861_AUTO] = "auto",
};
static const struct snd_pci_quirk alc861_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
/* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
* Any other models that need this preset?
*/
/* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
/* FIXME: the below seems conflict */
/* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
{}
};
static const struct alc_config_preset alc861_presets[] = {
[ALC861_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_3ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_6ST_DIG] = {
.mixers = { alc861_base_mixer },
.init_verbs = { alc861_base_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
.channel_mode = alc861_8ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC660_3ST] = {
.mixers = { alc861_3ST_mixer },
.init_verbs = { alc861_threestack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660_dac_nids),
.dac_nids = alc660_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
.channel_mode = alc861_threestack_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_UNIWILL_M31] = {
.mixers = { alc861_uniwill_m31_mixer },
.init_verbs = { alc861_uniwill_m31_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
.channel_mode = alc861_uniwill_m31_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_TOSHIBA] = {
.mixers = { alc861_toshiba_mixer },
.init_verbs = { alc861_base_init_verbs,
alc861_toshiba_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
.unsol_event = alc861_toshiba_unsol_event,
.init_hook = alc861_toshiba_automute,
},
[ALC861_ASUS] = {
.mixers = { alc861_asus_mixer },
.init_verbs = { alc861_asus_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
.channel_mode = alc861_asus_modes,
.need_dac_fix = 1,
.hp_nid = 0x06,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
[ALC861_ASUS_LAPTOP] = {
.mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
.init_verbs = { alc861_asus_init_verbs,
alc861_asus_laptop_init_verbs },
.num_dacs = ARRAY_SIZE(alc861_dac_nids),
.dac_nids = alc861_dac_nids,
.dig_out_nid = ALC861_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
.adc_nids = alc861_adc_nids,
.input_mux = &alc861_capture_source,
},
};

605
sound/pci/hda/alc861vd_quirks.c Normal file
View File

@ -0,0 +1,605 @@
/*
* ALC660-VD/ALC861-VD quirk models
* included by patch_realtek.c
*/
/* ALC861-VD models */
enum {
ALC861VD_AUTO,
ALC660VD_3ST,
ALC660VD_3ST_DIG,
ALC660VD_ASUS_V1S,
ALC861VD_3ST,
ALC861VD_3ST_DIG,
ALC861VD_6ST_DIG,
ALC861VD_LENOVO,
ALC861VD_DALLAS,
ALC861VD_HP,
ALC861VD_MODEL_LAST,
};
#define ALC861VD_DIGOUT_NID 0x06
static const hda_nid_t alc861vd_dac_nids[4] = {
/* front, surr, clfe, side surr */
0x02, 0x03, 0x04, 0x05
};
/* dac_nids for ALC660vd are in a different order - according to
* Realtek's driver.
* This should probably result in a different mixer for 6stack models
* of ALC660vd codecs, but for now there is only 3stack mixer
* - and it is the same as in 861vd.
* adc_nids in ALC660vd are (is) the same as in 861vd
*/
static const hda_nid_t alc660vd_dac_nids[3] = {
/* front, rear, clfe, rear_surr */
0x02, 0x04, 0x03
};
static const hda_nid_t alc861vd_adc_nids[1] = {
/* ADC0 */
0x09,
};
static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
static const struct hda_input_mux alc861vd_capture_source = {
.num_items = 4,
.items = {
{ "Mic", 0x0 },
{ "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
},
};
static const struct hda_input_mux alc861vd_dallas_capture_source = {
.num_items = 2,
.items = {
{ "Mic", 0x0 },
{ "Internal Mic", 0x1 },
},
};
static const struct hda_input_mux alc861vd_hp_capture_source = {
.num_items = 2,
.items = {
{ "Front Mic", 0x0 },
{ "ATAPI Mic", 0x1 },
},
};
/*
* 2ch mode
*/
static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
{ 2, NULL }
};
/*
* 6ch mode
*/
static const struct hda_verb alc861vd_6stack_ch6_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
/*
* 8ch mode
*/
static const struct hda_verb alc861vd_6stack_ch8_init[] = {
{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
{ } /* end */
};
static const struct hda_channel_mode alc861vd_6stack_modes[2] = {
{ 6, alc861vd_6stack_ch6_init },
{ 8, alc861vd_6stack_ch8_init },
};
static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
.info = alc_ch_mode_info,
.get = alc_ch_mode_get,
.put = alc_ch_mode_put,
},
{ } /* end */
};
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
static const struct snd_kcontrol_new alc861vd_6st_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0,
HDA_OUTPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861vd_3st_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
/*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ } /* end */
};
/* Pin assignment: Speaker=0x14, HP = 0x15,
* Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
*/
static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
/* Pin assignment: Speaker=0x14, Line-out = 0x15,
* Front Mic=0x18, ATAPI Mic = 0x19,
*/
static const struct snd_kcontrol_new alc861vd_hp_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static const struct hda_verb alc861vd_volume_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of
* the analog-loopback mixer widget
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
/*
* Set up output mixers (0x02 - 0x05)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
/*
* 3-stack pin configuration:
* front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
*/
static const struct hda_verb alc861vd_3stack_init_verbs[] = {
/*
* Set pin mode and muting
*/
/* set front pin widgets 0x14 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/*
* 6-stack pin configuration:
*/
static const struct hda_verb alc861vd_6stack_init_verbs[] = {
/*
* Set pin mode and muting
*/
/* set front pin widgets 0x14 for output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Rear Pin: output 1 (0x0d) */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
/* CLFE Pin: output 2 (0x0e) */
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
/* Side Pin: output 3 (0x0f) */
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Front Mic pin: input vref at 80% */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line In pin: input */
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
/* Line-2 In: Headphone output (output 0 - 0x0c) */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
/* CD pin widget for input */
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
static const struct hda_verb alc861vd_eapd_verbs[] = {
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
{}
};
static void alc861vd_lenovo_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
{
alc_hp_automute(codec);
alc88x_simple_mic_automute(codec);
}
static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
case ALC_MIC_EVENT:
alc88x_simple_mic_automute(codec);
break;
default:
alc_sku_unsol_event(codec, res);
break;
}
}
static const struct hda_verb alc861vd_dallas_verbs[] = {
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
{ } /* end */
};
/* toggle speaker-output according to the hp-jack state */
static void alc861vd_dallas_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->automute = 1;
spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/*
* configuration and preset
*/
static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
[ALC660VD_3ST] = "3stack-660",
[ALC660VD_3ST_DIG] = "3stack-660-digout",
[ALC660VD_ASUS_V1S] = "asus-v1s",
[ALC861VD_3ST] = "3stack",
[ALC861VD_3ST_DIG] = "3stack-digout",
[ALC861VD_6ST_DIG] = "6stack-digout",
[ALC861VD_LENOVO] = "lenovo",
[ALC861VD_DALLAS] = "dallas",
[ALC861VD_HP] = "hp",
[ALC861VD_AUTO] = "auto",
};
static const struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
/*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
{}
};
static const struct alc_config_preset alc861vd_presets[] = {
[ALC660VD_3ST] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC660VD_3ST_DIG] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_3ST] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_3ST_DIG] = {
.mixers = { alc861vd_3st_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_6ST_DIG] = {
.mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_6stack_init_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
.channel_mode = alc861vd_6stack_modes,
.input_mux = &alc861vd_capture_source,
},
[ALC861VD_LENOVO] = {
.mixers = { alc861vd_lenovo_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs,
alc861vd_eapd_verbs,
alc861vd_lenovo_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
.setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
[ALC861VD_DALLAS] = {
.mixers = { alc861vd_dallas_mixer },
.init_verbs = { alc861vd_dallas_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_dallas_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc861vd_dallas_setup,
.init_hook = alc_hp_automute,
},
[ALC861VD_HP] = {
.mixers = { alc861vd_hp_mixer },
.init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
.dac_nids = alc861vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_hp_capture_source,
.unsol_event = alc_sku_unsol_event,
.setup = alc861vd_dallas_setup,
.init_hook = alc_hp_automute,
},
[ALC660VD_ASUS_V1S] = {
.mixers = { alc861vd_lenovo_mixer },
.init_verbs = { alc861vd_volume_init_verbs,
alc861vd_3stack_init_verbs,
alc861vd_eapd_verbs,
alc861vd_lenovo_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
.dac_nids = alc660vd_dac_nids,
.dig_out_nid = ALC861VD_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
.setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
};

