diff --git a/[refs] b/[refs]
index 6ca4d0047be5..4d1d62538980 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
---
-refs/heads/master: 51e4152a969aa6d2306492ebf143932dcb535c9b
+refs/heads/master: b272cc769ac22014c0c60f2ebac46a2ae01300bf
diff --git a/trunk/Documentation/DocBook/writing-an-alsa-driver.tmpl b/trunk/Documentation/DocBook/writing-an-alsa-driver.tmpl
index 5de23c007078..598c22f3b3ac 100644
--- a/trunk/Documentation/DocBook/writing-an-alsa-driver.tmpl
+++ b/trunk/Documentation/DocBook/writing-an-alsa-driver.tmpl
@@ -4288,7 +4288,7 @@ struct _snd_pcm_runtime {
@@ -4343,13 +4343,6 @@ struct _snd_pcm_runtime {
by itself to start processing the output stream in the irq handler.
-
- If the MPU-401 interface shares its interrupt with the other logical
- devices on the card, set MPU401_INFO_IRQ_HOOK
- (see
- below).
-
-
Usually, the port address corresponds to the command port and
port + 1 corresponds to the data port. If not, you may change
@@ -4382,12 +4375,14 @@ struct _snd_pcm_runtime {
- The 6th argument specifies the ISA irq number that will be
- allocated. If no interrupt is to be allocated (because your
- code is already allocating a shared interrupt, or because the
- device does not use interrupts), pass -1 instead.
- For a MPU-401 device without an interrupt, a polling timer
- will be used instead.
+ The 6th argument specifies the irq number for UART. If the irq
+ is already allocated, pass 0 to the 7th argument
+ (irq_flags). Otherwise, pass the flags
+ for irq allocation
+ (SA_XXX bits) to it, and the irq will be
+ reserved by the mpu401-uart layer. If the card doesn't generate
+ UART interrupts, pass -1 as the irq number. Then a timer
+ interrupt will be invoked for polling.
@@ -4395,13 +4390,12 @@ struct _snd_pcm_runtime {
Interrupt Handler
When the interrupt is allocated in
- snd_mpu401_uart_new(), an exclusive ISA
- interrupt handler is automatically used, hence you don't have
- anything else to do than creating the mpu401 stuff. Otherwise, you
- have to set MPU401_INFO_IRQ_HOOK, and call
- snd_mpu401_uart_interrupt() explicitly from your
- own interrupt handler when it has determined that a UART interrupt
- has occurred.
+ snd_mpu401_uart_new(), the private
+ interrupt handler is used, hence you don't have anything else to do
+ than creating the mpu401 stuff. Otherwise, you have to call
+ snd_mpu401_uart_interrupt() explicitly when
+ a UART interrupt is invoked and checked in your own interrupt
+ handler.
diff --git a/trunk/Documentation/sound/alsa/ALSA-Configuration.txt b/trunk/Documentation/sound/alsa/ALSA-Configuration.txt
index 936699e4f04b..89757012c7ff 100644
--- a/trunk/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/trunk/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -886,12 +886,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
disable)
power_save_controller - Reset HD-audio controller in power-saving mode
(default = on)
- align_buffer_size - Force rounding of buffer/period sizes to multiples
- of 128 bytes. This is more efficient in terms of memory
- access but isn't required by the HDA spec and prevents
- users from specifying exact period/buffer sizes.
- (default = on)
- snoop - Enable/disable snooping (default = on)
This module supports multiple cards and autoprobe.
diff --git a/trunk/Documentation/sound/alsa/HD-Audio-Controls.txt b/trunk/Documentation/sound/alsa/HD-Audio-Controls.txt
index e9621e349e17..1482035243e6 100644
--- a/trunk/Documentation/sound/alsa/HD-Audio-Controls.txt
+++ b/trunk/Documentation/sound/alsa/HD-Audio-Controls.txt
@@ -98,19 +98,3 @@ Conexant codecs
* Auto-Mute Mode
See Reatek codecs.
-
-
-Analog 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" and "6ch".
- According to the configuration, this also controls the
- jack-retasking of multi-I/O 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.
diff --git a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
index 4f3443230d89..d70c93bdcadf 100644
--- a/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/trunk/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -29,6 +29,9 @@ ALC880
ALC260
======
+ hp HP machines
+ hp-3013 HP machines (3013-variant)
+ hp-dc7600 HP DC7600
fujitsu Fujitsu S7020
acer Acer TravelMate
will Will laptops (PB V7900)
@@ -43,10 +46,15 @@ ALC260
ALC262
======
fujitsu Fujitsu Laptop
+ hp-bpc HP xw4400/6400/8400/9400 laptops
+ hp-bpc-d7000 HP BPC D7000
+ hp-tc-t5735 HP Thin Client T5735
+ hp-rp5700 HP RP5700
benq Benq ED8
benq-t31 Benq T31
hippo Hippo (ATI) with jack detection, Sony UX-90s
hippo_1 Hippo (Benq) with jack detection
+ sony-assamd Sony ASSAMD
toshiba-s06 Toshiba S06
toshiba-rx1 Toshiba RX1
tyan Tyan Thunder n6650W (S2915-E)
@@ -58,15 +66,43 @@ ALC262
ALC267/268
==========
- N/A
+ quanta-il1 Quanta IL1 mini-notebook
+ 3stack 3-stack model
+ toshiba Toshiba A205
+ acer Acer laptops
+ acer-dmic Acer laptops with digital-mic
+ acer-aspire Acer Aspire One
+ dell Dell OEM laptops (Vostro 1200)
+ zepto Zepto laptops
+ test for testing/debugging purpose, almost all controls can
+ adjusted. Appearing only when compiled with
+ $CONFIG_SND_DEBUG=y
+ auto auto-config reading BIOS (default)
ALC269
======
+ basic Basic preset
+ quanta Quanta FL1
laptop-amic Laptops with analog-mic input
laptop-dmic Laptops with digital-mic input
+ fujitsu FSC Amilo
+ lifebook Fujitsu Lifebook S6420
+ auto auto-config reading BIOS (default)
ALC662/663/272
==============
+ 3stack-dig 3-stack (2-channel) with SPDIF
+ 3stack-6ch 3-stack (6-channel)
+ 3stack-6ch-dig 3-stack (6-channel) with SPDIF
+ 5stack-dig 5-stack with SPDIF
+ lenovo-101e Lenovo laptop
+ eeepc-p701 ASUS Eeepc P701
+ eeepc-ep20 ASUS Eeepc EP20
+ ecs ECS/Foxconn mobo
+ m51va ASUS M51VA
+ g71v ASUS G71V
+ h13 ASUS H13
+ g50v ASUS G50V
asus-mode1 ASUS
asus-mode2 ASUS
asus-mode3 ASUS
@@ -75,10 +111,15 @@ ALC662/663/272
asus-mode6 ASUS
asus-mode7 ASUS
asus-mode8 ASUS
+ dell Dell with ALC272
+ dell-zm1 Dell ZM1 with ALC272
+ samsung-nc10 Samsung NC10 mini notebook
+ auto auto-config reading BIOS (default)
ALC680
======
- N/A
+ base Base model (ASUS NX90)
+ auto auto-config reading BIOS (default)
ALC882/883/885/888/889
======================
@@ -134,11 +175,28 @@ ALC882/883/885/888/889
ALC861/660
==========
- N/A
+ 3stack 3-jack
+ 3stack-dig 3-jack with SPDIF I/O
+ 6stack-dig 6-jack with SPDIF I/O
+ 3stack-660 3-jack (for ALC660)
+ uniwill-m31 Uniwill M31 laptop
+ toshiba Toshiba laptop support
+ asus Asus laptop support
+ asus-laptop ASUS F2/F3 laptops
+ auto auto-config reading BIOS (default)
ALC861VD/660VD
==============
- N/A
+ 3stack 3-jack
+ 3stack-dig 3-jack with SPDIF OUT
+ 6stack-dig 6-jack with SPDIF OUT
+ 3stack-660 3-jack (for ALC660VD)
+ 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
+ lenovo Lenovo 3000 C200
+ dallas Dallas laptops
+ hp HP TX1000
+ asus-v1s ASUS V1Sn
+ auto auto-config reading BIOS (default)
CMI9880
=======
@@ -231,6 +289,7 @@ Conexant 5051
hp-dv6736 HP dv6736
hp-f700 HP Compaq Presario F700
ideapad Lenovo IdeaPad laptop
+ lenovo-x200 Lenovo X200 laptop
toshiba Toshiba Satellite M300
Conexant 5066
diff --git a/trunk/Documentation/sound/alsa/HD-Audio.txt b/trunk/Documentation/sound/alsa/HD-Audio.txt
index 03e2771ddeef..c82beb007634 100644
--- a/trunk/Documentation/sound/alsa/HD-Audio.txt
+++ b/trunk/Documentation/sound/alsa/HD-Audio.txt
@@ -447,10 +447,7 @@ The file needs to have a line `[codec]`. The next line should contain
three numbers indicating the codec vendor-id (0x12345678 in the
example), the codec subsystem-id (0xabcd1234) and the address (2) of
the codec. The rest patch entries are applied to this specified codec
-until another codec entry is given. Passing 0 or a negative number to
-the first or the second value will make the check of the corresponding
-field be skipped. It'll be useful for really broken devices that don't
-initialize SSID properly.
+until another codec entry is given.
The `[model]` line allows to change the model name of the each codec.
In the example above, it will be changed to model=auto.
@@ -494,7 +491,7 @@ Also, the codec chip name can be rewritten via `[chip_name]` line.
The hd-audio driver reads the file via request_firmware(). Thus,
a patch file has to be located on the appropriate firmware path,
typically, /lib/firmware. For example, when you pass the option
-`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be
+`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be
present.
The patch module option is specific to each card instance, and you
@@ -527,54 +524,6 @@ power-saving. See /sys/module/snd_hda_intel/parameters/power_save to
check the current value. If it's non-zero, the feature is turned on.
-Tracepoints
-~~~~~~~~~~~
-The hd-audio driver gives a few basic tracepoints.
-`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response`
-traces the response from RIRB (only when read from the codec driver).
-`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc,
-`hda:hda_unsol_event` traces the unsolicited events, and
-`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up
-via power-saving behavior.
-
-Enabling all tracepoints can be done like
-------------------------------------------------------------------------
- # echo 1 > /sys/kernel/debug/tracing/events/hda/enable
-------------------------------------------------------------------------
-then after some commands, you can traces from
-/sys/kernel/debug/tracing/trace file. For example, when you want to
-trace what codec command is sent, enable the tracepoint like:
-------------------------------------------------------------------------
- # cat /sys/kernel/debug/tracing/trace
- # tracer: nop
- #
- # TASK-PID CPU# TIMESTAMP FUNCTION
- # | | | | |
- <...>-7807 [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019
- <...>-7807 [002] 105147.774893: hda_send_cmd: [0:0] val=e39019
- <...>-7807 [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a
- <...>-7807 [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a
- <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019
- <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019
- <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a
- <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a
-------------------------------------------------------------------------
-Here `[0:0]` indicates the card number and the codec address, and
-`val` shows the value sent to the codec, respectively. The value is
-a packed value, and you can decode it via hda-decode-verb program
-included in hda-emu package below. For example, the value e3a019 is
-to set the left output-amp value to 25.
-------------------------------------------------------------------------
- % hda-decode-verb 0xe3a019
- raw value = 0x00e3a019
- cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19
- raw value: verb = 0x3a0, parm = 0x19
- verbname = set_amp_gain_mute
- amp raw val = 0xa019
- output, left, idx=0, mute=0, val=25
-------------------------------------------------------------------------
-
-
Development Tree
~~~~~~~~~~~~~~~~
The latest development codes for HD-audio are found on sound git tree:
diff --git a/trunk/MAINTAINERS b/trunk/MAINTAINERS
index 1dbbb278d218..bbf42cd74e2a 100644
--- a/trunk/MAINTAINERS
+++ b/trunk/MAINTAINERS
@@ -5991,7 +5991,7 @@ M: Jaroslav Kysela
M: Takashi Iwai
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
W: http://www.alsa-project.org/
-T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
T: git git://git.alsa-project.org/alsa-kernel.git
S: Maintained
F: Documentation/sound/
diff --git a/trunk/include/linux/input.h b/trunk/include/linux/input.h
index a514fb8faea3..a637e7814334 100644
--- a/trunk/include/linux/input.h
+++ b/trunk/include/linux/input.h
@@ -814,7 +814,6 @@ struct input_keymap_entry {
#define SW_KEYPAD_SLIDE 0x0a /* set = keypad slide out */
#define SW_FRONT_PROXIMITY 0x0b /* set = front proximity sensor active */
#define SW_ROTATE_LOCK 0x0c /* set = rotate locked/disabled */
-#define SW_LINEIN_INSERT 0x0d /* set = inserted */
#define SW_MAX 0x0f
#define SW_CNT (SW_MAX+1)
diff --git a/trunk/include/linux/usb/ch9.h b/trunk/include/linux/usb/ch9.h
index f30253599501..0fd3fbdd8283 100644
--- a/trunk/include/linux/usb/ch9.h
+++ b/trunk/include/linux/usb/ch9.h
@@ -377,23 +377,18 @@ struct usb_endpoint_descriptor {
#define USB_ENDPOINT_NUMBER_MASK 0x0f /* in bEndpointAddress */
#define USB_ENDPOINT_DIR_MASK 0x80
-#define USB_ENDPOINT_XFERTYPE_MASK 0x03 /* in bmAttributes */
-#define USB_ENDPOINT_XFER_CONTROL 0
-#define USB_ENDPOINT_XFER_ISOC 1
-#define USB_ENDPOINT_XFER_BULK 2
-#define USB_ENDPOINT_XFER_INT 3
-#define USB_ENDPOINT_MAX_ADJUSTABLE 0x80
-
#define USB_ENDPOINT_SYNCTYPE 0x0c
#define USB_ENDPOINT_SYNC_NONE (0 << 2)
#define USB_ENDPOINT_SYNC_ASYNC (1 << 2)
#define USB_ENDPOINT_SYNC_ADAPTIVE (2 << 2)
#define USB_ENDPOINT_SYNC_SYNC (3 << 2)
-#define USB_ENDPOINT_USAGE_MASK 0x30
-#define USB_ENDPOINT_USAGE_DATA 0x00
-#define USB_ENDPOINT_USAGE_FEEDBACK 0x10
-#define USB_ENDPOINT_USAGE_IMPLICIT_FB 0x20 /* Implicit feedback Data endpoint */
+#define USB_ENDPOINT_XFERTYPE_MASK 0x03 /* in bmAttributes */
+#define USB_ENDPOINT_XFER_CONTROL 0
+#define USB_ENDPOINT_XFER_ISOC 1
+#define USB_ENDPOINT_XFER_BULK 2
+#define USB_ENDPOINT_XFER_INT 3
+#define USB_ENDPOINT_MAX_ADJUSTABLE 0x80
/*-------------------------------------------------------------------------*/
diff --git a/trunk/include/sound/asound.h b/trunk/include/sound/asound.h
index a2e4ff5ba9e9..5d6074faa279 100644
--- a/trunk/include/sound/asound.h
+++ b/trunk/include/sound/asound.h
@@ -706,7 +706,7 @@ struct snd_timer_tread {
* *
****************************************************************************/
-#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 7)
+#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6)
struct snd_ctl_card_info {
int card; /* card number */
@@ -803,8 +803,6 @@ struct snd_ctl_elem_info {
unsigned int items; /* R: number of items */
unsigned int item; /* W: item number */
char name[64]; /* R: value name */
- __u64 names_ptr; /* W: names list (ELEM_ADD only) */
- unsigned int names_length;
} enumerated;
unsigned char reserved[128];
} value;
diff --git a/trunk/include/sound/initval.h b/trunk/include/sound/initval.h
index f99a0d2ddfe7..1daa6dff8297 100644
--- a/trunk/include/sound/initval.h
+++ b/trunk/include/sound/initval.h
@@ -62,7 +62,7 @@ static int snd_legacy_find_free_irq(int *irq_table)
{
while (*irq_table != -1) {
if (!request_irq(*irq_table, snd_legacy_empty_irq_handler,
- IRQF_PROBE_SHARED, "ALSA Test IRQ",
+ IRQF_DISABLED | IRQF_PROBE_SHARED, "ALSA Test IRQ",
(void *) irq_table)) {
free_irq(*irq_table, (void *) irq_table);
return *irq_table;
diff --git a/trunk/include/sound/jack.h b/trunk/include/sound/jack.h
index 63c790742db4..c140fc7cbd3f 100644
--- a/trunk/include/sound/jack.h
+++ b/trunk/include/sound/jack.h
@@ -42,7 +42,6 @@ enum snd_jack_types {
SND_JACK_MECHANICAL = 0x0008, /* If detected separately */
SND_JACK_VIDEOOUT = 0x0010,
SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT,
- SND_JACK_LINEIN = 0x0020,
/* Kept separate from switches to facilitate implementation */
SND_JACK_BTN_0 = 0x4000,
diff --git a/trunk/include/sound/mpu401.h b/trunk/include/sound/mpu401.h
index 20230db00ef1..1f1d53f8830b 100644
--- a/trunk/include/sound/mpu401.h
+++ b/trunk/include/sound/mpu401.h
@@ -50,10 +50,7 @@
#define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */
#define MPU401_INFO_MMIO (1 << 3) /* MMIO access */
#define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */
-#define MPU401_INFO_IRQ_HOOK (1 << 5) /* mpu401 irq handler is called
- from driver irq handler */
#define MPU401_INFO_NO_ACK (1 << 6) /* No ACK cmd needed */
-#define MPU401_INFO_USE_TIMER (1 << 15) /* internal */
#define MPU401_MODE_BIT_INPUT 0
#define MPU401_MODE_BIT_OUTPUT 1
@@ -76,7 +73,8 @@ struct snd_mpu401 {
unsigned long port; /* base port of MPU-401 chip */
unsigned long cport; /* port + 1 (usually) */
struct resource *res; /* port resource */
- int irq; /* IRQ number of MPU-401 chip */
+ int irq; /* IRQ number of MPU-401 chip (-1 = poll) */
+ int irq_flags;
unsigned long mode; /* MPU401_MODE_XXXX */
int timer_invoked;
@@ -133,6 +131,7 @@ int snd_mpu401_uart_new(struct snd_card *card,
unsigned long port,
unsigned int info_flags,
int irq,
+ int irq_flags,
struct snd_rawmidi ** rrawmidi);
#endif /* __SOUND_MPU401_H */
diff --git a/trunk/include/sound/pcm.h b/trunk/include/sound/pcm.h
index 3e7fda6e8164..57e71fa33f7c 100644
--- a/trunk/include/sound/pcm.h
+++ b/trunk/include/sound/pcm.h
@@ -825,8 +825,6 @@ int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime,
int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var);
-int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime,
- unsigned int base_rate);
int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime,
unsigned int cond,
int var,
@@ -1037,8 +1035,6 @@ static inline void snd_pcm_mmap_data_close(struct vm_area_struct *area)
atomic_dec(&substream->mmap_count);
}
-int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *area);
/* mmap for io-memory area */
#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA)
#define SNDRV_PCM_INFO_MMAP_IOMEM SNDRV_PCM_INFO_MMAP
diff --git a/trunk/sound/aoa/codecs/onyx.c b/trunk/sound/aoa/codecs/onyx.c
index 762af68c8996..3687a6cc9881 100644
--- a/trunk/sound/aoa/codecs/onyx.c
+++ b/trunk/sound/aoa/codecs/onyx.c
@@ -1067,6 +1067,7 @@ static int onyx_i2c_probe(struct i2c_client *client,
printk(KERN_DEBUG PFX "created and attached onyx instance\n");
return 0;
fail:
+ i2c_set_clientdata(client, NULL);
kfree(onyx);
return -ENODEV;
}
@@ -1111,7 +1112,8 @@ static int onyx_i2c_remove(struct i2c_client *client)
aoa_codec_unregister(&onyx->codec);
of_node_put(onyx->codec.node);
- kfree(onyx->codec_info);
+ if (onyx->codec_info)
+ kfree(onyx->codec_info);
kfree(onyx);
return 0;
}
diff --git a/trunk/sound/aoa/fabrics/layout.c b/trunk/sound/aoa/fabrics/layout.c
index 552b97afbca5..3fd1a7e24928 100644
--- a/trunk/sound/aoa/fabrics/layout.c
+++ b/trunk/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
sdev->pcmid = -1;
list_del(&ldev->list);
layouts_list_items--;
- kfree(ldev);
outnodev:
of_node_put(sound);
layout_device = NULL;
+ kfree(ldev);
return -ENODEV;
}
diff --git a/trunk/sound/arm/aaci.c b/trunk/sound/arm/aaci.c
index e518d38b1c74..d0cead38d5fb 100644
--- a/trunk/sound/arm/aaci.c
+++ b/trunk/sound/arm/aaci.c
@@ -443,7 +443,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream)
mutex_lock(&aaci->irq_lock);
if (!aaci->users++) {
ret = request_irq(aaci->dev->irq[0], aaci_irq,
- IRQF_SHARED, DRIVER_NAME, aaci);
+ IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci);
if (ret != 0)
aaci->users--;
}
diff --git a/trunk/sound/arm/pxa2xx-ac97-lib.c b/trunk/sound/arm/pxa2xx-ac97-lib.c
index 8ad65352bf91..88eec3847df2 100644
--- a/trunk/sound/arm/pxa2xx-ac97-lib.c
+++ b/trunk/sound/arm/pxa2xx-ac97-lib.c
@@ -359,7 +359,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
if (ret)
goto err_clk2;
- ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL);
+ ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL);
if (ret < 0)
goto err_irq;
diff --git a/trunk/sound/core/control.c b/trunk/sound/core/control.c
index 978fe1a8e9f0..f8c5be464510 100644
--- a/trunk/sound/core/control.c
+++ b/trunk/sound/core/control.c
@@ -989,6 +989,7 @@ struct user_element {
void *tlv_data; /* TLV data */
unsigned long tlv_data_size; /* TLV data size */
void *priv_data; /* private data (like strings for enumerated type) */
+ unsigned long priv_data_size; /* size of private data in bytes */
};
static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol,
@@ -1000,28 +1001,6 @@ static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol,
return 0;
}
-static int snd_ctl_elem_user_enum_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct user_element *ue = kcontrol->private_data;
- const char *names;
- unsigned int item;
-
- item = uinfo->value.enumerated.item;
-
- *uinfo = ue->info;
-
- item = min(item, uinfo->value.enumerated.items - 1);
- uinfo->value.enumerated.item = item;
-
- names = ue->priv_data;
- for (; item > 0; --item)
- names += strlen(names) + 1;
- strcpy(uinfo->value.enumerated.name, names);
-
- return 0;
-}
-
static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1076,46 +1055,11 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol,
return change;
}
-static int snd_ctl_elem_init_enum_names(struct user_element *ue)
-{
- char *names, *p;
- size_t buf_len, name_len;
- unsigned int i;
-
- if (ue->info.value.enumerated.names_length > 64 * 1024)
- return -EINVAL;
-
- names = memdup_user(
- (const void __user *)ue->info.value.enumerated.names_ptr,
- ue->info.value.enumerated.names_length);
- if (IS_ERR(names))
- return PTR_ERR(names);
-
- /* check that there are enough valid names */
- buf_len = ue->info.value.enumerated.names_length;
- p = names;
- for (i = 0; i < ue->info.value.enumerated.items; ++i) {
- name_len = strnlen(p, buf_len);
- if (name_len == 0 || name_len >= 64 || name_len == buf_len) {
- kfree(names);
- return -EINVAL;
- }
- p += name_len + 1;
- buf_len -= name_len + 1;
- }
-
- ue->priv_data = names;
- ue->info.value.enumerated.names_ptr = 0;
-
- return 0;
-}
-
static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol)
{
struct user_element *ue = kcontrol->private_data;
-
- kfree(ue->tlv_data);
- kfree(ue->priv_data);
+ if (ue->tlv_data)
+ kfree(ue->tlv_data);
kfree(ue);
}
@@ -1128,8 +1072,8 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
long private_size;
struct user_element *ue;
int idx, err;
-
- if (!replace && card->user_ctl_count >= MAX_USER_CONTROLS)
+
+ if (card->user_ctl_count >= MAX_USER_CONTROLS)
return -ENOMEM;
if (info->count < 1)
return -EINVAL;
@@ -1157,10 +1101,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
memcpy(&kctl.id, &info->id, sizeof(info->id));
kctl.count = info->owner ? info->owner : 1;
access |= SNDRV_CTL_ELEM_ACCESS_USER;
- if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED)
- kctl.info = snd_ctl_elem_user_enum_info;
- else
- kctl.info = snd_ctl_elem_user_info;
+ kctl.info = snd_ctl_elem_user_info;
if (access & SNDRV_CTL_ELEM_ACCESS_READ)
kctl.get = snd_ctl_elem_user_get;
if (access & SNDRV_CTL_ELEM_ACCESS_WRITE)
@@ -1181,11 +1122,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
if (info->count > 64)
return -EINVAL;
break;
- case SNDRV_CTL_ELEM_TYPE_ENUMERATED:
- private_size = sizeof(unsigned int);
- if (info->count > 128 || info->value.enumerated.items == 0)
- return -EINVAL;
- break;
case SNDRV_CTL_ELEM_TYPE_BYTES:
private_size = sizeof(unsigned char);
if (info->count > 512)
@@ -1207,17 +1143,9 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
ue->info.access = 0;
ue->elem_data = (char *)ue + sizeof(*ue);
ue->elem_data_size = private_size;
- if (ue->info.type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) {
- err = snd_ctl_elem_init_enum_names(ue);
- if (err < 0) {
- kfree(ue);
- return err;
- }
- }
kctl.private_free = snd_ctl_elem_user_free;
_kctl = snd_ctl_new(&kctl, access);
if (_kctl == NULL) {
- kfree(ue->priv_data);
kfree(ue);
return -ENOMEM;
}
diff --git a/trunk/sound/core/control_compat.c b/trunk/sound/core/control_compat.c
index 2bb95a7a8809..426874429a5e 100644
--- a/trunk/sound/core/control_compat.c
+++ b/trunk/sound/core/control_compat.c
@@ -83,8 +83,6 @@ struct snd_ctl_elem_info32 {
u32 items;
u32 item;
char name[64];
- u64 names_ptr;
- u32 names_length;
} enumerated;
unsigned char reserved[128];
} value;
@@ -374,8 +372,6 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file,
&data32->value.enumerated,
sizeof(data->value.enumerated)))
goto error;
- data->value.enumerated.names_ptr =
- (uintptr_t)compat_ptr(data->value.enumerated.names_ptr);
break;
default:
break;
diff --git a/trunk/sound/core/hwdep.c b/trunk/sound/core/hwdep.c
index 031e215b6dde..a70ee7f1ed98 100644
--- a/trunk/sound/core/hwdep.c
+++ b/trunk/sound/core/hwdep.c
@@ -272,14 +272,7 @@ static int snd_hwdep_control_ioctl(struct snd_card *card,
if (get_user(device, (int __user *)arg))
return -EFAULT;
mutex_lock(®ister_mutex);
-
- if (device < 0)
- device = 0;
- else if (device < SNDRV_MINOR_HWDEPS)
- device++;
- else
- device = SNDRV_MINOR_HWDEPS;
-
+ device = device < 0 ? 0 : device + 1;
while (device < SNDRV_MINOR_HWDEPS) {
if (snd_hwdep_search(card, device))
break;
diff --git a/trunk/sound/core/jack.c b/trunk/sound/core/jack.c
index 240a3e13470d..53b53e97c896 100644
--- a/trunk/sound/core/jack.c
+++ b/trunk/sound/core/jack.c
@@ -30,7 +30,6 @@ static int jack_switch_types[] = {
SW_LINEOUT_INSERT,
SW_JACK_PHYSICAL_INSERT,
SW_VIDEOOUT_INSERT,
- SW_LINEIN_INSERT,
};
static int snd_jack_dev_free(struct snd_device *device)
diff --git a/trunk/sound/core/oss/mixer_oss.c b/trunk/sound/core/oss/mixer_oss.c
index 1b5e0c49a0ad..d8359cfeca15 100644
--- a/trunk/sound/core/oss/mixer_oss.c
+++ b/trunk/sound/core/oss/mixer_oss.c
@@ -499,7 +499,7 @@ static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, c
memset(&id, 0, sizeof(id));
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- strlcpy(id.name, name, sizeof(id.name));
+ strcpy(id.name, name);
id.index = index;
return snd_ctl_find_id(card, &id);
}
diff --git a/trunk/sound/core/pcm_lib.c b/trunk/sound/core/pcm_lib.c
index 95d1e789715f..86d0caf91b35 100644
--- a/trunk/sound/core/pcm_lib.c
+++ b/trunk/sound/core/pcm_lib.c
@@ -1399,32 +1399,6 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2);
-static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params,
- struct snd_pcm_hw_rule *rule)
-{
- unsigned int base_rate = (unsigned int)(uintptr_t)rule->private;
- struct snd_interval *rate;
-
- rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
- return snd_interval_list(rate, 1, &base_rate, 0);
-}
-
-/**
- * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling
- * @runtime: PCM runtime instance
- * @base_rate: the rate at which the hardware does not resample
- */
-int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime,
- unsigned int base_rate)
-{
- return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE,
- SNDRV_PCM_HW_PARAM_RATE,
- snd_pcm_hw_rule_noresample_func,
- (void *)(uintptr_t)base_rate,
- SNDRV_PCM_HW_PARAM_RATE, -1);
-}
-EXPORT_SYMBOL(snd_pcm_hw_rule_noresample);
-
static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var)
{
@@ -1787,10 +1761,6 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
snd_pcm_uframes_t avail = 0;
long wait_time, tout;
- init_waitqueue_entry(&wait, current);
- set_current_state(TASK_INTERRUPTIBLE);
- add_wait_queue(&runtime->tsleep, &wait);
-
if (runtime->no_period_wakeup)
wait_time = MAX_SCHEDULE_TIMEOUT;
else {
@@ -1801,32 +1771,16 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
}
wait_time = msecs_to_jiffies(wait_time * 1000);
}
-
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&runtime->tsleep, &wait);
for (;;) {
if (signal_pending(current)) {
err = -ERESTARTSYS;
break;
}
-
- /*
- * We need to check if space became available already
- * (and thus the wakeup happened already) first to close
- * the race of space already having become available.
- * This check must happen after been added to the waitqueue
- * and having current state be INTERRUPTIBLE.
- */
- if (is_playback)
- avail = snd_pcm_playback_avail(runtime);
- else
- avail = snd_pcm_capture_avail(runtime);
- if (avail >= runtime->twake)
- break;
snd_pcm_stream_unlock_irq(substream);
-
- tout = schedule_timeout(wait_time);
-
+ tout = schedule_timeout_interruptible(wait_time);
snd_pcm_stream_lock_irq(substream);
- set_current_state(TASK_INTERRUPTIBLE);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_SUSPENDED:
err = -ESTRPIPE;
@@ -1852,9 +1806,14 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
err = -EIO;
break;
}
+ if (is_playback)
+ avail = snd_pcm_playback_avail(runtime);
+ else
+ avail = snd_pcm_capture_avail(runtime);
+ if (avail >= runtime->twake)
+ break;
}
_endloop:
- set_current_state(TASK_RUNNING);
remove_wait_queue(&runtime->tsleep, &wait);
*availp = avail;
return err;
diff --git a/trunk/sound/core/pcm_native.c b/trunk/sound/core/pcm_native.c
index 77d7df22e7c8..1c6be91dfb98 100644
--- a/trunk/sound/core/pcm_native.c
+++ b/trunk/sound/core/pcm_native.c
@@ -2058,12 +2058,16 @@ EXPORT_SYMBOL(snd_pcm_open_substream);
static int snd_pcm_open_file(struct file *file,
struct snd_pcm *pcm,
- int stream)
+ int stream,
+ struct snd_pcm_file **rpcm_file)
{
struct snd_pcm_file *pcm_file;
struct snd_pcm_substream *substream;
int err;
+ if (rpcm_file)
+ *rpcm_file = NULL;
+
err = snd_pcm_open_substream(pcm, stream, file, &substream);
if (err < 0)
return err;
@@ -2079,7 +2083,8 @@ static int snd_pcm_open_file(struct file *file,
substream->pcm_release = pcm_release_private;
}
file->private_data = pcm_file;
-
+ if (rpcm_file)
+ *rpcm_file = pcm_file;
return 0;
}
@@ -2108,6 +2113,7 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file)
static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream)
{
int err;
+ struct snd_pcm_file *pcm_file;
wait_queue_t wait;
if (pcm == NULL) {
@@ -2125,7 +2131,7 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream)
add_wait_queue(&pcm->open_wait, &wait);
mutex_lock(&pcm->open_mutex);
while (1) {
- err = snd_pcm_open_file(file, pcm, stream);
+ err = snd_pcm_open_file(file, pcm, stream, &pcm_file);
if (err >= 0)
break;
if (err == -EAGAIN) {
@@ -3150,8 +3156,8 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = {
/*
* mmap the DMA buffer on RAM
*/
-int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *area)
+static int snd_pcm_default_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area)
{
area->vm_flags |= VM_RESERVED;
#ifdef ARCH_HAS_DMA_MMAP_COHERENT
@@ -3171,7 +3177,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
area->vm_ops = &snd_pcm_vm_ops_data_fault;
return 0;
}
-EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap);
/*
* mmap the DMA buffer on I/O memory area
@@ -3237,7 +3242,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file,
if (substream->ops->mmap)
err = substream->ops->mmap(substream, area);
else
- err = snd_pcm_lib_default_mmap(substream, area);
+ err = snd_pcm_default_mmap(substream, area);
if (!err)
atomic_inc(&substream->mmap_count);
return err;
diff --git a/trunk/sound/core/timer.c b/trunk/sound/core/timer.c
index 67ebf1c21c04..7c1cbf0a0dc4 100644
--- a/trunk/sound/core/timer.c
+++ b/trunk/sound/core/timer.c
@@ -328,8 +328,6 @@ int snd_timer_close(struct snd_timer_instance *timeri)
mutex_unlock(®ister_mutex);
} else {
timer = timeri->timer;
- if (snd_BUG_ON(!timer))
- goto out;
/* wait, until the active callback is finished */
spin_lock_irq(&timer->lock);
while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
@@ -355,7 +353,6 @@ int snd_timer_close(struct snd_timer_instance *timeri)
}
mutex_unlock(®ister_mutex);
}
- out:
if (timeri->private_free)
timeri->private_free(timeri);
kfree(timeri->owner);
@@ -534,8 +531,6 @@ int snd_timer_stop(struct snd_timer_instance *timeri)
if (err < 0)
return err;
timer = timeri->timer;
- if (!timer)
- return -EINVAL;
spin_lock_irqsave(&timer->lock, flags);
timeri->cticks = timeri->ticks;
timeri->pticks = 0;
diff --git a/trunk/sound/drivers/aloop.c b/trunk/sound/drivers/aloop.c
index 4067f1548949..a0da7755fcea 100644
--- a/trunk/sound/drivers/aloop.c
+++ b/trunk/sound/drivers/aloop.c
@@ -575,8 +575,7 @@ static void loopback_runtime_free(struct snd_pcm_runtime *runtime)
static int loopback_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- return snd_pcm_lib_alloc_vmalloc_buffer(substream,
- params_buffer_bytes(params));
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
}
static int loopback_hw_free(struct snd_pcm_substream *substream)
@@ -588,7 +587,7 @@ static int loopback_hw_free(struct snd_pcm_substream *substream)
mutex_lock(&dpcm->loopback->cable_lock);
cable->valid &= ~(1 << substream->stream);
mutex_unlock(&dpcm->loopback->cable_lock);
- return snd_pcm_lib_free_vmalloc_buffer(substream);
+ return snd_pcm_lib_free_pages(substream);
}
static unsigned int get_cable_index(struct snd_pcm_substream *substream)
@@ -741,8 +740,6 @@ static struct snd_pcm_ops loopback_playback_ops = {
.prepare = loopback_prepare,
.trigger = loopback_trigger,
.pointer = loopback_pointer,
- .page = snd_pcm_lib_get_vmalloc_page,
- .mmap = snd_pcm_lib_mmap_vmalloc,
};
static struct snd_pcm_ops loopback_capture_ops = {
@@ -754,8 +751,6 @@ static struct snd_pcm_ops loopback_capture_ops = {
.prepare = loopback_prepare,
.trigger = loopback_trigger,
.pointer = loopback_pointer,
- .page = snd_pcm_lib_get_vmalloc_page,
- .mmap = snd_pcm_lib_mmap_vmalloc,
};
static int __devinit loopback_pcm_new(struct loopback *loopback,
@@ -776,6 +771,10 @@ static int __devinit loopback_pcm_new(struct loopback *loopback,
strcpy(pcm->name, "Loopback PCM");
loopback->pcm[device] = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 0, 2 * 1024 * 1024);
return 0;
}
diff --git a/trunk/sound/drivers/ml403-ac97cr.c b/trunk/sound/drivers/ml403-ac97cr.c
index 2c7a7636f472..5cfcb908c430 100644
--- a/trunk/sound/drivers/ml403-ac97cr.c
+++ b/trunk/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
"0x%x done\n", (unsigned int)ml403_ac97cr->port);
/* get irq */
irq = platform_get_irq(pfdev, 0);
- if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
+ if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
"unable to grab IRQ %d\n",
@@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
"request (playback) irq %d done\n",
ml403_ac97cr->irq);
irq = platform_get_irq(pfdev, 1);
- if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
+ if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
"unable to grab IRQ %d\n",
diff --git a/trunk/sound/drivers/mpu401/mpu401.c b/trunk/sound/drivers/mpu401/mpu401.c
index 1c02852aceea..149d05a8202d 100644
--- a/trunk/sound/drivers/mpu401/mpu401.c
+++ b/trunk/sound/drivers/mpu401/mpu401.c
@@ -86,7 +86,8 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard)
}
err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0,
- irq[dev], NULL);
+ irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0,
+ NULL);
if (err < 0) {
printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]);
goto _err;
diff --git a/trunk/sound/drivers/mpu401/mpu401_uart.c b/trunk/sound/drivers/mpu401/mpu401_uart.c
index e91698a634b2..2af09996a3d0 100644
--- a/trunk/sound/drivers/mpu401/mpu401_uart.c
+++ b/trunk/sound/drivers/mpu401/mpu401_uart.c
@@ -3,7 +3,7 @@
* Routines for control of MPU-401 in UART mode
*
* MPU-401 supports UART mode which is not capable generate transmit
- * interrupts thus output is done via polling. Without interrupt,
+ * interrupts thus output is done via polling. Also, if irq < 0, then
* input is done also via polling. Do not expect good performance.
*
*
@@ -374,7 +374,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up)
/* first time - flush FIFO */
while (max-- > 0)
mpu->read(mpu, MPU401D(mpu));
- if (mpu->info_flags & MPU401_INFO_USE_TIMER)
+ if (mpu->irq < 0)
snd_mpu401_uart_add_timer(mpu, 1);
}
@@ -383,7 +383,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up)
snd_mpu401_uart_input_read(mpu);
spin_unlock_irqrestore(&mpu->input_lock, flags);
} else {
- if (mpu->info_flags & MPU401_INFO_USE_TIMER)
+ if (mpu->irq < 0)
snd_mpu401_uart_remove_timer(mpu, 1);
clear_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode);
}
@@ -496,7 +496,7 @@ static struct snd_rawmidi_ops snd_mpu401_uart_input =
static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi)
{
struct snd_mpu401 *mpu = rmidi->private_data;
- if (mpu->irq >= 0)
+ if (mpu->irq_flags && mpu->irq >= 0)
free_irq(mpu->irq, (void *) mpu);
release_and_free_resource(mpu->res);
kfree(mpu);
@@ -509,7 +509,8 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi)
* @hardware: the hardware type, MPU401_HW_XXXX
* @port: the base address of MPU401 port
* @info_flags: bitflags MPU401_INFO_XXX
- * @irq: the ISA irq number, -1 if not to be allocated
+ * @irq: the irq number, -1 if no interrupt for mpu
+ * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved.
* @rrawmidi: the pointer to store the new rawmidi instance
*
* Creates a new MPU-401 instance.