1898
sound/pci/hda/alc880_quirks.c Normal file
View File

File diff suppressed because it is too large Load Diff

3755
sound/pci/hda/alc882_quirks.c Normal file
View File

File diff suppressed because it is too large Load Diff

467
sound/pci/hda/alc_quirks.c Normal file
View File

@ -0,0 +1,467 @@
/*
* Common codes for Realtek codec quirks
* included by patch_realtek.c
*/
/*
* configuration template - to be copied to the spec instance
*/
struct alc_config_preset {
const struct snd_kcontrol_new *mixers[5]; /* should be identical size
* with spec
*/
const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
const struct hda_verb *init_verbs[5];
unsigned int num_dacs;
const hda_nid_t *dac_nids;
hda_nid_t dig_out_nid; /* optional */
hda_nid_t hp_nid; /* optional */
const hda_nid_t *slave_dig_outs;
unsigned int num_adc_nids;
const hda_nid_t *adc_nids;
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid;
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
int need_dac_fix;
int const_channel_count;
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*setup)(struct hda_codec *);
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
const struct hda_amp_list *loopbacks;
void (*power_hook)(struct hda_codec *codec);
#endif
};
/*
* channel mode setting
*/
static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
spec->num_channel_mode);
}
static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
spec->ext_channel_count);
}
static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
&spec->ext_channel_count);
if (err >= 0 && !spec->const_channel_count) {
spec->multiout.max_channels = spec->ext_channel_count;
if (spec->need_dac_fix)
spec->multiout.num_dacs = spec->multiout.max_channels / 2;
}
return err;
}
/*
* Control the mode of pin widget settings via the mixer. "pc" is used
* instead of "%" to avoid consequences of accidentally treating the % as
* being part of a format specifier. Maximum allowed length of a value is
* 63 characters plus NULL terminator.
*
* Note: some retasking pin complexes seem to ignore requests for input
* states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
* are requested. Therefore order this list so that this behaviour will not
* cause problems when mixer clients move through the enum sequentially.
* NIDs 0x0f and 0x10 have been observed to have this behaviour as of
* March 2006.
*/
static const char * const alc_pin_mode_names[] = {
"Mic 50pc bias", "Mic 80pc bias",
"Line in", "Line out", "Headphone out",
};
static const unsigned char alc_pin_mode_values[] = {
PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
};
/* The control can present all 5 options, or it can limit the options based
* in the pin being assumed to be exclusively an input or an output pin. In
* addition, "input" pins may or may not process the mic bias option
* depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
* accept requests for bias as of chip versions up to March 2006) and/or
* wiring in the computer.
*/
#define ALC_PIN_DIR_IN 0x00
#define ALC_PIN_DIR_OUT 0x01
#define ALC_PIN_DIR_INOUT 0x02
#define ALC_PIN_DIR_IN_NOMICBIAS 0x03
#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
/* Info about the pin modes supported by the different pin direction modes.
* For each direction the minimum and maximum values are given.
*/
static const signed char alc_pin_mode_dir_info[5][2] = {
{ 0, 2 }, /* ALC_PIN_DIR_IN */
{ 3, 4 }, /* ALC_PIN_DIR_OUT */
{ 0, 4 }, /* ALC_PIN_DIR_INOUT */
{ 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
{ 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
};
#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
#define alc_pin_mode_n_items(_dir) \
(alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
unsigned int item_num = uinfo->value.enumerated.item;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
item_num = alc_pin_mode_min(dir);
strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
return 0;
}
static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
unsigned int i;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL,
0x00);
/* Find enumerated value for current pinctl setting */
i = alc_pin_mode_min(dir);
while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
i++;
*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
return 0;
}
static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL,
0x00);
if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
val = alc_pin_mode_min(dir);
change = pinctl != alc_pin_mode_values[val];
if (change) {
/* Set pin mode to that requested */
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
alc_pin_mode_values[val]);
/* Also enable the retasking pin's input/output as required
* for the requested pin mode. Enum values of 2 or less are
* input modes.
*
* Dynamically switching the input/output buffers probably
* reduces noise slightly (particularly on input) so we'll
* do it. However, having both input and output buffers
* enabled simultaneously doesn't seem to be problematic if
* this turns out to be necessary in the future.
*/
if (val <= 2) {
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, 0);
} else {
snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
}
return change;
}
#define ALC_PIN_MODE(xname, nid, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_pin_mode_info, \
.get = alc_pin_mode_get, \
.put = alc_pin_mode_put, \
.private_value = nid | (dir<<16) }
/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
* together using a mask with more than one bit set. This control is
* currently used only by the ALC260 test model. At this stage they are not
* needed for any "production" models.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_gpio_data_info snd_ctl_boolean_mono_info
static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_GPIO_DATA,
0x00);
/* Set/unset the masked GPIO bit(s) as needed */
change = (val == 0 ? 0 : mask) != (gpio_data & mask);
if (val == 0)
gpio_data &= ~mask;
else
gpio_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_GPIO_DATA, gpio_data);
return change;
}
#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_gpio_data_info, \
.get = alc_gpio_data_get, \
.put = alc_gpio_data_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling of the digital IO pins on the
* ALC260. This is incredibly simplistic; the intention of this control is
* to provide something in the test model allowing digital outputs to be
* identified if present. If models are found which can utilise these
* outputs a more complete mixer control can be devised for those models if
* necessary.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
signed int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
if (val==0)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
ctrl_data);
return change;
}
#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_spdif_ctrl_info, \
.get = alc_spdif_ctrl_get, \
.put = alc_spdif_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
* Again, this is only used in the ALC26x test models to help identify when
* the EAPD line must be asserted for features to work.
*/
#ifdef CONFIG_SND_DEBUG
#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long *valp = ucontrol->value.integer.value;
unsigned int val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE, 0x00);
*valp = (val & mask) != 0;
return 0;
}
static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int change;
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value & 0xffff;
unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
long val = *ucontrol->value.integer.value;
unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_EAPD_BTLENABLE,
0x00);
/* Set/unset the masked control bit(s) as needed */
change = (!val ? 0 : mask) != (ctrl_data & mask);
if (!val)
ctrl_data &= ~mask;
else
ctrl_data |= mask;
snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
ctrl_data);
return change;
}
#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
.subdevice = HDA_SUBDEV_NID_FLAG | nid, \
.info = alc_eapd_ctrl_info, \
.get = alc_eapd_ctrl_get, \
.put = alc_eapd_ctrl_put, \
.private_value = nid | (mask<<16) }
#endif /* CONFIG_SND_DEBUG */
static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
if (!cfg->line_outs) {
while (cfg->line_outs < AUTO_CFG_MAX_OUTS &&
cfg->line_out_pins[cfg->line_outs])
cfg->line_outs++;
}
if (!cfg->speaker_outs) {
while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS &&
cfg->speaker_pins[cfg->speaker_outs])
cfg->speaker_outs++;
}
if (!cfg->hp_outs) {
while (cfg->hp_outs < AUTO_CFG_MAX_OUTS &&
cfg->hp_pins[cfg->hp_outs])
cfg->hp_outs++;
}
}
/*
* set up from the preset table
*/
static void setup_preset(struct hda_codec *codec,
const struct alc_config_preset *preset)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
add_mixer(spec, preset->mixers[i]);
spec->cap_mixer = preset->cap_mixer;
for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
i++)
add_verb(spec, preset->init_verbs[i]);
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
spec->need_dac_fix = preset->need_dac_fix;
spec->const_channel_count = preset->const_channel_count;
if (preset->const_channel_count)
spec->multiout.max_channels = preset->const_channel_count;
else
spec->multiout.max_channels = spec->channel_mode[0].channels;
spec->ext_channel_count = spec->channel_mode[0].channels;
spec->multiout.num_dacs = preset->num_dacs;
spec->multiout.dac_nids = preset->dac_nids;
spec->multiout.dig_out_nid = preset->dig_out_nid;
spec->multiout.slave_dig_outs = preset->slave_dig_outs;
spec->multiout.hp_nid = preset->hp_nid;
spec->num_mux_defs = preset->num_mux_defs;
if (!spec->num_mux_defs)
spec->num_mux_defs = 1;
spec->input_mux = preset->input_mux;
spec->num_adc_nids = preset->num_adc_nids;
spec->adc_nids = preset->adc_nids;
spec->capsrc_nids = preset->capsrc_nids;
spec->dig_in_nid = preset->dig_in_nid;
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->power_hook = preset->power_hook;
spec->loopback.amplist = preset->loopbacks;
#endif
if (preset->setup)
preset->setup(codec);
alc_fixup_autocfg_pin_nums(codec);
}
/* auto-toggle front mic */
static void alc88x_simple_mic_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
present = snd_hda_jack_detect(codec, 0x18);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}