@@ -524,7 +525,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
unsigned short hardware,
unsigned long port,
unsigned int info_flags,
- int irq,
+ int irq, int irq_flags,
struct snd_rawmidi ** rrawmidi)
{
struct snd_mpu401 *mpu;
@@ -576,8 +577,8 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
mpu->cport = port + 2;
else
mpu->cport = port + 1;
- if (irq >= 0) {
- if (request_irq(irq, snd_mpu401_uart_interrupt, 0,
+ if (irq >= 0 && irq_flags) {
+ if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags,
"MPU401 UART", (void *) mpu)) {
snd_printk(KERN_ERR "mpu401_uart: "
"unable to grab IRQ %d\n", irq);
@@ -585,10 +586,9 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
return -EBUSY;
}
}
- if (irq < 0 && !(info_flags & MPU401_INFO_IRQ_HOOK))
- info_flags |= MPU401_INFO_USE_TIMER;
mpu->info_flags = info_flags;
mpu->irq = irq;
+ mpu->irq_flags = irq_flags;
if (card->shortname[0])
snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI",
card->shortname);
diff --git a/trunk/sound/drivers/mtpav.c b/trunk/sound/drivers/mtpav.c
index 1eef4ccebe4b..5c426df87678 100644
--- a/trunk/sound/drivers/mtpav.c
+++ b/trunk/sound/drivers/mtpav.c
@@ -589,7 +589,7 @@ static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard)
return -EBUSY;
}
mcard->port = port;
- if (request_irq(irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) {
+ if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) {
snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq);
return -EBUSY;
}
diff --git a/trunk/sound/drivers/serial-u16550.c b/trunk/sound/drivers/serial-u16550.c
index fc1d822802c3..a25fb7b1f441 100644
--- a/trunk/sound/drivers/serial-u16550.c
+++ b/trunk/sound/drivers/serial-u16550.c
@@ -816,7 +816,7 @@ static int __devinit snd_uart16550_create(struct snd_card *card,
if (irq >= 0 && irq != SNDRV_AUTO_IRQ) {
if (request_irq(irq, snd_uart16550_interrupt,
- 0, "Serial MIDI", uart)) {
+ IRQF_DISABLED, "Serial MIDI", uart)) {
snd_printk(KERN_WARNING
"irq %d busy. Using Polling.\n", irq);
} else {
diff --git a/trunk/sound/firewire/isight.c b/trunk/sound/firewire/isight.c
index cd094ecaca3b..440030818db7 100644
--- a/trunk/sound/firewire/isight.c
+++ b/trunk/sound/firewire/isight.c
@@ -51,6 +51,7 @@ struct isight {
struct fw_unit *unit;
struct fw_device *device;
u64 audio_base;
+ struct fw_address_handler iris_handler;
struct snd_pcm_substream *pcm;
struct mutex mutex;
struct iso_packets_buffer buffer;
diff --git a/trunk/sound/firewire/speakers.c b/trunk/sound/firewire/speakers.c
index cbe6bb9e53b6..3fc257da180c 100644
--- a/trunk/sound/firewire/speakers.c
+++ b/trunk/sound/firewire/speakers.c
@@ -778,10 +778,9 @@ static int __devexit fwspk_remove(struct device *dev)
{
struct fwspk *fwspk = dev_get_drvdata(dev);
+ mutex_lock(&fwspk->mutex);
amdtp_out_stream_pcm_abort(&fwspk->stream);
snd_card_disconnect(fwspk->card);
-
- mutex_lock(&fwspk->mutex);
fwspk_stop_stream(fwspk);
mutex_unlock(&fwspk->mutex);
@@ -797,8 +796,8 @@ static void fwspk_bus_reset(struct fw_unit *unit)
fcp_bus_reset(fwspk->unit);
if (cmp_connection_update(&fwspk->connection) < 0) {
- amdtp_out_stream_pcm_abort(&fwspk->stream);
mutex_lock(&fwspk->mutex);
+ amdtp_out_stream_pcm_abort(&fwspk->stream);
fwspk_stop_stream(fwspk);
mutex_unlock(&fwspk->mutex);
return;
diff --git a/trunk/sound/isa/ad1816a/ad1816a.c b/trunk/sound/isa/ad1816a/ad1816a.c
index a87a2b566e19..3cb75bc97699 100644
--- a/trunk/sound/isa/ad1816a/ad1816a.c
+++ b/trunk/sound/isa/ad1816a/ad1816a.c
@@ -204,7 +204,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard
if (mpu_port[dev] > 0) {
if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port[dev], 0, mpu_irq[dev],
+ mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED,
NULL) < 0)
printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]);
}
diff --git a/trunk/sound/isa/ad1816a/ad1816a_lib.c b/trunk/sound/isa/ad1816a/ad1816a_lib.c
index 177eed3271bc..05aef8b97e96 100644
--- a/trunk/sound/isa/ad1816a/ad1816a_lib.c
+++ b/trunk/sound/isa/ad1816a/ad1816a_lib.c
@@ -595,7 +595,7 @@ int __devinit snd_ad1816a_create(struct snd_card *card,
snd_ad1816a_free(chip);
return -EBUSY;
}
- if (request_irq(irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) {
+ if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) {
snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq);
snd_ad1816a_free(chip);
return -EBUSY;
diff --git a/trunk/sound/isa/als100.c b/trunk/sound/isa/als100.c
index 706effd6b3cd..20becc89f6f6 100644
--- a/trunk/sound/isa/als100.c
+++ b/trunk/sound/isa/als100.c
@@ -256,6 +256,7 @@ static int __devinit snd_card_als100_probe(int dev,
mpu_type,
mpu_port[dev], 0,
mpu_irq[dev],
+ mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0,
NULL) < 0)
snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
}
diff --git a/trunk/sound/isa/azt2320.c b/trunk/sound/isa/azt2320.c
index b7bdbf307740..aac8dc15c2fe 100644
--- a/trunk/sound/isa/azt2320.c
+++ b/trunk/sound/isa/azt2320.c
@@ -234,7 +234,8 @@ static int __devinit snd_card_azt2320_probe(int dev,
if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
if (snd_mpu401_uart_new(card, 0, MPU401_HW_AZT2320,
mpu_port[dev], 0,
- mpu_irq[dev], NULL) < 0)
+ mpu_irq[dev], IRQF_DISABLED,
+ NULL) < 0)
snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
}
diff --git a/trunk/sound/isa/cmi8330.c b/trunk/sound/isa/cmi8330.c
index dca69f80305f..fe79a169acb5 100644
--- a/trunk/sound/isa/cmi8330.c
+++ b/trunk/sound/isa/cmi8330.c
@@ -597,7 +597,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
if (mpuport[dev] != SNDRV_AUTO_PORT) {
if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
mpuport[dev], 0, mpuirq[dev],
- NULL) < 0)
+ IRQF_DISABLED, NULL) < 0)
printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n",
mpuport[dev]);
}
diff --git a/trunk/sound/isa/cs423x/cs4231.c b/trunk/sound/isa/cs423x/cs4231.c
index 409fa0ad7843..cb9153e75b82 100644
--- a/trunk/sound/isa/cs423x/cs4231.c
+++ b/trunk/sound/isa/cs423x/cs4231.c
@@ -131,6 +131,7 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n)
mpu_irq[n] = -1;
if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232,
mpu_port[n], 0, mpu_irq[n],
+ mpu_irq[n] >= 0 ? IRQF_DISABLED : 0,
NULL) < 0)
dev_warn(dev, "MPU401 not detected\n");
}
diff --git a/trunk/sound/isa/cs423x/cs4236.c b/trunk/sound/isa/cs423x/cs4236.c
index 0dbde461e6c1..999dc1e0fdbd 100644
--- a/trunk/sound/isa/cs423x/cs4236.c
+++ b/trunk/sound/isa/cs423x/cs4236.c
@@ -449,7 +449,8 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev)
mpu_irq[dev] = -1;
if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232,
mpu_port[dev], 0,
- mpu_irq[dev], NULL) < 0)
+ mpu_irq[dev],
+ mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0)
printk(KERN_WARNING IDENT ": MPU401 not detected\n");
}
diff --git a/trunk/sound/isa/es1688/es1688.c b/trunk/sound/isa/es1688/es1688.c
index 5493e9e4bcd5..0cde8131a575 100644
--- a/trunk/sound/isa/es1688/es1688.c
+++ b/trunk/sound/isa/es1688/es1688.c
@@ -174,7 +174,7 @@ static int __devinit snd_es1688_probe(struct snd_card *card, unsigned int n)
chip->mpu_port > 0) {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
chip->mpu_port, 0,
- mpu_irq[n], NULL);
+ mpu_irq[n], IRQF_DISABLED, NULL);
if (error < 0)
return error;
}
diff --git a/trunk/sound/isa/es1688/es1688_lib.c b/trunk/sound/isa/es1688/es1688_lib.c
index d3eab6fb0866..07676200496a 100644
--- a/trunk/sound/isa/es1688/es1688_lib.c
+++ b/trunk/sound/isa/es1688/es1688_lib.c
@@ -661,7 +661,7 @@ int snd_es1688_create(struct snd_card *card,
snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4);
return -EBUSY;
}
- if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) {
+ if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) {
snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq);
return -EBUSY;
}
diff --git a/trunk/sound/isa/es18xx.c b/trunk/sound/isa/es18xx.c
index bf6ad0bf51c6..fb4d6b34bbca 100644
--- a/trunk/sound/isa/es18xx.c
+++ b/trunk/sound/isa/es18xx.c
@@ -1805,7 +1805,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card,
return -EBUSY;
}
- if (request_irq(irq, snd_es18xx_interrupt, 0, "ES18xx",
+ if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx",
(void *) card)) {
snd_es18xx_free(card);
snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq);
@@ -2160,8 +2160,8 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev)
if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX,
- mpu_port[dev], MPU401_INFO_IRQ_HOOK,
- -1, &chip->rmidi);
+ mpu_port[dev], 0,
+ irq[dev], 0, &chip->rmidi);
if (err < 0)
return err;
}
diff --git a/trunk/sound/isa/galaxy/galaxy.c b/trunk/sound/isa/galaxy/galaxy.c
index e51d3244742a..ee54df082b9c 100644
--- a/trunk/sound/isa/galaxy/galaxy.c
+++ b/trunk/sound/isa/galaxy/galaxy.c
@@ -585,7 +585,8 @@ static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n)
if (mpu_port[n] >= 0) {
err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port[n], 0, mpu_irq[n], NULL);
+ mpu_port[n], 0, mpu_irq[n],
+ IRQF_DISABLED, NULL);
if (err < 0)
goto error;
}
diff --git a/trunk/sound/isa/gus/gus_main.c b/trunk/sound/isa/gus/gus_main.c
index 3167e5ac3699..12eb98f2f931 100644
--- a/trunk/sound/isa/gus/gus_main.c
+++ b/trunk/sound/isa/gus/gus_main.c
@@ -180,7 +180,7 @@ int snd_gus_create(struct snd_card *card,
snd_gus_free(gus);
return -EBUSY;
}
- if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) {
+ if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) {
snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq);
snd_gus_free(gus);
return -EBUSY;
diff --git a/trunk/sound/isa/gus/gusextreme.c b/trunk/sound/isa/gus/gusextreme.c
index c4733c08b60b..008e8e5bfa37 100644
--- a/trunk/sound/isa/gus/gusextreme.c
+++ b/trunk/sound/isa/gus/gusextreme.c
@@ -317,7 +317,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
if (es1688->mpu_port >= 0x300) {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
- es1688->mpu_port, 0, mpu_irq[n], NULL);
+ es1688->mpu_port, 0,
+ mpu_irq[n], IRQF_DISABLED, NULL);
if (error < 0)
goto out;
}
diff --git a/trunk/sound/isa/gus/gusmax.c b/trunk/sound/isa/gus/gusmax.c
index c43faa057ff6..3e4a58b72913 100644
--- a/trunk/sound/isa/gus/gusmax.c
+++ b/trunk/sound/isa/gus/gusmax.c
@@ -291,7 +291,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev)
goto _err;
}
- if (request_irq(xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) {
+ if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) {
snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
err = -EBUSY;
goto _err;
diff --git a/trunk/sound/isa/gus/interwave.c b/trunk/sound/isa/gus/interwave.c
index 5f869a32b48c..c7b80e4730fc 100644
--- a/trunk/sound/isa/gus/interwave.c
+++ b/trunk/sound/isa/gus/interwave.c
@@ -684,7 +684,7 @@ static int __devinit snd_interwave_probe(struct snd_card *card, int dev)
if ((err = snd_gus_initialize(gus)) < 0)
return err;
- if (request_irq(xirq, snd_interwave_interrupt, 0,
+ if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED,
"InterWave", iwcard)) {
snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
return -EBUSY;
diff --git a/trunk/sound/isa/msnd/msnd_pinnacle.c b/trunk/sound/isa/msnd/msnd_pinnacle.c
index 0961e2cf20ca..91d6023a63e5 100644
--- a/trunk/sound/isa/msnd/msnd_pinnacle.c
+++ b/trunk/sound/isa/msnd/msnd_pinnacle.c
@@ -600,7 +600,7 @@ static int __devinit snd_msnd_attach(struct snd_card *card)
mpu_io[0],
MPU401_MODE_INPUT |
MPU401_MODE_OUTPUT,
- mpu_irq[0],
+ mpu_irq[0], IRQF_DISABLED,
&chip->rmidi);
if (err < 0) {
printk(KERN_ERR LOGNAME
diff --git a/trunk/sound/isa/opl3sa2.c b/trunk/sound/isa/opl3sa2.c
index bbafb0b543ea..9b915e27b5bd 100644
--- a/trunk/sound/isa/opl3sa2.c
+++ b/trunk/sound/isa/opl3sa2.c
@@ -667,7 +667,7 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev)
err = snd_opl3sa2_detect(card);
if (err < 0)
return err;
- err = request_irq(xirq, snd_opl3sa2_interrupt, 0,
+ err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED,
"OPL3-SA2", card);
if (err) {
snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq);
@@ -707,9 +707,8 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev)
}
if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x340) {
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_OPL3SA2,
- midi_port[dev],
- MPU401_INFO_IRQ_HOOK, -1,
- &chip->rmidi)) < 0)
+ midi_port[dev], 0,
+ xirq, 0, &chip->rmidi)) < 0)
return err;
}
sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
diff --git a/trunk/sound/isa/opti9xx/miro.c b/trunk/sound/isa/opti9xx/miro.c
index d94d0f35cb76..8c24102d0d93 100644
--- a/trunk/sound/isa/opti9xx/miro.c
+++ b/trunk/sound/isa/opti9xx/miro.c
@@ -1377,7 +1377,8 @@ static int __devinit snd_miro_probe(struct snd_card *card)
rmidi = NULL;
else {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port, 0, miro->mpu_irq, &rmidi);
+ mpu_port, 0, miro->mpu_irq, IRQF_DISABLED,
+ &rmidi);
if (error < 0)
snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n",
mpu_port);
diff --git a/trunk/sound/isa/opti9xx/opti92x-ad1848.c b/trunk/sound/isa/opti9xx/opti92x-ad1848.c
index 6dbbfa76b440..c35dc68930dc 100644
--- a/trunk/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/trunk/sound/isa/opti9xx/opti92x-ad1848.c
@@ -892,7 +892,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
#endif
#ifdef OPTi93X
error = request_irq(irq, snd_opti93x_interrupt,
- 0, DEV_NAME" - WSS", chip);
+ IRQF_DISABLED, DEV_NAME" - WSS", chip);
if (error < 0) {
snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq);
return error;
@@ -914,7 +914,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
rmidi = NULL;
else {
error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- mpu_port, 0, mpu_irq, &rmidi);
+ mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi);
if (error)
snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n",
mpu_port);
diff --git a/trunk/sound/isa/sb/jazz16.c b/trunk/sound/isa/sb/jazz16.c
index 54e3c2c18060..8ccbcddf08e1 100644
--- a/trunk/sound/isa/sb/jazz16.c
+++ b/trunk/sound/isa/sb/jazz16.c
@@ -322,6 +322,7 @@ static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev)
MPU401_HW_MPU401,
mpu_port[dev], 0,
mpu_irq[dev],
+ mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0,
NULL) < 0)
snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n",
mpu_port[dev]);
diff --git a/trunk/sound/isa/sb/sb16.c b/trunk/sound/isa/sb/sb16.c
index 237f8bd7fbe4..4d1c5a300ff8 100644
--- a/trunk/sound/isa/sb/sb16.c
+++ b/trunk/sound/isa/sb/sb16.c
@@ -394,9 +394,8 @@ static int __devinit snd_sb16_probe(struct snd_card *card, int dev)
if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) {
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SB,
- chip->mpu_port,
- MPU401_INFO_IRQ_HOOK, -1,
- &chip->rmidi)) < 0)
+ chip->mpu_port, 0,
+ xirq, 0, &chip->rmidi)) < 0)
return err;
chip->rmidi_callback = snd_mpu401_uart_interrupt;
}
diff --git a/trunk/sound/isa/sb/sb_common.c b/trunk/sound/isa/sb/sb_common.c
index d2e19215813e..eae6c1c0eff9 100644
--- a/trunk/sound/isa/sb/sb_common.c
+++ b/trunk/sound/isa/sb/sb_common.c
@@ -240,7 +240,7 @@ int snd_sbdsp_create(struct snd_card *card,
if (request_irq(irq, irq_handler,
(hardware == SB_HW_ALS4000 ||
hardware == SB_HW_CS5530) ?
- IRQF_SHARED : 0,
+ IRQF_SHARED : IRQF_DISABLED,
"SoundBlaster", (void *) chip)) {
snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq);
snd_sbdsp_free(chip);
diff --git a/trunk/sound/isa/sc6000.c b/trunk/sound/isa/sc6000.c
index 207c161f100c..9a8bbf6dd62a 100644
--- a/trunk/sound/isa/sc6000.c
+++ b/trunk/sound/isa/sc6000.c
@@ -658,7 +658,8 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev)
if (snd_mpu401_uart_new(card, 0,
MPU401_HW_MPU401,
mpu_port[dev], 0,
- mpu_irq[dev], NULL) < 0)
+ mpu_irq[dev], IRQF_DISABLED,
+ NULL) < 0)
snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n",
mpu_port[dev]);
}
diff --git a/trunk/sound/isa/sscape.c b/trunk/sound/isa/sscape.c
index f2379e102b63..e2d5d2d3ed96 100644
--- a/trunk/sound/isa/sscape.c
+++ b/trunk/sound/isa/sscape.c
@@ -825,7 +825,8 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum,
int err;
err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port,
- MPU401_INFO_INTEGRATED, irq, &rawmidi);
+ MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED,
+ &rawmidi);
if (err == 0) {
struct snd_mpu401 *mpu = rawmidi->private_data;
mpu->open_input = mpu401_open;
diff --git a/trunk/sound/isa/wavefront/wavefront.c b/trunk/sound/isa/wavefront/wavefront.c
index 87142977335a..711670e4a425 100644
--- a/trunk/sound/isa/wavefront/wavefront.c
+++ b/trunk/sound/isa/wavefront/wavefront.c
@@ -418,7 +418,7 @@ snd_wavefront_probe (struct snd_card *card, int dev)
return -EBUSY;
}
if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt,
- 0, "ICS2115", acard)) {
+ IRQF_DISABLED, "ICS2115", acard)) {
snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]);
return -EBUSY;
}
@@ -449,7 +449,8 @@ snd_wavefront_probe (struct snd_card *card, int dev)
if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) {
err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232,
cs4232_mpu_port[dev], 0,
- cs4232_mpu_irq[dev], NULL);
+ cs4232_mpu_irq[dev], IRQF_DISABLED,
+ NULL);
if (err < 0) {
snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n");
return err;
diff --git a/trunk/sound/isa/wss/wss_lib.c b/trunk/sound/isa/wss/wss_lib.c
index 7277c5b7df6c..2a42cc377957 100644
--- a/trunk/sound/isa/wss/wss_lib.c
+++ b/trunk/sound/isa/wss/wss_lib.c
@@ -1833,7 +1833,7 @@ int snd_wss_create(struct snd_card *card,
}
chip->cport = cport;
if (!(hwshare & WSS_HWSHARE_IRQ))
- if (request_irq(irq, snd_wss_interrupt, 0,
+ if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED,
"WSS", (void *) chip)) {
snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq);
snd_wss_free(chip);
diff --git a/trunk/sound/mips/au1x00.c b/trunk/sound/mips/au1x00.c
index 7567ebd71913..446cf9748664 100644
--- a/trunk/sound/mips/au1x00.c
+++ b/trunk/sound/mips/au1x00.c
@@ -465,13 +465,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000)
flags = claim_dma_lock();
if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX,
- "AC97 TX", au1000_dma_interrupt, 0,
+ "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED,
au1000->stream[PLAYBACK])) < 0) {
release_dma_lock(flags);
return -EBUSY;
}
if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX,
- "AC97 RX", au1000_dma_interrupt, 0,
+ "AC97 RX", au1000_dma_interrupt, IRQF_DISABLED,
au1000->stream[CAPTURE])) < 0){
release_dma_lock(flags);
return -EBUSY;
diff --git a/trunk/sound/oss/pas2_pcm.c b/trunk/sound/oss/pas2_pcm.c
index 6f13ab4afc6b..8f7d175767a2 100644
--- a/trunk/sound/oss/pas2_pcm.c
+++ b/trunk/sound/oss/pas2_pcm.c
@@ -63,13 +63,13 @@ static int pcm_set_speed(int arg)
if (pcm_channels & 2)
{
- foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
- arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
+ foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
+ arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
}
else
{
- foo = (PIT_TICK_RATE + (arg / 2)) / arg;
- arg = (PIT_TICK_RATE + (foo / 2)) / foo;
+ foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
+ arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
}
pcm_speed = arg;
diff --git a/trunk/sound/oss/pss.c b/trunk/sound/oss/pss.c
index 2fc0624024b5..9b800ce5100e 100644
--- a/trunk/sound/oss/pss.c
+++ b/trunk/sound/oss/pss.c
@@ -673,8 +673,7 @@ static void configure_nonsound_components(void)
if (pss_cdrom_port == -1) { /* If cdrom port enablation wasn't requested */
printk(KERN_INFO "PSS: CDROM port not enabled.\n");
- } else if (!request_region(pss_cdrom_port, 2, "PSS CDROM")) {
- pss_cdrom_port = -1;
+ } else if (check_region(pss_cdrom_port, 2)) {
printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
} else {
set_io_base(devc, CONF_CDROM, pss_cdrom_port);
@@ -1233,8 +1232,7 @@ static void __exit cleanup_pss(void)
if(pssmpu)
unload_pss_mpu(&cfg_mpu);
unload_pss(&cfg);
- } else if (pss_cdrom_port != -1)
- release_region(pss_cdrom_port, 2);
+ }
if(!pss_keep_settings) /* Keep hardware settings if asked */
{
diff --git a/trunk/sound/oss/sound_timer.c b/trunk/sound/oss/sound_timer.c
index 8021c85f076d..48cda6c4c257 100644
--- a/trunk/sound/oss/sound_timer.c
+++ b/trunk/sound/oss/sound_timer.c
@@ -320,7 +320,7 @@ void sound_timer_init(struct sound_lowlev_timer *t, char *name)
n = sound_alloc_timerdev();
if (n == -1)
n = 0; /* Overwrite the system timer */
- strlcpy(sound_timer.info.name, name, sizeof(sound_timer.info.name));
+ strcpy(sound_timer.info.name, name);
sound_timer_devs[n] = &sound_timer;
}
EXPORT_SYMBOL(sound_timer_init);
diff --git a/trunk/sound/pci/Kconfig b/trunk/sound/pci/Kconfig
index 88168044375f..50abf5bf8e09 100644
--- a/trunk/sound/pci/Kconfig
+++ b/trunk/sound/pci/Kconfig
@@ -1,10 +1,5 @@
# ALSA PCI drivers
-config SND_TEA575X
- tristate
- depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
- default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
menuconfig SND_PCI
bool "PCI sound devices"
depends on PCI
@@ -568,6 +563,11 @@ config SND_FM801_TEA575X_BOOL
FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
SF64-PCR) into the snd-fm801 driver.
+config SND_TEA575X
+ tristate
+ depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+ default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
source "sound/pci/hda/Kconfig"
config SND_HDSP
diff --git a/trunk/sound/pci/ac97/ac97_patch.c b/trunk/sound/pci/ac97/ac97_patch.c
index a872d0a82976..200c9a1d48b7 100644
--- a/trunk/sound/pci/ac97/ac97_patch.c
+++ b/trunk/sound/pci/ac97/ac97_patch.c
@@ -1909,7 +1909,6 @@ static unsigned int ad1981_jacks_whitelist[] = {
0x103c0944, /* HP nc6220 */
0x103c0934, /* HP nc8220 */
0x103c006d, /* HP nx9105 */
- 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
0x17340088, /* FSC Scenic-W */
0 /* end */
};
diff --git a/trunk/sound/pci/als4000.c b/trunk/sound/pci/als4000.c
index 04628696eb08..a9c1af33f276 100644
--- a/trunk/sound/pci/als4000.c
+++ b/trunk/sound/pci/als4000.c
@@ -931,9 +931,8 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci,
if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000,
iobase + ALS4K_IOB_30_MIDI_DATA,
- MPU401_INFO_INTEGRATED |
- MPU401_INFO_IRQ_HOOK,
- -1, &chip->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED,
+ pci->irq, 0, &chip->rmidi)) < 0) {
printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n",
iobase + ALS4K_IOB_30_MIDI_DATA);
goto out_err;
diff --git a/trunk/sound/pci/asihpi/hpicmn.c b/trunk/sound/pci/asihpi/hpicmn.c
index bd47521b24ec..65b7ca13115b 100644
--- a/trunk/sound/pci/asihpi/hpicmn.c
+++ b/trunk/sound/pci/asihpi/hpicmn.c
@@ -631,12 +631,13 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
if (!p_cache)
return NULL;
- p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
- GFP_KERNEL);
+ p_cache->p_info =
+ kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
if (!p_cache->p_info) {
kfree(p_cache);
return NULL;
}
+ memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
p_cache->cache_size_in_bytes = size_in_bytes;
p_cache->control_count = control_count;
p_cache->p_cache = p_dsp_control_buffer;
diff --git a/trunk/sound/pci/au88x0/au88x0_mpu401.c b/trunk/sound/pci/au88x0/au88x0_mpu401.c
index e6c6a0febb75..0dc8d259d1ed 100644
--- a/trunk/sound/pci/au88x0/au88x0_mpu401.c
+++ b/trunk/sound/pci/au88x0/au88x0_mpu401.c
@@ -84,7 +84,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
#ifdef VORTEX_MPU401_LEGACY
if ((temp =
snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_MPU401, 0x330,
- MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) {
+ 0, 0, 0, &rmidi)) != 0) {
hwwrite(vortex->mmio, VORTEX_CTRL,
(hwread(vortex->mmio, VORTEX_CTRL) &
~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
@@ -94,8 +94,8 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA);
if ((temp =
snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port,
- MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO |
- MPU401_INFO_IRQ_HOOK, -1, &rmidi)) != 0) {
+ MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO,
+ 0, 0, &rmidi)) != 0) {
hwwrite(vortex->mmio, VORTEX_CTRL,
(hwread(vortex->mmio, VORTEX_CTRL) &
~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
diff --git a/trunk/sound/pci/azt3328.c b/trunk/sound/pci/azt3328.c
index d24fe425e87f..e4d76a270c9f 100644
--- a/trunk/sound/pci/azt3328.c
+++ b/trunk/sound/pci/azt3328.c
@@ -2625,19 +2625,16 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
int err;
snd_azf3328_dbgcallenter();
- if (dev >= SNDRV_CARDS) {
- err = -ENODEV;
- goto out;
- }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
if (!enable[dev]) {
dev++;
- err = -ENOENT;
- goto out;
+ return -ENOENT;
}
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
- goto out;
+ return err;
strcpy(card->driver, "AZF3328");
strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
@@ -2652,9 +2649,8 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
since our hardware ought to be similar, thus use same ID. */
err = snd_mpu401_uart_new(
card, 0,
- MPU401_HW_AZT2320, chip->mpu_io,
- MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
- -1, &chip->rmidi
+ MPU401_HW_AZT2320, chip->mpu_io, MPU401_INFO_INTEGRATED,
+ pci->irq, 0, &chip->rmidi
);
if (err < 0) {
snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n",
diff --git a/trunk/sound/pci/cmipci.c b/trunk/sound/pci/cmipci.c
index da9c73211eca..9cf99fb7eb9c 100644
--- a/trunk/sound/pci/cmipci.c
+++ b/trunk/sound/pci/cmipci.c
@@ -3228,9 +3228,8 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
iomidi,
(integrated_midi ?
- MPU401_INFO_INTEGRATED : 0) |
- MPU401_INFO_IRQ_HOOK,
- -1, &cm->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED : 0),
+ cm->irq, 0, &cm->rmidi)) < 0) {
printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi);
}
}
diff --git a/trunk/sound/pci/ctxfi/ctpcm.c b/trunk/sound/pci/ctxfi/ctpcm.c
index 2c8622617c8c..457d21189b0d 100644
--- a/trunk/sound/pci/ctxfi/ctpcm.c
+++ b/trunk/sound/pci/ctxfi/ctpcm.c
@@ -404,7 +404,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc,
int err;
int playback_count, capture_count;
- playback_count = (IEC958 == device) ? 1 : 256;
+ playback_count = (IEC958 == device) ? 1 : 8;
capture_count = (FRONT == device) ? 1 : 0;
err = snd_pcm_new(atc->card, "ctxfi", device,
playback_count, capture_count, &pcm);
diff --git a/trunk/sound/pci/ctxfi/ctsrc.c b/trunk/sound/pci/ctxfi/ctsrc.c
index e134b3a5780d..c749fa720889 100644
--- a/trunk/sound/pci/ctxfi/ctsrc.c
+++ b/trunk/sound/pci/ctxfi/ctsrc.c
@@ -20,7 +20,7 @@
#include "cthardware.h"
#include
-#define SRC_RESOURCE_NUM 256
+#define SRC_RESOURCE_NUM 64
#define SRCIMP_RESOURCE_NUM 256
static unsigned int conj_mask;
diff --git a/trunk/sound/pci/ctxfi/ctvmem.h b/trunk/sound/pci/ctxfi/ctvmem.h
index e6da60eb19ce..b23adfca4de6 100644
--- a/trunk/sound/pci/ctxfi/ctvmem.h
+++ b/trunk/sound/pci/ctxfi/ctvmem.h
@@ -18,7 +18,7 @@
#ifndef CTVMEM_H
#define CTVMEM_H
-#define CT_PTP_NUM 4 /* num of device page table pages */
+#define CT_PTP_NUM 1 /* num of device page table pages */
#include
#include
diff --git a/trunk/sound/pci/emu10k1/emupcm.c b/trunk/sound/pci/emu10k1/emupcm.c
index e22b8e2bbd88..622bace148e3 100644
--- a/trunk/sound/pci/emu10k1/emupcm.c
+++ b/trunk/sound/pci/emu10k1/emupcm.c
@@ -1146,11 +1146,6 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream)
kfree(epcm);
return err;
}
- err = snd_pcm_hw_rule_noresample(runtime, 48000);
- if (err < 0) {
- kfree(epcm);
- return err;
- }
mix = &emu->pcm_mixer[substream->number];
for (i = 0; i < 4; i++)
mix->send_routing[0][i] = mix->send_routing[1][i] = mix->send_routing[2][i] = i;
diff --git a/trunk/sound/pci/es1938.c b/trunk/sound/pci/es1938.c
index 718a2643474e..26a5a2f25d4b 100644
--- a/trunk/sound/pci/es1938.c
+++ b/trunk/sound/pci/es1938.c
@@ -1854,9 +1854,8 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci,
}
}
if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
- chip->mpu_port,
- MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
- -1, &chip->rmidi) < 0) {
+ chip->mpu_port, MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi) < 0) {
printk(KERN_ERR "es1938: unable to initialize MPU-401\n");
} else {
// this line is vital for MIDI interrupt handling on ess-solo1
diff --git a/trunk/sound/pci/es1968.c b/trunk/sound/pci/es1968.c
index 407e4abc4356..99ea9320c6b5 100644
--- a/trunk/sound/pci/es1968.c
+++ b/trunk/sound/pci/es1968.c
@@ -2843,9 +2843,8 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci,
if (enable_mpu[dev]) {
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
chip->io_port + ESM_MPU401_PORT,
- MPU401_INFO_INTEGRATED |
- MPU401_INFO_IRQ_HOOK,
- -1, &chip->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi)) < 0) {
printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n");
}
}
diff --git a/trunk/sound/pci/fm801.c b/trunk/sound/pci/fm801.c
index 136f7232bb7c..f9123f09e83e 100644
--- a/trunk/sound/pci/fm801.c
+++ b/trunk/sound/pci/fm801.c
@@ -68,7 +68,6 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
module_param_array(tea575x_tuner, int, NULL, 0444);
MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
-#define TUNER_DISABLED (1<<3)
#define TUNER_ONLY (1<<4)
#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF)
@@ -729,14 +728,11 @@ static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = {
{ .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" },
};
-#define get_tea575x_gpio(chip) \
- (&snd_fm801_tea575x_gpios[((chip)->tea575x_tuner & TUNER_TYPE_MASK) - 1])
-
static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins)
{
struct fm801 *chip = tea->private_data;
unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
- struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
+ struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
reg &= ~(FM801_GPIO_GP(gpio.data) |
FM801_GPIO_GP(gpio.clk) |
@@ -754,7 +750,7 @@ static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea)
{
struct fm801 *chip = tea->private_data;
unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
- struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
+ struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 |
(reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0;
@@ -764,7 +760,7 @@ static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output
{
struct fm801 *chip = tea->private_data;
unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL));
- struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip);
+ struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1];
/* use GPIO lines and set write enable bit */
reg |= FM801_GPIO_GS(gpio.data) |
@@ -1154,8 +1150,7 @@ static int snd_fm801_free(struct fm801 *chip)
__end_hw:
#ifdef CONFIG_SND_FM801_TEA575X_BOOL
- if (!(chip->tea575x_tuner & TUNER_DISABLED))
- snd_tea575x_exit(&chip->tea);
+ snd_tea575x_exit(&chip->tea);
#endif
if (chip->irq >= 0)
free_irq(chip->irq, chip);
@@ -1241,6 +1236,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
(tea575x_tuner & TUNER_TYPE_MASK) < 4) {
if (snd_tea575x_init(&chip->tea)) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
+ snd_fm801_free(chip);
return -ENODEV;
}
} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1249,19 +1245,17 @@ static int __devinit snd_fm801_create(struct snd_card *card,
chip->tea575x_tuner = tea575x_tuner;
if (!snd_tea575x_init(&chip->tea)) {
snd_printk(KERN_INFO "detected TEA575x radio type %s\n",
- get_tea575x_gpio(chip)->name);
+ snd_fm801_tea575x_gpios[tea575x_tuner - 1].name);
break;
}
}
if (tea575x_tuner == 4) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
- chip->tea575x_tuner = TUNER_DISABLED;
+ snd_fm801_free(chip);
+ return -ENODEV;
}
}
- if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
- strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name,
- sizeof(chip->tea.card));
- }
+ strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
#endif
*rchip = chip;
@@ -1312,9 +1306,8 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
}
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801,
FM801_REG(chip, MPU401_DATA),
- MPU401_INFO_INTEGRATED |
- MPU401_INFO_IRQ_HOOK,
- -1, &chip->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi)) < 0) {
snd_card_free(card);
return err;
}
diff --git a/trunk/sound/pci/hda/Makefile b/trunk/sound/pci/hda/Makefile
index f928d6634723..87365d5ea2a9 100644
--- a/trunk/sound/pci/hda/Makefile
+++ b/trunk/sound/pci/hda/Makefile
@@ -6,9 +6,6 @@ snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
-# for trace-points
-CFLAGS_hda_codec.o := -I$(src)
-
snd-hda-codec-realtek-objs := patch_realtek.o
snd-hda-codec-cmedia-objs := patch_cmedia.o
snd-hda-codec-analog-objs := patch_analog.o
diff --git a/trunk/sound/pci/hda/alc260_quirks.c b/trunk/sound/pci/hda/alc260_quirks.c
index 3b5170b9700f..21ec2cb100b0 100644
--- a/trunk/sound/pci/hda/alc260_quirks.c
+++ b/trunk/sound/pci/hda/alc260_quirks.c
@@ -7,6 +7,9 @@
enum {
ALC260_AUTO,
ALC260_BASIC,
+ ALC260_HP,
+ ALC260_HP_DC7600,
+ ALC260_HP_3013,
ALC260_FUJITSU_S702X,
ALC260_ACER,
ALC260_WILL,
@@ -139,6 +142,8 @@ static const struct hda_channel_mode alc260_modes[1] = {
/* Mixer combinations
*
* basic: base_output + input + pc_beep + capture
+ * HP: base_output + input + capture_alt
+ * HP_3013: hp_3013 + input + capture
* fujitsu: fujitsu + capture
* acer: acer + capture
*/
@@ -165,6 +170,145 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = {
{ } /* end */
};
+/* update HP, line and mono out pins according to the master switch */
+static void alc260_hp_master_update(struct hda_codec *codec)
+{
+ update_speakers(codec);
+}
+
+static int alc260_hp_master_sw_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;
+ *ucontrol->value.integer.value = !spec->master_mute;
+ return 0;
+}
+
+static int alc260_hp_master_sw_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 val = !*ucontrol->value.integer.value;
+
+ if (val == spec->master_mute)
+ return 0;
+ spec->master_mute = val;
+ alc260_hp_master_update(codec);
+ return 1;
+}
+
+static const struct snd_kcontrol_new alc260_hp_output_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc260_hp_unsol_verbs[] = {
+ {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x0f;
+ spec->autocfg.speaker_pins[0] = 0x10;
+ spec->autocfg.speaker_pins[1] = 0x11;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static const struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .subdevice = HDA_SUBDEV_NID_FLAG | 0x11,
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static void alc260_hp_3013_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x10;
+ spec->autocfg.speaker_pins[1] = 0x11;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
+static const struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc260_dc7600_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol),
+ HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc260_hp_3013_unsol_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_3012_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x10;
+ spec->autocfg.speaker_pins[0] = 0x0f;
+ spec->autocfg.speaker_pins[1] = 0x11;
+ spec->autocfg.speaker_pins[2] = 0x15;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+}
+
/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
* HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
*/
@@ -336,6 +480,106 @@ static const struct hda_verb alc260_init_verbs[] = {
{ }
};
+#if 0 /* should be identical with alc260_init_verbs? */
+static const struct hda_verb alc260_hp_init_verbs[] = {
+ /* Headphone and output */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ /* mono output */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* Line-2 pin widget for output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* unmute amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* unmute Line-Out mixer amp left and right (volume = 0) */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* unmute HP mixer amp left and right (volume = 0) */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* Unmute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ { }
+};
+#endif
+
+static const struct hda_verb alc260_hp_3013_init_verbs[] = {
+ /* Line out and output */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* mono output */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* Headphone pin widget for output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* unmute amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* unmute Line-Out mixer amp left and right (volume = 0) */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* unmute HP mixer amp left and right (volume = 0) */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* Unmute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ { }
+};
+
/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
* laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
* audio = 0x16, internal speaker = 0x10.