371
sound/pci/hda/hda_codec.c
View File

@ -243,7 +243,8 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
{
unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm);
unsigned int res;
codec_exec_verb(codec, cmd, &res);
if (codec_exec_verb(codec, cmd, &res))
return -1;
return res;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_read);
@ -307,14 +308,65 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
}
EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes);
static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
static bool add_conn_list(struct snd_array *array, hda_nid_t nid);
static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
hda_nid_t *src, int len);
/* look up the cached results */
static hda_nid_t *lookup_conn_list(struct snd_array *array, hda_nid_t nid)
{
int i, len;
for (i = 0; i < array->used; ) {
hda_nid_t *p = snd_array_elem(array, i);
if (nid == *p)
return p;
len = p[1];
i += len + 2;
}
return NULL;
}
/**
* snd_hda_get_connections - get connection list
* snd_hda_get_conn_list - get connection list
* @codec: the HDA codec
* @nid: NID to parse
* @listp: the pointer to store NID list
*
* Parses the connection list of the given widget and stores the list
* of NIDs.
*
* Returns the number of connections, or a negative error code.
*/
int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
const hda_nid_t **listp)
{
struct snd_array *array = &codec->conn_lists;
int len, err;
hda_nid_t list[HDA_MAX_CONNECTIONS];
hda_nid_t *p;
bool added = false;
again:
/* if the connection-list is already cached, read it */
p = lookup_conn_list(array, nid);
if (p) {
if (listp)
*listp = p + 2;
return p[1];
}
if (snd_BUG_ON(added))
return -EINVAL;
/* read the connection and add to the cache */
len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
if (len < 0)
return len;
err = snd_hda_override_conn_list(codec, nid, len, list);
if (err < 0)
return err;
added = true;
goto again;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
/**
* snd_hda_get_connections - copy connection list
* @codec: the HDA codec
* @nid: NID to parse
* @conn_list: connection list array
@ -328,42 +380,35 @@ static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns)
{
struct snd_array *array = &codec->conn_lists;
int i, len, old_used;
hda_nid_t list[HDA_MAX_CONNECTIONS];
const hda_nid_t *list;
int len = snd_hda_get_conn_list(codec, nid, &list);
/* look up the cached results */
for (i = 0; i < array->used; ) {
hda_nid_t *p = snd_array_elem(array, i);
len = p[1];
if (nid == *p)
return copy_conn_list(nid, conn_list, max_conns,
p + 2, len);
i += len + 2;
}
len = _hda_get_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
if (len < 0)
if (len <= 0)
return len;
/* add to the cache */
old_used = array->used;
if (!add_conn_list(array, nid) || !add_conn_list(array, len))
goto error_add;
for (i = 0; i < len; i++)
if (!add_conn_list(array, list[i]))
goto error_add;
return copy_conn_list(nid, conn_list, max_conns, list, len);
error_add:
array->used = old_used;
return -ENOMEM;
if (len > max_conns) {
snd_printk(KERN_ERR "hda_codec: "
"Too many connections %d for NID 0x%x\n",
len, nid);
return -EINVAL;
}
memcpy(conn_list, list, len * sizeof(hda_nid_t));
return len;
}
EXPORT_SYMBOL_HDA(snd_hda_get_connections);
static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns)
/**
* snd_hda_get_raw_connections - copy connection list without cache
* @codec: the HDA codec
* @nid: NID to parse
* @conn_list: connection list array
* @max_conns: max. number of connections to store
*
* Like snd_hda_get_connections(), copy the connection list but without
* checking through the connection-list cache.
* Currently called only from hda_proc.c, so not exported.
*/
int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns)
{
unsigned int parm;
int i, conn_len, conns;
@ -376,11 +421,8 @@ static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
wcaps = get_wcaps(codec, nid);
if (!(wcaps & AC_WCAP_CONN_LIST) &&
get_wcaps_type(wcaps) != AC_WID_VOL_KNB) {
snd_printk(KERN_WARNING "hda_codec: "
"connection list not available for 0x%x\n", nid);
return -EINVAL;
}
get_wcaps_type(wcaps) != AC_WID_VOL_KNB)
return 0;
parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
if (parm & AC_CLIST_LONG) {
@ -470,18 +512,77 @@ static bool add_conn_list(struct snd_array *array, hda_nid_t nid)
return true;
}
static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst,
hda_nid_t *src, int len)
/**
* snd_hda_override_conn_list - add/modify the connection-list to cache
* @codec: the HDA codec
* @nid: NID to parse
* @len: number of connection list entries
* @list: the list of connection entries
*
* Add or modify the given connection-list to the cache. If the corresponding
* cache already exists, invalidate it and append a new one.
*
* Returns zero or a negative error code.
*/
int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
const hda_nid_t *list)
{
if (len > max_dst) {
snd_printk(KERN_ERR "hda_codec: "
"Too many connections %d for NID 0x%x\n",
len, nid);
return -EINVAL;
}
memcpy(dst, src, len * sizeof(hda_nid_t));
return len;
struct snd_array *array = &codec->conn_lists;
hda_nid_t *p;
int i, old_used;
p = lookup_conn_list(array, nid);
if (p)
*p = -1; /* invalidate the old entry */
old_used = array->used;
if (!add_conn_list(array, nid) || !add_conn_list(array, len))
goto error_add;
for (i = 0; i < len; i++)
if (!add_conn_list(array, list[i]))
goto error_add;
return 0;
error_add:
array->used = old_used;
return -ENOMEM;
}
EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
/**
* snd_hda_get_conn_index - get the connection index of the given NID
* @codec: the HDA codec
* @mux: NID containing the list
* @nid: NID to select
* @recursive: 1 when searching NID recursively, otherwise 0
*
* Parses the connection list of the widget @mux and checks whether the
* widget @nid is present. If it is, return the connection index.
* Otherwise it returns -1.
*/
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid, int recursive)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
if (!recursive)
return -1;
if (recursive > 5) {
snd_printd("hda_codec: too deep connection for 0x%x\n", nid);
return -1;
}
recursive++;
for (i = 0; i < nums; i++)
if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
return i;
return -1;
}
EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
/**
* snd_hda_queue_unsol_event - add an unsolicited event to queue
@ -1083,6 +1184,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
snd_array_free(&codec->mixers);
snd_array_free(&codec->nids);
snd_array_free(&codec->conn_lists);
snd_array_free(&codec->spdif_out);
codec->bus->caddr_tbl[codec->addr] = NULL;
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
@ -1144,6 +1246,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16);
snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8);
snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64);
snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16);
if (codec->bus->modelname) {
codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
if (!codec->modelname) {
@ -2555,11 +2658,13 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff;
ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff;
ucontrol->value.iec958.status[0] = spdif->status & 0xff;
ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff;
return 0;
}
@ -2644,23 +2749,23 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value;
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
hda_nid_t nid = spdif->nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
codec->spdif_status = ucontrol->value.iec958.status[0] |
spdif->status = ucontrol->value.iec958.status[0] |
((unsigned int)ucontrol->value.iec958.status[1] << 8) |
((unsigned int)ucontrol->value.iec958.status[2] << 16) |
((unsigned int)ucontrol->value.iec958.status[3] << 24);
val = convert_from_spdif_status(codec->spdif_status);
val |= codec->spdif_ctls & 1;
change = codec->spdif_ctls != val;
codec->spdif_ctls = val;
if (change)
val = convert_from_spdif_status(spdif->status);
val |= spdif->ctls & 1;
change = spdif->ctls != val;
spdif->ctls = val;
if (change && nid != (u16)-1)
set_dig_out_convert(codec, nid, val & 0xff, (val >> 8) & 0xff);
mutex_unlock(&codec->spdif_mutex);
return change;
}
@ -2671,33 +2776,42 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.integer.value[0] = codec->spdif_ctls & AC_DIG1_ENABLE;
ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
return 0;
}
static inline void set_spdif_ctls(struct hda_codec *codec, hda_nid_t nid,
int dig1, int dig2)
{
set_dig_out_convert(codec, nid, dig1, dig2);
/* unmute amp switch (if any) */
if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
(dig1 & AC_DIG1_ENABLE))
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = kcontrol->private_value;
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
hda_nid_t nid = spdif->nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
val = codec->spdif_ctls & ~AC_DIG1_ENABLE;
val = spdif->ctls & ~AC_DIG1_ENABLE;
if (ucontrol->value.integer.value[0])
val |= AC_DIG1_ENABLE;
change = codec->spdif_ctls != val;
if (change) {
codec->spdif_ctls = val;
set_dig_out_convert(codec, nid, val & 0xff, -1);
/* unmute amp switch (if any) */
if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
(val & AC_DIG1_ENABLE))
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
}
change = spdif->ctls != val;
spdif->ctls = val;
if (change && nid != (u16)-1)
set_spdif_ctls(codec, nid, val & 0xff, -1);
mutex_unlock(&codec->spdif_mutex);
return change;
}
@ -2744,36 +2858,79 @@ static struct snd_kcontrol_new dig_mixes[] = {
*
* Returns 0 if successful, or a negative error code.
*/
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
hda_nid_t associated_nid,
hda_nid_t cvt_nid)
{
int err;
struct snd_kcontrol *kctl;
struct snd_kcontrol_new *dig_mix;
int idx;
struct hda_spdif_out *spdif;
idx = find_empty_mixer_ctl_idx(codec, "IEC958 Playback Switch");
if (idx < 0) {
printk(KERN_ERR "hda_codec: too many IEC958 outputs\n");
return -EBUSY;
}
spdif = snd_array_new(&codec->spdif_out);
for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
kctl = snd_ctl_new1(dig_mix, codec);
if (!kctl)
return -ENOMEM;
kctl->id.index = idx;
kctl->private_value = nid;
err = snd_hda_ctl_add(codec, nid, kctl);
kctl->private_value = codec->spdif_out.used - 1;
err = snd_hda_ctl_add(codec, associated_nid, kctl);
if (err < 0)
return err;
}
codec->spdif_ctls =
snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_DIGI_CONVERT_1, 0);
codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls);
spdif->nid = cvt_nid;
spdif->ctls = snd_hda_codec_read(codec, cvt_nid, 0,
AC_VERB_GET_DIGI_CONVERT_1, 0);
spdif->status = convert_to_spdif_status(spdif->ctls);
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
hda_nid_t nid)
{
int i;
for (i = 0; i < codec->spdif_out.used; i++) {
struct hda_spdif_out *spdif =
snd_array_elem(&codec->spdif_out, i);
if (spdif->nid == nid)
return spdif;
}
return NULL;
}
EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid);
void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
{
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
mutex_lock(&codec->spdif_mutex);
spdif->nid = (u16)-1;
mutex_unlock(&codec->spdif_mutex);
}
EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign);
void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
{
struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
unsigned short val;
mutex_lock(&codec->spdif_mutex);
if (spdif->nid != nid) {
spdif->nid = nid;
val = spdif->ctls;
set_spdif_ctls(codec, nid, val & 0xff, (val >> 8) & 0xff);
}
mutex_unlock(&codec->spdif_mutex);
}
EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_assign);
/*
* SPDIF sharing with analog output
*/
@ -3356,7 +3513,7 @@ static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
*
* Returns 0 if successful, otherwise a negative error code.
*/
static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
{
unsigned int i, val, wcaps;
@ -3448,6 +3605,7 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_query_supported_pcm);
/**
* snd_hda_is_supported_format - Check the validity of the format
@ -4177,10 +4335,12 @@ EXPORT_SYMBOL_HDA(snd_hda_input_mux_put);
static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
unsigned int stream_tag, unsigned int format)
{
struct hda_spdif_out *spdif = snd_hda_spdif_out_of_nid(codec, nid);
/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
set_dig_out_convert(codec, nid,
codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff,
spdif->ctls & ~AC_DIG1_ENABLE & 0xff,
-1);
snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
if (codec->slave_dig_outs) {
@ -4190,9 +4350,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
format);
}
/* turn on again (if needed) */
if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE))
set_dig_out_convert(codec, nid,
codec->spdif_ctls & 0xff, -1);
spdif->ctls & 0xff, -1);
}
static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
@ -4348,6 +4508,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
{
const hda_nid_t *nids = mout->dac_nids;
int chs = substream->runtime->channels;
struct hda_spdif_out *spdif =
snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
int i;
mutex_lock(&codec->spdif_mutex);
@ -4356,7 +4518,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
if (chs == 2 &&
snd_hda_is_supported_format(codec, mout->dig_out_nid,
format) &&
!(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
!(spdif->status & IEC958_AES0_NONAUDIO)) {
mout->dig_out_used = HDA_DIG_ANALOG_DUP;
setup_dig_out_stream(codec, mout->dig_out_nid,
stream_tag, format);
@ -4528,7 +4690,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
unsigned int wid_caps = get_wcaps(codec, nid);
unsigned int wid_type = get_wcaps_type(wid_caps);
unsigned int def_conf;
short assoc, loc;
short assoc, loc, conn, dev;
/* read all default configuration for pin complex */
if (wid_type != AC_WID_PIN)
@ -4538,10 +4700,19 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
continue;
def_conf = snd_hda_codec_get_pincfg(codec, nid);
if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
conn = get_defcfg_connect(def_conf);
if (conn == AC_JACK_PORT_NONE)
continue;
loc = get_defcfg_location(def_conf);
switch (get_defcfg_device(def_conf)) {
dev = get_defcfg_device(def_conf);
/* workaround for buggy BIOS setups */
if (dev == AC_JACK_LINE_OUT) {
if (conn == AC_JACK_PORT_FIXED)
dev = AC_JACK_SPEAKER;
}
switch (dev) {
case AC_JACK_LINE_OUT:
seq = get_defcfg_sequence(def_conf);
assoc = get_defcfg_association(def_conf);
@ -4957,17 +5128,15 @@ void *snd_array_new(struct snd_array *array)
{
if (array->used >= array->alloced) {
int num = array->alloced + array->alloc_align;
int size = (num + 1) * array->elem_size;
int oldsize = array->alloced * array->elem_size;
void *nlist;
if (snd_BUG_ON(num >= 4096))
return NULL;
nlist = kcalloc(num + 1, array->elem_size, GFP_KERNEL);
nlist = krealloc(array->list, size, GFP_KERNEL);
if (!nlist)
return NULL;
if (array->list) {
memcpy(nlist, array->list,
array->elem_size * array->alloced);
kfree(array->list);
}
memset(nlist + oldsize, 0, size - oldsize);
array->list = nlist;
array->alloced = num;
}