@@ -849,6 +1093,9 @@ static const struct hda_verb alc260_test_init_verbs[] = {
*/
static const char * const alc260_models[ALC260_MODEL_LAST] = {
[ALC260_BASIC] = "basic",
+ [ALC260_HP] = "hp",
+ [ALC260_HP_3013] = "hp-3013",
+ [ALC260_HP_DC7600] = "hp-dc7600",
[ALC260_FUJITSU_S702X] = "fujitsu",
[ALC260_ACER] = "acer",
[ALC260_WILL] = "will",
@@ -865,6 +1112,15 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
+ SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
+ SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
+ SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
@@ -888,6 +1144,54 @@ static const struct alc_config_preset alc260_presets[] = {
.channel_mode = alc260_modes,
.input_mux = &alc260_capture_source,
},
+ [ALC260_HP] = {
+ .mixers = { alc260_hp_output_mixer,
+ alc260_input_mixer },
+ .init_verbs = { alc260_init_verbs,
+ alc260_hp_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+ .adc_nids = alc260_adc_nids_alt,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc260_hp_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC260_HP_DC7600] = {
+ .mixers = { alc260_hp_dc7600_mixer,
+ alc260_input_mixer },
+ .init_verbs = { alc260_init_verbs,
+ alc260_hp_dc7600_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+ .adc_nids = alc260_adc_nids_alt,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc260_hp_3012_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC260_HP_3013] = {
+ .mixers = { alc260_hp_3013_mixer,
+ alc260_input_mixer },
+ .init_verbs = { alc260_hp_3013_init_verbs,
+ alc260_hp_3013_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt),
+ .adc_nids = alc260_adc_nids_alt,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc260_hp_3013_setup,
+ .init_hook = alc_inithook,
+ },
[ALC260_FUJITSU_S702X] = {
.mixers = { alc260_fujitsu_mixer },
.init_verbs = { alc260_fujitsu_init_verbs },
diff --git a/trunk/sound/pci/hda/alc262_quirks.c b/trunk/sound/pci/hda/alc262_quirks.c
index 7894b2b5aacf..8d2097d77642 100644
--- a/trunk/sound/pci/hda/alc262_quirks.c
+++ b/trunk/sound/pci/hda/alc262_quirks.c
@@ -10,7 +10,13 @@ enum {
ALC262_HIPPO,
ALC262_HIPPO_1,
ALC262_FUJITSU,
+ ALC262_HP_BPC,
+ ALC262_HP_BPC_D7000_WL,
+ ALC262_HP_BPC_D7000_WF,
+ ALC262_HP_TC_T5735,
+ ALC262_HP_RP5700,
ALC262_BENQ_ED8,
+ ALC262_SONY_ASSAMD,
ALC262_BENQ_T31,
ALC262_ULTRA,
ALC262_LENOVO_3000,
@@ -60,31 +66,164 @@ static const struct snd_kcontrol_new alc262_base_mixer[] = {
{ } /* end */
};
-/* bind hp and internal speaker mute (with plug check) as master switch */
+/* update HP, line and mono-out pins according to the master switch */
+#define alc262_hp_master_update alc260_hp_master_update
-static int alc262_hippo_master_sw_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static void alc262_hp_bpc_setup(struct hda_codec *codec)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- *ucontrol->value.integer.value = !spec->master_mute;
- return 0;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x16;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
}
-static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static void alc262_hp_wildwest_setup(struct hda_codec *codec)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- int val = !*ucontrol->value.integer.value;
- if (val == spec->master_mute)
- return 0;
- spec->master_mute = val;
- update_outputs(codec);
- return 1;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x16;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
}
+#define alc262_hp_master_sw_get alc260_hp_master_sw_get
+#define alc262_hp_master_sw_put alc260_hp_master_sw_put
+
+#define ALC262_HP_MASTER_SWITCH \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Master Playback Switch", \
+ .info = snd_ctl_boolean_mono_info, \
+ .get = alc262_hp_master_sw_get, \
+ .put = alc262_hp_master_sw_put, \
+ }, \
+ { \
+ .iface = NID_MAPPING, \
+ .name = "Master Playback Switch", \
+ .private_value = 0x15 | (0x16 << 8) | (0x1b << 16), \
+ }
+
+
+static const struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
+ ALC262_HP_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ 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("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_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),
+ HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
+ ALC262_HP_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x1a, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, 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 alc262_HP_BPC_WildWest_option_mixer[] = {
+ HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_hp_t5735_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_PIN;
+}
+
+static const struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, 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),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_hp_t5735_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static const struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_verb alc262_hp_rp5700_verbs[] = {
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {}
+};
+
+static const struct hda_input_mux alc262_hp_rp5700_capture_source = {
+ .num_items = 1,
+ .items = {
+ { "Line", 0x1 },
+ },
+};
+
+/* bind hp and internal speaker mute (with plug check) as master switch */
+#define alc262_hippo_master_update alc262_hp_master_update
+#define alc262_hippo_master_sw_get alc262_hp_master_sw_get
+#define alc262_hippo_master_sw_put alc262_hp_master_sw_put
+
#define ALC262_HIPPO_MASTER_SWITCH \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -100,9 +239,6 @@ static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol,
(SUBDEV_SPEAKER(0) << 16), \
}
-#define alc262_hp_master_sw_get alc262_hippo_master_sw_get
-#define alc262_hp_master_sw_put alc262_hippo_master_sw_put
-
static const struct snd_kcontrol_new alc262_hippo_mixer[] = {
ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -143,7 +279,8 @@ static void alc262_hippo_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc262_hippo1_setup(struct hda_codec *codec)
@@ -152,7 +289,8 @@ static void alc262_hippo1_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
@@ -215,7 +353,8 @@ static void alc262_tyan_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
@@ -357,7 +496,8 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec)
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x12;
spec->auto_mic = 1;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_PIN);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
}
/*
@@ -431,6 +571,27 @@ static const struct hda_input_mux alc262_fujitsu_capture_source = {
},
};
+static const struct hda_input_mux alc262_HP_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "AUX IN", 0x6 },
+ },
+};
+
+static const struct hda_input_mux alc262_HP_D7000_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x2 },
+ { "Line", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
static void alc262_fujitsu_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -438,7 +599,8 @@ static void alc262_fujitsu_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.hp_pins[1] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/* bind volumes of both NID 0x0c and 0x0d */
@@ -484,7 +646,8 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
@@ -589,8 +752,8 @@ static void alc262_ultra_automute(struct hda_codec *codec)
mute = 0;
/* auto-mute only when HP is used as HP */
if (!spec->cur_mux[0]) {
- spec->hp_jack_present = snd_hda_jack_detect(codec, 0x15);
- if (spec->hp_jack_present)
+ spec->jack_present = snd_hda_jack_detect(codec, 0x15);
+ if (spec->jack_present)
mute = HDA_AMP_MUTE;
}
/* mute/unmute internal speaker */
@@ -654,6 +817,206 @@ static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
{ } /* end */
};
+static const struct hda_verb alc262_HP_BPC_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* 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)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, 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)},
+
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {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},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */
+ /* Input mixer1: only unmute Mic */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
+
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { }
+};
+
+static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for front
+ * panel mic (mic 2)
+ */
+ /* 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)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, 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)},
+
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Mono */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* rear MIC */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* Line in */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Line out */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD in */
+
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, /*rear MIC*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/
+ /* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
+
+ { }
+};
+
static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */
@@ -679,8 +1042,13 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = {
[ALC262_HIPPO] = "hippo",
[ALC262_HIPPO_1] = "hippo_1",
[ALC262_FUJITSU] = "fujitsu",
+ [ALC262_HP_BPC] = "hp-bpc",
+ [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
+ [ALC262_HP_TC_T5735] = "hp-tc-t5735",
+ [ALC262_HP_RP5700] = "hp-rp5700",
[ALC262_BENQ_ED8] = "benq",
[ALC262_BENQ_T31] = "benq-t31",
+ [ALC262_SONY_ASSAMD] = "sony-assamd",
[ALC262_TOSHIBA_S06] = "toshiba-s06",
[ALC262_TOSHIBA_RX1] = "toshiba-rx1",
[ALC262_ULTRA] = "ultra",
@@ -693,6 +1061,41 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = {
static const struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+ ALC262_AUTO),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
+ ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
+ ALC262_HP_TC_T5735),
+ SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
+ SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
+ SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
+ SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+ SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
+#if 0 /* disable the quirk since model=auto works better in recent versions */
+ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
+ ALC262_SONY_ASSAMD),
+#endif
SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
ALC262_TOSHIBA_RX1),
SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
@@ -763,6 +1166,68 @@ static const struct alc_config_preset alc262_presets[] = {
.setup = alc262_fujitsu_setup,
.init_hook = alc_inithook,
},
+ [ALC262_HP_BPC] = {
+ .mixers = { alc262_HP_BPC_mixer },
+ .init_verbs = { alc262_HP_BPC_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_bpc_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_BPC_D7000_WF] = {
+ .mixers = { alc262_HP_BPC_WildWest_mixer },
+ .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_wildwest_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_BPC_D7000_WL] = {
+ .mixers = { alc262_HP_BPC_WildWest_mixer,
+ alc262_HP_BPC_WildWest_option_mixer },
+ .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_wildwest_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_TC_T5735] = {
+ .mixers = { alc262_hp_t5735_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hp_t5735_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC262_HP_RP5700] = {
+ .mixers = { alc262_hp_rp5700_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_hp_rp5700_capture_source,
+ },
[ALC262_BENQ_ED8] = {
.mixers = { alc262_base_mixer },
.init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
@@ -773,6 +1238,19 @@ static const struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
},
+ [ALC262_SONY_ASSAMD] = {
+ .mixers = { alc262_sony_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc_inithook,
+ },
[ALC262_BENQ_T31] = {
.mixers = { alc262_benq_t31_mixer },
.init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
diff --git a/trunk/sound/pci/hda/alc268_quirks.c b/trunk/sound/pci/hda/alc268_quirks.c
new file mode 100644
index 000000000000..be58bf2f3aec
--- /dev/null
+++ b/trunk/sound/pci/hda/alc268_quirks.c
@@ -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
+};
+
diff --git a/trunk/sound/pci/hda/alc269_quirks.c b/trunk/sound/pci/hda/alc269_quirks.c
new file mode 100644
index 000000000000..14fdcf29b154
--- /dev/null
+++ b/trunk/sound/pci/hda/alc269_quirks.c
@@ -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,
+ },
+};
+
diff --git a/trunk/sound/pci/hda/alc662_quirks.c b/trunk/sound/pci/hda/alc662_quirks.c
new file mode 100644
index 000000000000..e69a6ea3083a
--- /dev/null
+++ b/trunk/sound/pci/hda/alc662_quirks.c
@@ -0,0 +1,1408 @@
+/*
+ * ALC662/ALC663/ALC665/ALC670 quirk models
+ * included by patch_realtek.c
+ */
+
+/* ALC662 models */
+enum {
+ ALC662_AUTO,
+ ALC662_3ST_2ch_DIG,
+ ALC662_3ST_6ch_DIG,
+ ALC662_3ST_6ch,
+ ALC662_5ST_DIG,
+ ALC662_LENOVO_101E,
+ ALC662_ASUS_EEEPC_P701,
+ ALC662_ASUS_EEEPC_EP20,
+ ALC663_ASUS_M51VA,
+ ALC663_ASUS_G71V,
+ ALC663_ASUS_H13,
+ ALC663_ASUS_G50V,
+ ALC662_ECS,
+ ALC663_ASUS_MODE1,
+ ALC662_ASUS_MODE2,
+ ALC663_ASUS_MODE3,
+ ALC663_ASUS_MODE4,
+ ALC663_ASUS_MODE5,
+ ALC663_ASUS_MODE6,
+ ALC663_ASUS_MODE7,
+ ALC663_ASUS_MODE8,
+ ALC272_DELL,
+ ALC272_DELL_ZM1,
+ ALC272_SAMSUNG_NC10,
+ ALC662_MODEL_LAST,
+};
+
+#define ALC662_DIGOUT_NID 0x06
+#define ALC662_DIGIN_NID 0x0a
+
+static const hda_nid_t alc662_dac_nids[3] = {
+ /* front, rear, clfe */
+ 0x02, 0x03, 0x04
+};
+
+static const hda_nid_t alc272_dac_nids[2] = {
+ 0x02, 0x03
+};
+
+static const hda_nid_t alc662_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x09, 0x08
+};
+
+static const hda_nid_t alc272_adc_nids[1] = {
+ /* ADC1-2 */
+ 0x08,
+};
+
+static const hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
+static const hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
+
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+static const struct hda_input_mux alc662_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static const struct hda_input_mux alc662_lenovo_101e_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static const struct hda_input_mux alc663_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+#if 0 /* set to 1 for testing other input sources below */
+static const struct hda_input_mux alc272_nc10_capture_source = {
+ .num_items = 16,
+ .items = {
+ { "Autoselect Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "In-0x02", 0x2 },
+ { "In-0x03", 0x3 },
+ { "In-0x04", 0x4 },
+ { "In-0x05", 0x5 },
+ { "In-0x06", 0x6 },
+ { "In-0x07", 0x7 },
+ { "In-0x08", 0x8 },
+ { "In-0x09", 0x9 },
+ { "In-0x0a", 0x0a },
+ { "In-0x0b", 0x0b },
+ { "In-0x0c", 0x0c },
+ { "In-0x0d", 0x0d },
+ { "In-0x0e", 0x0e },
+ { "In-0x0f", 0x0f },
+ },
+};
+#endif
+
+/*
+ * 2ch mode
+ */
+static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
+ { 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc662_3ST_ch2_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc662_3ST_ch6_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
+ { 2, alc662_3ST_ch2_init },
+ { 6, alc662_3ST_ch6_init },
+};
+
+/*
+ * 2ch mode
+ */
+static const struct hda_verb alc662_sixstack_ch6_init[] = {
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static const struct hda_verb alc662_sixstack_ch8_init[] = {
+ { 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 alc662_5stack_modes[2] = {
+ { 2, alc662_sixstack_ch6_init },
+ { 6, alc662_sixstack_ch8_init },
+};
+
+/* 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 alc662_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, 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_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ /*Input mixer control */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, 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_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 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, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+
+ 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 */
+};
+
+static const struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ 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("MuteCtrl Playback Switch", 0x0c, 2, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_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 const struct hda_bind_ctls alc663_asus_one_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_m51va_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_tree_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_four_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc662_1bjd_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, 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("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc663_asus_two_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume",
+ &alc663_asus_two_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_g71v_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, 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("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_g50v_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, 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("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal 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),
+ { } /* end */
+};
+
+static const struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static const struct snd_kcontrol_new alc663_mode7_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("IntMic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("IntMic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc663_mode8_mixer[] = {
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch),
+ HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch),
+ HDA_CODEC_MUTE("Headphone1 Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone2 Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+static const struct snd_kcontrol_new alc662_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 */
+};
+
+static const struct hda_verb alc662_init_verbs[] = {
+ /* ADC: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {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)},
+
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* 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},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ { }
+};
+
+static const struct hda_verb alc662_eapd_init_verbs[] = {
+ /* always trun on EAPD */
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static const struct hda_verb alc662_sue_init_verbs[] = {
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc662_eeepc_sue_init_verbs[] = {
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+/* Set Unsolicited Event*/
+static const struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_m51va_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {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, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {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 alc663_21jd_amic_init_verbs[] = {
+ {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, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {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 alc662_1bjd_amic_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {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}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_15jd_amic_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {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 alc663_two_hp_amic_m1_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {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, 0x0}, /* Headphone */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {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, 0x01}, /* Headphone */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_g71v_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
+ /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */
+
+ {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}, /* Headphone */
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_FRONT_EVENT},
+ {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 alc663_g50v_init_verbs[] = {
+ {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}, /* Headphone */
+
+ {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 alc662_ecs_init_verbs[] = {
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc272_dell_zm1_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {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, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {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 alc272_dell_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {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, 0x01}, /* Headphone */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {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 alc663_mode7_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {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, 0x01},
+ {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, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {}
+};
+
+static const struct hda_verb alc663_mode8_init_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {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, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {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 snd_kcontrol_new alc662_auto_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new alc272_auto_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static void alc662_lenovo_101e_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.line_out_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->automute = 1;
+ spec->detect_line = 1;
+ spec->automute_lines = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc662_eeepc_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ alc262_hippo1_setup(codec);
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+}
+
+static void alc663_m51va_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;
+}
+
+/* ***************** Mode1 ******************************/
+static void alc663_mode1_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;
+}
+
+/* ***************** Mode2 ******************************/
+static void alc662_mode2_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_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode3 ******************************/
+static void alc663_mode3_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode4 ******************************/
+static void alc663_mode4_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute_mixer_nid[1] = 0x0e;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode5 ******************************/
+static void alc663_mode5_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->automute_mixer_nid[0] = 0x0c;
+ spec->automute_mixer_nid[1] = 0x0e;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_MIXER;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode6 ******************************/
+static void alc663_mode6_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x1b;
+ 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;
+}
+
+/* ***************** Mode7 ******************************/
+static void alc663_mode7_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x19;
+ spec->auto_mic = 1;
+}
+
+/* ***************** Mode8 ******************************/
+static void alc663_mode8_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.hp_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x17;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+static void alc663_g71v_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->autocfg.hp_pins[0] = 0x21;
+ spec->autocfg.line_out_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
+ spec->detect_line = 1;
+ spec->automute_lines = 1;
+ spec->ext_mic_pin = 0x18;
+ spec->int_mic_pin = 0x12;
+ spec->auto_mic = 1;
+}
+
+#define alc663_g50v_setup alc663_m51va_setup
+
+static const struct snd_kcontrol_new alc662_ecs_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+
+ HDA_CODEC_VOLUME("Mic/LineIn Boost Volume", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic/LineIn 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 */
+};
+
+static const struct snd_kcontrol_new alc272_nc10_mixer[] = {
+ /* Master Playback automatically created from Speaker and Headphone */
+ 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", 0x21, 0x0, 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, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+
+/*
+ * configuration and preset
+ */
+static const char * const alc662_models[ALC662_MODEL_LAST] = {
+ [ALC662_3ST_2ch_DIG] = "3stack-dig",
+ [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig",
+ [ALC662_3ST_6ch] = "3stack-6ch",
+ [ALC662_5ST_DIG] = "5stack-dig",
+ [ALC662_LENOVO_101E] = "lenovo-101e",
+ [ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
+ [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
+ [ALC662_ECS] = "ecs",
+ [ALC663_ASUS_M51VA] = "m51va",
+ [ALC663_ASUS_G71V] = "g71v",
+ [ALC663_ASUS_H13] = "h13",
+ [ALC663_ASUS_G50V] = "g50v",
+ [ALC663_ASUS_MODE1] = "asus-mode1",
+ [ALC662_ASUS_MODE2] = "asus-mode2",
+ [ALC663_ASUS_MODE3] = "asus-mode3",
+ [ALC663_ASUS_MODE4] = "asus-mode4",
+ [ALC663_ASUS_MODE5] = "asus-mode5",
+ [ALC663_ASUS_MODE6] = "asus-mode6",
+ [ALC663_ASUS_MODE7] = "asus-mode7",
+ [ALC663_ASUS_MODE8] = "asus-mode8",
+ [ALC272_DELL] = "dell",
+ [ALC272_DELL_ZM1] = "dell-zm1",
+ [ALC272_SAMSUNG_NC10] = "samsung-nc10",
+ [ALC662_AUTO] = "auto",
+};
+
+static const struct snd_pci_quirk alc662_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
+ SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL),
+ SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1),
+ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC663_ASUS_MODE7),
+ SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC663_ASUS_MODE8),
+ SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x17c3, "ASUS UX20", ALC663_ASUS_M51VA),
+ SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
+ SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
+ /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
+ SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
+ /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
+ SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x19d3, "ASUS NB", ALC663_ASUS_M51VA),
+ SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
+ SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
+ SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
+ SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
+ SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
+ SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
+ SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
+ SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
+ ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
+ {}
+};
+
+static const struct alc_config_preset alc662_presets[] = {
+ [ALC662_3ST_2ch_DIG] = {
+ .mixers = { alc662_3ST_2ch_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_3ST_6ch_DIG] = {
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_3ST_6ch] = {
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_5ST_DIG] = {
+ .mixers = { alc662_base_mixer, alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
+ .channel_mode = alc662_5stack_modes,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_LENOVO_101E] = {
+ .mixers = { alc662_lenovo_101e_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_lenovo_101e_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_lenovo_101e_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ASUS_EEEPC_P701] = {
+ .mixers = { alc662_eeepc_p701_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_eeepc_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_eeepc_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ASUS_EEEPC_EP20] = {
+ .mixers = { alc662_eeepc_ep20_mixer,
+ alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_eeepc_ep20_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .input_mux = &alc662_lenovo_101e_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_eeepc_ep20_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ECS] = {
+ .mixers = { alc662_ecs_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_ecs_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_eeepc_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_M51VA] = {
+ .mixers = { alc663_m51va_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_m51va_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_m51va_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_G71V] = {
+ .mixers = { alc663_g71v_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_g71v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_g71v_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_H13] = {
+ .mixers = { alc663_m51va_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_m51va_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .setup = alc663_m51va_setup,
+ .unsol_event = alc_sku_unsol_event,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_G50V] = {
+ .mixers = { alc663_g50v_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_g50v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .input_mux = &alc663_capture_source,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_g50v_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE1] = {
+ .mixers = { alc663_m51va_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_21jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode1_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC662_ASUS_MODE2] = {
+ .mixers = { alc662_1bjd_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc662_1bjd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc662_mode2_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE3] = {
+ .mixers = { alc663_two_hp_m1_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_two_hp_amic_m1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode3_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE4] = {
+ .mixers = { alc663_asus_21jd_clfe_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_21jd_amic_init_verbs},
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode4_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE5] = {
+ .mixers = { alc663_asus_15jd_clfe_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_15jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode5_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE6] = {
+ .mixers = { alc663_two_hp_m2_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_two_hp_amic_m2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode6_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE7] = {
+ .mixers = { alc663_mode7_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_mode7_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode7_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC663_ASUS_MODE8] = {
+ .mixers = { alc663_mode8_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_mode8_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode8_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC272_DELL] = {
+ .mixers = { alc663_m51va_mixer },
+ .cap_mixer = alc272_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc272_dell_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc272_dac_nids),
+ .dac_nids = alc272_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .adc_nids = alc272_adc_nids,
+ .num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
+ .capsrc_nids = alc272_capsrc_nids,
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_m51va_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC272_DELL_ZM1] = {
+ .mixers = { alc663_m51va_mixer },
+ .cap_mixer = alc662_auto_capture_mixer,
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc272_dell_zm1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc272_dac_nids),
+ .dac_nids = alc272_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .adc_nids = alc662_adc_nids,
+ .num_adc_nids = 1,
+ .capsrc_nids = alc662_capsrc_nids,
+ .channel_mode = alc662_3ST_2ch_modes,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_m51va_setup,
+ .init_hook = alc_inithook,
+ },
+ [ALC272_SAMSUNG_NC10] = {
+ .mixers = { alc272_nc10_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eapd_init_verbs,
+ alc663_21jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc272_dac_nids),
+ .dac_nids = alc272_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ /*.input_mux = &alc272_nc10_capture_source,*/
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc663_mode4_setup,
+ .init_hook = alc_inithook,
+ },
+};
+
+
diff --git a/trunk/sound/pci/hda/alc680_quirks.c b/trunk/sound/pci/hda/alc680_quirks.c
new file mode 100644
index 000000000000..0eeb227c7bc2
--- /dev/null
+++ b/trunk/sound/pci/hda/alc680_quirks.c
@@ -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,
+
+ },
+};
diff --git a/trunk/sound/pci/hda/alc861_quirks.c b/trunk/sound/pci/hda/alc861_quirks.c
new file mode 100644
index 000000000000..d719ec6350eb
--- /dev/null
+++ b/trunk/sound/pci/hda/alc861_quirks.c
@@ -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,
+ },
+};
+
diff --git a/trunk/sound/pci/hda/alc861vd_quirks.c b/trunk/sound/pci/hda/alc861vd_quirks.c
new file mode 100644
index 000000000000..8f28450f41f8
--- /dev/null
+++ b/trunk/sound/pci/hda/alc861vd_quirks.c
@@ -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,
+ },
+};
+
diff --git a/trunk/sound/pci/hda/alc880_quirks.c b/trunk/sound/pci/hda/alc880_quirks.c
index bea22edcfd8c..c844d2b59988 100644
--- a/trunk/sound/pci/hda/alc880_quirks.c
+++ b/trunk/sound/pci/hda/alc880_quirks.c
@@ -749,7 +749,8 @@ static void alc880_uniwill_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x16;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc880_uniwill_init_hook(struct hda_codec *codec)
@@ -780,7 +781,8 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -1049,7 +1051,8 @@ static void alc880_lg_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/*
@@ -1134,7 +1137,8 @@ static void alc880_lg_lw_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
@@ -1184,7 +1188,7 @@ static void alc880_medion_rim_automute(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
alc_hp_automute(codec);
/* toggle EAPD */
- if (spec->hp_jack_present)
+ if (spec->jack_present)
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
else
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
@@ -1206,7 +1210,8 @@ static void alc880_medion_rim_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/trunk/sound/pci/hda/alc882_quirks.c b/trunk/sound/pci/hda/alc882_quirks.c
index e251514a26a4..617d04723b82 100644
--- a/trunk/sound/pci/hda/alc882_quirks.c
+++ b/trunk/sound/pci/hda/alc882_quirks.c
@@ -173,7 +173,8 @@ static void alc889_automute_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[2] = 0x17;
spec->autocfg.speaker_pins[3] = 0x19;
spec->autocfg.speaker_pins[4] = 0x1a;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc889_intel_init_hook(struct hda_codec *codec)
@@ -190,7 +191,8 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[1] = 0x1b; /* hp */
spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
spec->autocfg.speaker_pins[1] = 0x15; /* bass */
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/*
@@ -473,7 +475,8 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
@@ -484,7 +487,8 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
@@ -495,7 +499,8 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
@@ -506,7 +511,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x1b;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
#define ALC882_DIGOUT_NID 0x06
@@ -1705,7 +1711,8 @@ static void alc885_imac24_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
#define alc885_mb5_setup alc885_imac24_setup
@@ -1714,11 +1721,12 @@ static void alc885_imac24_setup(struct hda_codec *codec)
/* Macbook Air 2,1 */
static void alc885_mba21_setup(struct hda_codec *codec)
{
- struct alc_spec *spec = codec->spec;
+ struct alc_spec *spec = codec->spec;
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x18;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x18;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
@@ -1729,7 +1737,8 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc885_imac91_setup(struct hda_codec *codec)
@@ -1739,7 +1748,8 @@ static void alc885_imac91_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct hda_verb alc882_targa_verbs[] = {
@@ -1763,7 +1773,7 @@ static void alc882_targa_automute(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
alc_hp_automute(codec);
snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
- spec->hp_jack_present ? 1 : 3);
+ spec->jack_present ? 1 : 3);
}
static void alc882_targa_setup(struct hda_codec *codec)
@@ -1772,7 +1782,8 @@ static void alc882_targa_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -2176,7 +2187,8 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1a;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
@@ -2329,7 +2341,8 @@ static void alc883_mitac_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct hda_verb alc883_mitac_verbs[] = {
@@ -2494,7 +2507,8 @@ static void alc888_3st_hp_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x18;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct hda_verb alc888_3st_hp_verbs[] = {
@@ -2554,7 +2568,8 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.line_out_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/* toggle speaker-output according to the hp-jack state */
@@ -2564,7 +2579,8 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/* toggle speaker-output according to the hp-jack state */
@@ -2577,7 +2593,8 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
@@ -2606,7 +2623,8 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc883_haier_w66_setup(struct hda_codec *codec)
@@ -2615,7 +2633,8 @@ static void alc883_haier_w66_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc883_lenovo_101e_setup(struct hda_codec *codec)
@@ -2625,7 +2644,10 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.line_out_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->detect_line = 1;
+ spec->automute_lines = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
/* toggle speaker-output according to the hp-jack state */
@@ -2636,7 +2658,8 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[1] = 0x16;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct hda_verb alc883_acer_eapd_verbs[] = {
@@ -2666,7 +2689,8 @@ static void alc888_6st_dell_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc888_lenovo_sky_setup(struct hda_codec *codec)
@@ -2679,7 +2703,8 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec)
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
spec->autocfg.speaker_pins[4] = 0x1a;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static void alc883_vaiott_setup(struct hda_codec *codec)
@@ -2689,7 +2714,8 @@ static void alc883_vaiott_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct hda_verb alc888_asus_m90v_verbs[] = {
@@ -2713,7 +2739,8 @@ static void alc883_mode2_setup(struct hda_codec *codec)
spec->ext_mic_pin = 0x18;
spec->int_mic_pin = 0x19;
spec->auto_mic = 1;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_AMP;
}
static const struct hda_verb alc888_asus_eee1601_verbs[] = {
diff --git a/trunk/sound/pci/hda/alc_quirks.c b/trunk/sound/pci/hda/alc_quirks.c
index a18952ed4311..2be1129cf458 100644
--- a/trunk/sound/pci/hda/alc_quirks.c
+++ b/trunk/sound/pci/hda/alc_quirks.c
@@ -453,19 +453,6 @@ static void setup_preset(struct hda_codec *codec,
alc_fixup_autocfg_pin_nums(codec);
}
-static void alc_simple_setup_automute(struct alc_spec *spec, int mode)
-{
- int lo_pin = spec->autocfg.line_out_pins[0];
-
- if (lo_pin == spec->autocfg.speaker_pins[0] ||
- lo_pin == spec->autocfg.hp_pins[0])
- lo_pin = 0;
- spec->automute_mode = mode;
- spec->detect_hp = !!spec->autocfg.hp_pins[0];
- spec->detect_lo = !!lo_pin;
- spec->automute_lo = spec->automute_lo_possible = !!lo_pin;
- spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0];
-}
/* auto-toggle front mic */
static void alc88x_simple_mic_automute(struct hda_codec *codec)
diff --git a/trunk/sound/pci/hda/hda_codec.c b/trunk/sound/pci/hda/hda_codec.c
index 1715e8b24ff0..3e7850c238c3 100644
--- a/trunk/sound/pci/hda/hda_codec.c
+++ b/trunk/sound/pci/hda/hda_codec.c
@@ -34,9 +34,6 @@
#include "hda_beep.h"
#include
-#define CREATE_TRACE_POINTS
-#include "hda_trace.h"
-
/*
* vendor / preset table
*/
@@ -211,19 +208,15 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd,
again:
snd_hda_power_up(codec);
mutex_lock(&bus->cmd_mutex);
- trace_hda_send_cmd(codec, cmd);
err = bus->ops.command(bus, cmd);
- if (!err && res) {
+ if (!err && res)
*res = bus->ops.get_response(bus, codec->addr);
- trace_hda_get_response(codec, *res);
- }
mutex_unlock(&bus->cmd_mutex);
snd_hda_power_down(codec);
if (res && *res == -1 && bus->rirb_error) {
if (bus->response_reset) {
snd_printd("hda_codec: resetting BUS due to "
"fatal communication error\n");
- trace_hda_bus_reset(bus);
bus->ops.bus_reset(bus);
}
goto again;
@@ -586,13 +579,9 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
return -1;
}
recursive++;
- for (i = 0; i < nums; i++) {
- unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
- if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
- continue;
+ 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);
@@ -614,7 +603,6 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
struct hda_bus_unsolicited *unsol;
unsigned int wp;
- trace_hda_unsol_event(bus, res, res_ex);
unsol = bus->unsol;
if (!unsol)
return 0;
@@ -1491,11 +1479,8 @@ static void really_cleanup_stream(struct hda_codec *codec,
struct hda_cvt_setup *q)
{
hda_nid_t nid = q->nid;
- if (q->stream_tag || q->channel_id)
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
- if (q->format_id)
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0
-);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
memset(q, 0, sizeof(*q));
q->nid = nid;
}
@@ -1699,29 +1684,6 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
}
EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
-/**
- * snd_hda_override_pin_caps - Override the pin capabilities
- * @codec: the CODEC
- * @nid: the NID to override
- * @caps: the capability bits to set
- *
- * Override the cached PIN capabilitiy bits value by the given one.
- *
- * Returns zero if successful or a negative error code.
- */
-int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
- unsigned int caps)
-{
- struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
- if (!info)
- return -ENOMEM;
- info->amp_caps = caps;
- info->head.val |= INFO_AMP_CAPS;
- return 0;
-}
-EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps);
-
/**
* snd_hda_pin_sense - execute pin sense measurement
* @codec: the CODEC to sense
@@ -4121,7 +4083,6 @@ static void hda_power_work(struct work_struct *work)
return;
}
- trace_hda_power_down(codec);
hda_call_codec_suspend(codec);
if (bus->ops.pm_notify)
bus->ops.pm_notify(bus);
@@ -4160,7 +4121,6 @@ void snd_hda_power_up(struct hda_codec *codec)
if (codec->power_on || codec->power_transition)
return;
- trace_hda_power_up(codec);
snd_hda_update_power_acct(codec);
codec->power_on = 1;
codec->power_jiffies = jiffies;
@@ -4573,11 +4533,6 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag,
0, format);
/* extra outputs copied from front */
- for (i = 0; i < ARRAY_SIZE(mout->hp_out_nid); i++)
- if (!mout->no_share_stream && mout->hp_out_nid[i])
- snd_hda_codec_setup_stream(codec,
- mout->hp_out_nid[i],
- stream_tag, 0, format);
for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
if (!mout->no_share_stream && mout->extra_out_nid[i])
snd_hda_codec_setup_stream(codec,
@@ -4610,10 +4565,6 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
snd_hda_codec_cleanup_stream(codec, nids[i]);
if (mout->hp_nid)
snd_hda_codec_cleanup_stream(codec, mout->hp_nid);
- for (i = 0; i < ARRAY_SIZE(mout->hp_out_nid); i++)
- if (mout->hp_out_nid[i])
- snd_hda_codec_cleanup_stream(codec,
- mout->hp_out_nid[i]);
for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
if (mout->extra_out_nid[i])
snd_hda_codec_cleanup_stream(codec,
@@ -4694,27 +4645,6 @@ static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
}
}
-/* Reorder the surround channels
- * ALSA sequence is front/surr/clfe/side
- * HDA sequence is:
- * 4-ch: front/surr => OK as it is
- * 6-ch: front/clfe/surr
- * 8-ch: front/clfe/rear/side|fc
- */
-static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
-{
- hda_nid_t nid;
-
- switch (nums) {
- case 3:
- case 4:
- nid = pins[1];
- pins[1] = pins[2];
- pins[2] = nid;
- break;
- }
-}
-
/*
* Parse all pin widgets and store the useful pin nids to cfg
*
@@ -4732,13 +4662,12 @@ static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
* The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
* respectively.
*/
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids,
- unsigned int cond_flags)
+int snd_hda_parse_pin_def_config(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ const hda_nid_t *ignore_nids)
{
hda_nid_t nid, end_nid;
- short seq, assoc_line_out;
+ short seq, assoc_line_out, assoc_speaker;
short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
@@ -4749,7 +4678,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
memset(sequences_line_out, 0, sizeof(sequences_line_out));
memset(sequences_speaker, 0, sizeof(sequences_speaker));
memset(sequences_hp, 0, sizeof(sequences_hp));
- assoc_line_out = 0;
+ assoc_line_out = assoc_speaker = 0;
end_nid = codec->start_nid + codec->num_nodes;
for (nid = codec->start_nid; nid < end_nid; nid++) {
@@ -4801,10 +4730,16 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
case AC_JACK_SPEAKER:
seq = get_defcfg_sequence(def_conf);
assoc = get_defcfg_association(def_conf);
+ if (!assoc)
+ continue;
+ if (!assoc_speaker)
+ assoc_speaker = assoc;
+ else if (assoc_speaker != assoc)
+ continue;
if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
continue;
cfg->speaker_pins[cfg->speaker_outs] = nid;
- sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
+ sequences_speaker[cfg->speaker_outs] = seq;
cfg->speaker_outs++;
break;
case AC_JACK_HP_OUT:
@@ -4853,8 +4788,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
* If no line-out is defined but multiple HPs are found,
* some of them might be the real line-outs.
*/
- if (!cfg->line_outs && cfg->hp_outs > 1 &&
- !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
+ if (!cfg->line_outs && cfg->hp_outs > 1) {
int i = 0;
while (i < cfg->hp_outs) {
/* The real HPs should have the sequence 0x0f */
@@ -4891,8 +4825,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
* FIX-UP: if no line-outs are detected, try to use speaker or HP pin
* as a primary output
*/
- if (!cfg->line_outs &&
- !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
+ if (!cfg->line_outs) {
if (cfg->speaker_outs) {
cfg->line_outs = cfg->speaker_outs;
memcpy(cfg->line_out_pins, cfg->speaker_pins,
@@ -4910,9 +4843,21 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
}
}
- reorder_outputs(cfg->line_outs, cfg->line_out_pins);
- reorder_outputs(cfg->hp_outs, cfg->hp_pins);
- reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
+ /* Reorder the surround channels
+ * ALSA sequence is front/surr/clfe/side
+ * HDA sequence is:
+ * 4-ch: front/surr => OK as it is
+ * 6-ch: front/clfe/surr
+ * 8-ch: front/clfe/rear/side|fc
+ */
+ switch (cfg->line_outs) {
+ case 3:
+ case 4:
+ nid = cfg->line_out_pins[1];
+ cfg->line_out_pins[1] = cfg->line_out_pins[2];
+ cfg->line_out_pins[2] = nid;
+ break;
+ }
sort_autocfg_input_pins(cfg);
@@ -4950,7 +4895,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
return 0;
}
-EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
+EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config);
int snd_hda_get_input_pin_attr(unsigned int def_conf)
{
@@ -5208,6 +5153,30 @@ void snd_array_free(struct snd_array *array)
}
EXPORT_SYMBOL_HDA(snd_array_free);
+/**
+ * snd_print_pcm_rates - Print the supported PCM rates to the string buffer
+ * @pcm: PCM caps bits
+ * @buf: the string buffer to write
+ * @buflen: the max buffer length
+ *
+ * used by hda_proc.c and hda_eld.c
+ */
+void snd_print_pcm_rates(int pcm, char *buf, int buflen)
+{
+ static unsigned int rates[] = {
+ 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
+ 96000, 176400, 192000, 384000
+ };
+ int i, j;
+
+ for (i = 0, j = 0; i < ARRAY_SIZE(rates); i++)
+ if (pcm & (1 << i))
+ j += snprintf(buf + j, buflen - j, " %d", rates[i]);
+
+ buf[j] = '\0'; /* necessary when j == 0 */
+}
+EXPORT_SYMBOL_HDA(snd_print_pcm_rates);
+
/**
* snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer
* @pcm: PCM caps bits
@@ -5249,8 +5218,6 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid,
return "Mic";
case SND_JACK_LINEOUT:
return "Line-out";
- case SND_JACK_LINEIN:
- return "Line-in";
case SND_JACK_HEADSET:
return "Headset";
case SND_JACK_VIDEOOUT:
diff --git a/trunk/sound/pci/hda/hda_eld.c b/trunk/sound/pci/hda/hda_eld.c
index 1c8ddf547a2d..28ce17d09c33 100644
--- a/trunk/sound/pci/hda/hda_eld.c
+++ b/trunk/sound/pci/hda/hda_eld.c
@@ -144,17 +144,25 @@ static int cea_sampling_frequencies[8] = {
SNDRV_PCM_RATE_192000, /* 7: 192000Hz */
};
-static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
+static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
int byte_index)
{
unsigned int val;
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_HDMI_ELDD, byte_index);
+
#ifdef BE_PARANOID
printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
#endif
- return val;
+
+ if ((val & AC_ELDD_ELD_VALID) == 0) {
+ snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
+ byte_index);
+ val = 0;
+ }
+
+ return val & AC_ELDD_ELD_DATA;
}
#define GRAB_BITS(buf, byte, lowbit, bits) \
@@ -318,11 +326,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
int size;
unsigned char *buf;
- /*
- * ELD size is initialized to zero in caller function. If no errors and
- * ELD is valid, actual eld_size is assigned in hdmi_update_eld()
- */
-
if (!eld->eld_valid)
return -ENOENT;
@@ -332,59 +335,24 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n");
size = 128;
}
- if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) {
+ if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) {
snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size);
return -ERANGE;
}
- /* set ELD buffer */
- buf = eld->eld_buffer;
-
- for (i = 0; i < size; i++) {
- unsigned int val = hdmi_get_eld_data(codec, nid, i);
- if (!(val & AC_ELDD_ELD_VALID)) {
- if (!i) {
- snd_printd(KERN_INFO
- "HDMI: invalid ELD data\n");
- ret = -EINVAL;
- goto error;
- }
- snd_printd(KERN_INFO
- "HDMI: invalid ELD data byte %d\n", i);
- val = 0;
- } else
- val &= AC_ELDD_ELD_DATA;
- buf[i] = val;
- }
+ buf = kmalloc(size, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ for (i = 0; i < size; i++)
+ buf[i] = hdmi_get_eld_byte(codec, nid, i);
ret = hdmi_update_eld(eld, buf, size);
-error:
+ kfree(buf);
return ret;
}
-/**
- * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with
- * hdmi-specific routine.
- */
-static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen)
-{
- static unsigned int alsa_rates[] = {
- 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
- 96000, 176400, 192000, 384000
- };
- int i, j;
-
- for (i = 0, j = 0; i < ARRAY_SIZE(alsa_rates); i++)
- if (pcm & (1 << i))
- j += snprintf(buf + j, buflen - j, " %d",
- alsa_rates[i]);
-
- buf[j] = '\0'; /* necessary when j == 0 */
-}
-
-#define SND_PRINT_RATES_ADVISED_BUFSIZE 80
-
static void hdmi_show_short_audio_desc(struct cea_sad *a)
{
char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
@@ -393,7 +361,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
if (!a->format)
return;
- hdmi_print_pcm_rates(a->rates, buf, sizeof(buf));
+ snd_print_pcm_rates(a->rates, buf, sizeof(buf));
if (a->format == AUDIO_CODING_TYPE_LPCM)
snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
@@ -452,7 +420,7 @@ static void hdmi_print_sad_info(int i, struct cea_sad *a,
i, a->format, cea_audio_coding_type_names[a->format]);
snd_iprintf(buffer, "sad%d_channels\t\t%d\n", i, a->channels);
- hdmi_print_pcm_rates(a->rates, buf, sizeof(buf));
+ snd_print_pcm_rates(a->rates, buf, sizeof(buf));
snd_iprintf(buffer, "sad%d_rates\t\t[0x%x]%s\n", i, a->rates, buf);
if (a->format == AUDIO_CODING_TYPE_LPCM) {
diff --git a/trunk/sound/pci/hda/hda_hwdep.c b/trunk/sound/pci/hda/hda_hwdep.c
index 7e7d0788ddcf..bf3ced51e0f8 100644
--- a/trunk/sound/pci/hda/hda_hwdep.c
+++ b/trunk/sound/pci/hda/hda_hwdep.c
@@ -643,14 +643,14 @@ static inline int strmatch(const char *a, const char *b)
static void parse_codec_mode(char *buf, struct hda_bus *bus,
struct hda_codec **codecp)
{
- int vendorid, subid, caddr;
+ unsigned int vendorid, subid, caddr;
struct hda_codec *codec;
*codecp = NULL;
if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) {
list_for_each_entry(codec, &bus->codec_list, list) {
- if ((vendorid <= 0 || codec->vendor_id == vendorid) &&
- (subid <= 0 || codec->subsystem_id == subid) &&
+ if (codec->vendor_id == vendorid &&
+ codec->subsystem_id == subid &&
codec->addr == caddr) {
*codecp = codec;
break;
@@ -756,6 +756,8 @@ static int get_line_from_fw(char *buf, int size, struct firmware *fw)
}
if (!fw->size)
return 0;
+ if (size < fw->size)
+ size = fw->size;
for (len = 0; len < fw->size; len++) {
if (!*p)
diff --git a/trunk/sound/pci/hda/hda_intel.c b/trunk/sound/pci/hda/hda_intel.c
index bd7fc99af187..be6982289c0d 100644
--- a/trunk/sound/pci/hda/hda_intel.c
+++ b/trunk/sound/pci/hda/hda_intel.c
@@ -34,6 +34,7 @@
*
*/
+#include
#include
#include
#include
@@ -45,12 +46,6 @@
#include
#include
#include
-#include
-#ifdef CONFIG_X86
-/* for snoop control */
-#include
-#include
-#endif
#include
#include
#include "hda_codec.h"
@@ -121,22 +116,6 @@ module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif
-static int align_buffer_size = 1;
-module_param(align_buffer_size, bool, 0644);
-MODULE_PARM_DESC(align_buffer_size,
- "Force buffer and period sizes to be multiple of 128 bytes.");
-
-#ifdef CONFIG_X86
-static bool hda_snoop = true;
-module_param_named(snoop, hda_snoop, bool, 0444);
-MODULE_PARM_DESC(snoop, "Enable/disable snooping");
-#define azx_snoop(chip) (chip)->snoop
-#else
-#define hda_snoop true
-#define azx_snoop(chip) true
-#endif
-
-
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, ICH6M},"
@@ -381,7 +360,7 @@ struct azx_dev {
*/
unsigned char stream_tag; /* assigned stream */
unsigned char index; /* stream index */
- int assigned_key; /* last device# key assigned to */
+ int device; /* last device number assigned to */
unsigned int opened :1;
unsigned int running :1;
@@ -392,7 +371,6 @@ struct azx_dev {
* when link position is not greater than FIFO size
*/
unsigned int insufficient :1;
- unsigned int wc_marked:1;
};
/* CORB/RIRB */
@@ -460,7 +438,6 @@ struct azx {
unsigned int msi :1;
unsigned int irq_pending_warned :1;
unsigned int probing :1; /* codec probing phase */
- unsigned int snoop:1;
/* for debugging */
unsigned int last_cmd[AZX_MAX_CODECS];
@@ -504,7 +481,6 @@ enum {
#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 */
-#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@@ -566,45 +542,6 @@ static char *driver_short_names[] __devinitdata = {
/* for pcm support */
#define get_azx_dev(substream) (substream->runtime->private_data)
-#ifdef CONFIG_X86
-static void __mark_pages_wc(struct azx *chip, void *addr, size_t size, bool on)
-{
- if (azx_snoop(chip))
- return;
- if (addr && size) {
- int pages = (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
- if (on)
- set_memory_wc((unsigned long)addr, pages);
- else
- set_memory_wb((unsigned long)addr, pages);
- }
-}
-
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
- bool on)
-{
- __mark_pages_wc(chip, buf->area, buf->bytes, on);
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
- struct snd_pcm_runtime *runtime, bool on)
-{
- if (azx_dev->wc_marked != on) {
- __mark_pages_wc(chip, runtime->dma_area, runtime->dma_bytes, on);
- azx_dev->wc_marked = on;
- }
-}
-#else
-/* NOP for other archs */
-static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf,
- bool on)
-{
-}
-static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev,
- struct snd_pcm_runtime *runtime, bool on)
-{
-}
-#endif
-
static int azx_acquire_irq(struct azx *chip, int do_disconnect);
static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
/*
@@ -626,7 +563,6 @@ static int azx_alloc_cmd_io(struct azx *chip)
snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n");
return err;
}
- mark_pages_wc(chip, &chip->rb, true);
return 0;
}
@@ -1143,15 +1079,7 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg,
static void azx_init_pci(struct azx *chip)
{
- /* force to non-snoop mode for a new VIA controller when BIOS is set */
- if (chip->snoop && chip->driver_type == AZX_DRIVER_VIA) {
- u8 snoop;
- pci_read_config_byte(chip->pci, 0x42, &snoop);
- if (!(snoop & 0x80) && chip->pci->revision == 0x30) {
- chip->snoop = 0;
- snd_printdd(SFX "Force to non-snoop mode\n");
- }
- }
+ unsigned short snoop;
/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
* TCSEL == Traffic Class Select Register, which sets PCI express QOS
@@ -1168,15 +1096,15 @@ static void azx_init_pci(struct azx *chip)
* we need to enable snoop.
*/
if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) {
- snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip));
+ snd_printdd(SFX "Enabling ATI snoop\n");
update_pci_byte(chip->pci,
- ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07,
- azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0);
+ ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
+ 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP);
}
/* For NVIDIA HDA, enable snoop */
if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) {
- snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip));
+ snd_printdd(SFX "Enabling Nvidia snoop\n");
update_pci_byte(chip->pci,
NVIDIA_HDA_TRANSREG_ADDR,
0x0f, NVIDIA_HDA_ENABLE_COHBITS);
@@ -1190,20 +1118,16 @@ static void azx_init_pci(struct azx *chip)
/* Enable SCH/PCH snoop if needed */
if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) {
- unsigned short snoop;
pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
- if ((!azx_snoop(chip) && !(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)) ||
- (azx_snoop(chip) && (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP))) {
- snoop &= ~INTEL_SCH_HDA_DEVC_NOSNOOP;
- if (!azx_snoop(chip))
- snoop |= INTEL_SCH_HDA_DEVC_NOSNOOP;
- pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, snoop);
+ if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) {
+ pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC,
+ snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP));
pci_read_config_word(chip->pci,
INTEL_SCH_HDA_DEVC, &snoop);
- }
- snd_printdd(SFX "SCH snoop: %s\n",
+ snd_printdd(SFX "HDA snoop disabled, enabling ... %s\n",
(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)
- ? "Disabled" : "Enabled");
+ ? "Failed" : "OK");
+ }
}
}
@@ -1410,16 +1334,12 @@ static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev)
*/
static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
{
- unsigned int val;
/* make sure the run bit is zero for SD */
azx_stream_clear(chip, azx_dev);
/* program the stream_tag */
- val = azx_sd_readl(azx_dev, SD_CTL);
- val = (val & ~SD_CTL_STREAM_TAG_MASK) |
- (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT);
- if (!azx_snoop(chip))
- val |= SD_CTL_TRAFFIC_PRIO;
- azx_sd_writel(azx_dev, SD_CTL, val);
+ azx_sd_writel(azx_dev, SD_CTL,
+ (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)|
+ (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT));
/* program the length of samples in cyclic buffer */
azx_sd_writel(azx_dev, SD_CBL, azx_dev->bufsize);
@@ -1613,9 +1533,6 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
{
int dev, i, nums;
struct azx_dev *res = NULL;
- /* make a non-zero unique key for the substream */
- int key = (substream->pcm->device << 16) | (substream->number << 2) |
- (substream->stream + 1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dev = chip->playback_index_offset;
@@ -1627,12 +1544,12 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
for (i = 0; i < nums; i++, dev++)
if (!chip->azx_dev[dev].opened) {
res = &chip->azx_dev[dev];
- if (res->assigned_key == key)
+ if (res->device == substream->pcm->device)
break;
}
if (res) {
res->opened = 1;
- res->assigned_key = key;
+ res->device = substream->pcm->device;
}
return res;
}
@@ -1682,7 +1599,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned long flags;
int err;
- int buff_step;
mutex_lock(&chip->open_mutex);
azx_dev = azx_assign_device(chip, substream);
@@ -1697,25 +1613,10 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
runtime->hw.rates = hinfo->rates;
snd_pcm_limit_hw_rates(runtime);
snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
- if (align_buffer_size)
- /* constrain buffer sizes to be multiple of 128
- bytes. This is more efficient in terms of memory
- access but isn't required by the HDA spec and
- prevents users from specifying exact period/buffer
- sizes. For example for 44.1kHz, a period size set
- to 20ms will be rounded to 19.59ms. */
- buff_step = 128;
- else
- /* Don't enforce steps on buffer sizes, still need to
- be multiple of 4 bytes (HDA spec). Tested on Intel
- HDA controllers, may not work on all devices where
- option needs to be disabled */
- buff_step = 4;
-
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
- buff_step);
+ 128);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
- buff_step);
+ 128);
snd_hda_power_up(apcm->codec);
err = hinfo->ops.open(hinfo, apcm->codec, substream);
if (err < 0) {
@@ -1770,30 +1671,19 @@ static int azx_pcm_close(struct snd_pcm_substream *substream)
static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
- struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
- struct azx *chip = apcm->chip;
- struct snd_pcm_runtime *runtime = substream->runtime;
struct azx_dev *azx_dev = get_azx_dev(substream);
- int ret;
- mark_runtime_wc(chip, azx_dev, runtime, false);
azx_dev->bufsize = 0;
azx_dev->period_bytes = 0;
azx_dev->format_val = 0;
- ret = snd_pcm_lib_malloc_pages(substream,
+ return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
- if (ret < 0)
- return ret;
- mark_runtime_wc(chip, azx_dev, runtime, true);
- return ret;
}
static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
struct azx_dev *azx_dev = get_azx_dev(substream);
- struct azx *chip = apcm->chip;
- struct snd_pcm_runtime *runtime = substream->runtime;
struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
/* reset BDL address */
@@ -1806,7 +1696,6 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
snd_hda_codec_cleanup(apcm->codec, hinfo, substream);
- mark_runtime_wc(chip, azx_dev, runtime, false);
return snd_pcm_lib_free_pages(substream);
}
@@ -2035,8 +1924,7 @@ static unsigned int azx_via_get_position(struct azx *chip,
}
static unsigned int azx_get_position(struct azx *chip,
- struct azx_dev *azx_dev,
- bool with_check)
+ struct azx_dev *azx_dev)
{
unsigned int pos;
int stream = azx_dev->substream->stream;
@@ -2052,7 +1940,7 @@ static unsigned int azx_get_position(struct azx *chip,
default:
/* use the position buffer */
pos = le32_to_cpu(*azx_dev->posbuf);
- if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) {
+ if (chip->position_fix[stream] == POS_FIX_AUTO) {
if (!pos || pos == (u32)-1) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
@@ -2076,7 +1964,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
struct azx *chip = apcm->chip;
struct azx_dev *azx_dev = get_azx_dev(substream);
return bytes_to_frames(substream->runtime,
- azx_get_position(chip, azx_dev, false));
+ azx_get_position(chip, azx_dev));
}
/*
@@ -2099,7 +1987,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
return -1; /* bogus (too early) interrupt */
stream = azx_dev->substream->stream;
- pos = azx_get_position(chip, azx_dev, true);
+ pos = azx_get_position(chip, azx_dev);
if (WARN_ONCE(!azx_dev->period_bytes,
"hda-intel: zero azx_dev->period_bytes"))
@@ -2166,20 +2054,6 @@ static void azx_clear_irq_pending(struct azx *chip)
spin_unlock_irq(&chip->reg_lock);
}
-#ifdef CONFIG_X86
-static int azx_pcm_mmap(struct snd_pcm_substream *substream,
- struct vm_area_struct *area)
-{
- struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
- struct azx *chip = apcm->chip;
- if (!azx_snoop(chip))
- area->vm_page_prot = pgprot_writecombine(area->vm_page_prot);
- return snd_pcm_lib_default_mmap(substream, area);
-}
-#else
-#define azx_pcm_mmap NULL
-#endif
-
static struct snd_pcm_ops azx_pcm_ops = {
.open = azx_pcm_open,
.close = azx_pcm_close,
@@ -2189,7 +2063,6 @@ static struct snd_pcm_ops azx_pcm_ops = {
.prepare = azx_pcm_prepare,
.trigger = azx_pcm_trigger,
.pointer = azx_pcm_pointer,
- .mmap = azx_pcm_mmap,
.page = snd_pcm_sgbuf_ops_page,
};
@@ -2470,19 +2343,13 @@ static int azx_free(struct azx *chip)
if (chip->azx_dev) {
for (i = 0; i < chip->num_streams; i++)
- if (chip->azx_dev[i].bdl.area) {
- mark_pages_wc(chip, &chip->azx_dev[i].bdl, false);
+ if (chip->azx_dev[i].bdl.area)
snd_dma_free_pages(&chip->azx_dev[i].bdl);
- }
}
- if (chip->rb.area) {
- mark_pages_wc(chip, &chip->rb, false);
+ if (chip->rb.area)
snd_dma_free_pages(&chip->rb);
- }
- if (chip->posbuf.area) {
- mark_pages_wc(chip, &chip->posbuf, false);
+ if (chip->posbuf.area)
snd_dma_free_pages(&chip->posbuf);
- }
pci_release_regions(chip->pci);
pci_disable_device(chip->pci);
kfree(chip->azx_dev);
@@ -2502,7 +2369,6 @@ static int azx_dev_free(struct snd_device *device)
static struct snd_pci_quirk position_fix_list[] __devinitdata = {
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, 0x02c6, "Dell Inspiron 1010", 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),
@@ -2678,7 +2544,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
check_probe_mask(chip, dev);
chip->single_cmd = single_cmd;
- chip->snoop = hda_snoop;
if (bdl_pos_adj[dev] < 0) {
switch (chip->driver_type) {
@@ -2751,10 +2616,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
gcap &= ~ICH6_GCAP_64OK;
}
- /* disable buffer size rounding to 128-byte multiples if supported */
- if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
- align_buffer_size = 0;
-
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
@@ -2806,7 +2667,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
goto errout;
}
- mark_pages_wc(chip, &chip->azx_dev[i].bdl, true);
}
/* allocate memory for the position buffer */
err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
@@ -2816,7 +2676,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
goto errout;
}
- mark_pages_wc(chip, &chip->posbuf, true);
/* allocate CORB/RIRB */
err = azx_alloc_cmd_io(chip);
if (err < 0)
@@ -2958,49 +2817,37 @@ static void __devexit azx_remove(struct pci_dev *pci)
static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
/* CPT */
{ PCI_DEVICE(0x8086, 0x1c20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
- AZX_DCAPS_BUFSIZE },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
/* PBG */
{ PCI_DEVICE(0x8086, 0x1d20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
- AZX_DCAPS_BUFSIZE},
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
/* Panther Point */
{ PCI_DEVICE(0x8086, 0x1e20),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
- AZX_DCAPS_BUFSIZE},
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP },
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
- .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
- AZX_DCAPS_BUFSIZE},
+ .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP },
{ PCI_DEVICE(0x8086, 0x2668),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ICH6 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH6 */
{ PCI_DEVICE(0x8086, 0x27d8),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ICH7 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH7 */
{ PCI_DEVICE(0x8086, 0x269a),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ESB2 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ESB2 */
{ PCI_DEVICE(0x8086, 0x284b),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ICH8 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH8 */
{ PCI_DEVICE(0x8086, 0x293e),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ICH9 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x293f),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ICH9 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH9 */
{ PCI_DEVICE(0x8086, 0x3a3e),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ICH10 */
+ .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC }, /* ICH10 */
{ PCI_DEVICE(0x8086, 0x3a6e),
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |
- AZX_DCAPS_BUFSIZE }, /* ICH10 */
+ .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,
.class_mask = 0xffffff,
- .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE },
+ .driver_data = AZX_DRIVER_ICH },
/* ATI SB 450/600/700/800/900 */
{ PCI_DEVICE(0x1002, 0x437b),
.driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB },
diff --git a/trunk/sound/pci/hda/hda_local.h b/trunk/sound/pci/hda/hda_local.h
index 79f49e2e8cbc..2e7ac31afa8d 100644
--- a/trunk/sound/pci/hda/hda_local.h
+++ b/trunk/sound/pci/hda/hda_local.h
@@ -267,14 +267,11 @@ int snd_hda_ch_mode_put(struct hda_codec *codec,
enum { HDA_FRONT, HDA_REAR, HDA_CLFE, HDA_SIDE }; /* index for dac_nidx */
enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */
-#define HDA_MAX_OUTS 5
-
struct hda_multi_out {
int num_dacs; /* # of DACs, must be more than 1 */
const hda_nid_t *dac_nids; /* DAC list */
hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */
- hda_nid_t hp_out_nid[HDA_MAX_OUTS]; /* DACs for multiple HPs */
- hda_nid_t extra_out_nid[HDA_MAX_OUTS]; /* other (e.g. speaker) DACs */
+ hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */
hda_nid_t dig_out_nid; /* digital out audio widget */
const hda_nid_t *slave_dig_outs;
int max_channels; /* currently supported analog channels */
@@ -336,6 +333,9 @@ int snd_hda_codec_proc_new(struct hda_codec *codec);
static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
#endif
+#define SND_PRINT_RATES_ADVISED_BUFSIZE 80
+void snd_print_pcm_rates(int pcm, char *buf, int buflen);
+
#define SND_PRINT_BITS_ADVISED_BUFSIZE 16
void snd_print_pcm_bits(int pcm, char *buf, int buflen);
@@ -385,7 +385,7 @@ enum {
AUTO_PIN_HP_OUT
};
-#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS
+#define AUTO_CFG_MAX_OUTS 5
#define AUTO_CFG_MAX_INS 8
struct auto_pin_cfg_item {
@@ -442,21 +442,10 @@ struct auto_pin_cfg {
(cfg & AC_DEFCFG_SEQUENCE)
#define get_defcfg_device(cfg) \
((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
-#define get_defcfg_misc(cfg) \
- ((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT)
-
-/* bit-flags for snd_hda_parse_pin_def_config() behavior */
-#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */
-#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids,
- unsigned int cond_flags);
-
-/* older function */
-#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
- snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
+int snd_hda_parse_pin_def_config(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ const hda_nid_t *ignore_nids);
/* amp values */
#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
@@ -503,16 +492,12 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps);
u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
- unsigned int caps);
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
{
return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) &&
- !(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid) &
- AC_DEFCFG_MISC_NO_PRESENCE)) &&
(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP);
}
@@ -604,8 +589,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
#define get_amp_nid_(pv) ((pv) & 0xffff)
#define get_amp_nid(kc) get_amp_nid_((kc)->private_value)
#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3)
-#define get_amp_direction_(pv) (((pv) >> 18) & 0x1)
-#define get_amp_direction(kc) get_amp_direction_((kc)->private_value)
+#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1)
#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf)
#define get_amp_offset(kc) (((kc)->private_value >> 23) & 0x3f)
#define get_amp_min_mute(kc) (((kc)->private_value >> 29) & 0x1)
@@ -623,7 +607,6 @@ struct cea_sad {
};
#define ELD_FIXED_BYTES 20
-#define ELD_MAX_SIZE 256
#define ELD_MAX_MNL 16
#define ELD_MAX_SAD 16
@@ -648,7 +631,6 @@ struct hdmi_eld {
int spk_alloc;
int sad_count;
struct cea_sad sad[ELD_MAX_SAD];
- char eld_buffer[ELD_MAX_SIZE];
#ifdef CONFIG_PROC_FS
struct snd_info_entry *proc_entry;
#endif
diff --git a/trunk/sound/pci/hda/hda_proc.c b/trunk/sound/pci/hda/hda_proc.c
index 2c981b55940b..2be57b051aa2 100644
--- a/trunk/sound/pci/hda/hda_proc.c
+++ b/trunk/sound/pci/hda/hda_proc.c
@@ -152,18 +152,12 @@ static void print_amp_vals(struct snd_info_buffer *buffer,
static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm)
{
- static unsigned int rates[] = {
- 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
- 96000, 176400, 192000, 384000
- };
- int i;
+ char buf[SND_PRINT_RATES_ADVISED_BUFSIZE];
pcm &= AC_SUPPCM_RATES;
snd_iprintf(buffer, " rates [0x%x]:", pcm);
- for (i = 0; i < ARRAY_SIZE(rates); i++)
- if (pcm & (1 << i))
- snd_iprintf(buffer, " %d", rates[i]);
- snd_iprintf(buffer, "\n");
+ snd_print_pcm_rates(pcm, buf, sizeof(buf));
+ snd_iprintf(buffer, "%s\n", buf);
}
static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm)
diff --git a/trunk/sound/pci/hda/hda_trace.h b/trunk/sound/pci/hda/hda_trace.h
deleted file mode 100644
index 9884871ddb00..000000000000
--- a/trunk/sound/pci/hda/hda_trace.h
+++ /dev/null
@@ -1,117 +0,0 @@
-#undef TRACE_SYSTEM
-#define TRACE_SYSTEM hda
-#define TRACE_INCLUDE_FILE hda_trace
-
-#if !defined(_TRACE_HDA_H) || defined(TRACE_HEADER_MULTI_READ)
-#define _TRACE_HDA_H
-
-#include
-
-struct hda_bus;
-struct hda_codec;
-
-DECLARE_EVENT_CLASS(hda_cmd,
-
- TP_PROTO(struct hda_codec *codec, unsigned int val),
-
- TP_ARGS(codec, val),
-
- TP_STRUCT__entry(
- __field( unsigned int, card )
- __field( unsigned int, addr )
- __field( unsigned int, val )
- ),
-
- TP_fast_assign(
- __entry->card = (codec)->bus->card->number;
- __entry->addr = (codec)->addr;
- __entry->val = (val);
- ),
-
- TP_printk("[%d:%d] val=%x", __entry->card, __entry->addr, __entry->val)
-);
-
-DEFINE_EVENT(hda_cmd, hda_send_cmd,
- TP_PROTO(struct hda_codec *codec, unsigned int val),
- TP_ARGS(codec, val)
-);
-
-DEFINE_EVENT(hda_cmd, hda_get_response,
- TP_PROTO(struct hda_codec *codec, unsigned int val),
- TP_ARGS(codec, val)
-);
-
-TRACE_EVENT(hda_bus_reset,
-
- TP_PROTO(struct hda_bus *bus),
-
- TP_ARGS(bus),
-
- TP_STRUCT__entry(
- __field( unsigned int, card )
- ),
-
- TP_fast_assign(
- __entry->card = (bus)->card->number;
- ),
-
- TP_printk("[%d]", __entry->card)
-);
-
-DECLARE_EVENT_CLASS(hda_power,
-
- TP_PROTO(struct hda_codec *codec),
-
- TP_ARGS(codec),
-
- TP_STRUCT__entry(
- __field( unsigned int, card )
- __field( unsigned int, addr )
- ),
-
- TP_fast_assign(
- __entry->card = (codec)->bus->card->number;
- __entry->addr = (codec)->addr;
- ),
-
- TP_printk("[%d:%d]", __entry->card, __entry->addr)
-);
-
-DEFINE_EVENT(hda_power, hda_power_down,
- TP_PROTO(struct hda_codec *codec),
- TP_ARGS(codec)
-);
-
-DEFINE_EVENT(hda_power, hda_power_up,
- TP_PROTO(struct hda_codec *codec),
- TP_ARGS(codec)
-);
-
-TRACE_EVENT(hda_unsol_event,
-
- TP_PROTO(struct hda_bus *bus, u32 res, u32 res_ex),
-
- TP_ARGS(bus, res, res_ex),
-
- TP_STRUCT__entry(
- __field( unsigned int, card )
- __field( u32, res )
- __field( u32, res_ex )
- ),
-
- TP_fast_assign(
- __entry->card = (bus)->card->number;
- __entry->res = res;
- __entry->res_ex = res_ex;
- ),
-
- TP_printk("[%d] res=%x, res_ex=%x", __entry->card,
- __entry->res, __entry->res_ex)
-);
-
-#endif /* _TRACE_HDA_H */
-
-/* This part must be outside protection */
-#undef TRACE_INCLUDE_PATH
-#define TRACE_INCLUDE_PATH .
-#include
diff --git a/trunk/sound/pci/hda/patch_analog.c b/trunk/sound/pci/hda/patch_analog.c
index d8aac588f23b..8648917acffb 100644
--- a/trunk/sound/pci/hda/patch_analog.c
+++ b/trunk/sound/pci/hda/patch_analog.c
@@ -48,8 +48,6 @@ struct ad198x_spec {
const hda_nid_t *alt_dac_nid;
const struct hda_pcm_stream *stream_analog_alt_playback;
- int independent_hp;
- int num_active_streams;
/* capture */
unsigned int num_adc_nids;
@@ -304,72 +302,6 @@ static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
}
#endif
-static void activate_ctl(struct hda_codec *codec, const char *name, int active)
-{
- struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name);
- if (ctl) {
- ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
- ctl->vd[0].access |= active ? 0 :
- SNDRV_CTL_ELEM_ACCESS_INACTIVE;
- ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_WRITE;
- ctl->vd[0].access |= active ?
- SNDRV_CTL_ELEM_ACCESS_WRITE : 0;
- snd_ctl_notify(codec->bus->card,
- SNDRV_CTL_EVENT_MASK_INFO, &ctl->id);
- }
-}
-
-static void set_stream_active(struct hda_codec *codec, bool active)
-{
- struct ad198x_spec *spec = codec->spec;
- if (active)
- spec->num_active_streams++;
- else
- spec->num_active_streams--;
- activate_ctl(codec, "Independent HP", spec->num_active_streams == 0);
-}
-
-static int ad1988_independent_hp_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = { "OFF", "ON", NULL};
- int index;
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- index = uinfo->value.enumerated.item;
- if (index >= 2)
- index = 1;
- strcpy(uinfo->value.enumerated.name, texts[index]);
- return 0;
-}
-
-static int ad1988_independent_hp_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- ucontrol->value.enumerated.item[0] = spec->independent_hp;
- return 0;
-}
-
-static int ad1988_independent_hp_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *spec = codec->spec;
- unsigned int select = ucontrol->value.enumerated.item[0];
- if (spec->independent_hp != select) {
- spec->independent_hp = select;
- if (spec->independent_hp)
- spec->multiout.hp_nid = 0;
- else
- spec->multiout.hp_nid = spec->alt_dac_nid[0];
- return 1;
- }
- return 0;
-}
-
/*
* Analog playback callbacks
*/
@@ -378,15 +310,8 @@ static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct ad198x_spec *spec = codec->spec;
- int err;
- set_stream_active(codec, true);
- err = snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
hinfo);
- if (err < 0) {
- set_stream_active(codec, false);
- return err;
- }
- return 0;
}
static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -408,41 +333,11 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
}
-static int ad198x_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- set_stream_active(codec, false);
- return 0;
-}
-
-static int ad1988_alt_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct ad198x_spec *spec = codec->spec;
- if (!spec->independent_hp)
- return -EBUSY;
- set_stream_active(codec, true);
- return 0;
-}
-
-static int ad1988_alt_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- set_stream_active(codec, false);
- return 0;
-}
-
static const struct hda_pcm_stream ad198x_pcm_analog_alt_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
- .ops = {
- .open = ad1988_alt_playback_pcm_open,
- .close = ad1988_alt_playback_pcm_close
- },
+ /* NID is set in ad198x_build_pcms */
};
/*
@@ -507,6 +402,7 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
return 0;
}
+
/*
*/
static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
@@ -517,8 +413,7 @@ static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
.ops = {
.open = ad198x_playback_pcm_open,
.prepare = ad198x_playback_pcm_prepare,
- .cleanup = ad198x_playback_pcm_cleanup,
- .close = ad198x_playback_pcm_close
+ .cleanup = ad198x_playback_pcm_cleanup
},
};
@@ -2163,6 +2058,7 @@ static int patch_ad1981(struct hda_codec *codec)
enum {
AD1988_6STACK,
AD1988_6STACK_DIG,
+ AD1988_6STACK_DIG_FP,
AD1988_3STACK,
AD1988_3STACK_DIG,
AD1988_LAPTOP,
@@ -2272,17 +2168,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
return err;
}
-static const struct snd_kcontrol_new ad1988_hp_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Independent HP",
- .info = ad1988_independent_hp_info,
- .get = ad1988_independent_hp_get,
- .put = ad1988_independent_hp_put,
- },
- { } /* end */
-};
-
/* 6-stack mode */
static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
@@ -2303,7 +2188,6 @@ static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
};
static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
@@ -2326,6 +2210,13 @@ static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
+
+ { } /* end */
+};
+
+static const struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+
{ } /* end */
};
@@ -2347,7 +2238,6 @@ static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
};
static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
@@ -2382,7 +2272,6 @@ static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
/* laptop mode */
static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
@@ -2557,7 +2446,7 @@ static const struct hda_verb ad1988_6stack_init_verbs[] = {
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2705,7 +2594,7 @@ static const struct hda_verb ad1988_3stack_init_verbs[] = {
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2780,7 +2669,7 @@ static const struct hda_verb ad1988_laptop_init_verbs[] = {
{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Port-A front headphon path */
- {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -2893,11 +2782,11 @@ static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx)
{
static const hda_nid_t idx_to_dac[8] = {
/* A B C D E F G H */
- 0x03, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a
+ 0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a
};
static const hda_nid_t idx_to_dac_rev2[8] = {
/* A B C D E F G H */
- 0x03, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
+ 0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
};
if (is_rev2(codec))
return idx_to_dac_rev2[idx];
@@ -3134,8 +3023,8 @@ static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec,
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
switch (nid) {
- case 0x11: /* port-A - DAC 03 */
- snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x00);
+ case 0x11: /* port-A - DAC 04 */
+ snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x01);
break;
case 0x14: /* port-B - DAC 06 */
snd_hda_codec_write(codec, 0x30, 0, AC_VERB_SET_CONNECT_SEL, 0x02);
@@ -3261,6 +3150,7 @@ static int ad1988_auto_init(struct hda_codec *codec)
static const char * const ad1988_models[AD1988_MODEL_LAST] = {
[AD1988_6STACK] = "6stack",
[AD1988_6STACK_DIG] = "6stack-dig",
+ [AD1988_6STACK_DIG_FP] = "6stack-dig-fp",
[AD1988_3STACK] = "3stack",
[AD1988_3STACK_DIG] = "3stack-dig",
[AD1988_LAPTOP] = "laptop",
@@ -3318,11 +3208,10 @@ static int patch_ad1988(struct hda_codec *codec)
}
set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = ad1988_alt_dac_nid[0];
switch (board_config) {
case AD1988_6STACK:
case AD1988_6STACK_DIG:
+ case AD1988_6STACK_DIG_FP:
spec->multiout.max_channels = 8;
spec->multiout.num_dacs = 4;
if (is_rev2(codec))
@@ -3338,7 +3227,19 @@ static int patch_ad1988(struct hda_codec *codec)
spec->mixers[1] = ad1988_6stack_mixers2;
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1988_6stack_init_verbs;
- if (board_config == AD1988_6STACK_DIG) {
+ if (board_config == AD1988_6STACK_DIG_FP) {
+ spec->num_mixers++;
+ spec->mixers[2] = ad1988_6stack_fp_mixers;
+ spec->num_init_verbs++;
+ spec->init_verbs[1] = ad1988_6stack_fp_init_verbs;
+ spec->slave_vols = ad1988_6stack_fp_slave_vols;
+ spec->slave_sws = ad1988_6stack_fp_slave_sws;
+ spec->alt_dac_nid = ad1988_alt_dac_nid;
+ spec->stream_analog_alt_playback =
+ &ad198x_pcm_analog_alt_playback;
+ }
+ if ((board_config == AD1988_6STACK_DIG) ||
+ (board_config == AD1988_6STACK_DIG_FP)) {
spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
spec->dig_in_nid = AD1988_SPDIF_IN;
}
@@ -3381,15 +3282,6 @@ static int patch_ad1988(struct hda_codec *codec)
break;
}
- if (spec->autocfg.hp_pins[0]) {
- spec->mixers[spec->num_mixers++] = ad1988_hp_mixers;
- spec->slave_vols = ad1988_6stack_fp_slave_vols;
- spec->slave_sws = ad1988_6stack_fp_slave_sws;
- spec->alt_dac_nid = ad1988_alt_dac_nid;
- spec->stream_analog_alt_playback =
- &ad198x_pcm_analog_alt_playback;
- }
-
spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
spec->adc_nids = ad1988_adc_nids;
spec->capsrc_nids = ad1988_capsrc_nids;
diff --git a/trunk/sound/pci/hda/patch_cirrus.c b/trunk/sound/pci/hda/patch_cirrus.c
index c45f3e69bcf0..47d6ffc9b5b5 100644
--- a/trunk/sound/pci/hda/patch_cirrus.c
+++ b/trunk/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
unsigned int *idxp)
{
- int i, idx;
+ int i;
hda_nid_t nid;
nid = codec->start_nid;
@@ -384,11 +384,9 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
type = get_wcaps_type(get_wcaps(codec, nid));
if (type != AC_WID_AUD_IN)
continue;
- idx = snd_hda_get_conn_index(codec, nid, pin, false);
- if (idx >= 0) {
- *idxp = idx;
+ *idxp = snd_hda_get_conn_index(codec, nid, pin, false);
+ if (*idxp >= 0)
return nid;
- }
}
return 0;
}
@@ -535,7 +533,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
int index, unsigned int pval, int dir,
struct snd_kcontrol **kctlp)
{
- char tmp[44];
+ char tmp[32];
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
knew.private_value = pval;
diff --git a/trunk/sound/pci/hda/patch_conexant.c b/trunk/sound/pci/hda/patch_conexant.c
index 0c8b5a1993ed..502fc9499453 100644
--- a/trunk/sound/pci/hda/patch_conexant.c
+++ b/trunk/sound/pci/hda/patch_conexant.c
@@ -136,8 +136,6 @@ struct conexant_spec {
unsigned int thinkpad:1;
unsigned int hp_laptop:1;
unsigned int asus:1;
- unsigned int pin_eapd_ctrls:1;
- unsigned int single_adc_amp:1;
unsigned int adc_switching:1;
@@ -1869,6 +1867,39 @@ static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
{ } /* end */
};
+static const struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
+ /* Line in, Mic */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+ /* SPK */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP, Amp */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Docking HP */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* DAC1 */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Record selector: Internal mic */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+ /* SPDIF route: PCM */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* needed for W500 Advanced Mini Dock 250410 */
+ {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* EAPD */
+ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+ { } /* end */
+};
+
static const struct hda_verb cxt5051_f700_init_verbs[] = {
/* Line in, Mic */
{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
@@ -1937,6 +1968,7 @@ enum {
CXT5051_LAPTOP, /* Laptops w/ EAPD support */
CXT5051_HP, /* no docking */
CXT5051_HP_DV6736, /* HP without mic switch */
+ CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */
CXT5051_F700, /* HP Compaq Presario F700 */
CXT5051_TOSHIBA, /* Toshiba M300 & co */
CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */
@@ -1948,6 +1980,7 @@ static const char *const cxt5051_models[CXT5051_MODELS] = {
[CXT5051_LAPTOP] = "laptop",
[CXT5051_HP] = "hp",
[CXT5051_HP_DV6736] = "hp-dv6736",
+ [CXT5051_LENOVO_X200] = "lenovo-x200",
[CXT5051_F700] = "hp-700",
[CXT5051_TOSHIBA] = "toshiba",
[CXT5051_IDEAPAD] = "ideapad",
@@ -1962,6 +1995,7 @@ static const struct snd_pci_quirk cxt5051_cfg_tbl[] = {
SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
CXT5051_LAPTOP),
SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
+ SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD),
{}
};
@@ -2019,6 +2053,13 @@ static int patch_cxt5051(struct hda_codec *codec)
spec->mixers[0] = cxt5051_hp_dv6736_mixers;
spec->auto_mic = 0;
break;
+ case CXT5051_LENOVO_X200:
+ spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
+ /* Thinkpad X301 does not have S/PDIF wired and no ability
+ to use a docking station. */
+ if (codec->subsystem_id == 0x17aa211f)
+ spec->multiout.dig_out_nid = 0;
+ break;
case CXT5051_F700:
spec->init_verbs[0] = cxt5051_f700_init_verbs;
spec->mixers[0] = cxt5051_f700_mixers;
@@ -3069,7 +3110,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
- SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
@@ -3308,8 +3348,6 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
#define MAX_AUTO_DACS 5
-#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */
-
/* fill analog DAC list from the widget tree */
static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
{
@@ -3332,26 +3370,16 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
/* fill pin_dac_pair list from the pin and dac list */
static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
int num_pins, hda_nid_t *dacs, int *rest,
- struct pin_dac_pair *filled, int nums,
- int type)
+ struct pin_dac_pair *filled, int type)
{
- int i, start = nums;
+ int i, nums;
- for (i = 0; i < num_pins; i++, nums++) {
+ nums = 0;
+ for (i = 0; i < num_pins; i++) {
filled[nums].pin = pins[i];
filled[nums].type = type;
filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
- if (filled[nums].dac)
- continue;
- if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
- filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
- continue;
- }
- if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
- filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
- continue;
- }
- snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
+ nums++;
}
return nums;
}
@@ -3367,19 +3395,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
rest = fill_cx_auto_dacs(codec, dacs);
/* parse all analog output pins */
nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
- dacs, &rest, spec->dac_info, 0,
- AUTO_PIN_LINE_OUT);
- nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
- dacs, &rest, spec->dac_info, nums,
- AUTO_PIN_HP_OUT);
- nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
- dacs, &rest, spec->dac_info, nums,
- AUTO_PIN_SPEAKER_OUT);
+ dacs, &rest, spec->dac_info,
+ AUTO_PIN_LINE_OUT);
+ nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+ dacs, &rest, spec->dac_info + nums,
+ AUTO_PIN_HP_OUT);
+ nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+ dacs, &rest, spec->dac_info + nums,
+ AUTO_PIN_SPEAKER_OUT);
spec->dac_info_filled = nums;
/* fill multiout struct */
for (i = 0; i < nums; i++) {
hda_nid_t dac = spec->dac_info[i].dac;
- if (!dac || (dac & DAC_SLAVE_FLAG))
+ if (!dac)
continue;
switch (spec->dac_info[i].type) {
case AUTO_PIN_LINE_OUT:
@@ -3432,14 +3460,12 @@ static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins,
static void do_automute(struct hda_codec *codec, int num_pins,
hda_nid_t *pins, bool on)
{
- struct conexant_spec *spec = codec->spec;
int i;
for (i = 0; i < num_pins; i++)
snd_hda_codec_write(codec, pins[i], 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
on ? PIN_OUT : 0);
- if (spec->pin_eapd_ctrls)
- cx_auto_turn_eapd(codec, num_pins, pins, on);
+ cx_auto_turn_eapd(codec, num_pins, pins, on);
}
static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
@@ -3464,12 +3490,9 @@ static void cx_auto_update_speakers(struct hda_codec *codec)
int on = 1;
/* turn on HP EAPD when HP jacks are present */
- if (spec->pin_eapd_ctrls) {
- if (spec->auto_mute)
- on = spec->hp_present;
- cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on);
- }
-
+ if (spec->auto_mute)
+ on = spec->hp_present;
+ cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on);
/* mute speakers in auto-mode if HP or LO jacks are plugged */
if (spec->auto_mute)
on = !(spec->hp_present ||
@@ -3839,7 +3862,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
}
if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
cx_auto_check_auto_mic(codec);
- if (imux->num_items > 1) {
+ if (imux->num_items > 1 && !spec->auto_mic) {
for (i = 1; i < imux->num_items; i++) {
if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
spec->adc_switching = 1;
@@ -3896,10 +3919,20 @@ static void cx_auto_parse_beep(struct hda_codec *codec)
#define cx_auto_parse_beep(codec)
#endif
-/* parse EAPDs */
+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;
@@ -3908,18 +3941,14 @@ static void cx_auto_parse_eapd(struct hda_codec *codec)
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;
}
-
- /* NOTE: below is a wild guess; if we have more than two EAPDs,
- * it's a new chip, where EAPDs are supposed to be associated to
- * pins, and we can control EAPD per pin.