30
sound/pci/hda/hda_codec.h
View File

@ -829,8 +829,7 @@ struct hda_codec {
struct mutex spdif_mutex;
struct mutex control_mutex;
unsigned int spdif_status; /* IEC958 status bits */
unsigned short spdif_ctls; /* SPDIF control bits */
struct snd_array spdif_out;
unsigned int spdif_in_enable; /* SPDIF input enable? */
const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
struct snd_array init_pins; /* initial (BIOS) pin configurations */
@ -904,6 +903,16 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *start_id);
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
const hda_nid_t **listp);
int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
const hda_nid_t *list);
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid, int recursive);
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
struct hda_verb {
hda_nid_t nid;
@ -947,6 +956,17 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
hda_nid_t nid, unsigned int cfg); /* for hwdep */
void snd_hda_shutup_pins(struct hda_codec *codec);
/* SPDIF controls */
struct hda_spdif_out {
hda_nid_t nid; /* Converter nid values relate to */
unsigned int status; /* IEC958 status bits */
unsigned short ctls; /* SPDIF control bits */
};
struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
hda_nid_t nid);
void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx);
void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid);
/*
* Mixer
*/
@ -997,17 +1017,15 @@ int snd_hda_suspend(struct hda_bus *bus);
int snd_hda_resume(struct hda_bus *bus);
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
static inline
int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (codec->patch_ops.check_power_status)
return codec->patch_ops.check_power_status(codec, nid);
#endif
return 0;
}
#else
#define hda_call_check_power_status(codec, nid) 0
#endif
/*
* get widget information

46
sound/pci/hda/hda_eld.c
View File

@ -580,43 +580,45 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
#endif /* CONFIG_PROC_FS */
/* update PCM info based on ELD */
void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
struct hda_pcm_stream *codec_pars)
void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
struct hda_pcm_stream *hinfo)
{
u32 rates;
u64 formats;
unsigned int maxbps;
unsigned int channels_max;
int i;
/* assume basic audio support (the basic audio flag is not in ELD;
* however, all audio capable sinks are required to support basic
* audio) */
pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
pcm->formats = SNDRV_PCM_FMTBIT_S16_LE;
pcm->maxbps = 16;
pcm->channels_max = 2;
rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000;
formats = SNDRV_PCM_FMTBIT_S16_LE;
maxbps = 16;
channels_max = 2;
for (i = 0; i < eld->sad_count; i++) {
struct cea_sad *a = &eld->sad[i];
pcm->rates |= a->rates;
if (a->channels > pcm->channels_max)
pcm->channels_max = a->channels;
rates |= a->rates;
if (a->channels > channels_max)
channels_max = a->channels;
if (a->format == AUDIO_CODING_TYPE_LPCM) {
if (a->sample_bits & AC_SUPPCM_BITS_20) {
pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (pcm->maxbps < 20)
pcm->maxbps = 20;
formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (maxbps < 20)
maxbps = 20;
}
if (a->sample_bits & AC_SUPPCM_BITS_24) {
pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (pcm->maxbps < 24)
pcm->maxbps = 24;
formats |= SNDRV_PCM_FMTBIT_S32_LE;
if (maxbps < 24)
maxbps = 24;
}
}
}
if (!codec_pars)
return;
/* restrict the parameters by the values the codec provides */
pcm->rates &= codec_pars->rates;
pcm->formats &= codec_pars->formats;
pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max);
pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
hinfo->rates &= rates;
hinfo->formats &= formats;
hinfo->maxbps = min(hinfo->maxbps, maxbps);
hinfo->channels_max = min(hinfo->channels_max, channels_max);
}