- * OTOH, if only one or two EAPDs are found, it's an old chip,
- * thus it might control over all pins.
- */
- spec->pin_eapd_ctrls = spec->num_eapds > 2;
}
static int cx_auto_parse_auto_config(struct hda_codec *codec)
@@ -4006,8 +4035,6 @@ static void cx_auto_init_output(struct hda_codec *codec)
nid = spec->dac_info[i].dac;
if (!nid)
nid = spec->multiout.dac_nids[0];
- else if (nid & DAC_SLAVE_FLAG)
- nid &= ~DAC_SLAVE_FLAG;
select_connection(codec, spec->dac_info[i].pin, nid);
}
if (spec->auto_mute) {
@@ -4025,9 +4052,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
}
}
cx_auto_update_speakers(codec);
- /* turn on all EAPDs if no individual EAPD control is available */
- if (!spec->pin_eapd_ctrls)
- cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+ /* 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)
@@ -4141,11 +4167,9 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
hda_nid_t pin, const char *name, int idx)
{
unsigned int caps;
- if (dac && !(dac & DAC_SLAVE_FLAG)) {
- caps = query_amp_caps(codec, dac, HDA_OUTPUT);
- if (caps & AC_AMPCAP_NUM_STEPS)
- return cx_auto_add_pb_volume(codec, dac, name, idx);
- }
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ if (caps & AC_AMPCAP_NUM_STEPS)
+ return cx_auto_add_pb_volume(codec, dac, name, idx);
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (caps & AC_AMPCAP_NUM_STEPS)
return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4167,7 +4191,8 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
- hda_nid_t dac = spec->dac_info[i].dac;
+ if (!spec->dac_info[i].dac)
+ continue;
type = spec->dac_info[i].type;
if (type == AUTO_PIN_LINE_OUT)
type = spec->autocfg.line_out_type;
@@ -4186,7 +4211,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
idx = num_spk++;
break;
}
- err = try_add_pb_volume(codec, dac,
+ err = try_add_pb_volume(codec, spec->dac_info[i].dac,
spec->dac_info[i].pin,
label, idx);
if (err < 0)
@@ -4214,8 +4239,6 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
int idx = get_input_connection(codec, adc_nid, nid);
if (idx < 0)
continue;
- if (spec->single_adc_amp)
- idx = 0;
return cx_auto_add_volume_idx(codec, label, pfx,
cidx, adc_nid, HDA_INPUT, idx);
}
@@ -4256,21 +4279,14 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
struct hda_input_mux *imux = &spec->private_imux;
const char *prev_label;
int input_conn[HDA_MAX_NUM_INPUTS];
- int i, j, err, cidx;
+ int i, err, cidx;
int multi_connection;
- if (!imux->num_items)
- return 0;
-
multi_connection = 0;
for (i = 0; i < imux->num_items; i++) {
cidx = get_input_connection(codec, spec->imux_info[i].adc,
spec->imux_info[i].pin);
- if (cidx < 0)
- continue;
- input_conn[i] = spec->imux_info[i].adc;
- if (!spec->single_adc_amp)
- input_conn[i] |= cidx << 8;
+ input_conn[i] = (spec->imux_info[i].adc << 8) | cidx;
if (i > 0 && input_conn[i] != input_conn[0])
multi_connection = 1;
}
@@ -4299,15 +4315,6 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
err = cx_auto_add_capture_volume(codec, nid,
"Capture", "", cidx);
} else {
- bool dup_found = false;
- for (j = 0; j < i; j++) {
- if (input_conn[j] == input_conn[i]) {
- dup_found = true;
- break;
- }
- }
- if (dup_found)
- continue;
err = cx_auto_add_capture_volume(codec, nid,
label, " Capture", cidx);
}
@@ -4371,53 +4378,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
.reboot_notify = snd_hda_shutup_pins,
};
-/*
- * pin fix-up
- */
-struct cxt_pincfg {
- hda_nid_t nid;
- u32 val;
-};
-
-static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg)
-{
- for (; cfg->nid; cfg++)
- snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-
-}
-
-static void apply_pin_fixup(struct hda_codec *codec,
- const struct snd_pci_quirk *quirk,
- const struct cxt_pincfg **table)
-{
- quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
- if (quirk) {
- snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n",
- quirk->name);
- apply_pincfg(codec, table[quirk->value]);
- }
-}
-
-enum {
- CXT_PINCFG_LENOVO_X200,
-};
-
-static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
- { 0x16, 0x042140ff }, /* HP (seq# overridden) */
- { 0x17, 0x21a11000 }, /* dock-mic */
- { 0x19, 0x2121103f }, /* dock-HP */
- {}
-};
-
-static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
- [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
-};
-
-static const struct snd_pci_quirk cxt_fixups[] = {
- SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
- {}
-};
-
static int patch_conexant_auto(struct hda_codec *codec)
{
struct conexant_spec *spec;
@@ -4431,15 +4391,6 @@ static int patch_conexant_auto(struct hda_codec *codec)
return -ENOMEM;
codec->spec = spec;
codec->pin_amp_workaround = 1;
-
- switch (codec->vendor_id) {
- case 0x14f15045:
- spec->single_adc_amp = 1;
- break;
- }
-
- apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
-
err = cx_auto_search_adcs(codec);
if (err < 0)
return err;
diff --git a/trunk/sound/pci/hda/patch_hdmi.c b/trunk/sound/pci/hda/patch_hdmi.c
index aac3bfacda3f..19cb72db9c38 100644
--- a/trunk/sound/pci/hda/patch_hdmi.c
+++ b/trunk/sound/pci/hda/patch_hdmi.c
@@ -324,66 +324,6 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid)
return -EINVAL;
}
-static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct hdmi_spec *spec;
- int pin_idx;
-
- spec = codec->spec;
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
-
- pin_idx = kcontrol->private_value;
- uinfo->count = spec->pins[pin_idx].sink_eld.eld_size;
-
- return 0;
-}
-
-static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct hdmi_spec *spec;
- int pin_idx;
-
- spec = codec->spec;
- pin_idx = kcontrol->private_value;
-
- memcpy(ucontrol->value.bytes.data,
- spec->pins[pin_idx].sink_eld.eld_buffer, ELD_MAX_SIZE);
-
- return 0;
-}
-
-static struct snd_kcontrol_new eld_bytes_ctl = {
- .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
- .iface = SNDRV_CTL_ELEM_IFACE_PCM,
- .name = "ELD",
- .info = hdmi_eld_ctl_info,
- .get = hdmi_eld_ctl_get,
-};
-
-static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx,
- int device)
-{
- struct snd_kcontrol *kctl;
- struct hdmi_spec *spec = codec->spec;
- int err;
-
- kctl = snd_ctl_new1(&eld_bytes_ctl, codec);
- if (!kctl)
- return -ENOMEM;
- kctl->private_value = pin_idx;
- kctl->id.device = device;
-
- err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl);
- if (err < 0)
- return err;
-
- return 0;
-}
-
#ifdef BE_PARANOID
static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid,
int *packet_index, int *byte_index)
@@ -1006,6 +946,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
unsigned int caps, config;
int pin_idx;
struct hdmi_spec_per_pin *per_pin;
+ struct hdmi_eld *eld;
int err;
caps = snd_hda_param_read(codec, pin_nid, AC_PAR_PIN_CAP);
@@ -1022,15 +963,23 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
pin_idx = spec->num_pins;
per_pin = &spec->pins[pin_idx];
+ eld = &per_pin->sink_eld;
per_pin->pin_nid = pin_nid;
+ err = snd_hda_input_jack_add(codec, pin_nid,
+ SND_JACK_VIDEOOUT, NULL);
+ if (err < 0)
+ return err;
+
err = hdmi_read_pin_conn(codec, pin_idx);
if (err < 0)
return err;
spec->num_pins++;
+ hdmi_present_sense(codec, pin_nid, eld);
+
return 0;
}
@@ -1213,25 +1162,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec)
return 0;
}
-static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx)
-{
- int err;
- char hdmi_str[32];
- struct hdmi_spec *spec = codec->spec;
- struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
- int pcmdev = spec->pcm_rec[pin_idx].device;
-
- snprintf(hdmi_str, sizeof(hdmi_str), "HDMI/DP,pcm=%d", pcmdev);
-
- err = snd_hda_input_jack_add(codec, per_pin->pin_nid,
- SND_JACK_VIDEOOUT, pcmdev > 0 ? hdmi_str : NULL);
- if (err < 0)
- return err;
-
- hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld);
- return 0;
-}
-
static int generic_hdmi_build_controls(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
@@ -1240,25 +1170,12 @@ static int generic_hdmi_build_controls(struct hda_codec *codec)
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
-
- err = generic_hdmi_build_jack(codec, pin_idx);
- if (err < 0)
- return err;
-
err = snd_hda_create_spdif_out_ctls(codec,
per_pin->pin_nid,
per_pin->mux_nids[0]);
if (err < 0)
return err;
snd_hda_spdif_ctls_unassign(codec, pin_idx);
-
- /* add control for ELD Bytes */
- err = hdmi_create_eld_ctl(codec,
- pin_idx,
- spec->pcm_rec[pin_idx].device);
-
- if (err < 0)
- return err;
}
return 0;
@@ -1574,7 +1491,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
int chs;
- unsigned int dataDCC2, channel_id;
+ unsigned int dataDCC1, dataDCC2, channel_id;
int i;
struct hdmi_spec *spec = codec->spec;
struct hda_spdif_out *spdif =
@@ -1584,6 +1501,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
chs = substream->runtime->channels;
+ dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT;
dataDCC2 = 0x2;
/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c
index 80d6add8a620..e125c60fe352 100644
--- a/trunk/sound/pci/hda/patch_realtek.c
+++ b/trunk/sound/pci/hda/patch_realtek.c
@@ -116,8 +116,6 @@ struct alc_spec {
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid; /* digital-in NID; optional */
hda_nid_t mixer_nid; /* analog-mixer NID */
- DECLARE_BITMAP(vol_ctls, 0x20 << 1);
- DECLARE_BITMAP(sw_ctls, 0x20 << 1);
/* capture setup for dynamic dual-adc switch */
hda_nid_t cur_adc;
@@ -161,27 +159,23 @@ struct alc_spec {
void (*power_hook)(struct hda_codec *codec);
#endif
void (*shutup)(struct hda_codec *codec);
- void (*automute_hook)(struct hda_codec *codec);
/* for pin sensing */
- unsigned int hp_jack_present:1;
+ unsigned int jack_present: 1;
unsigned int line_jack_present:1;
unsigned int master_mute:1;
unsigned int auto_mic:1;
unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */
- unsigned int automute_speaker:1; /* automute speaker outputs */
- unsigned int automute_lo:1; /* automute LO outputs */
- unsigned int detect_hp:1; /* Headphone detection enabled */
- unsigned int detect_lo:1; /* Line-out detection enabled */
- unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */
- unsigned int automute_lo_possible:1; /* there are line outs and HP */
+ unsigned int automute:1; /* HP automute enabled */
+ unsigned int detect_line:1; /* Line-out detection enabled */
+ unsigned int automute_lines:1; /* automute line-out as well */
+ unsigned int automute_hp_lo:1; /* both HP and LO available */
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
unsigned int dyn_adc_switch:1; /* switch ADCs (for ALC275) */
unsigned int single_input_src:1;
unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
- unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
/* auto-mute control */
int automute_mode;
@@ -199,7 +193,6 @@ struct alc_spec {
/* for PLL fix */
hda_nid_t pll_nid;
unsigned int pll_coef_idx, pll_coef_bit;
- unsigned int coef0;
/* fix-up list */
int fixup_id;
@@ -209,9 +202,6 @@ struct alc_spec {
/* multi-io */
int multi_ios;
struct alc_multi_io multi_io[4];
-
- /* bind volumes */
- struct snd_array bind_ctls;
};
#define ALC_MODEL_AUTO 0 /* common for all chips */
@@ -535,8 +525,8 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
}
}
-/* Toggle outputs muting */
-static void update_outputs(struct hda_codec *codec)
+/* Toggle internal speakers muting */
+static void update_speakers(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int on;
@@ -548,10 +538,10 @@ static void update_outputs(struct hda_codec *codec)
do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins, spec->master_mute, true);
- if (!spec->automute_speaker)
+ if (!spec->automute)
on = 0;
else
- on = spec->hp_jack_present | spec->line_jack_present;
+ on = spec->jack_present | spec->line_jack_present;
on |= spec->master_mute;
do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
spec->autocfg.speaker_pins, on, false);
@@ -561,35 +551,26 @@ static void update_outputs(struct hda_codec *codec)
if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
return;
- if (!spec->automute_lo)
+ if (!spec->automute_lines || !spec->automute)
on = 0;
else
- on = spec->hp_jack_present;
+ on = spec->jack_present;
on |= spec->master_mute;
do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins, on, false);
}
-static void call_update_outputs(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- if (spec->automute_hook)
- spec->automute_hook(codec);
- else
- update_outputs(codec);
-}
-
/* standard HP-automute helper */
static void alc_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- spec->hp_jack_present =
+ if (!spec->automute)
+ return;
+ spec->jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins);
- if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo))
- return;
- call_update_outputs(codec);
+ update_speakers(codec);
}
/* standard line-out-automute helper */
@@ -597,16 +578,12 @@ static void alc_line_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- /* check LO jack only when it's different from HP */
- if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
+ if (!spec->automute || !spec->detect_line)
return;
-
spec->line_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins);
- if (!spec->automute_speaker || !spec->detect_lo)
- return;
- call_update_outputs(codec);
+ update_speakers(codec);
}
#define get_connection_index(codec, mux, nid) \
@@ -804,7 +781,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
- if (spec->automute_speaker_possible && spec->automute_lo_possible) {
+ if (spec->automute_hp_lo) {
uinfo->value.enumerated.items = 3;
texts = texts3;
} else {
@@ -823,12 +800,13 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
- unsigned int val = 0;
- if (spec->automute_speaker)
- val++;
- if (spec->automute_lo)
- val++;
-
+ unsigned int val;
+ if (!spec->automute)
+ val = 0;
+ else if (!spec->automute_lines)
+ val = 1;
+ else
+ val = 2;
ucontrol->value.enumerated.item[0] = val;
return 0;
}
@@ -841,36 +819,28 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
switch (ucontrol->value.enumerated.item[0]) {
case 0:
- if (!spec->automute_speaker && !spec->automute_lo)
+ if (!spec->automute)
return 0;
- spec->automute_speaker = 0;
- spec->automute_lo = 0;
+ spec->automute = 0;
break;
case 1:
- if (spec->automute_speaker_possible) {
- if (!spec->automute_lo && spec->automute_speaker)
- return 0;
- spec->automute_speaker = 1;
- spec->automute_lo = 0;
- } else if (spec->automute_lo_possible) {
- if (spec->automute_lo)
- return 0;
- spec->automute_lo = 1;
- } else
- return -EINVAL;
+ if (spec->automute && !spec->automute_lines)
+ return 0;
+ spec->automute = 1;
+ spec->automute_lines = 0;
break;
case 2:
- if (!spec->automute_lo_possible || !spec->automute_speaker_possible)
+ if (!spec->automute_hp_lo)
return -EINVAL;
- if (spec->automute_speaker && spec->automute_lo)
+ if (spec->automute && spec->automute_lines)
return 0;
- spec->automute_speaker = 1;
- spec->automute_lo = 1;
+ spec->automute = 1;
+ spec->automute_lines = 1;
break;
default:
return -EINVAL;
}
- call_update_outputs(codec);
+ update_speakers(codec);
return 1;
}
@@ -907,7 +877,7 @@ static int alc_add_automute_mode_enum(struct hda_codec *codec)
* Check the availability of HP/line-out auto-mute;
* Set up appropriately if really supported
*/
-static void alc_init_automute(struct hda_codec *codec)
+static void alc_init_auto_hp(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
@@ -922,6 +892,8 @@ static void alc_init_automute(struct hda_codec *codec)
present++;
if (present < 2) /* need two different output types */
return;
+ if (present == 3)
+ spec->automute_hp_lo = 1; /* both HP and LO automute */
if (!cfg->speaker_pins[0] &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
@@ -937,8 +909,6 @@ static void alc_init_automute(struct hda_codec *codec)
cfg->hp_outs = cfg->line_outs;
}
- spec->automute_mode = ALC_AUTOMUTE_PIN;
-
for (i = 0; i < cfg->hp_outs; i++) {
hda_nid_t nid = cfg->hp_pins[i];
if (!is_jack_detectable(codec, nid))
@@ -948,32 +918,28 @@ static void alc_init_automute(struct hda_codec *codec)
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | ALC_HP_EVENT);
- spec->detect_hp = 1;
- }
-
- if (cfg->line_out_type == AUTO_PIN_LINE_OUT && cfg->line_outs) {
- if (cfg->speaker_outs)
- for (i = 0; i < cfg->line_outs; i++) {
- hda_nid_t nid = cfg->line_out_pins[i];
- if (!is_jack_detectable(codec, nid))
- continue;
- snd_printdd("realtek: Enable Line-Out "
- "auto-muting on NID 0x%x\n", nid);
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | ALC_FRONT_EVENT);
- spec->detect_lo = 1;
+ spec->automute = 1;
+ spec->automute_mode = ALC_AUTOMUTE_PIN;
+ }
+ if (spec->automute && cfg->line_out_pins[0] &&
+ cfg->speaker_pins[0] &&
+ cfg->line_out_pins[0] != cfg->hp_pins[0] &&
+ cfg->line_out_pins[0] != cfg->speaker_pins[0]) {
+ for (i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t nid = cfg->line_out_pins[i];
+ if (!is_jack_detectable(codec, nid))
+ continue;
+ snd_printdd("realtek: Enable Line-Out auto-muting "
+ "on NID 0x%x\n", nid);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC_FRONT_EVENT);
+ spec->detect_line = 1;
}
- spec->automute_lo_possible = spec->detect_hp;
+ spec->automute_lines = spec->detect_line;
}
- spec->automute_speaker_possible = cfg->speaker_outs &&
- (spec->detect_hp || spec->detect_lo);
-
- spec->automute_lo = spec->automute_lo_possible;
- spec->automute_speaker = spec->automute_speaker_possible;
-
- if (spec->automute_speaker_possible || spec->automute_lo_possible) {
+ if (spec->automute) {
/* create a control for automute mode */
alc_add_automute_mode_enum(codec);
spec->unsol_event = alc_sku_unsol_event;
@@ -1174,7 +1140,7 @@ static void alc_init_auto_mic(struct hda_codec *codec)
/* check the availabilities of auto-mute and auto-mic switches */
static void alc_auto_check_switches(struct hda_codec *codec)
{
- alc_init_automute(codec);
+ alc_init_auto_hp(codec);
alc_init_auto_mic(codec);
}
@@ -1354,9 +1320,7 @@ static int alc_subsystem_id(struct hda_codec *codec,
* 15 : 1 --> enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
- if (!spec->autocfg.hp_pins[0] &&
- !(spec->autocfg.line_out_pins[0] &&
- spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
+ if (!spec->autocfg.hp_pins[0]) {
hda_nid_t nid;
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
@@ -1557,15 +1521,6 @@ static void alc_write_coef_idx(struct hda_codec *codec, unsigned int coef_idx,
coef_val);
}
-/* a special bypass for COEF 0; read the cached value at the second time */
-static unsigned int alc_get_coef0(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- if (!spec->coef0)
- spec->coef0 = alc_read_coef_idx(codec, 0);
- return spec->coef0;
-}
-
/*
* Digital I/O handling
*/
@@ -1604,29 +1559,27 @@ static void alc_auto_init_digital(struct hda_codec *codec)
static void alc_auto_parse_digital(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- int i, err, nums;
+ int i, err;
hda_nid_t dig_nid;
/* support multiple SPDIFs; the secondary is set up as a slave */
- nums = 0;
for (i = 0; i < spec->autocfg.dig_outs; i++) {
hda_nid_t conn[4];
err = snd_hda_get_connections(codec,
spec->autocfg.dig_out_pins[i],
conn, ARRAY_SIZE(conn));
- if (err <= 0)
+ if (err < 0)
continue;
dig_nid = conn[0]; /* assume the first element is audio-out */
- if (!nums) {
+ if (!i) {
spec->multiout.dig_out_nid = dig_nid;
spec->dig_out_type = spec->autocfg.dig_out_type[0];
} else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- if (nums >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
- spec->slave_dig_outs[nums - 1] = dig_nid;
+ spec->slave_dig_outs[i - 1] = dig_nid;
}
- nums++;
}
if (spec->autocfg.dig_in_pin) {
@@ -1831,7 +1784,6 @@ static const char * const alc_slave_vols[] = {
"Speaker Playback Volume",
"Mono Playback Volume",
"Line-Out Playback Volume",
- "PCM Playback Volume",
NULL,
};
@@ -1846,7 +1798,6 @@ static const char * const alc_slave_sws[] = {
"Mono Playback Switch",
"IEC958 Playback Switch",
"Line-Out Playback Switch",
- "PCM Playback Switch",
NULL,
};
@@ -2272,7 +2223,6 @@ static int alc_build_pcms(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
struct hda_pcm *info = spec->pcm_rec;
const struct hda_pcm_stream *p;
- bool have_multi_adcs;
int i;
codec->num_pcms = 1;
@@ -2351,11 +2301,8 @@ static int alc_build_pcms(struct hda_codec *codec)
/* If the use of more than one ADC is requested for the current
* model, configure a second analog capture-only PCM.
*/
- have_multi_adcs = (spec->num_adc_nids > 1) &&
- !spec->dyn_adc_switch && !spec->auto_mic &&
- (!spec->input_mux || spec->input_mux->num_items > 1);
/* Additional Analaog capture for index #2 */
- if (spec->alt_dac_nid || have_multi_adcs) {
+ if (spec->alt_dac_nid || spec->num_adc_nids > 1) {
codec->num_pcms = 3;
info = spec->pcm_rec + 2;
info->name = spec->stream_name_analog;
@@ -2371,7 +2318,7 @@ static int alc_build_pcms(struct hda_codec *codec)
alc_pcm_null_stream;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
}
- if (have_multi_adcs) {
+ if (spec->num_adc_nids > 1) {
p = spec->stream_analog_alt_capture;
if (!p)
p = &alc_pcm_analog_alt_capture;
@@ -2412,18 +2359,6 @@ static void alc_free_kctls(struct hda_codec *codec)
snd_array_free(&spec->kctls);
}
-static void alc_free_bind_ctls(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- if (spec->bind_ctls.list) {
- struct hda_bind_ctls **ctl = spec->bind_ctls.list;
- int i;
- for (i = 0; i < spec->bind_ctls.used; i++)
- kfree(ctl[i]);
- }
- snd_array_free(&spec->bind_ctls);
-}
-
static void alc_free(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -2434,7 +2369,6 @@ static void alc_free(struct hda_codec *codec)
alc_shutup(codec);
snd_hda_input_jack_free(codec);
alc_free_kctls(codec);
- alc_free_bind_ctls(codec);
kfree(spec);
snd_hda_detach_beep_device(codec);
}
@@ -2497,47 +2431,6 @@ static int alc_codec_rename(struct hda_codec *codec, const char *name)
return 0;
}
-/*
- * Rename codecs appropriately from COEF value
- */
-struct alc_codec_rename_table {
- unsigned int vendor_id;
- unsigned short coef_mask;
- unsigned short coef_bits;
- const char *name;
-};
-
-static struct alc_codec_rename_table rename_tbl[] = {
- { 0x10ec0269, 0xfff0, 0x3010, "ALC277" },
- { 0x10ec0269, 0xf0f0, 0x2010, "ALC259" },
- { 0x10ec0269, 0xf0f0, 0x3010, "ALC258" },
- { 0x10ec0269, 0x00f0, 0x0010, "ALC269VB" },
- { 0x10ec0269, 0xffff, 0xa023, "ALC259" },
- { 0x10ec0269, 0xffff, 0x6023, "ALC281X" },
- { 0x10ec0269, 0x00f0, 0x0020, "ALC269VC" },
- { 0x10ec0887, 0x00f0, 0x0030, "ALC887-VD" },
- { 0x10ec0888, 0x00f0, 0x0030, "ALC888-VD" },
- { 0x10ec0888, 0xf0f0, 0x3020, "ALC886" },
- { 0x10ec0899, 0x2000, 0x2000, "ALC899" },
- { 0x10ec0892, 0xffff, 0x8020, "ALC661" },
- { 0x10ec0892, 0xffff, 0x8011, "ALC661" },
- { 0x10ec0892, 0xffff, 0x4011, "ALC656" },
- { } /* terminator */
-};
-
-static int alc_codec_rename_from_preset(struct hda_codec *codec)
-{
- const struct alc_codec_rename_table *p;
-
- for (p = rename_tbl; p->vendor_id; p++) {
- if (p->vendor_id != codec->vendor_id)
- continue;
- if ((alc_get_coef0(codec) & p->coef_mask) == p->coef_bits)
- return alc_codec_rename(codec, p->name);
- }
- return 0;
-}
-
/*
* Automatic parse of I/O pins from the BIOS configuration
*/
@@ -2546,15 +2439,11 @@ enum {
ALC_CTL_WIDGET_VOL,
ALC_CTL_WIDGET_MUTE,
ALC_CTL_BIND_MUTE,
- ALC_CTL_BIND_VOL,
- ALC_CTL_BIND_SW,
};
static const struct snd_kcontrol_new alc_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
HDA_BIND_MUTE(NULL, 0, 0, 0),
- HDA_BIND_VOL(NULL, 0),
- HDA_BIND_SW(NULL, 0),
};
/* add dynamic controls */
@@ -2595,14 +2484,13 @@ static int add_control_with_pfx(struct alc_spec *spec, int type,
#define __add_pb_sw_ctrl(spec, type, pfx, cidx, val) \
add_control_with_pfx(spec, type, pfx, "Playback", "Switch", cidx, val)
-static const char * const channel_name[4] = {
- "Front", "Surround", "CLFE", "Side"
-};
-
static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
bool can_be_master, int *index)
{
struct auto_pin_cfg *cfg = &spec->autocfg;
+ static const char * const chname[4] = {
+ "Front", "Surround", NULL /*CLFE*/, "Side"
+ };
*index = 0;
if (cfg->line_outs == 1 && !spec->multi_ios &&
@@ -2625,10 +2513,7 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
return "PCM";
break;
}
- if (snd_BUG_ON(ch >= ARRAY_SIZE(channel_name)))
- return "PCM";
-
- return channel_name[ch];
+ return chname[ch];
}
/* create input playback/capture controls for the given pin */
@@ -2663,6 +2548,7 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
hda_nid_t *adc_nids = spec->private_adc_nids;
hda_nid_t *cap_nids = spec->private_capsrc_nids;
int max_nums = ARRAY_SIZE(spec->private_adc_nids);
+ bool indep_capsrc = false;
int i, nums = 0;
nid = codec->start_nid;
@@ -2684,11 +2570,13 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec)
break;
if (type == AC_WID_AUD_SEL) {
cap_nids[nums] = src;
+ indep_capsrc = true;
break;
}
n = snd_hda_get_conn_list(codec, src, &list);
if (n > 1) {
cap_nids[nums] = src;
+ indep_capsrc = true;
break;
} else if (n != 1)
break;
@@ -2889,9 +2777,8 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
if (found_in_nid_list(nid, spec->multiout.dac_nids,
spec->multiout.num_dacs))
continue;
- if (found_in_nid_list(nid, spec->multiout.hp_out_nid,
- ARRAY_SIZE(spec->multiout.hp_out_nid)))
- continue;
+ if (spec->multiout.hp_nid == nid)
+ continue;
if (found_in_nid_list(nid, spec->multiout.extra_out_nid,
ARRAY_SIZE(spec->multiout.extra_out_nid)))
continue;
@@ -2908,29 +2795,6 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin)
return 0;
}
-static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
- const hda_nid_t *pins, hda_nid_t *dacs)
-{
- int i;
-
- if (num_outs && !dacs[0]) {
- dacs[0] = alc_auto_look_for_dac(codec, pins[0]);
- if (!dacs[0])
- return 0;
- }
-
- for (i = 1; i < num_outs; i++)
- dacs[i] = get_dac_if_single(codec, pins[i]);
- for (i = 1; i < num_outs; i++) {
- if (!dacs[i])
- dacs[i] = alc_auto_look_for_dac(codec, pins[i]);
- }
- return 0;
-}
-
-static int alc_auto_fill_multi_ios(struct hda_codec *codec,
- unsigned int location);
-
/* fill in the dac_nids table from the parsed pin configuration */
static int alc_auto_fill_dac_nids(struct hda_codec *codec)
{
@@ -2942,7 +2806,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
again:
/* set num_dacs once to full for alc_auto_look_for_dac() */
spec->multiout.num_dacs = cfg->line_outs;
- spec->multiout.hp_out_nid[0] = 0;
+ spec->multiout.hp_nid = 0;
spec->multiout.extra_out_nid[0] = 0;
memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
spec->multiout.dac_nids = spec->private_dac_nids;
@@ -2953,7 +2817,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
spec->private_dac_nids[i] =
get_dac_if_single(codec, cfg->line_out_pins[i]);
if (cfg->hp_outs)
- spec->multiout.hp_out_nid[0] =
+ spec->multiout.hp_nid =
get_dac_if_single(codec, cfg->hp_pins[0]);
if (cfg->speaker_outs)
spec->multiout.extra_out_nid[0] =
@@ -2985,58 +2849,24 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
sizeof(hda_nid_t) * (cfg->line_outs - i - 1));
}
- if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
- /* try to fill multi-io first */
- unsigned int location, defcfg;
- int num_pins;
-
- defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]);
- location = get_defcfg_location(defcfg);
-
- num_pins = alc_auto_fill_multi_ios(codec, location);
- if (num_pins > 0) {
- spec->multi_ios = num_pins;
- spec->ext_channel_count = 2;
- spec->multiout.num_dacs = num_pins + 1;
- }
- }
-
- if (cfg->line_out_type != AUTO_PIN_HP_OUT)
- alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins,
- spec->multiout.hp_out_nid);
- if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT)
- alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins,
- spec->multiout.extra_out_nid);
+ if (cfg->hp_outs && !spec->multiout.hp_nid)
+ spec->multiout.hp_nid =
+ alc_auto_look_for_dac(codec, cfg->hp_pins[0]);
+ if (cfg->speaker_outs && !spec->multiout.extra_out_nid[0])
+ spec->multiout.extra_out_nid[0] =
+ alc_auto_look_for_dac(codec, cfg->speaker_pins[0]);
return 0;
}
-static inline unsigned int get_ctl_pos(unsigned int data)
-{
- hda_nid_t nid = get_amp_nid_(data);
- unsigned int dir = get_amp_direction_(data);
- return (nid << 1) | dir;
-}
-
-#define is_ctl_used(bits, data) \
- test_bit(get_ctl_pos(data), bits)
-#define mark_ctl_usage(bits, data) \
- set_bit(get_ctl_pos(data), bits)
-
static int alc_auto_add_vol_ctl(struct hda_codec *codec,
const char *pfx, int cidx,
hda_nid_t nid, unsigned int chs)
{
- struct alc_spec *spec = codec->spec;
- unsigned int val;
if (!nid)
return 0;
- val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT);
- if (is_ctl_used(spec->vol_ctls, val) && chs != 2) /* exclude LFE */
- return 0;
- mark_ctl_usage(spec->vol_ctls, val);
return __add_pb_vol_ctrl(codec->spec, ALC_CTL_WIDGET_VOL, pfx, cidx,
- val);
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
}
#define alc_auto_add_stereo_vol(codec, pfx, cidx, nid) \
@@ -3049,7 +2879,6 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec,
const char *pfx, int cidx,
hda_nid_t nid, unsigned int chs)
{
- struct alc_spec *spec = codec->spec;
int wid_type;
int type;
unsigned long val;
@@ -3066,9 +2895,6 @@ static int alc_auto_add_sw_ctl(struct hda_codec *codec,
type = ALC_CTL_BIND_MUTE;
val = HDA_COMPOSE_AMP_VAL(nid, chs, 2, HDA_INPUT);
}
- if (is_ctl_used(spec->sw_ctls, val) && chs != 2) /* exclude LFE */
- return 0;
- mark_ctl_usage(spec->sw_ctls, val);
return __add_pb_sw_ctrl(codec->spec, type, pfx, cidx, val);
}
@@ -3129,7 +2955,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
sw = alc_look_for_out_mute_nid(codec, pin, dac);
vol = alc_look_for_out_vol_nid(codec, pin, dac);
name = alc_get_line_out_pfx(spec, i, true, &index);
- if (!name || !strcmp(name, "CLFE")) {
+ if (!name) {
/* Center/LFE */
err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1);
if (err < 0)
@@ -3155,24 +2981,23 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
return 0;
}
+/* add playback controls for speaker and HP outputs */
static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
- hda_nid_t dac, const char *pfx)
+ hda_nid_t dac, const char *pfx)
{
struct alc_spec *spec = codec->spec;
hda_nid_t sw, vol;
int err;
+ if (!pin)
+ return 0;
if (!dac) {
- unsigned int val;
/* the corresponding DAC is already occupied */
if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
return 0; /* no way */
/* create a switch only */
- val = HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT);
- if (is_ctl_used(spec->sw_ctls, val))
- return 0; /* already created */
- mark_ctl_usage(spec->sw_ctls, val);
- return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val);
+ return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
}
sw = alc_look_for_out_mute_nid(codec, pin, dac);
@@ -3186,112 +3011,20 @@ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
return 0;
}
-static struct hda_bind_ctls *new_bind_ctl(struct hda_codec *codec,
- unsigned int nums,
- struct hda_ctl_ops *ops)
-{
- struct alc_spec *spec = codec->spec;
- struct hda_bind_ctls **ctlp, *ctl;
- snd_array_init(&spec->bind_ctls, sizeof(ctl), 8);
- ctlp = snd_array_new(&spec->bind_ctls);
- if (!ctlp)
- return NULL;
- ctl = kzalloc(sizeof(*ctl) + sizeof(long) * (nums + 1), GFP_KERNEL);
- *ctlp = ctl;
- if (ctl)
- ctl->ops = ops;
- return ctl;
-}
-
-/* add playback controls for speaker and HP outputs */
-static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins,
- const hda_nid_t *pins,
- const hda_nid_t *dacs,
- const char *pfx)
-{
- struct alc_spec *spec = codec->spec;
- struct hda_bind_ctls *ctl;
- char name[32];
- int i, n, err;
-
- if (!num_pins || !pins[0])
- return 0;
-
- if (num_pins == 1) {
- hda_nid_t dac = *dacs;
- if (!dac)
- dac = spec->multiout.dac_nids[0];
- return alc_auto_create_extra_out(codec, *pins, dac, pfx);
- }
-
- if (dacs[num_pins - 1]) {
- /* OK, we have a multi-output system with individual volumes */
- for (i = 0; i < num_pins; i++) {
- snprintf(name, sizeof(name), "%s %s",
- pfx, channel_name[i]);
- err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
- name);
- if (err < 0)
- return err;
- }
- return 0;
- }
-
- /* Let's create a bind-controls */
- ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_sw);
- if (!ctl)
- return -ENOMEM;
- n = 0;
- for (i = 0; i < num_pins; i++) {
- if (get_wcaps(codec, pins[i]) & AC_WCAP_OUT_AMP)
- ctl->values[n++] =
- HDA_COMPOSE_AMP_VAL(pins[i], 3, 0, HDA_OUTPUT);
- }
- if (n) {
- snprintf(name, sizeof(name), "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_SW, name, 0, (long)ctl);
- if (err < 0)
- return err;
- }
-
- ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_vol);
- if (!ctl)
- return -ENOMEM;
- n = 0;
- for (i = 0; i < num_pins; i++) {
- hda_nid_t vol;
- if (!pins[i] || !dacs[i])
- continue;
- vol = alc_look_for_out_vol_nid(codec, pins[i], dacs[i]);
- if (vol)
- ctl->values[n++] =
- HDA_COMPOSE_AMP_VAL(vol, 3, 0, HDA_OUTPUT);
- }
- if (n) {
- snprintf(name, sizeof(name), "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_BIND_VOL, name, 0, (long)ctl);
- if (err < 0)
- return err;
- }
- return 0;
-}
-
static int alc_auto_create_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- return alc_auto_create_extra_outs(codec, spec->autocfg.hp_outs,
- spec->autocfg.hp_pins,
- spec->multiout.hp_out_nid,
- "Headphone");
+ return alc_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
+ spec->multiout.hp_nid,
+ "Headphone");
}
static int alc_auto_create_speaker_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- return alc_auto_create_extra_outs(codec, spec->autocfg.speaker_outs,
- spec->autocfg.speaker_pins,
- spec->multiout.extra_out_nid,
- "Speaker");
+ return alc_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0],
+ spec->multiout.extra_out_nid[0],
+ "Speaker");
}
static void alc_auto_set_output_and_unmute(struct hda_codec *codec,
@@ -3348,39 +3081,16 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- int i;
- hda_nid_t pin, dac;
+ hda_nid_t pin;
- for (i = 0; i < spec->autocfg.hp_outs; i++) {
- if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
- break;
- pin = spec->autocfg.hp_pins[i];
- if (!pin)
- break;
- dac = spec->multiout.hp_out_nid[i];
- if (!dac) {
- if (i > 0 && spec->multiout.hp_out_nid[0])
- dac = spec->multiout.hp_out_nid[0];
- else
- dac = spec->multiout.dac_nids[0];
- }
- alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
- }
- for (i = 0; i < spec->autocfg.speaker_outs; i++) {
- if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
- break;
- pin = spec->autocfg.speaker_pins[i];
- if (!pin)
- break;
- dac = spec->multiout.extra_out_nid[i];
- if (!dac) {
- if (i > 0 && spec->multiout.extra_out_nid[0])
- dac = spec->multiout.extra_out_nid[0];
- else
- dac = spec->multiout.dac_nids[0];
- }
- alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
- }
+ pin = spec->autocfg.hp_pins[0];
+ if (pin)
+ alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
+ spec->multiout.hp_nid);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+ spec->multiout.extra_out_nid[0]);
}
/*
@@ -3391,7 +3101,6 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec,
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
- hda_nid_t prime_dac = spec->private_dac_nids[0];
int type, i, num_pins = 0;
for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) {
@@ -3419,13 +3128,8 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec,
}
}
spec->multiout.num_dacs = 1;
- if (num_pins < 2) {
- /* clear up again */
- memset(spec->private_dac_nids, 0,
- sizeof(spec->private_dac_nids));
- spec->private_dac_nids[0] = prime_dac;
+ if (num_pins < 2)
return 0;
- }
return num_pins;
}
@@ -3511,11 +3215,36 @@ static const struct snd_kcontrol_new alc_auto_channel_mode_enum = {
.put = alc_auto_ch_mode_put,
};
-static int alc_auto_add_multi_channel_mode(struct hda_codec *codec)
+static int alc_auto_add_multi_channel_mode(struct hda_codec *codec,
+ int (*fill_dac)(struct hda_codec *))
{
struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int location, defcfg;
+ int num_pins;
+
+ if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->hp_outs == 1) {
+ /* use HP as primary out */
+ cfg->speaker_outs = cfg->line_outs;
+ memcpy(cfg->speaker_pins, cfg->line_out_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->line_outs = cfg->hp_outs;
+ memcpy(cfg->line_out_pins, cfg->hp_pins, sizeof(cfg->hp_pins));
+ cfg->hp_outs = 0;
+ memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+ if (fill_dac)
+ fill_dac(codec);
+ }
+ if (cfg->line_outs != 1 ||
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ return 0;
- if (spec->multi_ios > 0) {
+ defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]);
+ location = get_defcfg_location(defcfg);
+
+ num_pins = alc_auto_fill_multi_ios(codec, location);
+ if (num_pins > 0) {
struct snd_kcontrol_new *knew;
knew = alc_kcontrol_new(spec);
@@ -3525,6 +3254,10 @@ static int alc_auto_add_multi_channel_mode(struct hda_codec *codec)
knew->name = kstrdup("Channel Mode", GFP_KERNEL);
if (!knew->name)
return -ENOMEM;
+
+ spec->multi_ios = num_pins;
+ spec->ext_channel_count = 2;
+ spec->multiout.num_dacs = num_pins + 1;
}
return 0;
}
@@ -3807,42 +3540,27 @@ static int alc_parse_auto_config(struct hda_codec *codec,
const hda_nid_t *ssid_nids)
{
struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
int err;
- err = snd_hda_parse_pin_defcfg(codec, cfg, ignore_nids,
- spec->parse_flags);
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ ignore_nids);
if (err < 0)
return err;
- if (!cfg->line_outs) {
- if (cfg->dig_outs || cfg->dig_in_pin) {
+ if (!spec->autocfg.line_outs) {
+ if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
spec->multiout.max_channels = 2;
spec->no_analog = 1;
goto dig_only;
}
return 0; /* can't find valid BIOS pin config */
}
-
- if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
- cfg->line_outs <= cfg->hp_outs) {
- /* use HP as primary out */
- cfg->speaker_outs = cfg->line_outs;
- memcpy(cfg->speaker_pins, cfg->line_out_pins,
- sizeof(cfg->speaker_pins));
- cfg->line_outs = cfg->hp_outs;
- memcpy(cfg->line_out_pins, cfg->hp_pins, sizeof(cfg->hp_pins));
- cfg->hp_outs = 0;
- memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
- cfg->line_out_type = AUTO_PIN_HP_OUT;
- }
-
err = alc_auto_fill_dac_nids(codec);
if (err < 0)
return err;
- err = alc_auto_add_multi_channel_mode(codec);
+ err = alc_auto_add_multi_channel_mode(codec, alc_auto_fill_dac_nids);
if (err < 0)
return err;
- err = alc_auto_create_multi_out_ctls(codec, cfg);
+ err = alc_auto_create_multi_out_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
err = alc_auto_create_hp_out(codec);
@@ -3945,8 +3663,10 @@ static int patch_alc880(struct hda_codec *codec)
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc880_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -3971,8 +3691,10 @@ static int patch_alc880(struct hda_codec *codec)
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -3987,10 +3709,6 @@ static int patch_alc880(struct hda_codec *codec)
#endif
return 0;
-
- error:
- alc_free(codec);
- return err;
}
@@ -4072,8 +3790,10 @@ static int patch_alc260(struct hda_codec *codec)
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc260_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -4098,8 +3818,10 @@ static int patch_alc260(struct hda_codec *codec)
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
}
@@ -4117,10 +3839,6 @@ static int patch_alc260(struct hda_codec *codec)
#endif
return 0;
-
- error:
- alc_free(codec);
- return err;
}
@@ -4147,7 +3865,6 @@ enum {
PINFIX_LENOVO_Y530,
PINFIX_PB_M5210,
PINFIX_ACER_ASPIRE_7736,
- PINFIX_ASUS_W90V,
};
static const struct alc_fixup alc882_fixups[] = {
@@ -4179,18 +3896,10 @@ static const struct alc_fixup alc882_fixups[] = {
.type = ALC_FIXUP_SKU,
.v.sku = ALC_FIXUP_SKU_IGNORE,
},
- [PINFIX_ASUS_W90V] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x16, 0x99130110 }, /* fix sequence for CLFE */
- { }
- }
- },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210),
- SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", PINFIX_ASUS_W90V),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
@@ -4237,10 +3946,6 @@ static int patch_alc882(struct hda_codec *codec)
break;
}
- err = alc_codec_rename_from_preset(codec);
- if (err < 0)
- goto error;
-
board_config = alc_board_config(codec, ALC882_MODEL_LAST,
alc882_models, alc882_cfg_tbl);
@@ -4264,8 +3969,10 @@ static int patch_alc882(struct hda_codec *codec)
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc882_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -4290,8 +3997,10 @@ static int patch_alc882(struct hda_codec *codec)
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -4310,10 +4019,6 @@ static int patch_alc882(struct hda_codec *codec)
#endif
return 0;
-
- error:
- alc_free(codec);
- return err;
}
@@ -4418,8 +4123,10 @@ static int patch_alc262(struct hda_codec *codec)
if (board_config == ALC_MODEL_AUTO) {
/* automatic parse from the BIOS config */
err = alc262_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
else if (!err) {
printk(KERN_INFO
@@ -4444,8 +4151,10 @@ static int patch_alc262(struct hda_codec *codec)
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -4465,10 +4174,6 @@ static int patch_alc262(struct hda_codec *codec)
#endif
return 0;
-
- error:
- alc_free(codec);
- return err;
}
/*
@@ -4517,9 +4222,14 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
/*
*/
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc268_quirks.c"
+#endif
+
static int patch_alc268(struct hda_codec *codec)
{
struct alc_spec *spec;
+ int board_config;
int i, has_beep, err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -4530,10 +4240,38 @@ static int patch_alc268(struct hda_codec *codec)
/* ALC268 has no aa-loopback mixer */
- /* automatic parse from the BIOS config */
- err = alc268_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ board_config = alc_board_config(codec, ALC268_MODEL_LAST,
+ alc268_models, alc268_cfg_tbl);
+
+ if (board_config < 0)
+ board_config = alc_board_codec_sid_config(codec,
+ ALC268_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc268_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC268_3ST;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc268_presets[board_config]);
has_beep = 0;
for (i = 0; i < spec->num_mixers; i++) {
@@ -4545,8 +4283,10 @@ static int patch_alc268(struct hda_codec *codec)
if (has_beep) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
/* override the amp caps for beep generator */
snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
@@ -4568,16 +4308,13 @@ static int patch_alc268(struct hda_codec *codec)
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
alc_init_jacks(codec);
return 0;
-
- error:
- alc_free(codec);
- return err;
}
/*
@@ -4671,9 +4408,9 @@ static void alc269_toggle_power_output(struct hda_codec *codec, int power_up)
static void alc269_shutup(struct hda_codec *codec)
{
- if ((alc_get_coef0(codec) & 0x00ff) == 0x017)
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017)
alc269_toggle_power_output(codec, 0);
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
alc269_toggle_power_output(codec, 0);
msleep(150);
}
@@ -4682,19 +4419,19 @@ static void alc269_shutup(struct hda_codec *codec)
#ifdef CONFIG_PM
static int alc269_resume(struct hda_codec *codec)
{
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
alc269_toggle_power_output(codec, 0);
msleep(150);
}
codec->patch_ops.init(codec);
- if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
alc269_toggle_power_output(codec, 1);
msleep(200);
}
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018)
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018)
alc269_toggle_power_output(codec, 1);
snd_hda_codec_resume_amp(codec);
@@ -4747,46 +4484,6 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
}
-static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
- const struct alc_fixup *fix, int action)
-{
- int coef;
-
- if (action != ALC_FIXUP_ACT_INIT)
- return;
- /* The digital-mic unit sends PDM (differential signal) instead of
- * the standard PCM, thus you can't record a valid mono stream as is.