80
sound/pci/hda/hda_intel.c
View File

@ -177,7 +177,8 @@ MODULE_DESCRIPTION("Intel HDA driver");
#define ICH6_REG_INTCTL 0x20
#define ICH6_REG_INTSTS 0x24
#define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */
#define ICH6_REG_SYNC 0x34
#define ICH6_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */
#define ICH6_REG_SSYNC 0x38
#define ICH6_REG_CORBLBASE 0x40
#define ICH6_REG_CORBUBASE 0x44
#define ICH6_REG_CORBWP 0x48
@ -479,6 +480,7 @@ enum {
#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */
#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */
#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@ -1706,13 +1708,16 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned int bufsize, period_bytes, format_val, stream_tag;
int err;
struct hda_spdif_out *spdif =
snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid);
unsigned short ctls = spdif ? spdif->ctls : 0;
azx_stream_reset(chip, azx_dev);
format_val = snd_hda_calc_stream_format(runtime->rate,
runtime->channels,
runtime->format,
hinfo->maxbps,
apcm->codec->spdif_ctls);
ctls);
if (!format_val) {
snd_printk(KERN_ERR SFX
"invalid format_val, rate=%d, ch=%d, format=%d\n",
@ -1792,7 +1797,11 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
spin_lock(&chip->reg_lock);
if (nsync > 1) {
/* first, set SYNC bits of corresponding streams */
azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits);
if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
azx_writel(chip, OLD_SSYNC,
azx_readl(chip, OLD_SSYNC) | sbits);
else
azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits);
}
snd_pcm_group_for_each_entry(s, substream) {
if (s->pcm->card != substream->pcm->card)
@ -1848,7 +1857,11 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
if (nsync > 1) {
spin_lock(&chip->reg_lock);
/* reset SYNC bits */
azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits);
if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC)
azx_writel(chip, OLD_SSYNC,
azx_readl(chip, OLD_SSYNC) & ~sbits);
else
azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits);
spin_unlock(&chip->reg_lock);
}
return 0;
@ -1863,7 +1876,7 @@ static unsigned int azx_via_get_position(struct azx *chip,
unsigned int fifo_size;
link_pos = azx_sd_readl(azx_dev, SD_LPIB);
if (azx_dev->index >= 4) {
if (azx_dev->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* Playback, no problem using link position */
return link_pos;
}
@ -1927,6 +1940,17 @@ static unsigned int azx_get_position(struct azx *chip,
default:
/* use the position buffer */
pos = le32_to_cpu(*azx_dev->posbuf);
if (chip->position_fix[stream] == POS_FIX_AUTO) {
if (!pos || pos == (u32)-1) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
"using LPIB read method instead.\n");
chip->position_fix[stream] = POS_FIX_LPIB;
pos = azx_sd_readl(azx_dev, SD_LPIB);
} else
chip->position_fix[stream] = POS_FIX_POSBUF;
}
break;
}
if (pos >= azx_dev->bufsize)
@ -1964,16 +1988,6 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
stream = azx_dev->substream->stream;
pos = azx_get_position(chip, azx_dev);
if (chip->position_fix[stream] == POS_FIX_AUTO) {
if (!pos) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
"using LPIB read method instead.\n");
chip->position_fix[stream] = POS_FIX_LPIB;
pos = azx_get_position(chip, azx_dev);
} else
chip->position_fix[stream] = POS_FIX_POSBUF;
}
if (WARN_ONCE(!azx_dev->period_bytes,
"hda-intel: zero azx_dev->period_bytes"))
@ -2061,6 +2075,8 @@ static void azx_pcm_free(struct snd_pcm *pcm)
}
}
#define MAX_PREALLOC_SIZE (32 * 1024 * 1024)
static int
azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
struct hda_pcm *cpcm)
@ -2069,6 +2085,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
struct snd_pcm *pcm;
struct azx_pcm *apcm;
int pcm_dev = cpcm->device;
unsigned int size;
int s, err;
if (pcm_dev >= HDA_MAX_PCMS) {
@ -2104,9 +2121,12 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
snd_pcm_set_ops(pcm, s, &azx_pcm_ops);
}
/* buffer pre-allocation */
size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024;
if (size > MAX_PREALLOC_SIZE)
size = MAX_PREALLOC_SIZE;
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
snd_dma_pci_data(chip->pci),
1024 * 64, 32 * 1024 * 1024);
size, MAX_PREALLOC_SIZE);
return 0;
}
@ -2149,7 +2169,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
{
if (request_irq(chip->pci->irq, azx_interrupt,
chip->msi ? 0 : IRQF_SHARED,
"hda_intel", chip)) {
KBUILD_MODNAME, chip)) {
printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
"disabling device\n", chip->pci->irq);
if (do_disconnect)
@ -2347,28 +2367,20 @@ static int azx_dev_free(struct snd_device *device)
* white/black-listing for position_fix
*/
static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB),
SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
{}
};
@ -2815,6 +2827,22 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
{ PCI_DEVICE(0x8086, 0x2668),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH6 */
{ PCI_DEVICE(0x8086, 0x27d8),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH7 */
{ PCI_DEVICE(0x8086, 0x269a),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ESB2 */
{ PCI_DEVICE(0x8086, 0x284b),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH8 */
{ PCI_DEVICE(0x8086, 0x293e),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x293f),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x3a3e),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
{ PCI_DEVICE(0x8086, 0x3a6e),
.driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
/* Generic Intel */
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID),
.class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
@ -2908,7 +2936,7 @@ MODULE_DEVICE_TABLE(pci, azx_ids);
/* pci_driver definition */
static struct pci_driver driver = {
.name = "HDA Intel",
.name = KBUILD_MODNAME,
.id_table = azx_ids,
.probe = azx_probe,
.remove = __devexit_p(azx_remove),

10
sound/pci/hda/hda_local.h
View File

@ -212,7 +212,9 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
/*
* SPDIF I/O
*/
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
hda_nid_t associated_nid,
hda_nid_t cvt_nid);
int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
/*
@ -563,7 +565,6 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
* power-management
*/
#ifdef CONFIG_SND_HDA_POWER_SAVE
void snd_hda_schedule_power_save(struct hda_codec *codec);
struct hda_amp_list {
@ -580,7 +581,6 @@ struct hda_loopback_check {
int snd_hda_check_amp_list_power(struct hda_codec *codec,
struct hda_loopback_check *check,
hda_nid_t nid);
#endif /* CONFIG_SND_HDA_POWER_SAVE */
/*
* AMP control callbacks
@ -639,8 +639,8 @@ struct hdmi_eld {
int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
void snd_hdmi_show_eld(struct hdmi_eld *eld);
void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
struct hda_pcm_stream *codec_pars);
void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
struct hda_pcm_stream *hinfo);
#ifdef CONFIG_PROC_FS
int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,

2
sound/pci/hda/hda_proc.c
View File

@ -636,7 +636,7 @@ static void print_codec_info(struct snd_info_entry *entry,
wid_caps |= AC_WCAP_CONN_LIST;
if (wid_caps & AC_WCAP_CONN_LIST)
conn_len = snd_hda_get_connections(codec, nid, conn,
conn_len = snd_hda_get_raw_connections(codec, nid, conn,
HDA_MAX_CONNECTIONS);
if (wid_caps & AC_WCAP_IN_AMP) {

7
sound/pci/hda/patch_analog.c
View File

@ -213,7 +213,9 @@ static int ad198x_build_controls(struct hda_codec *codec)
return err;
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@ -1920,7 +1922,8 @@ static int patch_ad1981(struct hda_codec *codec)
spec->mixers[0] = ad1981_hp_mixers;
spec->num_init_verbs = 2;
spec->init_verbs[1] = ad1981_hp_init_verbs;
spec->multiout.dig_out_nid = 0;
if (!is_jack_available(codec, 0x0a))
spec->multiout.dig_out_nid = 0;
spec->input_mux = &ad1981_hp_capture_source;
codec->patch_ops.init = ad1981_hp_init;

3
sound/pci/hda/patch_ca0110.c
View File

@ -240,7 +240,8 @@ static int ca0110_build_controls(struct hda_codec *codec)
}
if (spec->dig_out) {
err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out);
err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
spec->dig_out);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);