- * Below is a workaround specific to ALC269 to control the dmic
- * signal source as mono.
- */
- coef = alc_read_coef_idx(codec, 0x07);
- alc_write_coef_idx(codec, 0x07, coef | 0x80);
-}
-
-static void alc269_quanta_automute(struct hda_codec *codec)
-{
- update_outputs(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);
-}
-
-static void alc269_fixup_quanta_mute(struct hda_codec *codec,
- const struct alc_fixup *fix, int action)
-{
- struct alc_spec *spec = codec->spec;
- if (action != ALC_FIXUP_ACT_PROBE)
- return;
- spec->automute_hook = alc269_quanta_automute;
-}
-
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4797,13 +4494,6 @@ enum {
ALC275_FIXUP_SONY_HWEQ,
ALC271_FIXUP_DMIC,
ALC269_FIXUP_PCM_44K,
- ALC269_FIXUP_STEREO_DMIC,
- ALC269_FIXUP_QUANTA_MUTE,
- ALC269_FIXUP_LIFEBOOK,
- ALC269_FIXUP_AMIC,
- ALC269_FIXUP_DMIC,
- ALC269VB_FIXUP_AMIC,
- ALC269VB_FIXUP_DMIC,
};
static const struct alc_fixup alc269_fixups[] = {
@@ -4866,144 +4556,23 @@ static const struct alc_fixup alc269_fixups[] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc269_fixup_pcm_44k,
},
- [ALC269_FIXUP_STEREO_DMIC] = {
- .type = ALC_FIXUP_FUNC,
- .v.func = alc269_fixup_stereo_dmic,
- },
- [ALC269_FIXUP_QUANTA_MUTE] = {
- .type = ALC_FIXUP_FUNC,
- .v.func = alc269_fixup_quanta_mute,
- },
- [ALC269_FIXUP_LIFEBOOK] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x1a, 0x2101103f }, /* dock line-out */
- { 0x1b, 0x23a11040 }, /* dock mic-in */
- { }
- },
- .chained = true,
- .chain_id = ALC269_FIXUP_QUANTA_MUTE
- },
- [ALC269_FIXUP_AMIC] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x15, 0x0121401f }, /* HP out */
- { 0x18, 0x01a19c20 }, /* mic */
- { 0x19, 0x99a3092f }, /* int-mic */
- { }
- },
- },
- [ALC269_FIXUP_DMIC] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x12, 0x99a3092f }, /* int-mic */
- { 0x14, 0x99130110 }, /* speaker */
- { 0x15, 0x0121401f }, /* HP out */
- { 0x18, 0x01a19c20 }, /* mic */
- { }
- },
- },
- [ALC269VB_FIXUP_AMIC] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x18, 0x01a19c20 }, /* mic */
- { 0x19, 0x99a3092f }, /* int-mic */
- { 0x21, 0x0121401f }, /* HP out */
- { }
- },
- },
- [ALC269_FIXUP_DMIC] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x12, 0x99a3092f }, /* int-mic */
- { 0x14, 0x99130110 }, /* speaker */
- { 0x18, 0x01a19c20 }, /* mic */
- { 0x21, 0x0121401f }, /* HP out */
- { }
- },
- },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
- SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
- SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
-
-#if 1
- /* Below is a quirk table taken from the old code.
- * Basically the device should work as is without the fixup table.
- * If BIOS doesn't give a proper info, enable the corresponding
- * fixup entry.
- */
- SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
- ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x11b3, "ASUS K52DR", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x11e3, "ASUS U33Jc", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1273, "ASUS UL80Jt", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1283, "ASUS U53Jc", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS N82JV", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x12d3, "ASUS N61Jv", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x13a3, "ASUS UL30Vt", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1373, "ASUS G73JX", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1383, "ASUS UJ30Jc", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x13d3, "ASUS N61JA", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1413, "ASUS UL50", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1443, "ASUS UL30", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1453, "ASUS M60Jv", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1483, "ASUS UL80", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x14f3, "ASUS F83Vf", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS UL20", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1513, "ASUS UX30", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1593, "ASUS N51Vn", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15a3, "ASUS N60Jv", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15b3, "ASUS N60Dp", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15c3, "ASUS N70De", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x15e3, "ASUS F83T", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1643, "ASUS M60J", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_FIXUP_DMIC),
- SND_PCI_QUIRK(0x17aa, 0x3be9, "Quanta Wistron", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x17ff, 0x059a, "Quanta EL3", ALC269_FIXUP_DMIC),
- SND_PCI_QUIRK(0x17ff, 0x059b, "Quanta JR1", ALC269_FIXUP_DMIC),
-#endif
- {}
-};
-
-static const struct alc_model_fixup alc269_fixup_models[] = {
- {.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"},
- {.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"},
{}
};
@@ -5012,23 +4581,23 @@ static int alc269_fill_coef(struct hda_codec *codec)
{
int val;
- if ((alc_get_coef0(codec) & 0x00ff) < 0x015) {
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) < 0x015) {
alc_write_coef_idx(codec, 0xf, 0x960b);
alc_write_coef_idx(codec, 0xe, 0x8817);
}
- if ((alc_get_coef0(codec) & 0x00ff) == 0x016) {
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x016) {
alc_write_coef_idx(codec, 0xf, 0x960b);
alc_write_coef_idx(codec, 0xe, 0x8814);
}
- if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) {
val = alc_read_coef_idx(codec, 0x04);
/* Power up output pin */
alc_write_coef_idx(codec, 0x04, val | (1<<11));
}
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
+ if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) {
val = alc_read_coef_idx(codec, 0xd);
if ((val & 0x0c00) >> 10 != 0x1) {
/* Capless ramp up clock control */
@@ -5052,10 +4621,15 @@ static int alc269_fill_coef(struct hda_codec *codec)
/*
*/
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc269_quirks.c"
+#endif
+
static int patch_alc269(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err = 0;
+ int board_config, coef;
+ int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -5067,41 +4641,72 @@ static int patch_alc269(struct hda_codec *codec)
alc_auto_parse_customize_define(codec);
- err = alc_codec_rename_from_preset(codec);
- if (err < 0)
- goto error;
-
if (codec->vendor_id == 0x10ec0269) {
spec->codec_variant = ALC269_TYPE_ALC269VA;
- switch (alc_get_coef0(codec) & 0x00f0) {
- case 0x0010:
+ coef = alc_read_coef_idx(codec, 0);
+ if ((coef & 0x00f0) == 0x0010) {
if (codec->bus->pci->subsystem_vendor == 0x1025 &&
- spec->cdefine.platform_type == 1)
- err = alc_codec_rename(codec, "ALC271X");
+ spec->cdefine.platform_type == 1) {
+ alc_codec_rename(codec, "ALC271X");
+ } else if ((coef & 0xf000) == 0x2000) {
+ alc_codec_rename(codec, "ALC259");
+ } else if ((coef & 0xf000) == 0x3000) {
+ alc_codec_rename(codec, "ALC258");
+ } else if ((coef & 0xfff0) == 0x3010) {
+ alc_codec_rename(codec, "ALC277");
+ } else {
+ alc_codec_rename(codec, "ALC269VB");
+ }
spec->codec_variant = ALC269_TYPE_ALC269VB;
- break;
- case 0x0020:
- if (codec->bus->pci->subsystem_vendor == 0x17aa &&
- codec->bus->pci->subsystem_device == 0x21f3)
- err = alc_codec_rename(codec, "ALC3202");
+ } else if ((coef & 0x00f0) == 0x0020) {
+ if (coef == 0xa023)
+ alc_codec_rename(codec, "ALC259");
+ else if (coef == 0x6023)
+ alc_codec_rename(codec, "ALC281X");
+ else if (codec->bus->pci->subsystem_vendor == 0x17aa &&
+ codec->bus->pci->subsystem_device == 0x21f3)
+ alc_codec_rename(codec, "ALC3202");
+ else
+ alc_codec_rename(codec, "ALC269VC");
spec->codec_variant = ALC269_TYPE_ALC269VC;
- break;
- default:
+ } else
alc_fix_pll_init(codec, 0x20, 0x04, 15);
- }
- if (err < 0)
- goto error;
alc269_fill_coef(codec);
}
- alc_pick_fixup(codec, alc269_fixup_models,
- alc269_fixup_tbl, alc269_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ board_config = alc_board_config(codec, ALC269_MODEL_LAST,
+ alc269_models, alc269_cfg_tbl);
- /* automatic parse from the BIOS config */
- err = alc269_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc269_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC269_BASIC;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc269_presets[board_config]);
if (!spec->no_analog && !spec->adc_nids) {
alc_auto_fill_adc_caps(codec);
@@ -5114,8 +4719,10 @@ static int patch_alc269(struct hda_codec *codec)
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
}
@@ -5127,7 +4734,8 @@ static int patch_alc269(struct hda_codec *codec)
#ifdef CONFIG_PM
codec->patch_ops.resume = alc269_resume;
#endif
- spec->init_hook = alc_auto_init_std;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc269_shutup;
alc_init_jacks(codec);
@@ -5139,10 +4747,6 @@ static int patch_alc269(struct hda_codec *codec)
#endif
return 0;
-
- error:
- alc_free(codec);
- return err;
}
/*
@@ -5190,9 +4794,14 @@ static const struct snd_pci_quirk alc861_fixup_tbl[] = {
/*
*/
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc861_quirks.c"
+#endif
+
static int patch_alc861(struct hda_codec *codec)
{
struct alc_spec *spec;
+ int board_config;
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5203,13 +4812,39 @@ static int patch_alc861(struct hda_codec *codec)
spec->mixer_nid = 0x15;
- alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ board_config = alc_board_config(codec, ALC861_MODEL_LAST,
+ alc861_models, alc861_cfg_tbl);
- /* automatic parse from the BIOS config */
- err = alc861_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc861_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC861_3ST_DIG;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc861_presets[board_config]);
if (!spec->no_analog && !spec->adc_nids) {
alc_auto_fill_adc_caps(codec);
@@ -5222,8 +4857,10 @@ static int patch_alc861(struct hda_codec *codec)
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
}
@@ -5232,18 +4869,18 @@ static int patch_alc861(struct hda_codec *codec)
alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
+ if (board_config == ALC_MODEL_AUTO) {
+ spec->init_hook = alc_auto_init_std;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->power_hook = alc_power_eapd;
+#endif
+ }
#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->power_hook = alc_power_eapd;
if (!spec->loopback.amplist)
spec->loopback.amplist = alc861_loopbacks;
#endif
return 0;
-
- error:
- alc_free(codec);
- return err;
}
/*
@@ -5265,41 +4902,24 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
}
enum {
- ALC660VD_FIX_ASUS_GPIO1,
- ALC861VD_FIX_DALLAS,
+ ALC660VD_FIX_ASUS_GPIO1
};
-/* exclude VREF80 */
-static void alc861vd_fixup_dallas(struct hda_codec *codec,
- const struct alc_fixup *fix, int action)
-{
- if (action == ALC_FIXUP_ACT_PRE_PROBE) {
- snd_hda_override_pin_caps(codec, 0x18, 0x00001714);
- snd_hda_override_pin_caps(codec, 0x19, 0x0000171c);
- }
-}
-
+/* reset GPIO1 */
static const struct alc_fixup alc861vd_fixups[] = {
[ALC660VD_FIX_ASUS_GPIO1] = {
.type = ALC_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
- /* reset GPIO1 */
{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
{ }
}
},
- [ALC861VD_FIX_DALLAS] = {
- .type = ALC_FIXUP_FUNC,
- .v.func = alc861vd_fixup_dallas,
- },
};
static const struct snd_pci_quirk alc861vd_fixup_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_FIX_DALLAS),
SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
- SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_FIX_DALLAS),
{}
};
@@ -5311,10 +4931,14 @@ static const struct hda_verb alc660vd_eapd_verbs[] = {
/*
*/
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc861vd_quirks.c"
+#endif
+
static int patch_alc861vd(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err;
+ int err, board_config;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -5324,13 +4948,39 @@ static int patch_alc861vd(struct hda_codec *codec)
spec->mixer_nid = 0x0b;
- alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ board_config = alc_board_config(codec, ALC861VD_MODEL_LAST,
+ alc861vd_models, alc861vd_cfg_tbl);
- /* automatic parse from the BIOS config */
- err = alc861vd_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc861vd_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC861VD_3ST;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc861vd_presets[board_config]);
if (codec->vendor_id == 0x10ec0660) {
/* always turn on EAPD */
@@ -5348,8 +4998,10 @@ static int patch_alc861vd(struct hda_codec *codec)
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x23);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
@@ -5359,7 +5011,8 @@ static int patch_alc861vd(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
@@ -5367,10 +5020,6 @@ static int patch_alc861vd(struct hda_codec *codec)
#endif
return 0;
-
- error:
- alc_free(codec);
- return err;
}
/*
@@ -5428,14 +5077,6 @@ enum {
ALC662_FIXUP_CZC_P10T,
ALC662_FIXUP_SKU_IGNORE,
ALC662_FIXUP_HP_RP5800,
- ALC662_FIXUP_ASUS_MODE1,
- ALC662_FIXUP_ASUS_MODE2,
- ALC662_FIXUP_ASUS_MODE3,
- ALC662_FIXUP_ASUS_MODE4,
- ALC662_FIXUP_ASUS_MODE5,
- ALC662_FIXUP_ASUS_MODE6,
- ALC662_FIXUP_ASUS_MODE7,
- ALC662_FIXUP_ASUS_MODE8,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -5477,204 +5118,37 @@ static const struct alc_fixup alc662_fixups[] = {
.chained = true,
.chain_id = ALC662_FIXUP_SKU_IGNORE
},
- [ALC662_FIXUP_ASUS_MODE1] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x18, 0x01a19c20 }, /* mic */
- { 0x19, 0x99a3092f }, /* int-mic */
- { 0x21, 0x0121401f }, /* HP out */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
- [ALC662_FIXUP_ASUS_MODE2] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x18, 0x01a19820 }, /* mic */
- { 0x19, 0x99a3092f }, /* int-mic */
- { 0x1b, 0x0121401f }, /* HP out */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
- [ALC662_FIXUP_ASUS_MODE3] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x15, 0x0121441f }, /* HP */
- { 0x18, 0x01a19840 }, /* mic */
- { 0x19, 0x99a3094f }, /* int-mic */
- { 0x21, 0x01211420 }, /* HP2 */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
- [ALC662_FIXUP_ASUS_MODE4] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x16, 0x99130111 }, /* speaker */
- { 0x18, 0x01a19840 }, /* mic */
- { 0x19, 0x99a3094f }, /* int-mic */
- { 0x21, 0x0121441f }, /* HP */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
- [ALC662_FIXUP_ASUS_MODE5] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x15, 0x0121441f }, /* HP */
- { 0x16, 0x99130111 }, /* speaker */
- { 0x18, 0x01a19840 }, /* mic */
- { 0x19, 0x99a3094f }, /* int-mic */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
- [ALC662_FIXUP_ASUS_MODE6] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x15, 0x01211420 }, /* HP2 */
- { 0x18, 0x01a19840 }, /* mic */
- { 0x19, 0x99a3094f }, /* int-mic */
- { 0x1b, 0x0121441f }, /* HP */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
- [ALC662_FIXUP_ASUS_MODE7] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x17, 0x99130111 }, /* speaker */
- { 0x18, 0x01a19840 }, /* mic */
- { 0x19, 0x99a3094f }, /* int-mic */
- { 0x1b, 0x01214020 }, /* HP */
- { 0x21, 0x0121401f }, /* HP */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
- [ALC662_FIXUP_ASUS_MODE8] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x99130110 }, /* speaker */
- { 0x12, 0x99a30970 }, /* int-mic */
- { 0x15, 0x01214020 }, /* HP */
- { 0x17, 0x99130111 }, /* speaker */
- { 0x18, 0x01a19840 }, /* mic */
- { 0x21, 0x0121401f }, /* HP */
- { }
- },
- .chained = true,
- .chain_id = ALC662_FIXUP_SKU_IGNORE
- },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
- SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
-
-#if 0
- /* Below is a quirk table taken from the old code.
- * Basically the device should work as is without the fixup table.
- * If BIOS doesn't give a proper info, enable the corresponding
- * fixup entry.
- */
- SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC662_FIXUP_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1303, "ASUS G60J", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1333, "ASUS G60Jx", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x13e3, "ASUS N71JA", ALC662_FIXUP_ASUS_MODE7),
- SND_PCI_QUIRK(0x1043, 0x1463, "ASUS N71", ALC662_FIXUP_ASUS_MODE7),
- SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G72", ALC662_FIXUP_ASUS_MODE8),
- SND_PCI_QUIRK(0x1043, 0x1563, "ASUS N90", ALC662_FIXUP_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x15d3, "ASUS N50SF F50SF", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS K40C K50C", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1733, "ASUS N81De", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC662_FIXUP_ASUS_MODE6),
- SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC662_FIXUP_ASUS_MODE6),
- SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1793, "ASUS F50GX", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x17b3, "ASUS F70SL", ALC662_FIXUP_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x17f3, "ASUS X58LE", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC662_FIXUP_ASUS_MODE5),
- SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC662_FIXUP_ASUS_MODE6),
- SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1853, "ASUS F50Z", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC662_FIXUP_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC662_FIXUP_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x18b3, "ASUS N80Vc", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x18c3, "ASUS VX5", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS N81Te", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x18f3, "ASUS N505Tp", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x1943, "ASUS Vx3V", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC662_FIXUP_ASUS_MODE3),
- SND_PCI_QUIRK(0x1043, 0x1983, "ASUS N5051A", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_FIXUP_ASUS_MODE2),
- SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC662_FIXUP_ASUS_MODE1),
- SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC662_FIXUP_ASUS_MODE4),
-#endif
{}
};
static const struct alc_model_fixup alc662_fixup_models[] = {
{.id = ALC272_FIXUP_MARIO, .name = "mario"},
- {.id = ALC662_FIXUP_ASUS_MODE1, .name = "asus-mode1"},
- {.id = ALC662_FIXUP_ASUS_MODE2, .name = "asus-mode2"},
- {.id = ALC662_FIXUP_ASUS_MODE3, .name = "asus-mode3"},
- {.id = ALC662_FIXUP_ASUS_MODE4, .name = "asus-mode4"},
- {.id = ALC662_FIXUP_ASUS_MODE5, .name = "asus-mode5"},
- {.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"},
- {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"},
- {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"},
{}
};
/*
*/
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc662_quirks.c"
+#endif
+
static int patch_alc662(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err = 0;
+ int err, board_config;
+ int coef;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (!spec)
@@ -5684,31 +5158,50 @@ static int patch_alc662(struct hda_codec *codec)
spec->mixer_nid = 0x0b;
- /* handle multiple HPs as is */
- spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
-
alc_auto_parse_customize_define(codec);
alc_fix_pll_init(codec, 0x20, 0x04, 15);
- err = alc_codec_rename_from_preset(codec);
- if (err < 0)
- goto error;
+ coef = alc_read_coef_idx(codec, 0);
+ if (coef == 0x8020 || coef == 0x8011)
+ alc_codec_rename(codec, "ALC661");
+ else if (coef & (1 << 14) &&
+ codec->bus->pci->subsystem_vendor == 0x1025 &&
+ spec->cdefine.platform_type == 1)
+ alc_codec_rename(codec, "ALC272X");
+ else if (coef == 0x4011)
+ alc_codec_rename(codec, "ALC656");
+
+ board_config = alc_board_config(codec, ALC662_MODEL_LAST,
+ alc662_models, alc662_cfg_tbl);
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
- if ((alc_get_coef0(codec) & (1 << 14)) &&
- codec->bus->pci->subsystem_vendor == 0x1025 &&
- spec->cdefine.platform_type == 1) {
- if (alc_codec_rename(codec, "ALC272X") < 0)
- goto error;
+ if (board_config == ALC_MODEL_AUTO) {
+ alc_pick_fixup(codec, alc662_fixup_models,
+ alc662_fixup_tbl, alc662_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+ /* automatic parse from the BIOS config */
+ err = alc662_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC662_3ST_2ch_DIG;
+ }
+#endif
}
- alc_pick_fixup(codec, alc662_fixup_models,
- alc662_fixup_tbl, alc662_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- /* automatic parse from the BIOS config */
- err = alc662_parse_auto_config(codec);
- if (err < 0)
- goto error;
+ if (board_config != ALC_MODEL_AUTO)
+ setup_preset(codec, &alc662_presets[board_config]);
if (!spec->no_analog && !spec->adc_nids) {
alc_auto_fill_adc_caps(codec);
@@ -5721,8 +5214,10 @@ static int patch_alc662(struct hda_codec *codec)
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0)
- goto error;
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
switch (codec->vendor_id) {
case 0x10ec0662:
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -5742,7 +5237,8 @@ static int patch_alc662(struct hda_codec *codec)
alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
alc_init_jacks(codec);
@@ -5753,10 +5249,32 @@ static int patch_alc662(struct hda_codec *codec)
#endif
return 0;
+}
- error:
- alc_free(codec);
- return err;
+static int patch_alc888(struct hda_codec *codec)
+{
+ if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){
+ kfree(codec->chip_name);
+ if (codec->vendor_id == 0x10ec0887)
+ codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL);
+ else
+ codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
+ if (!codec->chip_name) {
+ alc_free(codec);
+ return -ENOMEM;
+ }
+ return patch_alc662(codec);
+ }
+ return patch_alc882(codec);
+}
+
+static int patch_alc899(struct hda_codec *codec)
+{
+ if ((alc_read_coef_idx(codec, 0) & 0x2000) != 0x2000) {
+ kfree(codec->chip_name);
+ codec->chip_name = kstrdup("ALC898", GFP_KERNEL);
+ }
+ return patch_alc882(codec);
}
/*
@@ -5770,9 +5288,14 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
/*
*/
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+#include "alc680_quirks.c"
+#endif
+
static int patch_alc680(struct hda_codec *codec)
{
struct alc_spec *spec;
+ int board_config;
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5783,11 +5306,43 @@ static int patch_alc680(struct hda_codec *codec)
/* ALC680 has no aa-loopback mixer */
- /* automatic parse from the BIOS config */
- err = alc680_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
+ board_config = alc_board_config(codec, ALC680_MODEL_LAST,
+ alc680_models, alc680_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC_MODEL_AUTO;
+ }
+
+ if (board_config == ALC_MODEL_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc680_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ }
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC680_BASE;
+ }
+#endif
+ }
+
+ if (board_config != ALC_MODEL_AUTO) {
+ setup_preset(codec, &alc680_presets[board_config]);
+#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
+ spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
+#endif
+ }
+
+ if (!spec->no_analog && !spec->adc_nids) {
+ alc_auto_fill_adc_caps(codec);
+ alc_rebuild_imux_for_auto_mic(codec);
+ alc_remove_invalid_adc_nids(codec);
}
if (!spec->no_analog && !spec->cap_mixer)
@@ -5796,7 +5351,8 @@ static int patch_alc680(struct hda_codec *codec)
spec->vmaster_nid = 0x02;
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
+ if (board_config == ALC_MODEL_AUTO)
+ spec->init_hook = alc_auto_init_std;
return 0;
}
@@ -5824,8 +5380,6 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
.patch = patch_alc882 },
{ .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
.patch = patch_alc662 },
- { .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3",
- .patch = patch_alc662 },
{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
@@ -5838,13 +5392,13 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
.patch = patch_alc882 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
- { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
+ { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
.patch = patch_alc882 },
- { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
+ { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
{ .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
- { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 },
+ { .id = 0x10ec0899, .name = "ALC899", .patch = patch_alc899 },
{} /* terminator */
};
diff --git a/trunk/sound/pci/hda/patch_sigmatel.c b/trunk/sound/pci/hda/patch_sigmatel.c
index de4c36027cbe..aa376b59c006 100644
--- a/trunk/sound/pci/hda/patch_sigmatel.c
+++ b/trunk/sound/pci/hda/patch_sigmatel.c
@@ -673,7 +673,6 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
return 0;
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac_vrefout_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
{
@@ -697,7 +696,6 @@ static int stac_vrefout_set(struct hda_codec *codec,
return 1;
}
-#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@@ -2972,9 +2970,8 @@ static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
{
struct sigmatel_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
int j, conn_len;
- hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
unsigned int wcaps, wtype;
conn_len = snd_hda_get_connections(codec, nid, conn,
@@ -3002,21 +2999,10 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
return conn[j];
}
}
-
- /* if all DACs are already assigned, connect to the primary DAC,
- unless we're assigning a secondary headphone */
- fallback_dac = spec->multiout.dac_nids[0];
- if (spec->multiout.hp_nid) {
- for (j = 0; j < cfg->hp_outs; j++)
- if (cfg->hp_pins[j] == nid) {
- fallback_dac = spec->multiout.hp_nid;
- break;
- }
- }
-
+ /* if all DACs are already assigned, connect to the primary DAC */
if (conn_len > 1) {
for (j = 0; j < conn_len; j++) {
- if (conn[j] == fallback_dac) {
+ if (conn[j] == spec->multiout.dac_nids[0]) {
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL, j);
break;
@@ -3791,10 +3777,9 @@ static int is_dual_headphones(struct hda_codec *codec)
}
-static int stac92xx_parse_auto_config(struct hda_codec *codec)
+static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
{
struct sigmatel_spec *spec = codec->spec;
- hda_nid_t dig_out = 0, dig_in = 0;
int hp_swap = 0;
int i, err;
@@ -3977,22 +3962,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec)
if (spec->multiout.max_channels > 2)
spec->surr_switch = 1;
- /* find digital out and in converters */
- for (i = codec->start_nid; i < codec->start_nid + codec->num_nodes; i++) {
- unsigned int wid_caps = get_wcaps(codec, i);
- if (wid_caps & AC_WCAP_DIGITAL) {
- switch (get_wcaps_type(wid_caps)) {
- case AC_WID_AUD_OUT:
- if (!dig_out)
- dig_out = i;
- break;
- case AC_WID_AUD_IN:
- if (!dig_in)
- dig_in = i;
- break;
- }
- }
- }
if (spec->autocfg.dig_outs)
spec->multiout.dig_out_nid = dig_out;
if (dig_in && spec->autocfg.dig_in_pin)
@@ -4159,14 +4128,22 @@ static int stac92xx_add_jack(struct hda_codec *codec,
#ifdef CONFIG_SND_HDA_INPUT_JACK
int def_conf = snd_hda_codec_get_pincfg(codec, nid);
int connectivity = get_defcfg_connect(def_conf);
+ char name[32];
+ int err;
if (connectivity && connectivity != AC_JACK_PORT_FIXED)
return 0;
- return snd_hda_input_jack_add(codec, nid, type, NULL);
-#else
- return 0;
+ snprintf(name, sizeof(name), "%s at %s %s Jack",
+ snd_hda_get_jack_type(def_conf),
+ snd_hda_get_jack_connectivity(def_conf),
+ snd_hda_get_jack_location(def_conf));
+
+ err = snd_hda_input_jack_add(codec, nid, type, name);
+ if (err < 0)
+ return err;
#endif /* CONFIG_SND_HDA_INPUT_JACK */
+ return 0;
}
static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid,
@@ -5296,7 +5273,7 @@ static int patch_stac925x(struct hda_codec *codec)
spec->capvols = stac925x_capvols;
spec->capsws = stac925x_capsws;
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x8, 0x7);
if (!err) {
if (spec->board_config < 0) {
printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -5437,7 +5414,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids);
spec->pwr_nids = stac92hd73xx_pwr_nids;
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x25, 0x27);
if (!err) {
if (spec->board_config < 0) {
@@ -5606,7 +5583,9 @@ static void stac92hd8x_fill_auto_spec(struct hda_codec *codec)
static int patch_stac92hd83xxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
+ hda_nid_t conn[STAC92HD83_DAC_COUNT + 1];
int err;
+ int num_dacs;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -5646,8 +5625,25 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
stac92xx_set_config_regs(codec,
stac92hd83xxx_brd_tbl[spec->board_config]);
- if (spec->board_config != STAC_92HD83XXX_PWR_REF)
+ switch (codec->vendor_id) {
+ case 0x111d76d1:
+ case 0x111d76d9:
+ case 0x111d76e5:
+ case 0x111d7666:
+ case 0x111d7667:
+ case 0x111d7668:
+ case 0x111d7669:
+ case 0x111d76e3:
+ case 0x111d7604:
+ case 0x111d76d4:
+ case 0x111d7605:
+ case 0x111d76d5:
+ case 0x111d76e7:
+ if (spec->board_config == STAC_92HD83XXX_PWR_REF)
+ break;
spec->num_pwrs = 0;
+ break;
+ }
codec->patch_ops = stac92xx_patch_ops;
@@ -5674,7 +5670,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
}
#endif
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x1d, 0);
if (!err) {
if (spec->board_config < 0) {
printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -5690,6 +5686,22 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
return err;
}
+ /* docking output support */
+ num_dacs = snd_hda_get_connections(codec, 0xF,
+ conn, STAC92HD83_DAC_COUNT + 1) - 1;
+ /* skip non-DAC connections */
+ while (num_dacs >= 0 &&
+ (get_wcaps_type(get_wcaps(codec, conn[num_dacs]))
+ != AC_WID_AUD_OUT))
+ num_dacs--;
+ /* set port E and F to select the last DAC */
+ if (num_dacs >= 0) {
+ snd_hda_codec_write_cache(codec, 0xE, 0,
+ AC_VERB_SET_CONNECT_SEL, num_dacs);
+ snd_hda_codec_write_cache(codec, 0xF, 0,
+ AC_VERB_SET_CONNECT_SEL, num_dacs);
+ }
+
codec->proc_widget_hook = stac92hd_proc_hook;
return 0;
@@ -5995,7 +6007,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->multiout.dac_nids = spec->dac_nids;
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x21, 0);
if (!err) {
if (spec->board_config < 0) {
printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6104,7 +6116,7 @@ static int patch_stac922x(struct hda_codec *codec)
spec->multiout.dac_nids = spec->dac_nids;
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x08, 0x09);
if (!err) {
if (spec->board_config < 0) {
printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6229,7 +6241,7 @@ static int patch_stac927x(struct hda_codec *codec)
spec->aloopback_shift = 0;
spec->eapd_switch = 1;
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
if (!err) {
if (spec->board_config < 0) {
printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6354,7 +6366,7 @@ static int patch_stac9205(struct hda_codec *codec)
break;
}
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x1f, 0x20);
if (!err) {
if (spec->board_config < 0) {
printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -6459,7 +6471,7 @@ static int patch_stac9872(struct hda_codec *codec)
spec->capvols = stac9872_capvols;
spec->capsws = stac9872_capsws;
- err = stac92xx_parse_auto_config(codec);
+ err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
if (err < 0) {
stac92xx_free(codec);
return -EINVAL;
@@ -6559,23 +6571,10 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
{ .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e8, .name = "92HD66B1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e9, .name = "92HD66B2X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ea, .name = "92HD66B3X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76eb, .name = "92HD66C1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ec, .name = "92HD66C2X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ed, .name = "92HD66C3X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ee, .name = "92HD66B1X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ef, .name = "92HD66B2X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f0, .name = "92HD66B3X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f1, .name = "92HD66C1X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f2, .name = "92HD66C2X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f3, .name = "92HD66C3/65", .patch = patch_stac92hd83xxx},
{} /* terminator */
};
diff --git a/trunk/sound/pci/hda/patch_via.c b/trunk/sound/pci/hda/patch_via.c
index 0b020a93a8ed..84d8798bf33a 100644
--- a/trunk/sound/pci/hda/patch_via.c
+++ b/trunk/sound/pci/hda/patch_via.c
@@ -1506,49 +1506,39 @@ static int via_build_pcms(struct hda_codec *codec)
struct via_spec *spec = codec->spec;
struct hda_pcm *info = spec->pcm_rec;
- codec->num_pcms = 0;
+ codec->num_pcms = 1;
codec->pcm_info = info;
- if (spec->multiout.num_dacs || spec->num_adc_nids) {
- snprintf(spec->stream_name_analog,
- sizeof(spec->stream_name_analog),
- "%s Analog", codec->chip_name);
- info->name = spec->stream_name_analog;
+ snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog),
+ "%s Analog", codec->chip_name);
+ info->name = spec->stream_name_analog;
- if (spec->multiout.num_dacs) {
- if (!spec->stream_analog_playback)
- spec->stream_analog_playback =
- &via_pcm_analog_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
- *spec->stream_analog_playback;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
- spec->multiout.dac_nids[0];
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
- spec->multiout.max_channels;
- }
+ if (!spec->stream_analog_playback)