1097
sound/pci/hda/patch_ca0132.c Normal file
View File

File diff suppressed because it is too large Load Diff

19
sound/pci/hda/patch_cirrus.c
View File

@ -346,21 +346,15 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
nid = codec->start_nid;
for (i = 0; i < codec->num_nodes; i++, nid++) {
hda_nid_t pins[2];
unsigned int type;
int j, nums;
int idx;
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
nums = snd_hda_get_connections(codec, nid, pins,
ARRAY_SIZE(pins));
if (nums <= 0)
continue;
for (j = 0; j < nums; j++) {
if (pins[j] == pin) {
*idxp = j;
return nid;
}
idx = snd_hda_get_conn_index(codec, nid, pin, 0);
if (idx >= 0) {
*idxp = idx;
return nid;
}
}
return 0;
@ -821,7 +815,8 @@ static int build_digital_output(struct hda_codec *codec)
if (!spec->multiout.dig_out_nid)
return 0;
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);

17
sound/pci/hda/patch_cmedia.c
View File

@ -327,7 +327,9 @@ static int cmi9880_build_controls(struct hda_codec *codec)
return err;
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@ -396,12 +398,11 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi
{
struct cmi_spec *spec = codec->spec;
hda_nid_t nid;
int i, j, k, len;
int i, j, k;
/* clear the table, only one c-media dac assumed here */
memset(spec->multi_init, 0, sizeof(spec->multi_init));
for (j = 0, i = 0; i < cfg->line_outs; i++) {
hda_nid_t conn[4];
nid = cfg->line_out_pins[i];
/* set as output */
spec->multi_init[j].nid = nid;
@ -414,12 +415,10 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi
spec->multi_init[j].verb = AC_VERB_SET_CONNECT_SEL;
spec->multi_init[j].param = 0;
/* find the index in connect list */
len = snd_hda_get_connections(codec, nid, conn, 4);
for (k = 0; k < len; k++)
if (conn[k] == spec->dac_nids[i]) {
spec->multi_init[j].param = k;
break;
}
k = snd_hda_get_conn_index(codec, nid,
spec->dac_nids[i], 0);
if (k >= 0)
spec->multi_init[j].param = k;
j++;
}
}

71
sound/pci/hda/patch_conexant.c
View File

@ -155,6 +155,10 @@ struct conexant_spec {
unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
unsigned int beep_amp;
/* extra EAPD pins */
unsigned int num_eapds;
hda_nid_t eapds[4];
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@ -510,6 +514,7 @@ static int conexant_build_controls(struct hda_codec *codec)
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
@ -1123,10 +1128,8 @@ static int patch_cxt5045(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5045_MODELS,
cxt5045_models,
cxt5045_cfg_tbl);
#if 0 /* use the old method just for safety */
if (board_config < 0)
board_config = CXT5045_AUTO;
#endif
board_config = CXT5045_AUTO; /* model=auto as default */
if (board_config == CXT5045_AUTO)
return patch_conexant_auto(codec);
@ -1564,10 +1567,8 @@ static int patch_cxt5047(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5047_MODELS,
cxt5047_models,
cxt5047_cfg_tbl);
#if 0 /* not enabled as default, as BIOS often broken for this codec */
if (board_config < 0)
board_config = CXT5047_AUTO;
#endif
board_config = CXT5047_AUTO; /* model=auto as default */
if (board_config == CXT5047_AUTO)
return patch_conexant_auto(codec);
@ -1993,10 +1994,8 @@ static int patch_cxt5051(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
cxt5051_models,
cxt5051_cfg_tbl);
#if 0 /* use the old method just for safety */
if (board_config < 0)
board_config = CXT5051_AUTO;
#endif
board_config = CXT5051_AUTO; /* model=auto as default */
if (board_config == CXT5051_AUTO)
return patch_conexant_auto(codec);
@ -3114,10 +3113,8 @@ static int patch_cxt5066(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
cxt5066_models, cxt5066_cfg_tbl);
#if 0 /* use the old method just for safety */
if (board_config < 0)
board_config = CXT5066_AUTO;
#endif
board_config = CXT5066_AUTO; /* model=auto as default */
if (board_config == CXT5066_AUTO)
return patch_conexant_auto(codec);
@ -3308,19 +3305,8 @@ static const struct hda_pcm_stream cx_auto_pcm_analog_capture = {
static const hda_nid_t cx_auto_adc_nids[] = { 0x14 };
/* get the connection index of @nid in the widget @mux */
static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
return -1;
}
#define get_connection_index(codec, mux, nid)\
snd_hda_get_conn_index(codec, mux, nid, 0)
/* get an unassigned DAC from the given list.
* Return the nid if found and reduce the DAC list, or return zero if
@ -3919,6 +3905,38 @@ static void cx_auto_parse_beep(struct hda_codec *codec)
#define cx_auto_parse_beep(codec)
#endif
static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
{
int i;
for (i = 0; i < nums; i++)
if (list[i] == nid)
return true;
return false;
}
/* parse extra-EAPD that aren't assigned to any pins */
static void cx_auto_parse_eapd(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
hda_nid_t nid, end_nid;
end_nid = codec->start_nid + codec->num_nodes;
for (nid = codec->start_nid; nid < end_nid; nid++) {
if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
continue;
if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD))
continue;
if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) ||
found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) ||
found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs))
continue;
spec->eapds[spec->num_eapds++] = nid;
if (spec->num_eapds >= ARRAY_SIZE(spec->eapds))
break;
}
}
static int cx_auto_parse_auto_config(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@ -3932,6 +3950,7 @@ static int cx_auto_parse_auto_config(struct hda_codec *codec)
cx_auto_parse_input(codec);
cx_auto_parse_digital(codec);
cx_auto_parse_beep(codec);
cx_auto_parse_eapd(codec);
return 0;
}
@ -4019,6 +4038,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
}
}
cx_auto_update_speakers(codec);
/* turn on/off extra EAPDs, too */
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
}
static void cx_auto_init_input(struct hda_codec *codec)

698
sound/pci/hda/patch_hdmi.c
View File

File diff suppressed because it is too large Load Diff

19880
sound/pci/hda/patch_realtek.c
View File

File diff suppressed because it is too large Load Diff

31
sound/pci/hda/patch_sigmatel.c
View File

@ -1112,7 +1112,9 @@ static int stac92xx_build_controls(struct hda_codec *codec)
}
if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
err = snd_hda_create_spdif_out_ctls(codec,
spec->multiout.dig_out_nid,
spec->multiout.dig_out_nid);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec,
@ -3406,30 +3408,9 @@ static hda_nid_t get_connected_node(struct hda_codec *codec, hda_nid_t mux,
return 0;
}
static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
hda_nid_t nid)
{
hda_nid_t conn[HDA_MAX_NUM_INPUTS];
int i, nums;
if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST))
return -1;
nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
for (i = 0; i < nums; i++)
if (conn[i] == nid)
return i;
for (i = 0; i < nums; i++) {
unsigned int wid_caps = get_wcaps(codec, conn[i]);
unsigned int wid_type = get_wcaps_type(wid_caps);
if (wid_type != AC_WID_PIN && wid_type != AC_WID_AUD_MIX)
if (get_connection_index(codec, conn[i], nid) >= 0)
return i;
}
return -1;
}
/* look for NID recursively */
#define get_connection_index(codec, mux, nid) \
snd_hda_get_conn_index(codec, mux, nid, 1)
/* create a volume assigned to the given pin (only if supported) */
/* return 1 if the volume control is created */

6161
sound/pci/hda/patch_via.c
View File

File diff suppressed because it is too large Load Diff

4
sound/pci/ice1712/ice1712.c
View File

@ -2607,7 +2607,7 @@ static int __devinit snd_ice1712_create(struct snd_card *card,
ice->profi_port = pci_resource_start(pci, 3);
if (request_irq(pci->irq, snd_ice1712_interrupt, IRQF_SHARED,
"ICE1712", ice)) {
KBUILD_MODNAME, ice)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_ice1712_free(ice);
return -EIO;
@ -2802,7 +2802,7 @@ static void __devexit snd_ice1712_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "ICE1712",
.name = KBUILD_MODNAME,
.id_table = snd_ice1712_ids,
.probe = snd_ice1712_probe,
.remove = __devexit_p(snd_ice1712_remove),