+ spec->stream_analog_playback = &via_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ *spec->stream_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dac_nids[0];
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+ spec->multiout.max_channels;
- if (!spec->stream_analog_capture) {
- if (spec->dyn_adc_switch)
- spec->stream_analog_capture =
- &via_pcm_dyn_adc_analog_capture;
- else
- spec->stream_analog_capture =
- &via_pcm_analog_capture;
- }
- if (spec->num_adc_nids) {
- info->stream[SNDRV_PCM_STREAM_CAPTURE] =
- *spec->stream_analog_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
- spec->adc_nids[0];
- if (!spec->dyn_adc_switch)
- info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
- spec->num_adc_nids;
- }
- codec->num_pcms++;
- info++;
+ if (!spec->stream_analog_capture) {
+ if (spec->dyn_adc_switch)
+ spec->stream_analog_capture =
+ &via_pcm_dyn_adc_analog_capture;
+ else
+ spec->stream_analog_capture = &via_pcm_analog_capture;
}
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ *spec->stream_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+ if (!spec->dyn_adc_switch)
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
+ spec->num_adc_nids;
if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+ codec->num_pcms++;
+ info++;
snprintf(spec->stream_name_digital,
sizeof(spec->stream_name_digital),
"%s Digital", codec->chip_name);
@@ -1572,19 +1562,17 @@ static int via_build_pcms(struct hda_codec *codec)
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
spec->dig_in_nid;
}
- codec->num_pcms++;
- info++;
}
if (spec->hp_dac_nid) {
+ codec->num_pcms++;
+ info++;
snprintf(spec->stream_name_hp, sizeof(spec->stream_name_hp),
"%s HP", codec->chip_name);
info->name = spec->stream_name_hp;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = via_pcm_hp_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
spec->hp_dac_nid;
- codec->num_pcms++;
- info++;
}
return 0;
}
@@ -2096,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
struct via_spec *spec = codec->spec;
struct nid_path *path;
bool check_dac;
- hda_nid_t pin, dac = 0;
+ hda_nid_t pin, dac;
int err;
pin = spec->autocfg.speaker_pins[0];
@@ -3700,8 +3688,13 @@ static const struct hda_verb vt1812_init_verbs[] = {
static void set_widgets_power_state_vt1812(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
+ int imux_is_smixer =
+ snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
unsigned int parm;
unsigned int present;
+ /* MUX10 (1eh) = stereo mixer */
+ imux_is_smixer =
+ snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
/* inputs */
/* PW 5/6/7 (29h/2ah/2bh) */
parm = AC_PWRST_D3;
diff --git a/trunk/sound/pci/ice1712/ice1712.c b/trunk/sound/pci/ice1712/ice1712.c
index 8531b983f3af..0ccc0eb75775 100644
--- a/trunk/sound/pci/ice1712/ice1712.c
+++ b/trunk/sound/pci/ice1712/ice1712.c
@@ -2748,9 +2748,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
if (!c->no_mpu401) {
err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
ICEREG(ice, MPU1_CTRL),
- c->mpu401_1_info_flags |
- MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
- -1, &ice->rmidi[0]);
+ (c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED),
+ ice->irq, 0, &ice->rmidi[0]);
if (err < 0) {
snd_card_free(card);
return err;
@@ -2765,9 +2764,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
/* 2nd port used */
err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712,
ICEREG(ice, MPU2_CTRL),
- c->mpu401_2_info_flags |
- MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
- -1, &ice->rmidi[1]);
+ (c->mpu401_2_info_flags | MPU401_INFO_INTEGRATED),
+ ice->irq, 0, &ice->rmidi[1]);
if (err < 0) {
snd_card_free(card);
diff --git a/trunk/sound/pci/intel8x0.c b/trunk/sound/pci/intel8x0.c
index 45b2055f5a76..6a5b387b97fd 100644
--- a/trunk/sound/pci/intel8x0.c
+++ b/trunk/sound/pci/intel8x0.c
@@ -42,12 +42,6 @@
#include
#include
-#ifdef CONFIG_KVM_GUEST
-#include
-#else
-#define kvm_para_available() (0)
-#endif
-
MODULE_AUTHOR("Jaroslav Kysela ");
MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
MODULE_LICENSE("GPL");
@@ -83,7 +77,6 @@ static int buggy_semaphore;
static int buggy_irq = -1; /* auto-check */
static int xbox;
static int spdif_aclink = -1;
-static int inside_vm = -1;
module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for Intel i8x0 soundcard.");
@@ -101,8 +94,6 @@ module_param(xbox, bool, 0444);
MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection.");
module_param(spdif_aclink, int, 0444);
MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link.");
-module_param(inside_vm, bool, 0444);
-MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization.");
/* just for backward compatibility */
static int enable;
@@ -409,7 +400,6 @@ struct intel8x0 {
unsigned buggy_irq: 1; /* workaround for buggy mobos */
unsigned xbox: 1; /* workaround for Xbox AC'97 detection */
unsigned buggy_semaphore: 1; /* workaround for buggy codec semaphore */
- unsigned inside_vm: 1; /* enable VM optimization */
int spdif_idx; /* SPDIF BAR index; *_SPBAR or -1 if use PCMOUT */
unsigned int sdm_saved; /* SDM reg value */
@@ -1075,11 +1065,8 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs
udelay(10);
continue;
}
- if (civ != igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV))
- continue;
- if (chip->inside_vm)
- break;
- if (ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb))
+ if (civ == igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV) &&
+ ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb))
break;
} while (timeout--);
ptr = ichdev->last_pos;
@@ -2997,10 +2984,6 @@ static int __devinit snd_intel8x0_create(struct snd_card *card,
if (xbox)
chip->xbox = 1;
- chip->inside_vm = inside_vm;
- if (inside_vm)
- printk(KERN_INFO "intel8x0: enable KVM optimization\n");
-
if (pci->vendor == PCI_VENDOR_ID_INTEL &&
pci->device == PCI_DEVICE_ID_INTEL_440MX)
chip->fix_nocache = 1; /* enable workaround */
@@ -3243,14 +3226,6 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci,
buggy_irq = 0;
}
- if (inside_vm < 0) {
- /* detect KVM and Parallels virtual environments */
- inside_vm = kvm_para_available();
-#if defined(__i386__) || defined(__x86_64__)
- inside_vm = inside_vm || boot_cpu_has(X86_FEATURE_HYPERVISOR);
-#endif
- }
-
if ((err = snd_intel8x0_create(card, pci, pci_id->driver_data,
&chip)) < 0) {
snd_card_free(card);
diff --git a/trunk/sound/pci/maestro3.c b/trunk/sound/pci/maestro3.c
index 2fd4bf2d6653..0378126e6272 100644
--- a/trunk/sound/pci/maestro3.c
+++ b/trunk/sound/pci/maestro3.c
@@ -2820,8 +2820,8 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
/* TODO enable MIDI IRQ and I/O */
err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401,
chip->iobase + MPU401_DATA_PORT,
- MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK,
- -1, &chip->rmidi);
+ MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi);
if (err < 0)
printk(KERN_WARNING "maestro3: no MIDI support.\n");
#endif
diff --git a/trunk/sound/pci/oxygen/oxygen_lib.c b/trunk/sound/pci/oxygen/oxygen_lib.c
index 53e5508abcbf..82311fcb86f6 100644
--- a/trunk/sound/pci/oxygen/oxygen_lib.c
+++ b/trunk/sound/pci/oxygen/oxygen_lib.c
@@ -678,15 +678,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
goto err_card;
if (chip->model.device_config & (MIDI_OUTPUT | MIDI_INPUT)) {
- unsigned int info_flags =
- MPU401_INFO_INTEGRATED | MPU401_INFO_IRQ_HOOK;
+ unsigned int info_flags = MPU401_INFO_INTEGRATED;
if (chip->model.device_config & MIDI_OUTPUT)
info_flags |= MPU401_INFO_OUTPUT;
if (chip->model.device_config & MIDI_INPUT)
info_flags |= MPU401_INFO_INPUT;
err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
chip->addr + OXYGEN_MPU401,
- info_flags, -1, &chip->midi);
+ info_flags, 0, 0,
+ &chip->midi);
if (err < 0)
goto err_card;
}
diff --git a/trunk/sound/pci/oxygen/xonar_pcm179x.c b/trunk/sound/pci/oxygen/xonar_pcm179x.c
index 8433aa7c3d75..32d096c98f5b 100644
--- a/trunk/sound/pci/oxygen/xonar_pcm179x.c
+++ b/trunk/sound/pci/oxygen/xonar_pcm179x.c
@@ -1074,7 +1074,6 @@ static const struct oxygen_model model_xonar_st = {
.device_config = PLAYBACK_0_TO_I2S |
PLAYBACK_1_TO_SPDIF |
CAPTURE_0_FROM_I2S_2 |
- CAPTURE_1_FROM_SPDIF |
AC97_FMIC_SWITCH,
.dac_channels_pcm = 2,
.dac_channels_mixer = 2,
diff --git a/trunk/sound/pci/riptide/riptide.c b/trunk/sound/pci/riptide/riptide.c
index 88cc776aa38b..e34ae14908b3 100644
--- a/trunk/sound/pci/riptide/riptide.c
+++ b/trunk/sound/pci/riptide/riptide.c
@@ -2109,7 +2109,7 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
val = mpu_port[dev];
pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, val);
err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE,
- val, MPU401_INFO_IRQ_HOOK, -1,
+ val, 0, chip->irq, 0,
&chip->rmidi);
if (err < 0)
snd_printk(KERN_WARNING
diff --git a/trunk/sound/pci/rme9652/hdsp.c b/trunk/sound/pci/rme9652/hdsp.c
index f74220292254..1c6d1e1c27c1 100644
--- a/trunk/sound/pci/rme9652/hdsp.c
+++ b/trunk/sound/pci/rme9652/hdsp.c
@@ -151,7 +151,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
#define HDSP_PROGRAM 0x020
#define HDSP_CONFIG_MODE_0 0x040
#define HDSP_CONFIG_MODE_1 0x080
-#define HDSP_VERSION_BIT (0x100 | HDSP_S_LOAD)
+#define HDSP_VERSION_BIT 0x100
#define HDSP_BIGENDIAN_MODE 0x200
#define HDSP_RD_MULTIPLE 0x400
#define HDSP_9652_ENABLE_MIXER 0x800
diff --git a/trunk/sound/pci/rme9652/hdspm.c b/trunk/sound/pci/rme9652/hdspm.c
index 15a6c3b9bc9a..6edc67ced905 100644
--- a/trunk/sound/pci/rme9652/hdspm.c
+++ b/trunk/sound/pci/rme9652/hdspm.c
@@ -520,9 +520,16 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_DMA_AREA_BYTES (HDSPM_MAX_CHANNELS * HDSPM_CHANNEL_BUFFER_BYTES)
#define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
+/* revisions >= 230 indicate AES32 card */
+#define HDSPM_MADI_ANCIENT_REV 204
+#define HDSPM_MADI_OLD_REV 207
+#define HDSPM_MADI_REV 210
#define HDSPM_RAYDAT_REV 211
#define HDSPM_AIO_REV 212
#define HDSPM_MADIFACE_REV 213
+#define HDSPM_AES_REV 240
+#define HDSPM_AES32_REV 234
+#define HDSPM_AES32_OLD_REV 233
/* speed factor modes */
#define HDSPM_SPEED_SINGLE 0
@@ -1234,30 +1241,10 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
return rate;
}
-/* return latency in samples per period */
-static int hdspm_get_latency(struct hdspm *hdspm)
-{
- int n;
-
- n = hdspm_decode_latency(hdspm->control_register);
-
- /* Special case for new RME cards with 32 samples period size.
- * The three latency bits in the control register
- * (HDSP_LatencyMask) encode latency values of 64 samples as
- * 0, 128 samples as 1 ... 4096 samples as 6. For old cards, 7
- * denotes 8192 samples, but on new cards like RayDAT or AIO,
- * it corresponds to 32 samples.
- */
- if ((7 == n) && (RayDAT == hdspm->io_type || AIO == hdspm->io_type))
- n = -1;
-
- return 1 << (n + 6);
-}
-
/* Latency function */
static inline void hdspm_compute_period_size(struct hdspm *hdspm)
{
- hdspm->period_bytes = 4 * hdspm_get_latency(hdspm);
+ hdspm->period_bytes = 1 << ((hdspm_decode_latency(hdspm->control_register) + 8));
}
@@ -1316,27 +1303,12 @@ static int hdspm_set_interrupt_interval(struct hdspm *s, unsigned int frames)
spin_lock_irq(&s->lock);
- if (32 == frames) {
- /* Special case for new RME cards like RayDAT/AIO which
- * support period sizes of 32 samples. Since latency is
- * encoded in the three bits of HDSP_LatencyMask, we can only
- * have values from 0 .. 7. While 0 still means 64 samples and
- * 6 represents 4096 samples on all cards, 7 represents 8192
- * on older cards and 32 samples on new cards.
- *
- * In other words, period size in samples is calculated by
- * 2^(n+6) with n ranging from 0 .. 7.
- */
- n = 7;
- } else {
- frames >>= 7;
- n = 0;
- while (frames) {
- n++;
- frames >>= 1;
- }
+ frames >>= 7;
+ n = 0;
+ while (frames) {
+ n++;
+ frames >>= 1;
}
-
s->control_register &= ~HDSPM_LatencyMask;
s->control_register |= hdspm_encode_latency(n);
@@ -1367,10 +1339,6 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
break;
case MADIface:
freq_const = 131072000000000ULL;
- break;
- default:
- snd_BUG();
- return 0;
}
return div_u64(freq_const, period);
@@ -1388,19 +1356,16 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
switch (hdspm->io_type) {
case MADIface:
- n = 131072000000000ULL; /* 125 MHz */
- break;
+ n = 131072000000000ULL; /* 125 MHz */
+ break;
case MADI:
case AES32:
- n = 110069313433624ULL; /* 105 MHz */
- break;
+ n = 110069313433624ULL; /* 105 MHz */
+ break;
case RayDAT:
case AIO:
- n = 104857600000000ULL; /* 100 MHz */
- break;
- default:
- snd_BUG();
- return;
+ n = 104857600000000ULL; /* 100 MHz */
+ break;
}
n = div_u64(n, rate);
@@ -4829,7 +4794,8 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
snd_iprintf(buffer, "--- Settings ---\n");
- x = hdspm_get_latency(hdspm);
+ x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
+ HDSPM_LatencyMask));
snd_iprintf(buffer,
"Size (Latency): %d samples (2 periods of %lu bytes)\n",
@@ -4992,7 +4958,8 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
snd_iprintf(buffer, "--- Settings ---\n");
- x = hdspm_get_latency(hdspm);
+ x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
+ HDSPM_LatencyMask));
snd_iprintf(buffer,
"Size (Latency): %d samples (2 periods of %lu bytes)\n",
@@ -5698,6 +5665,19 @@ static int snd_hdspm_prepare(struct snd_pcm_substream *substream)
return 0;
}
+static unsigned int period_sizes_old[] = {
+ 64, 128, 256, 512, 1024, 2048, 4096
+};
+
+static unsigned int period_sizes_new[] = {
+ 32, 64, 128, 256, 512, 1024, 2048, 4096
+};
+
+/* RayDAT and AIO always have a buffer of 16384 samples per channel */
+static unsigned int raydat_aio_buffer_sizes[] = {
+ 16384
+};
+
static struct snd_pcm_hardware snd_hdspm_playback_subinfo = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
@@ -5716,8 +5696,8 @@ static struct snd_pcm_hardware snd_hdspm_playback_subinfo = {
.channels_max = HDSPM_MAX_CHANNELS,
.buffer_bytes_max =
HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
- .period_bytes_min = (32 * 4),
- .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4),
+ .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS,
.periods_min = 2,
.periods_max = 512,
.fifo_size = 0
@@ -5741,13 +5721,31 @@ static struct snd_pcm_hardware snd_hdspm_capture_subinfo = {
.channels_max = HDSPM_MAX_CHANNELS,
.buffer_bytes_max =
HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
- .period_bytes_min = (32 * 4),
- .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4),
+ .period_bytes_max = (4096 * 4) * HDSPM_MAX_CHANNELS,
.periods_min = 2,
.periods_max = 512,
.fifo_size = 0
};
+static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_old = {
+ .count = ARRAY_SIZE(period_sizes_old),
+ .list = period_sizes_old,
+ .mask = 0
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes_new = {
+ .count = ARRAY_SIZE(period_sizes_new),
+ .list = period_sizes_new,
+ .mask = 0
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_raydat_io_buffer = {
+ .count = ARRAY_SIZE(raydat_aio_buffer_sizes),
+ .list = raydat_aio_buffer_sizes,
+ .mask = 0
+};
+
static int snd_hdspm_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
@@ -5948,29 +5946,26 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream)
spin_unlock_irq(&hdspm->lock);
snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
- snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
switch (hdspm->io_type) {
case AIO:
case RayDAT:
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- 32, 4096);
- /* RayDAT & AIO have a fixed buffer of 16384 samples per channel */
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
- 16384, 16384);
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes_new);
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ &hw_constraints_raydat_io_buffer);
+
break;
default:
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- 64, 8192);
- break;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes_old);
}
if (AES32 == hdspm->io_type) {
- runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&hdspm_hw_constraints_aes32_sample_rates);
} else {
@@ -6023,28 +6018,24 @@ static int snd_hdspm_capture_open(struct snd_pcm_substream *substream)
spin_unlock_irq(&hdspm->lock);
snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
- snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
-
switch (hdspm->io_type) {
case AIO:
case RayDAT:
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- 32, 4096);
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
- 16384, 16384);
- break;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes_new);
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ &hw_constraints_raydat_io_buffer);
+ break;
default:
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- 64, 8192);
- break;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes_old);
}
if (AES32 == hdspm->io_type) {
- runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&hdspm_hw_constraints_aes32_sample_rates);
} else {
@@ -6090,7 +6081,7 @@ static inline int copy_u32_le(void __user *dest, void __iomem *src)
}
static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
- unsigned int cmd, unsigned long arg)
+ unsigned int cmd, unsigned long __user arg)
{
void __user *argp = (void __user *)arg;
struct hdspm *hdspm = hw->private_data;
@@ -6215,13 +6206,11 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
info.line_out = hdspm_line_out(hdspm);
info.passthru = 0;
spin_unlock_irq(&hdspm->lock);
- if (copy_to_user(argp, &info, sizeof(info)))
+ if (copy_to_user((void __user *) arg, &info, sizeof(info)))
return -EFAULT;
break;
case SNDRV_HDSPM_IOCTL_GET_STATUS:
- memset(&status, 0, sizeof(status));
-
status.card_type = hdspm->io_type;
status.autosync_source = hdspm_autosync_ref(hdspm);
@@ -6246,7 +6235,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
status.card_specific.madi.madi_input =
(statusregister & HDSPM_AB_int) ? 1 : 0;
status.card_specific.madi.channel_format =
- (statusregister & HDSPM_RX_64ch) ? 1 : 0;
+ (statusregister & HDSPM_TX_64ch) ? 1 : 0;
/* TODO: Mac driver sets it when f_s>48kHz */
status.card_specific.madi.frame_format = 0;
@@ -6254,15 +6243,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
break;
}
- if (copy_to_user(argp, &status, sizeof(status)))
+ if (copy_to_user((void __user *) arg, &status, sizeof(status)))
return -EFAULT;
break;
case SNDRV_HDSPM_IOCTL_GET_VERSION:
- memset(&hdspm_version, 0, sizeof(hdspm_version));
-
hdspm_version.card_type = hdspm->io_type;
strncpy(hdspm_version.cardname, hdspm->card_name,
sizeof(hdspm_version.cardname));
@@ -6273,13 +6260,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
if (hdspm->tco)
hdspm_version.addons |= HDSPM_ADDON_TCO;
- if (copy_to_user(argp, &hdspm_version,
+ if (copy_to_user((void __user *) arg, &hdspm_version,
sizeof(hdspm_version)))
return -EFAULT;
break;
case SNDRV_HDSPM_IOCTL_GET_MIXER:
- if (copy_from_user(&mixer, argp, sizeof(mixer)))
+ if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer)))
return -EFAULT;
if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer,
sizeof(struct hdspm_mixer)))
@@ -6496,6 +6483,13 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
strcpy(card->driver, "HDSPM");
switch (hdspm->firmware_rev) {
+ case HDSPM_MADI_REV:
+ case HDSPM_MADI_OLD_REV:
+ case HDSPM_MADI_ANCIENT_REV:
+ hdspm->io_type = MADI;
+ hdspm->card_name = "RME MADI";
+ hdspm->midiPorts = 3;
+ break;
case HDSPM_RAYDAT_REV:
hdspm->io_type = RayDAT;
hdspm->card_name = "RME RayDAT";
@@ -6511,25 +6505,17 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
hdspm->card_name = "RME MADIface";
hdspm->midiPorts = 1;
break;
+ case HDSPM_AES_REV:
+ case HDSPM_AES32_REV:
+ case HDSPM_AES32_OLD_REV:
+ hdspm->io_type = AES32;
+ hdspm->card_name = "RME AES32";
+ hdspm->midiPorts = 2;
+ break;
default:
- if ((hdspm->firmware_rev == 0xf0) ||
- ((hdspm->firmware_rev >= 0xe6) &&
- (hdspm->firmware_rev <= 0xea))) {
- hdspm->io_type = AES32;
- hdspm->card_name = "RME AES32";
- hdspm->midiPorts = 2;
- } else if ((hdspm->firmware_rev == 0xd5) ||
- ((hdspm->firmware_rev >= 0xc8) &&
- (hdspm->firmware_rev <= 0xcf))) {
- hdspm->io_type = MADI;
- hdspm->card_name = "RME MADI";
- hdspm->midiPorts = 3;
- } else {
- snd_printk(KERN_ERR
- "HDSPM: unknown firmware revision %x\n",
+ snd_printk(KERN_ERR "HDSPM: unknown firmware revision %x\n",
hdspm->firmware_rev);
- return -ENODEV;
- }
+ return -ENODEV;
}
err = pci_enable_device(pci);
diff --git a/trunk/sound/pci/sis7019.c b/trunk/sound/pci/sis7019.c
index 5ffb20b18786..bcf61524a13f 100644
--- a/trunk/sound/pci/sis7019.c
+++ b/trunk/sound/pci/sis7019.c
@@ -1234,7 +1234,7 @@ static int sis_resume(struct pci_dev *pci)
goto error;
}
- if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
+ if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
KBUILD_MODNAME, sis)) {
printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq);
goto error;
@@ -1340,7 +1340,7 @@ static int __devinit sis_chip_create(struct snd_card *card,
if (rc)
goto error_out_cleanup;
- if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
+ if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
KBUILD_MODNAME, sis)) {
printk(KERN_ERR "unable to allocate irq %d\n", sis->irq);
goto error_out_cleanup;
diff --git a/trunk/sound/pci/sonicvibes.c b/trunk/sound/pci/sonicvibes.c
index c5008166cf1f..2571a67b389a 100644
--- a/trunk/sound/pci/sonicvibes.c
+++ b/trunk/sound/pci/sonicvibes.c
@@ -1493,10 +1493,9 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci,
return err;
}
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES,
- sonic->midi_port,
- MPU401_INFO_INTEGRATED |
- MPU401_INFO_IRQ_HOOK,
- -1, &midi_uart)) < 0) {
+ sonic->midi_port, MPU401_INFO_INTEGRATED,
+ sonic->irq, 0,
+ &midi_uart)) < 0) {
snd_card_free(card);
return err;
}
diff --git a/trunk/sound/pci/trident/trident.c b/trunk/sound/pci/trident/trident.c
index 5e707effdc7c..d8a128f6fc02 100644
--- a/trunk/sound/pci/trident/trident.c
+++ b/trunk/sound/pci/trident/trident.c
@@ -148,9 +148,8 @@ static int __devinit snd_trident_probe(struct pci_dev *pci,
if (trident->device != TRIDENT_DEVICE_ID_SI7018 &&
(err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE,
trident->midi_port,
- MPU401_INFO_INTEGRATED |
- MPU401_INFO_IRQ_HOOK,
- -1, &trident->rmidi)) < 0) {
+ MPU401_INFO_INTEGRATED,
+ trident->irq, 0, &trident->rmidi)) < 0) {
snd_card_free(card);
return err;
}
diff --git a/trunk/sound/pci/via82xx.c b/trunk/sound/pci/via82xx.c
index c3656fffdb50..f03fd620a2a0 100644
--- a/trunk/sound/pci/via82xx.c
+++ b/trunk/sound/pci/via82xx.c
@@ -1175,7 +1175,6 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
struct snd_pcm_runtime *runtime = substream->runtime;
int err;
struct via_rate_lock *ratep;
- bool use_src = false;
runtime->hw = snd_via82xx_hw;
@@ -1197,7 +1196,6 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
SNDRV_PCM_RATE_8000_48000);
runtime->hw.rate_min = 8000;
runtime->hw.rate_max = 48000;
- use_src = true;
} else if (! ratep->rate) {
int idx = viadev->direction ? AC97_RATES_ADC : AC97_RATES_FRONT_DAC;
runtime->hw.rates = chip->ac97->rates[idx];
@@ -1214,12 +1212,6 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
- if (use_src) {
- err = snd_pcm_hw_rule_noresample(runtime, 48000);
- if (err < 0)
- return err;
- }
-
runtime->private_data = viadev;
viadev->substream = substream;
@@ -2076,9 +2068,8 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip)
pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg);
if (chip->mpu_res) {
if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A,
- mpu_port, MPU401_INFO_INTEGRATED |
- MPU401_INFO_IRQ_HOOK, -1,
- &chip->rmidi) < 0) {
+ mpu_port, MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi) < 0) {
printk(KERN_WARNING "unable to initialize MPU-401"
" at 0x%lx, skipping\n", mpu_port);
legacy &= ~VIA_FUNC_ENABLE_MIDI;
diff --git a/trunk/sound/pci/ymfpci/ymfpci.c b/trunk/sound/pci/ymfpci/ymfpci.c
index 3253b04da184..511d57653124 100644
--- a/trunk/sound/pci/ymfpci/ymfpci.c
+++ b/trunk/sound/pci/ymfpci/ymfpci.c
@@ -305,9 +305,8 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
if (chip->mpu_res) {
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI,
mpu_port[dev],
- MPU401_INFO_INTEGRATED |
- MPU401_INFO_IRQ_HOOK,
- -1, &chip->rawmidi)) < 0) {
+ MPU401_INFO_INTEGRATED,
+ pci->irq, 0, &chip->rawmidi)) < 0) {
printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]);
legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */
pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl);
diff --git a/trunk/sound/pci/ymfpci/ymfpci_main.c b/trunk/sound/pci/ymfpci/ymfpci_main.c
index 66ea71b2a70d..f3260e658b8a 100644
--- a/trunk/sound/pci/ymfpci/ymfpci_main.c
+++ b/trunk/sound/pci/ymfpci/ymfpci_main.c
@@ -897,18 +897,6 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream)
struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_ymfpci_pcm *ypcm;
- int err;
-
- runtime->hw = snd_ymfpci_playback;
- /* FIXME? True value is 256/48 = 5.33333 ms */
- err = snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_TIME,
- 5334, UINT_MAX);
- if (err < 0)
- return err;
- err = snd_pcm_hw_rule_noresample(runtime, 48000);
- if (err < 0)
- return err;
ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL);
if (ypcm == NULL)
@@ -916,8 +904,11 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream)
ypcm->chip = chip;
ypcm->type = PLAYBACK_VOICE;
ypcm->substream = substream;
+ runtime->hw = snd_ymfpci_playback;
runtime->private_data = ypcm;
runtime->private_free = snd_ymfpci_pcm_free_substream;
+ /* FIXME? True value is 256/48 = 5.33333 ms */
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX);
return 0;
}
@@ -1022,18 +1013,6 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream,
struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_ymfpci_pcm *ypcm;
- int err;
-
- runtime->hw = snd_ymfpci_capture;
- /* FIXME? True value is 256/48 = 5.33333 ms */
- err = snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_TIME,
- 5334, UINT_MAX);
- if (err < 0)
- return err;
- err = snd_pcm_hw_rule_noresample(runtime, 48000);
- if (err < 0)
- return err;
ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL);
if (ypcm == NULL)
@@ -1043,6 +1022,9 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream,
ypcm->substream = substream;
ypcm->capture_bank_number = capture_bank_number;
chip->capture_substream[capture_bank_number] = substream;
+ runtime->hw = snd_ymfpci_capture;
+ /* FIXME? True value is 256/48 = 5.33333 ms */
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX);
runtime->private_data = ypcm;
runtime->private_free = snd_ymfpci_pcm_free_substream;
snd_ymfpci_hw_start(chip);
@@ -1633,7 +1615,7 @@ YMFPCI_DOUBLE("ADC Playback Volume", 0, YDSXGR_PRIADCOUTVOL),
YMFPCI_DOUBLE("ADC Capture Volume", 0, YDSXGR_PRIADCLOOPVOL),
YMFPCI_DOUBLE("ADC Playback Volume", 1, YDSXGR_SECADCOUTVOL),
YMFPCI_DOUBLE("ADC Capture Volume", 1, YDSXGR_SECADCLOOPVOL),
-YMFPCI_DOUBLE("FM Legacy Playback Volume", 0, YDSXGR_LEGACYOUTVOL),
+YMFPCI_DOUBLE("FM Legacy Volume", 0, YDSXGR_LEGACYOUTVOL),
YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ", PLAYBACK,VOLUME), 0, YDSXGR_ZVOUTVOL),
YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("", CAPTURE,VOLUME), 0, YDSXGR_ZVLOOPVOL),
YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ",PLAYBACK,VOLUME), 1, YDSXGR_SPDIFOUTVOL),
diff --git a/trunk/sound/ppc/keywest.c b/trunk/sound/ppc/keywest.c
index 4080becf4cef..8f064c7ce745 100644
--- a/trunk/sound/ppc/keywest.c
+++ b/trunk/sound/ppc/keywest.c
@@ -82,6 +82,7 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
static int keywest_remove(struct i2c_client *client)
{
+ i2c_set_clientdata(client, NULL);
if (! keywest_ctx)
return 0;
if (client == keywest_ctx->client)
diff --git a/trunk/sound/ppc/snd_ps3.c b/trunk/sound/ppc/snd_ps3.c
index 775bd95d4be6..bc823a547550 100644
--- a/trunk/sound/ppc/snd_ps3.c
+++ b/trunk/sound/ppc/snd_ps3.c
@@ -845,7 +845,7 @@ static int __devinit snd_ps3_allocate_irq(void)
return ret;
}
- ret = request_irq(the_card.irq_no, snd_ps3_interrupt, 0,
+ ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED,
SND_PS3_DRIVER_NAME, &the_card);
if (ret) {
pr_info("%s: request_irq failed (%d)\n", __func__, ret);
diff --git a/trunk/sound/soc/au1x/dma.c b/trunk/sound/soc/au1x/dma.c
index 177f7137a9c8..7aa5b7606777 100644
--- a/trunk/sound/soc/au1x/dma.c
+++ b/trunk/sound/soc/au1x/dma.c
@@ -211,7 +211,7 @@ static int alchemy_pcm_open(struct snd_pcm_substream *substream)
/* DMA setup */
name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx";
ctx->stream[s].dma = request_au1000_dma(dmaids[s], name,
- au1000_dma_interrupt, 0,
+ au1000_dma_interrupt, IRQF_DISABLED,
&ctx->stream[s]);
set_dma_mode(ctx->stream[s].dma,
get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
diff --git a/trunk/sound/soc/codecs/tlv320dac33.c b/trunk/sound/soc/codecs/tlv320dac33.c
index dc8a2b2bdc1c..3f4920d5456d 100644
--- a/trunk/sound/soc/codecs/tlv320dac33.c
+++ b/trunk/sound/soc/codecs/tlv320dac33.c
@@ -1419,7 +1419,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
/* Check if the IRQ number is valid and request it */
if (dac33->irq >= 0) {
ret = request_irq(dac33->irq, dac33_interrupt_handler,
- IRQF_TRIGGER_RISING,
+ IRQF_TRIGGER_RISING | IRQF_DISABLED,
codec->name, codec);
if (ret < 0) {
dev_err(codec->dev, "Could not request IRQ%d (%d)\n",
diff --git a/trunk/sound/soc/codecs/wm8940.c b/trunk/sound/soc/codecs/wm8940.c
index dc5cb3150857..de9ec9b8b7d9 100644
--- a/trunk/sound/soc/codecs/wm8940.c
+++ b/trunk/sound/soc/codecs/wm8940.c
@@ -621,7 +621,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8940_BCLKDIV:
- reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3;
+ reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFE3;
ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2));
break;
case WM8940_MCLKDIV:
diff --git a/trunk/sound/soc/nuc900/nuc900-pcm.c b/trunk/sound/soc/nuc900/nuc900-pcm.c
index ae8d6806966b..4e3626b9d8f9 100644
--- a/trunk/sound/soc/nuc900/nuc900-pcm.c
+++ b/trunk/sound/soc/nuc900/nuc900-pcm.c
@@ -268,7 +268,7 @@ static int nuc900_dma_open(struct snd_pcm_substream *substream)
nuc900_audio = nuc900_ac97_data;
if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt,
- 0, "nuc900-dma", substream))
+ IRQF_DISABLED, "nuc900-dma", substream))
return -EBUSY;
runtime->private_data = nuc900_audio;
diff --git a/trunk/sound/soc/samsung/ac97.c b/trunk/sound/soc/samsung/ac97.c
index b5e922f469d5..65ea53884806 100644
--- a/trunk/sound/soc/samsung/ac97.c
+++ b/trunk/sound/soc/samsung/ac97.c
@@ -444,7 +444,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev)
}
ret = request_irq(irq_res->start, s3c_ac97_irq,
- 0, "AC97", NULL);
+ IRQF_DISABLED, "AC97", NULL);
if (ret < 0) {
dev_err(&pdev->dev, "ac97: interrupt request failed.\n");
goto err4;
diff --git a/trunk/sound/soc/sh/fsi.c b/trunk/sound/soc/sh/fsi.c
index a32fd16ad668..916b9f99b7e7 100644
--- a/trunk/sound/soc/sh/fsi.c
+++ b/trunk/sound/soc/sh/fsi.c
@@ -1285,7 +1285,7 @@ static int fsi_probe(struct platform_device *pdev)
pm_runtime_enable(&pdev->dev);
dev_set_drvdata(&pdev->dev, master);
- ret = request_irq(irq, &fsi_interrupt, 0,
+ ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
id_entry->name, master);
if (ret) {
dev_err(&pdev->dev, "irq request err\n");
diff --git a/trunk/sound/soc/txx9/txx9aclc-ac97.c b/trunk/sound/soc/txx9/txx9aclc-ac97.c
index a4e3f5501847..743d07b82c06 100644
--- a/trunk/sound/soc/txx9/txx9aclc-ac97.c
+++ b/trunk/sound/soc/txx9/txx9aclc-ac97.c
@@ -201,7 +201,7 @@ static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev)
if (!drvdata->base)
return -EBUSY;
err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq,
- 0, dev_name(&pdev->dev), drvdata);
+ IRQF_DISABLED, dev_name(&pdev->dev), drvdata);
if (err < 0)
return err;
diff --git a/trunk/sound/sparc/amd7930.c b/trunk/sound/sparc/amd7930.c
index f036776380b5..ad7d4d7d9237 100644
--- a/trunk/sound/sparc/amd7930.c
+++ b/trunk/sound/sparc/amd7930.c
@@ -962,7 +962,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
amd7930_idle(amd);
if (request_irq(irq, snd_amd7930_interrupt,
- IRQF_SHARED, "amd7930", amd)) {
+ IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) {
snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n",
dev, irq);
snd_amd7930_free(amd);
diff --git a/trunk/sound/usb/6fire/firmware.c b/trunk/sound/usb/6fire/firmware.c
index 07bcfe4d18a7..1e3ae3327dd3 100644
--- a/trunk/sound/usb/6fire/firmware.c
+++ b/trunk/sound/usb/6fire/firmware.c
@@ -16,7 +16,6 @@
#include
#include
-#include
#include "firmware.h"
#include "chip.h"
@@ -60,19 +59,21 @@ struct ihex_record {
unsigned int txt_offset; /* current position in txt_data */
};
-static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc)
+static u8 usb6fire_fw_ihex_nibble(const u8 n)
{
- u8 val = 0;
- int hval;
-
- hval = hex_to_bin(data[0]);
- if (hval >= 0)
- val |= (hval << 4);
-
- hval = hex_to_bin(data[1]);
- if (hval >= 0)
- val |= hval;
+ if (n >= '0' && n <= '9')
+ return n - '0';
+ else if (n >= 'A' && n <= 'F')
+ return n - ('A' - 10);
+ else if (n >= 'a' && n <= 'f')
+ return n - ('a' - 10);
+ return 0;
+}
+static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc)
+{
+ u8 val = (usb6fire_fw_ihex_nibble(data[0]) << 4) |
+ usb6fire_fw_ihex_nibble(data[1]);
*crc += val;
return val;
}
diff --git a/trunk/sound/usb/Kconfig b/trunk/sound/usb/Kconfig
index 3efc21c3d67c..8beb77563da2 100644
--- a/trunk/sound/usb/Kconfig
+++ b/trunk/sound/usb/Kconfig
@@ -67,7 +67,6 @@ config SND_USB_CAIAQ
* Native Instruments Guitar Rig mobile
* Native Instruments Traktor Kontrol X1
* Native Instruments Traktor Kontrol S4
- * Native Instruments Maschine Controller
To compile this driver as a module, choose M here: the module
will be called snd-usb-caiaq.