4
sound/pci/ice1712/ice1724.c
View File

@ -2509,7 +2509,7 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
ice->profi_port = pci_resource_start(pci, 1);
if (request_irq(pci->irq, snd_vt1724_interrupt,
IRQF_SHARED, "ICE1724", ice)) {
IRQF_SHARED, KBUILD_MODNAME, ice)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_vt1724_free(ice);
return -EIO;
@ -2802,7 +2802,7 @@ static int snd_vt1724_resume(struct pci_dev *pci)
#endif
static struct pci_driver driver = {
.name = "ICE1724",
.name = KBUILD_MODNAME,
.id_table = snd_vt1724_ids,
.probe = snd_vt1724_probe,
.remove = __devexit_p(snd_vt1724_remove),

12
sound/pci/intel8x0.c
View File

@ -1882,6 +1882,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
.name = "Dell Inspiron 6000",
.type = AC97_TUNE_HP_MUTE_LED /* cf. Malone #41015 */
},
{
.subvendor = 0x1028,
.subdevice = 0x0189,
.name = "Dell Inspiron 9300",
.type = AC97_TUNE_HP_MUTE_LED
},
{
.subvendor = 0x1028,
.subdevice = 0x0191,
@ -2647,7 +2653,7 @@ static int intel8x0_resume(struct pci_dev *pci)
pci_set_master(pci);
snd_intel8x0_chip_init(chip, 0);
if (request_irq(pci->irq, snd_intel8x0_interrupt,
IRQF_SHARED, card->shortname, chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "intel8x0: unable to grab IRQ %d, "
"disabling device\n", pci->irq);
snd_card_disconnect(card);
@ -3106,7 +3112,7 @@ static int __devinit snd_intel8x0_create(struct snd_card *card,
/* request irq after initializaing int_sta_mask, etc */
if (request_irq(pci->irq, snd_intel8x0_interrupt,
IRQF_SHARED, card->shortname, chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_intel8x0_free(chip);
return -EBUSY;
@ -3266,7 +3272,7 @@ static void __devexit snd_intel8x0_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "Intel ICH",
.name = KBUILD_MODNAME,
.id_table = snd_intel8x0_ids,
.probe = snd_intel8x0_probe,
.remove = __devexit_p(snd_intel8x0_remove),

6
sound/pci/intel8x0m.c
View File

@ -1047,7 +1047,7 @@ static int intel8x0m_resume(struct pci_dev *pci)
}
pci_set_master(pci);
if (request_irq(pci->irq, snd_intel8x0m_interrupt,
IRQF_SHARED, card->shortname, chip)) {
IRQF_SHARED, KBUILD_MODNAME, chip)) {
printk(KERN_ERR "intel8x0m: unable to grab IRQ %d, "
"disabling device\n", pci->irq);
snd_card_disconnect(card);
@ -1174,7 +1174,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card,
port_inited:
if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED,
card->shortname, chip)) {
KBUILD_MODNAME, chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
snd_intel8x0m_free(chip);
return -EBUSY;
@ -1325,7 +1325,7 @@ static void __devexit snd_intel8x0m_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "Intel ICH Modem",
.name = KBUILD_MODNAME,
.id_table = snd_intel8x0m_ids,
.probe = snd_intel8x0m_probe,
.remove = __devexit_p(snd_intel8x0m_remove),

4
sound/pci/korg1212/korg1212.c
View File

@ -2241,7 +2241,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev *
err = request_irq(pci->irq, snd_korg1212_interrupt,
IRQF_SHARED,
"korg1212", korg1212);
KBUILD_MODNAME, korg1212);
if (err) {
snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq);
@ -2477,7 +2477,7 @@ static void __devexit snd_korg1212_remove(struct pci_dev *pci)
}
static struct pci_driver driver = {
.name = "korg1212",
.name = KBUILD_MODNAME,
.id_table = snd_korg1212_ids,
.probe = snd_korg1212_probe,
.remove = __devexit_p(snd_korg1212_remove),

4
sound/pci/lola/lola.c
View File

@ -648,7 +648,7 @@ static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci,
goto errout;
if (request_irq(pci->irq, lola_interrupt, IRQF_SHARED,
DRVNAME, chip)) {
KBUILD_MODNAME, chip)) {
printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq);
err = -EBUSY;
goto errout;
@ -771,7 +771,7 @@ MODULE_DEVICE_TABLE(pci, lola_ids);
/* pci_driver definition */
static struct pci_driver driver = {
.name = DRVNAME,
.name = KBUILD_MODNAME,
.id_table = lola_ids,
.probe = lola_probe,
.remove = __devexit_p(lola_remove),

2
sound/pci/lola/lola.h
View File

@ -480,7 +480,7 @@ struct lola {
/* count values in the Vendor Specific Mixer Widget's Audio Widget Capabilities */
#define LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(res) ((res >> 2) & 0x1f)
#define LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(res) ((res >> 7) & 0x1f)
#define LOLA_MIXER_DEST_REC_OUTPUT_SEPARATION(res) ((res >> 7) & 0x1f)
int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb,
unsigned int data, unsigned int extdata);

128
sound/pci/lola/lola_mixer.c
View File

@ -144,40 +144,61 @@ int __devinit lola_init_mixer_widget(struct lola *chip, int nid)
chip->mixer.dest_stream_ins = chip->pcm[CAPT].num_streams;
chip->mixer.dest_phys_outs = chip->pin[PLAY].num_pins;
/* mixer matrix can have unused areas between PhysIn and
/* mixer matrix may have unused areas between PhysIn and
* Play or Record and PhysOut zones
*/
chip->mixer.src_stream_out_ofs = chip->mixer.src_phys_ins +
LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(val);
chip->mixer.dest_phys_out_ofs = chip->mixer.dest_stream_ins +
LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(val);
LOLA_MIXER_DEST_REC_OUTPUT_SEPARATION(val);
/* example : MixerMatrix of LoLa881
* 0-------8------16-------8------16
* | | | | |
* | INPUT | | INPUT | |
* | -> |unused | -> |unused |
* | RECORD| | OUTPUT| |
* | | | | |
* 8--------------------------------
* | | | | |
* | | | | |
* |unused |unused |unused |unused |
* | | | | |
* | | | | |
* 16-------------------------------
* | | | | |
* | PLAY | | PLAY | |
* | -> |unused | -> |unused |
* | RECORD| | OUTPUT| |
* | | | | |
* 8--------------------------------
* | | | | |
* | | | | |
* |unused |unused |unused |unused |
* | | | | |
* | | | | |
* 16-------------------------------
/* example : MixerMatrix of LoLa881 (LoLa16161 uses unused zones)
* +-+ 0-------8------16-------8------16
* | | | | | | |
* |s| | INPUT | | INPUT | |
* | |->| -> |unused | -> |unused |
* |r| |CAPTURE| | OUTPUT| |
* | | | MIX | | MIX | |
* |c| 8--------------------------------
* | | | | | | |
* | | | | | | |
* |g| |unused |unused |unused |unused |
* | | | | | | |
* |a| | | | | |
* | | 16-------------------------------
* |i| | | | | |
* | | | PLAYBK| | PLAYBK| |
* |n|->| -> |unused | -> |unused |
* | | |CAPTURE| | OUTPUT| |
* | | | MIX | | MIX | |
* |a| 8--------------------------------
* |r| | | | | |
* |r| | | | | |
* |a| |unused |unused |unused |unused |
* |y| | | | | |
* | | | | | | |
* +++ 16--|---------------|------------
* +---V---------------V-----------+
* | dest_mix_gain_enable array |
* +-------------------------------+
*/
/* example : MixerMatrix of LoLa280
* +-+ 0-------8-2
* | | | | |
* |s| | INPUT | | INPUT
* |r|->| -> | | ->
* |c| |CAPTURE| | <- OUTPUT
* | | | MIX | | MIX
* |g| 8----------
* |a| | | |
* |i| | PLAYBK| | PLAYBACK
* |n|->| -> | | ->
* | | |CAPTURE| | <- OUTPUT
* |a| | MIX | | MIX
* |r| 8---|----|-
* |r| +---V----V-------------------+
* |a| | dest_mix_gain_enable array |
* |y| +----------------------------+
*/
if (chip->mixer.src_stream_out_ofs > MAX_AUDIO_INOUT_COUNT ||
chip->mixer.dest_phys_out_ofs > MAX_STREAM_IN_COUNT) {
@ -192,6 +213,9 @@ int __devinit lola_init_mixer_widget(struct lola *chip, int nid)
(((1U << chip->mixer.dest_phys_outs) - 1)
<< chip->mixer.dest_phys_out_ofs);
snd_printdd("Mixer src_mask=%x, dest_mask=%x\n",
chip->mixer.src_mask, chip->mixer.dest_mask);
return 0;
}
@ -202,12 +226,19 @@ static int lola_mixer_set_src_gain(struct lola *chip, unsigned int id,
if (!(chip->mixer.src_mask & (1 << id)))
return -EINVAL;
writew(gain, &chip->mixer.array->src_gain[id]);
oldval = val = readl(&chip->mixer.array->src_gain_enable);
if (on)
val |= (1 << id);
else
val &= ~(1 << id);
/* test if values unchanged */
if ((val == oldval) &&
(gain == readw(&chip->mixer.array->src_gain[id])))
return 0;
snd_printdd("lola_mixer_set_src_gain (id=%d, gain=%d) enable=%x\n",
id, gain, val);
writew(gain, &chip->mixer.array->src_gain[id]);
writel(val, &chip->mixer.array->src_gain_enable);
lola_codec_flush(chip);
/* inform micro-controller about the new source gain */
@ -269,6 +300,7 @@ static int lola_mixer_set_mapping_gain(struct lola *chip,
src, dest);
}
#if 0 /* not used */
static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id,
unsigned int mask, unsigned short *gains)
{
@ -289,6 +321,7 @@ static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id,
return lola_codec_write(chip, chip->mixer.nid,
LOLA_VERB_SET_DESTINATION_GAIN, id, 0);
}
#endif /* not used */
/*
*/
@ -376,6 +409,8 @@ static int set_analog_volume(struct lola *chip, int dir,
return 0;
if (external_call)
lola_codec_flush(chip);
snd_printdd("set_analog_volume (dir=%d idx=%d, volume=%d)\n",
dir, idx, val);
err = lola_codec_write(chip, pin->nid,
LOLA_VERB_SET_AMP_GAIN_MUTE, val, 0);
if (err < 0)
@ -427,23 +462,40 @@ static int init_mixer_values(struct lola *chip)
{
int i;
/* all src on */
/* all sample rate converters on */
lola_set_src_config(chip, (1 << chip->pin[CAPT].num_pins) - 1, false);
/* clear all matrix */
/* clear all mixer matrix settings */
memset_io(chip->mixer.array, 0, sizeof(*chip->mixer.array));
/* set src gain to 0dB */
/* inform firmware about all updated matrix columns - capture part */
for (i = 0; i < chip->mixer.dest_stream_ins; i++)
lola_codec_write(chip, chip->mixer.nid,
LOLA_VERB_SET_DESTINATION_GAIN,
i, 0);
/* inform firmware about all updated matrix columns - output part */
for (i = 0; i < chip->mixer.dest_phys_outs; i++)
lola_codec_write(chip, chip->mixer.nid,
LOLA_VERB_SET_DESTINATION_GAIN,
chip->mixer.dest_phys_out_ofs + i, 0);
/* set all digital input source (master) gains to 0dB */
for (i = 0; i < chip->mixer.src_phys_ins; i++)
lola_mixer_set_src_gain(chip, i, 336, true); /* 0dB */
/* set all digital playback source (master) gains to 0dB */
for (i = 0; i < chip->mixer.src_stream_outs; i++)
lola_mixer_set_src_gain(chip,
i + chip->mixer.src_stream_out_ofs,
336, true); /* 0dB */
/* set 1:1 dest gain */
/* set gain value 0dB diagonally in matrix - part INPUT -> CAPTURE */
for (i = 0; i < chip->mixer.dest_stream_ins; i++) {
int src = i % chip->mixer.src_phys_ins;
lola_mixer_set_mapping_gain(chip, src, i, 336, true);
}
/* set gain value 0dB diagonally in matrix , part PLAYBACK -> OUTPUT
* (LoLa280 : playback channel 0,2,4,6 linked to output channel 0)
* (LoLa280 : playback channel 1,3,5,7 linked to output channel 1)
*/
for (i = 0; i < chip->mixer.src_stream_outs; i++) {
int src = chip->mixer.src_stream_out_ofs + i;
int dst = chip->mixer.dest_phys_out_ofs +
@ -693,6 +745,7 @@ static int __devinit create_src_gain_mixer(struct lola *chip,
snd_ctl_new1(&lola_src_gain_mixer, chip));
}
#if 0 /* not used */
/*
* destination gain (matrix-like) mixer
*/
@ -781,6 +834,7 @@ static int __devinit create_dest_gain_mixer(struct lola *chip,
return snd_ctl_add(chip->card,
snd_ctl_new1(&lola_dest_gain_mixer, chip));
}
#endif /* not used */
/*
*/
@ -798,14 +852,16 @@ int __devinit lola_create_mixer(struct lola *chip)
if (err < 0)
return err;
err = create_src_gain_mixer(chip, chip->mixer.src_phys_ins, 0,
"Line Source Gain Volume");
"Digital Capture Volume");
if (err < 0)
return err;
err = create_src_gain_mixer(chip, chip->mixer.src_stream_outs,
chip->mixer.src_stream_out_ofs,
"Stream Source Gain Volume");
"Digital Playback Volume");
if (err < 0)
return err;
#if 0
/* FIXME: buggy mixer matrix handling */
err = create_dest_gain_mixer(chip,
chip->mixer.src_phys_ins, 0,
chip->mixer.dest_stream_ins, 0,
@ -834,6 +890,6 @@ int __devinit lola_create_mixer(struct lola *chip)
"Stream Playback Volume");
if (err < 0)
return err;
#endif /* FIXME */
return init_mixer_values(chip);
}

25
sound/pci/lx6464es/lx6464es.c
View File

@ -762,7 +762,6 @@ static int lx_set_granularity(struct lx6464es *chip, u32 gran)
static int __devinit lx_init_dsp(struct lx6464es *chip)
{
int err;
u8 mac_address[6];
int i;
snd_printdd("->lx_init_dsp\n");
@ -787,11 +786,11 @@ static int __devinit lx_init_dsp(struct lx6464es *chip)
/** \todo the mac address should be ready by not, but it isn't,
* so we wait for it */
for (i = 0; i != 1000; ++i) {
err = lx_dsp_get_mac(chip, mac_address);
err = lx_dsp_get_mac(chip);
if (err)
return err;
if (mac_address[0] || mac_address[1] || mac_address[2] ||
mac_address[3] || mac_address[4] || mac_address[5])
if (chip->mac_address[0] || chip->mac_address[1] || chip->mac_address[2] ||
chip->mac_address[3] || chip->mac_address[4] || chip->mac_address[5])
goto mac_ready;
msleep(1);
}
@ -800,8 +799,8 @@ static int __devinit lx_init_dsp(struct lx6464es *chip)
mac_ready:
snd_printd(LXP "mac address ready read after: %dms\n", i);
snd_printk(LXP "mac address: %02X.%02X.%02X.%02X.%02X.%02X\n",
mac_address[0], mac_address[1], mac_address[2],
mac_address[3], mac_address[4], mac_address[5]);
chip->mac_address[0], chip->mac_address[1], chip->mac_address[2],
chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
err = lx_init_get_version_features(chip);
if (err)
@ -1031,7 +1030,7 @@ static int __devinit snd_lx6464es_create(struct snd_card *card,
chip->port_dsp_bar = pci_ioremap_bar(pci, 2);
err = request_irq(pci->irq, lx_interrupt, IRQF_SHARED,
card_name, chip);
KBUILD_MODNAME, chip);
if (err) {
snd_printk(KERN_ERR LXP "unable to grab IRQ %d\n", pci->irq);
goto request_irq_failed;
@ -1108,8 +1107,14 @@ static int __devinit snd_lx6464es_probe(struct pci_dev *pci,
goto out_free;
}
strcpy(card->driver, "lx6464es");
strcpy(card->shortname, "Digigram LX6464ES");
strcpy(card->driver, "LX6464ES");
sprintf(card->id, "LX6464ES_%02X%02X%02X",
chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
sprintf(card->shortname, "LX6464ES %02X.%02X.%02X.%02X.%02X.%02X",
chip->mac_address[0], chip->mac_address[1], chip->mac_address[2],
chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]);
sprintf(card->longname, "%s at 0x%lx, 0x%p, irq %i",
card->shortname, chip->port_plx,
chip->port_dsp_bar, chip->irq);
@ -1137,7 +1142,7 @@ static void __devexit snd_lx6464es_remove(struct pci_dev *pci)
static struct pci_driver driver = {
.name = "Digigram LX6464ES",
.name = KBUILD_MODNAME,
.id_table = snd_lx6464es_ids,
.probe = snd_lx6464es_probe,
.remove = __devexit_p(snd_lx6464es_remove),

2
sound/pci/lx6464es/lx6464es.h
View File

@ -69,6 +69,8 @@ struct lx6464es {
struct pci_dev *pci;
int irq;
u8 mac_address[6];
spinlock_t lock; /* interrupt spinlock */
struct mutex setup_mutex; /* mutex used in hw_params, open
* and close */

Some files were not shown because too many files have changed in this diff Show More