@@ -86,7 +85,6 @@ config SND_USB_CAIAQ_INPUT
* Native Instruments Kore Controller 2
* Native Instruments Audio Kontrol 1
* Native Instruments Traktor Kontrol S4
- * Native Instruments Maschine Controller
config SND_USB_US122L
tristate "Tascam US-122L USB driver"
diff --git a/trunk/sound/usb/Makefile b/trunk/sound/usb/Makefile
index ac256dc4c6be..cf9ed66445fa 100644
--- a/trunk/sound/usb/Makefile
+++ b/trunk/sound/usb/Makefile
@@ -3,16 +3,16 @@
#
snd-usb-audio-objs := card.o \
- clock.o \
- endpoint.o \
- format.o \
- helper.o \
mixer.o \
mixer_quirks.o \
- pcm.o \
proc.o \
quirks.o \
- stream.o
+ format.o \
+ endpoint.o \
+ urb.o \
+ pcm.o \
+ helper.o \
+ clock.o
snd-usbmidi-lib-objs := midi.o
diff --git a/trunk/sound/usb/caiaq/audio.c b/trunk/sound/usb/caiaq/audio.c
index 2cf87f5afed4..d0d493ca28ae 100644
--- a/trunk/sound/usb/caiaq/audio.c
+++ b/trunk/sound/usb/caiaq/audio.c
@@ -139,12 +139,8 @@ static void stream_stop(struct snd_usb_caiaqdev *dev)
for (i = 0; i < N_URBS; i++) {
usb_kill_urb(dev->data_urbs_in[i]);
-
- if (test_bit(i, &dev->outurb_active_mask))
- usb_kill_urb(dev->data_urbs_out[i]);
+ usb_kill_urb(dev->data_urbs_out[i]);
}
-
- dev->outurb_active_mask = 0;
}
static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
@@ -616,9 +612,8 @@ static void read_completed(struct urb *urb)
{
struct snd_usb_caiaq_cb_info *info = urb->context;
struct snd_usb_caiaqdev *dev;
- struct urb *out = NULL;
- int i, frame, len, send_it = 0, outframe = 0;
- size_t offset = 0;
+ struct urb *out;
+ int frame, len, send_it = 0, outframe = 0;
if (urb->status || !info)
return;
@@ -628,17 +623,7 @@ static void read_completed(struct urb *urb)
if (!dev->streaming)
return;
- /* find an unused output urb that is unused */
- for (i = 0; i < N_URBS; i++)
- if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) {
- out = dev->data_urbs_out[i];
- break;
- }
-
- if (!out) {
- log("Unable to find an output urb to use\n");
- goto requeue;
- }
+ out = dev->data_urbs_out[info->index];
/* read the recently received packet and send back one which has
* the same layout */
@@ -649,8 +634,7 @@ static void read_completed(struct urb *urb)
len = urb->iso_frame_desc[outframe].actual_length;
out->iso_frame_desc[outframe].length = len;
out->iso_frame_desc[outframe].actual_length = 0;
- out->iso_frame_desc[outframe].offset = offset;
- offset += len;
+ out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
if (len > 0) {
spin_lock(&dev->spinlock);
@@ -666,15 +650,11 @@ static void read_completed(struct urb *urb)
}
if (send_it) {
- out->number_of_packets = outframe;
+ out->number_of_packets = FRAMES_PER_URB;
out->transfer_flags = URB_ISO_ASAP;
usb_submit_urb(out, GFP_ATOMIC);
- } else {
- struct snd_usb_caiaq_cb_info *oinfo = out->context;
- clear_bit(oinfo->index, &dev->outurb_active_mask);
}
-requeue:
/* re-submit inbound urb */
for (frame = 0; frame < FRAMES_PER_URB; frame++) {
urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
@@ -696,8 +676,6 @@ static void write_completed(struct urb *urb)
dev->output_running = 1;
wake_up(&dev->prepare_wait_queue);
}
-
- clear_bit(info->index, &dev->outurb_active_mask);
}
static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
@@ -849,9 +827,6 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
if (!dev->data_cb_info)
return -ENOMEM;
- dev->outurb_active_mask = 0;
- BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8));
-
for (i = 0; i < N_URBS; i++) {
dev->data_cb_info[i].dev = dev;
dev->data_cb_info[i].index = i;
diff --git a/trunk/sound/usb/caiaq/device.c b/trunk/sound/usb/caiaq/device.c
index 3eb605bd9503..45bc4a2dc6f0 100644
--- a/trunk/sound/usb/caiaq/device.c
+++ b/trunk/sound/usb/caiaq/device.c
@@ -50,8 +50,7 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, Session I/O},"
"{Native Instruments, GuitarRig mobile}"
"{Native Instruments, Traktor Kontrol X1}"
- "{Native Instruments, Traktor Kontrol S4}"
- "{Native Instruments, Maschine Controller}");
+ "{Native Instruments, Traktor Kontrol S4}");
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
@@ -147,11 +146,6 @@ static struct usb_device_id snd_usb_id_table[] = {
.idVendor = USB_VID_NATIVEINSTRUMENTS,
.idProduct = USB_PID_TRAKTORAUDIO2
},
- {
- .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
- .idVendor = USB_VID_NATIVEINSTRUMENTS,
- .idProduct = USB_PID_MASCHINECONTROLLER
- },
{ /* terminator */ }
};
diff --git a/trunk/sound/usb/caiaq/device.h b/trunk/sound/usb/caiaq/device.h
index 562b0bff9c41..b2b310194ffa 100644
--- a/trunk/sound/usb/caiaq/device.h
+++ b/trunk/sound/usb/caiaq/device.h
@@ -18,7 +18,6 @@
#define USB_PID_TRAKTORKONTROLX1 0x2305
#define USB_PID_TRAKTORKONTROLS4 0xbaff
#define USB_PID_TRAKTORAUDIO2 0x041d
-#define USB_PID_MASCHINECONTROLLER 0x0808
#define EP1_BUFSIZE 64
#define EP4_BUFSIZE 512
@@ -97,7 +96,6 @@ struct snd_usb_caiaqdev {
int input_panic, output_panic, warned;
char *audio_in_buf, *audio_out_buf;
unsigned int samplerates, bpp;
- unsigned long outurb_active_mask;
struct snd_pcm_substream *sub_playback[MAX_STREAMS];
struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/trunk/sound/usb/caiaq/input.c b/trunk/sound/usb/caiaq/input.c
index 26a121b42c3c..4432ef7a70a9 100644
--- a/trunk/sound/usb/caiaq/input.c
+++ b/trunk/sound/usb/caiaq/input.c
@@ -30,7 +30,7 @@ static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A };
static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4,
KEY_5, KEY_6, KEY_7 };
static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4,
- KEY_5, KEY_6, KEY_7, KEY_8, KEY_9 };
+ KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
static unsigned short keycode_kore[] = {
KEY_FN_F1, /* "menu" */
@@ -67,61 +67,6 @@ static unsigned short keycode_kore[] = {
KEY_BRL_DOT5
};
-#define MASCHINE_BUTTONS (42)
-#define MASCHINE_BUTTON(X) ((X) + BTN_MISC)
-#define MASCHINE_PADS (16)
-#define MASCHINE_PAD(X) ((X) + ABS_PRESSURE)
-
-static unsigned short keycode_maschine[] = {
- MASCHINE_BUTTON(40), /* mute */
- MASCHINE_BUTTON(39), /* solo */
- MASCHINE_BUTTON(38), /* select */
- MASCHINE_BUTTON(37), /* duplicate */
- MASCHINE_BUTTON(36), /* navigate */
- MASCHINE_BUTTON(35), /* pad mode */
- MASCHINE_BUTTON(34), /* pattern */
- MASCHINE_BUTTON(33), /* scene */
- KEY_RESERVED, /* spacer */
-
- MASCHINE_BUTTON(30), /* rec */
- MASCHINE_BUTTON(31), /* erase */
- MASCHINE_BUTTON(32), /* shift */
- MASCHINE_BUTTON(28), /* grid */
- MASCHINE_BUTTON(27), /* > */
- MASCHINE_BUTTON(26), /* < */
- MASCHINE_BUTTON(25), /* restart */
-
- MASCHINE_BUTTON(21), /* E */
- MASCHINE_BUTTON(22), /* F */
- MASCHINE_BUTTON(23), /* G */
- MASCHINE_BUTTON(24), /* H */
- MASCHINE_BUTTON(20), /* D */
- MASCHINE_BUTTON(19), /* C */
- MASCHINE_BUTTON(18), /* B */
- MASCHINE_BUTTON(17), /* A */
-
- MASCHINE_BUTTON(0), /* control */
- MASCHINE_BUTTON(2), /* browse */
- MASCHINE_BUTTON(4), /* < */
- MASCHINE_BUTTON(6), /* snap */
- MASCHINE_BUTTON(7), /* autowrite */
- MASCHINE_BUTTON(5), /* > */
- MASCHINE_BUTTON(3), /* sampling */
- MASCHINE_BUTTON(1), /* step */
-
- MASCHINE_BUTTON(15), /* 8 softkeys */
- MASCHINE_BUTTON(14),
- MASCHINE_BUTTON(13),
- MASCHINE_BUTTON(12),
- MASCHINE_BUTTON(11),
- MASCHINE_BUTTON(10),
- MASCHINE_BUTTON(9),
- MASCHINE_BUTTON(8),
-
- MASCHINE_BUTTON(16), /* note repeat */
- MASCHINE_BUTTON(29) /* play */
-};
-
#define KONTROLX1_INPUTS (40)
#define KONTROLS4_BUTTONS (12 * 8)
#define KONTROLS4_AXIS (46)
@@ -273,29 +218,6 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev,
input_report_abs(input_dev, ABS_HAT3Y, i);
input_sync(input_dev);
break;
-
- case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
- /* 4 under the left screen */
- input_report_abs(input_dev, ABS_HAT0X, decode_erp(buf[21], buf[20]));
- input_report_abs(input_dev, ABS_HAT0Y, decode_erp(buf[15], buf[14]));
- input_report_abs(input_dev, ABS_HAT1X, decode_erp(buf[9], buf[8]));
- input_report_abs(input_dev, ABS_HAT1Y, decode_erp(buf[3], buf[2]));
-
- /* 4 under the right screen */
- input_report_abs(input_dev, ABS_HAT2X, decode_erp(buf[19], buf[18]));
- input_report_abs(input_dev, ABS_HAT2Y, decode_erp(buf[13], buf[12]));
- input_report_abs(input_dev, ABS_HAT3X, decode_erp(buf[7], buf[6]));
- input_report_abs(input_dev, ABS_HAT3Y, decode_erp(buf[1], buf[0]));
-
- /* volume */
- input_report_abs(input_dev, ABS_RX, decode_erp(buf[17], buf[16]));
- /* tempo */
- input_report_abs(input_dev, ABS_RY, decode_erp(buf[11], buf[10]));
- /* swing */
- input_report_abs(input_dev, ABS_RZ, decode_erp(buf[5], buf[4]));
-
- input_sync(input_dev);
- break;
}
}
@@ -478,25 +400,6 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev,
input_sync(dev->input_dev);
}
-#define MASCHINE_MSGBLOCK_SIZE 2
-
-static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *dev,
- const unsigned char *buf,
- unsigned int len)
-{
- unsigned int i, pad_id;
- uint16_t pressure;
-
- for (i = 0; i < MASCHINE_PADS; i++) {
- pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]);
- pad_id = pressure >> 12;
-
- input_report_abs(dev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff);
- }
-
- input_sync(dev->input_dev);
-}
-
static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb)
{
struct snd_usb_caiaqdev *dev = urb->context;
@@ -522,13 +425,6 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb)
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4):
snd_usb_caiaq_tks4_dispatch(dev, buf, urb->actual_length);
break;
-
- case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
- if (urb->actual_length < (MASCHINE_PADS * MASCHINE_MSGBLOCK_SIZE))
- goto requeue;
-
- snd_usb_caiaq_maschine_dispatch(dev, buf, urb->actual_length);
- break;
}
requeue:
@@ -548,7 +444,6 @@ static int snd_usb_caiaq_input_open(struct input_dev *idev)
switch (dev->chip.usb_id) {
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4):
- case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0)
return -EIO;
break;
@@ -567,7 +462,6 @@ static void snd_usb_caiaq_input_close(struct input_dev *idev)
switch (dev->chip.usb_id) {
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4):
- case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
usb_kill_urb(dev->ep4_in_urb);
break;
}
@@ -758,50 +652,6 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev)
break;
- case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER):
- input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS);
- input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) |
- BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) |
- BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) |
- BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) |
- BIT_MASK(ABS_RX) | BIT_MASK(ABS_RY) |
- BIT_MASK(ABS_RZ);
-
- BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_maschine));
- memcpy(dev->keycode, keycode_maschine, sizeof(keycode_maschine));
- input->keycodemax = ARRAY_SIZE(keycode_maschine);
-
- for (i = 0; i < MASCHINE_PADS; i++) {
- input->absbit[0] |= MASCHINE_PAD(i);
- input_set_abs_params(input, MASCHINE_PAD(i), 0, 0xfff, 5, 10);
- }
-
- input_set_abs_params(input, ABS_HAT0X, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_HAT0Y, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_HAT1X, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_HAT1Y, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_HAT2X, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_HAT2Y, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_HAT3X, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_HAT3Y, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_RX, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_RY, 0, 999, 0, 10);
- input_set_abs_params(input, ABS_RZ, 0, 999, 0, 10);
-
- dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL);
- if (!dev->ep4_in_urb) {
- ret = -ENOMEM;
- goto exit_free_idev;
- }
-
- usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev,
- usb_rcvbulkpipe(usb_dev, 0x4),
- dev->ep4_in_buf, EP4_BUFSIZE,
- snd_usb_caiaq_ep4_reply_dispatch, dev);
-
- snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5);
- break;
-
default:
/* no input methods supported on this device */
goto exit_free_idev;
@@ -814,17 +664,15 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev)
for (i = 0; i < input->keycodemax; i++)
__set_bit(dev->keycode[i], input->keybit);
- dev->input_dev = input;
-
ret = input_register_device(input);
if (ret < 0)
goto exit_free_idev;
+ dev->input_dev = input;
return 0;
exit_free_idev:
input_free_device(input);
- dev->input_dev = NULL;
return ret;
}
@@ -840,3 +688,4 @@ void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev)
input_unregister_device(dev->input_dev);
dev->input_dev = NULL;
}
+
diff --git a/trunk/sound/usb/card.c b/trunk/sound/usb/card.c
index c1575eafff12..781d9e61adfb 100644
--- a/trunk/sound/usb/card.c
+++ b/trunk/sound/usb/card.c
@@ -65,9 +65,9 @@
#include "helper.h"
#include "debug.h"
#include "pcm.h"
+#include "urb.h"
#include "format.h"
#include "power.h"
-#include "stream.h"
MODULE_AUTHOR("Takashi Iwai ");
MODULE_DESCRIPTION("USB Audio");
@@ -185,7 +185,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int
return -EINVAL;
}
- if (! snd_usb_parse_audio_interface(chip, interface)) {
+ if (! snd_usb_parse_audio_endpoints(chip, interface)) {
usb_set_interface(dev, interface, 0); /* reset the current interface */
usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
return -EINVAL;
@@ -530,11 +530,8 @@ snd_usb_audio_probe(struct usb_device *dev,
return chip;
__error:
- if (chip) {
- if (!chip->num_interfaces)
- snd_card_free(chip->card);
- chip->probing = 0;
- }
+ if (chip && !chip->num_interfaces)
+ snd_card_free(chip->card);
mutex_unlock(®ister_mutex);
__err_val:
return NULL;
diff --git a/trunk/sound/usb/card.h b/trunk/sound/usb/card.h
index a39edcc32a93..ae4251d5abf7 100644
--- a/trunk/sound/usb/card.h
+++ b/trunk/sound/usb/card.h
@@ -94,8 +94,6 @@ struct snd_usb_substream {
spinlock_t lock;
struct snd_urb_ops ops; /* callbacks (must be filled at init) */
- int last_frame_number; /* stored frame number */
- int last_delay; /* stored delay */
};
struct snd_usb_stream {
diff --git a/trunk/sound/usb/clock.c b/trunk/sound/usb/clock.c
index 379baad3d5ad..075195e8661a 100644
--- a/trunk/sound/usb/clock.c
+++ b/trunk/sound/usb/clock.c
@@ -91,7 +91,7 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
UAC2_CX_CLOCK_SELECTOR << 8,
snd_usb_ctrl_intf(chip) | (selector_id << 8),
- &buf, sizeof(buf));
+ &buf, sizeof(buf), 1000);
if (ret < 0)
return ret;
@@ -118,7 +118,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_CLOCK_VALID << 8,
snd_usb_ctrl_intf(chip) | (source_id << 8),
- &data, sizeof(data));
+ &data, sizeof(data), 1000);
if (err < 0) {
snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n",
@@ -222,7 +222,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
- data, sizeof(data))) < 0) {
+ data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
dev->devnum, iface, fmt->altsetting, rate, ep);
return err;
@@ -231,7 +231,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
- data, sizeof(data))) < 0) {
+ data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
dev->devnum, iface, fmt->altsetting, ep);
return 0; /* some devices don't support reading */
@@ -273,7 +273,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
- data, sizeof(data))) < 0) {
+ data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
dev->devnum, iface, fmt->altsetting, rate);
return err;
@@ -283,7 +283,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
- data, sizeof(data))) < 0) {
+ data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
dev->devnum, iface, fmt->altsetting);
return err;
diff --git a/trunk/sound/usb/endpoint.c b/trunk/sound/usb/endpoint.c
index 81c6edecd862..7c0d21ecd821 100644
--- a/trunk/sound/usb/endpoint.c
+++ b/trunk/sound/usb/endpoint.c
@@ -15,951 +15,436 @@
*
*/
-#include
#include
+#include
#include
#include
+#include
#include
#include
#include "usbaudio.h"
-#include "helper.h"
#include "card.h"
+#include "proc.h"
+#include "quirks.h"
#include "endpoint.h"
+#include "urb.h"
#include "pcm.h"
+#include "helper.h"
+#include "format.h"
+#include "clock.h"
/*
- * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
- * this will overflow at approx 524 kHz
- */
-static inline unsigned get_usb_full_speed_rate(unsigned int rate)
-{
- return ((rate << 13) + 62) / 125;
-}
-
-/*
- * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
- * this will overflow at approx 4 MHz
- */
-static inline unsigned get_usb_high_speed_rate(unsigned int rate)
-{
- return ((rate << 10) + 62) / 125;
-}
-
-/*
- * unlink active urbs.
- */
-static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
-{
- struct snd_usb_audio *chip = subs->stream->chip;
- unsigned int i;
- int async;
-
- subs->running = 0;
-
- if (!force && subs->stream->chip->shutdown) /* to be sure... */
- return -EBADFD;
-
- async = !can_sleep && chip->async_unlink;
-
- if (!async && in_interrupt())
- return 0;
-
- for (i = 0; i < subs->nurbs; i++) {
- if (test_bit(i, &subs->active_mask)) {
- if (!test_and_set_bit(i, &subs->unlink_mask)) {
- struct urb *u = subs->dataurb[i].urb;
- if (async)
- usb_unlink_urb(u);
- else
- usb_kill_urb(u);
- }
- }
- }
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- if (test_bit(i+16, &subs->active_mask)) {
- if (!test_and_set_bit(i+16, &subs->unlink_mask)) {
- struct urb *u = subs->syncurb[i].urb;
- if (async)
- usb_unlink_urb(u);
- else
- usb_kill_urb(u);
- }
- }
- }
- }
- return 0;
-}
-
-
-/*
- * release a urb data
+ * free a substream
*/
-static void release_urb_ctx(struct snd_urb_ctx *u)
+static void free_substream(struct snd_usb_substream *subs)
{
- if (u->urb) {
- if (u->buffer_size)
- usb_free_coherent(u->subs->dev, u->buffer_size,
- u->urb->transfer_buffer,
- u->urb->transfer_dma);
- usb_free_urb(u->urb);
- u->urb = NULL;
+ struct list_head *p, *n;
+
+ if (!subs->num_formats)
+ return; /* not initialized */
+ list_for_each_safe(p, n, &subs->fmt_list) {
+ struct audioformat *fp = list_entry(p, struct audioformat, list);
+ kfree(fp->rate_table);
+ kfree(fp);
}
+ kfree(subs->rate_list.list);
}
-/*
- * wait until all urbs are processed.
- */
-static int wait_clear_urbs(struct snd_usb_substream *subs)
-{
- unsigned long end_time = jiffies + msecs_to_jiffies(1000);
- unsigned int i;
- int alive;
-
- do {
- alive = 0;
- for (i = 0; i < subs->nurbs; i++) {
- if (test_bit(i, &subs->active_mask))
- alive++;
- }
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- if (test_bit(i + 16, &subs->active_mask))
- alive++;
- }
- }
- if (! alive)
- break;
- schedule_timeout_uninterruptible(1);
- } while (time_before(jiffies, end_time));
- if (alive)
- snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive);
- return 0;
-}
-
-/*
- * release a substream
- */
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force)
-{
- int i;
-
- /* stop urbs (to be sure) */
- deactivate_urbs(subs, force, 1);
- wait_clear_urbs(subs);
-
- for (i = 0; i < MAX_URBS; i++)
- release_urb_ctx(&subs->dataurb[i]);
- for (i = 0; i < SYNC_URBS; i++)
- release_urb_ctx(&subs->syncurb[i]);
- usb_free_coherent(subs->dev, SYNC_URBS * 4,
- subs->syncbuf, subs->sync_dma);
- subs->syncbuf = NULL;
- subs->nurbs = 0;
-}
/*
- * complete callback from data urb
+ * free a usb stream instance
*/
-static void snd_complete_urb(struct urb *urb)
+static void snd_usb_audio_stream_free(struct snd_usb_stream *stream)
{
- struct snd_urb_ctx *ctx = urb->context;
- struct snd_usb_substream *subs = ctx->subs;
- struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
- int err = 0;
-
- if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) ||
- !subs->running || /* can be stopped during retire callback */
- (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 ||
- (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
- clear_bit(ctx->index, &subs->active_mask);
- if (err < 0) {
- snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err);
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- }
- }
+ free_substream(&stream->substream[0]);
+ free_substream(&stream->substream[1]);
+ list_del(&stream->list);
+ kfree(stream);
}
-
-/*
- * complete callback from sync urb
- */
-static void snd_complete_sync_urb(struct urb *urb)
+static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
{
- struct snd_urb_ctx *ctx = urb->context;
- struct snd_usb_substream *subs = ctx->subs;
- struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
- int err = 0;
-
- if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) ||
- !subs->running || /* can be stopped during retire callback */
- (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 ||
- (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
- clear_bit(ctx->index + 16, &subs->active_mask);
- if (err < 0) {
- snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err);
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- }
+ struct snd_usb_stream *stream = pcm->private_data;
+ if (stream) {
+ stream->pcm = NULL;
+ snd_usb_audio_stream_free(stream);
}
}
/*
- * initialize a substream for plaback/capture
+ * add this endpoint to the chip instance.
+ * if a stream with the same endpoint already exists, append to it.
+ * if not, create a new pcm stream.
*/
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
- unsigned int period_bytes,
- unsigned int rate,
- unsigned int frame_bits)
+int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp)
{
- unsigned int maxsize, i;
- int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
- unsigned int urb_packs, total_packs, packs_per_ms;
- struct snd_usb_audio *chip = subs->stream->chip;
-
- /* calculate the frequency in 16.16 format */
- if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
- subs->freqn = get_usb_full_speed_rate(rate);
- else
- subs->freqn = get_usb_high_speed_rate(rate);
- subs->freqm = subs->freqn;
- subs->freqshift = INT_MIN;
- /* calculate max. frequency */
- if (subs->maxpacksize) {
- /* whatever fits into a max. size packet */
- maxsize = subs->maxpacksize;
- subs->freqmax = (maxsize / (frame_bits >> 3))
- << (16 - subs->datainterval);
- } else {
- /* no max. packet size: just take 25% higher than nominal */
- subs->freqmax = subs->freqn + (subs->freqn >> 2);
- maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3))
- >> (16 - subs->datainterval);
- }
- subs->phase = 0;
-
- if (subs->fill_max)
- subs->curpacksize = subs->maxpacksize;
- else
- subs->curpacksize = maxsize;
+ struct list_head *p;
+ struct snd_usb_stream *as;
+ struct snd_usb_substream *subs;
+ struct snd_pcm *pcm;
+ int err;
- if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL)
- packs_per_ms = 8 >> subs->datainterval;
- else
- packs_per_ms = 1;
-
- if (is_playback) {
- urb_packs = max(chip->nrpacks, 1);
- urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
- } else
- urb_packs = 1;
- urb_packs *= packs_per_ms;
- if (subs->syncpipe)
- urb_packs = min(urb_packs, 1U << subs->syncinterval);
-
- /* decide how many packets to be used */
- if (is_playback) {
- unsigned int minsize, maxpacks;
- /* determine how small a packet can be */
- minsize = (subs->freqn >> (16 - subs->datainterval))
- * (frame_bits >> 3);
- /* with sync from device, assume it can be 12% lower */
- if (subs->syncpipe)
- minsize -= minsize >> 3;
- minsize = max(minsize, 1u);
- total_packs = (period_bytes + minsize - 1) / minsize;
- /* we need at least two URBs for queueing */
- if (total_packs < 2) {
- total_packs = 2;
- } else {
- /* and we don't want too long a queue either */
- maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
- total_packs = min(total_packs, maxpacks);
+ list_for_each(p, &chip->pcm_list) {
+ as = list_entry(p, struct snd_usb_stream, list);
+ if (as->fmt_type != fp->fmt_type)
+ continue;
+ subs = &as->substream[stream];
+ if (!subs->endpoint)
+ continue;
+ if (subs->endpoint == fp->endpoint) {
+ list_add_tail(&fp->list, &subs->fmt_list);
+ subs->num_formats++;
+ subs->formats |= fp->formats;
+ return 0;
}
- } else {
- while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
- urb_packs >>= 1;
- total_packs = MAX_URBS * urb_packs;
}
- subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
- if (subs->nurbs > MAX_URBS) {
- /* too much... */
- subs->nurbs = MAX_URBS;
- total_packs = MAX_URBS * urb_packs;
- } else if (subs->nurbs < 2) {
- /* too little - we need at least two packets
- * to ensure contiguous playback/capture
- */
- subs->nurbs = 2;
- }
-
- /* allocate and initialize data urbs */
- for (i = 0; i < subs->nurbs; i++) {
- struct snd_urb_ctx *u = &subs->dataurb[i];
- u->index = i;
- u->subs = subs;
- u->packets = (i + 1) * total_packs / subs->nurbs
- - i * total_packs / subs->nurbs;
- u->buffer_size = maxsize * u->packets;
- if (subs->fmt_type == UAC_FORMAT_TYPE_II)
- u->packets++; /* for transfer delimiter */
- u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
- if (!u->urb)
- goto out_of_memory;
- u->urb->transfer_buffer =
- usb_alloc_coherent(subs->dev, u->buffer_size,
- GFP_KERNEL, &u->urb->transfer_dma);
- if (!u->urb->transfer_buffer)
- goto out_of_memory;
- u->urb->pipe = subs->datapipe;
- u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
- u->urb->interval = 1 << subs->datainterval;
- u->urb->context = u;
- u->urb->complete = snd_complete_urb;
+ /* look for an empty stream */
+ list_for_each(p, &chip->pcm_list) {
+ as = list_entry(p, struct snd_usb_stream, list);
+ if (as->fmt_type != fp->fmt_type)
+ continue;
+ subs = &as->substream[stream];
+ if (subs->endpoint)
+ continue;
+ err = snd_pcm_new_stream(as->pcm, stream, 1);
+ if (err < 0)
+ return err;
+ snd_usb_init_substream(as, stream, fp);
+ return 0;
}
- if (subs->syncpipe) {
- /* allocate and initialize sync urbs */
- subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4,
- GFP_KERNEL, &subs->sync_dma);
- if (!subs->syncbuf)
- goto out_of_memory;
- for (i = 0; i < SYNC_URBS; i++) {
- struct snd_urb_ctx *u = &subs->syncurb[i];
- u->index = i;
- u->subs = subs;
- u->packets = 1;
- u->urb = usb_alloc_urb(1, GFP_KERNEL);
- if (!u->urb)
- goto out_of_memory;
- u->urb->transfer_buffer = subs->syncbuf + i * 4;
- u->urb->transfer_dma = subs->sync_dma + i * 4;
- u->urb->transfer_buffer_length = 4;
- u->urb->pipe = subs->syncpipe;
- u->urb->transfer_flags = URB_ISO_ASAP |
- URB_NO_TRANSFER_DMA_MAP;
- u->urb->number_of_packets = 1;
- u->urb->interval = 1 << subs->syncinterval;
- u->urb->context = u;
- u->urb->complete = snd_complete_sync_urb;
- }
+ /* create a new pcm */
+ as = kzalloc(sizeof(*as), GFP_KERNEL);
+ if (!as)
+ return -ENOMEM;
+ as->pcm_index = chip->pcm_devs;
+ as->chip = chip;
+ as->fmt_type = fp->fmt_type;
+ err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs,
+ stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0,
+ stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1,
+ &pcm);
+ if (err < 0) {
+ kfree(as);
+ return err;
}
- return 0;
-
-out_of_memory:
- snd_usb_release_substream_urbs(subs, 0);
- return -ENOMEM;
-}
+ as->pcm = pcm;
+ pcm->private_data = as;
+ pcm->private_free = snd_usb_audio_pcm_free;
+ pcm->info_flags = 0;
+ if (chip->pcm_devs > 0)
+ sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs);
+ else
+ strcpy(pcm->name, "USB Audio");
-/*
- * prepare urb for full speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 10.14 frequency is passed through the pipe.
- */
-static int prepare_capture_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned char *cp = urb->transfer_buffer;
- struct snd_urb_ctx *ctx = urb->context;
-
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->iso_frame_desc[0].length = 3;
- urb->iso_frame_desc[0].offset = 0;
- cp[0] = subs->freqn >> 2;
- cp[1] = subs->freqn >> 10;
- cp[2] = subs->freqn >> 18;
- return 0;
-}
+ snd_usb_init_substream(as, stream, fp);
-/*
- * prepare urb for high speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 12.13 frequency is passed as 16.16 through the pipe.
- */
-static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned char *cp = urb->transfer_buffer;
- struct snd_urb_ctx *ctx = urb->context;
-
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->iso_frame_desc[0].length = 4;
- urb->iso_frame_desc[0].offset = 0;
- cp[0] = subs->freqn;
- cp[1] = subs->freqn >> 8;
- cp[2] = subs->freqn >> 16;
- cp[3] = subs->freqn >> 24;
- return 0;
-}
+ list_add(&as->list, &chip->pcm_list);
+ chip->pcm_devs++;
-/*
- * process after capture sync complete
- * - nothing to do
- */
-static int retire_capture_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- return 0;
-}
+ snd_usb_proc_pcm_format_add(as);
-/*
- * prepare urb for capture data pipe
- *
- * fill the offset and length of each descriptor.
- *
- * we use a temporary buffer to write the captured data.
- * since the length of written data is determined by host, we cannot
- * write onto the pcm buffer directly... the data is thus copied
- * later at complete callback to the global buffer.
- */
-static int prepare_capture_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- int i, offs;
- struct snd_urb_ctx *ctx = urb->context;
-
- offs = 0;
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- for (i = 0; i < ctx->packets; i++) {
- urb->iso_frame_desc[i].offset = offs;
- urb->iso_frame_desc[i].length = subs->curpacksize;
- offs += subs->curpacksize;
- }
- urb->transfer_buffer_length = offs;
- urb->number_of_packets = ctx->packets;
return 0;
}
-/*
- * process after capture complete
- *
- * copy the data from each desctiptor to the pcm buffer, and
- * update the current position.
- */
-static int retire_capture_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
+static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
+ struct usb_host_interface *alts,
+ int protocol, int iface_no)
{
- unsigned long flags;
- unsigned char *cp;
- int i;
- unsigned int stride, frames, bytes, oldptr;
- int period_elapsed = 0;
-
- stride = runtime->frame_bits >> 3;
-
- for (i = 0; i < urb->number_of_packets; i++) {
- cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
- if (urb->iso_frame_desc[i].status) {
- snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
- // continue;
- }
- bytes = urb->iso_frame_desc[i].actual_length;
- frames = bytes / stride;
- if (!subs->txfr_quirk)
- bytes = frames * stride;
- if (bytes % (runtime->sample_bits >> 3) != 0) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- int oldbytes = bytes;
-#endif
- bytes = frames * stride;
- snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
- oldbytes, bytes);
- }
- /* update the current pointer */
- spin_lock_irqsave(&subs->lock, flags);
- oldptr = subs->hwptr_done;
- subs->hwptr_done += bytes;
- if (subs->hwptr_done >= runtime->buffer_size * stride)
- subs->hwptr_done -= runtime->buffer_size * stride;
- frames = (bytes + (oldptr % stride)) / stride;
- subs->transfer_done += frames;
- if (subs->transfer_done >= runtime->period_size) {
- subs->transfer_done -= runtime->period_size;
- period_elapsed = 1;
- }
- spin_unlock_irqrestore(&subs->lock, flags);
- /* copy a data chunk */
- if (oldptr + bytes > runtime->buffer_size * stride) {
- unsigned int bytes1 =
- runtime->buffer_size * stride - oldptr;
- memcpy(runtime->dma_area + oldptr, cp, bytes1);
- memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
- } else {
- memcpy(runtime->dma_area + oldptr, cp, bytes);
- }
+ /* parsed with a v1 header here. that's ok as we only look at the
+ * header first which is the same for both versions */
+ struct uac_iso_endpoint_descriptor *csep;
+ struct usb_interface_descriptor *altsd = get_iface_desc(alts);
+ int attributes = 0;
+
+ csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+ /* Creamware Noah has this descriptor after the 2nd endpoint */
+ if (!csep && altsd->bNumEndpoints >= 2)
+ csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+ if (!csep || csep->bLength < 7 ||
+ csep->bDescriptorSubtype != UAC_EP_GENERAL) {
+ snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
+ " class specific endpoint descriptor\n",
+ chip->dev->devnum, iface_no,
+ altsd->bAlternateSetting);
+ return 0;
}
- if (period_elapsed)
- snd_pcm_period_elapsed(subs->pcm_substream);
- return 0;
-}
-/*
- * Process after capture complete when paused. Nothing to do.
- */
-static int retire_paused_capture_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- return 0;
-}
+ if (protocol == UAC_VERSION_1) {
+ attributes = csep->bmAttributes;
+ } else {
+ struct uac2_iso_endpoint_descriptor *csep2 =
+ (struct uac2_iso_endpoint_descriptor *) csep;
+ attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX;
-/*
- * prepare urb for playback sync pipe
- *
- * set up the offset and length to receive the current frequency.
- */
-static int prepare_playback_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- struct snd_urb_ctx *ctx = urb->context;
+ /* emulate the endpoint attributes of a v1 device */
+ if (csep2->bmControls & UAC2_CONTROL_PITCH)
+ attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL;
+ }
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize);
- urb->iso_frame_desc[0].offset = 0;
- return 0;
+ return attributes;
}
-/*
- * process after playback sync complete
- *
- * Full speed devices report feedback values in 10.14 format as samples per
- * frame, high speed devices in 16.16 format as samples per microframe.
- * Because the Audio Class 1 spec was written before USB 2.0, many high speed
- * devices use a wrong interpretation, some others use an entirely different
- * format. Therefore, we cannot predict what format any particular device uses
- * and must detect it automatically.
- */
-static int retire_playback_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
+static struct uac2_input_terminal_descriptor *
+ snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+ int terminal_id)
{
- unsigned int f;
- int shift;
- unsigned long flags;
-
- if (urb->iso_frame_desc[0].status != 0 ||
- urb->iso_frame_desc[0].actual_length < 3)
- return 0;
-
- f = le32_to_cpup(urb->transfer_buffer);
- if (urb->iso_frame_desc[0].actual_length == 3)
- f &= 0x00ffffff;
- else
- f &= 0x0fffffff;
- if (f == 0)
- return 0;
-
- if (unlikely(subs->freqshift == INT_MIN)) {
- /*
- * The first time we see a feedback value, determine its format
- * by shifting it left or right until it matches the nominal
- * frequency value. This assumes that the feedback does not
- * differ from the nominal value more than +50% or -25%.
- */
- shift = 0;
- while (f < subs->freqn - subs->freqn / 4) {
- f <<= 1;
- shift++;
- }
- while (f > subs->freqn + subs->freqn / 2) {
- f >>= 1;
- shift--;
- }
- subs->freqshift = shift;
- }
- else if (subs->freqshift >= 0)
- f <<= subs->freqshift;
- else
- f >>= -subs->freqshift;
+ struct uac2_input_terminal_descriptor *term = NULL;
- if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) {
- /*
- * If the frequency looks valid, set it.
- * This value is referred to in prepare_playback_urb().
- */
- spin_lock_irqsave(&subs->lock, flags);
- subs->freqm = f;
- spin_unlock_irqrestore(&subs->lock, flags);
- } else {
- /*
- * Out of range; maybe the shift value is wrong.
- * Reset it so that we autodetect again the next time.
- */
- subs->freqshift = INT_MIN;
+ while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+ ctrl_iface->extralen,
+ term, UAC_INPUT_TERMINAL))) {
+ if (term->bTerminalID == terminal_id)
+ return term;
}
- return 0;
+ return NULL;
}
-/* determine the number of frames in the next packet */
-static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
+static struct uac2_output_terminal_descriptor *
+ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+ int terminal_id)
{
- if (subs->fill_max)
- return subs->maxframesize;
- else {
- subs->phase = (subs->phase & 0xffff)
- + (subs->freqm << subs->datainterval);
- return min(subs->phase >> 16, subs->maxframesize);
- }
-}
+ struct uac2_output_terminal_descriptor *term = NULL;
-/*
- * Prepare urb for streaming before playback starts or when paused.
- *
- * We don't have any data, so we send silence.
- */
-static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned int i, offs, counts;
- struct snd_urb_ctx *ctx = urb->context;
- int stride = runtime->frame_bits >> 3;
-
- offs = 0;
- urb->dev = ctx->subs->dev;
- for (i = 0; i < ctx->packets; ++i) {
- counts = snd_usb_audio_next_packet_size(subs);
- urb->iso_frame_desc[i].offset = offs * stride;
- urb->iso_frame_desc[i].length = counts * stride;
- offs += counts;
+ while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+ ctrl_iface->extralen,
+ term, UAC_OUTPUT_TERMINAL))) {
+ if (term->bTerminalID == terminal_id)
+ return term;
}
- urb->number_of_packets = ctx->packets;
- urb->transfer_buffer_length = offs * stride;
- memset(urb->transfer_buffer,
- runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
- offs * stride);
- return 0;
-}
-/*
- * prepare urb for playback data pipe
- *
- * Since a URB can handle only a single linear buffer, we must use double
- * buffering when the data to be transferred overflows the buffer boundary.
- * To avoid inconsistencies when updating hwptr_done, we use double buffering
- * for all URBs.
- */
-static int prepare_playback_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- int i, stride;
- unsigned int counts, frames, bytes;
- unsigned long flags;
- int period_elapsed = 0;
- struct snd_urb_ctx *ctx = urb->context;
-
- stride = runtime->frame_bits >> 3;
-
- frames = 0;
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->number_of_packets = 0;
- spin_lock_irqsave(&subs->lock, flags);
- for (i = 0; i < ctx->packets; i++) {
- counts = snd_usb_audio_next_packet_size(subs);
- /* set up descriptor */
- urb->iso_frame_desc[i].offset = frames * stride;
- urb->iso_frame_desc[i].length = counts * stride;
- frames += counts;
- urb->number_of_packets++;
- subs->transfer_done += counts;
- if (subs->transfer_done >= runtime->period_size) {
- subs->transfer_done -= runtime->period_size;
- period_elapsed = 1;
- if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
- if (subs->transfer_done > 0) {
- /* FIXME: fill-max mode is not
- * supported yet */
- frames -= subs->transfer_done;
- counts -= subs->transfer_done;
- urb->iso_frame_desc[i].length =
- counts * stride;
- subs->transfer_done = 0;
- }
- i++;
- if (i < ctx->packets) {
- /* add a transfer delimiter */
- urb->iso_frame_desc[i].offset =
- frames * stride;
- urb->iso_frame_desc[i].length = 0;
- urb->number_of_packets++;
- }
- break;
- }
- }
- if (period_elapsed) /* finish at the period boundary */
- break;
- }
- bytes = frames * stride;
- if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
- /* err, the transferred area goes over buffer boundary. */
- unsigned int bytes1 =
- runtime->buffer_size * stride - subs->hwptr_done;
- memcpy(urb->transfer_buffer,
- runtime->dma_area + subs->hwptr_done, bytes1);
- memcpy(urb->transfer_buffer + bytes1,
- runtime->dma_area, bytes - bytes1);
- } else {
- memcpy(urb->transfer_buffer,
- runtime->dma_area + subs->hwptr_done, bytes);
- }
- subs->hwptr_done += bytes;
- if (subs->hwptr_done >= runtime->buffer_size * stride)
- subs->hwptr_done -= runtime->buffer_size * stride;
-
- /* update delay with exact number of samples queued */
- runtime->delay = subs->last_delay;
- runtime->delay += frames;
- subs->last_delay = runtime->delay;
-
- /* realign last_frame_number */
- subs->last_frame_number = usb_get_current_frame_number(subs->dev);
- subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
-
- spin_unlock_irqrestore(&subs->lock, flags);
- urb->transfer_buffer_length = bytes;
- if (period_elapsed)
- snd_pcm_period_elapsed(subs->pcm_substream);
- return 0;
+ return NULL;
}
-/*
- * process after playback data complete
- * - decrease the delay count again
- */
-static int retire_playback_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
+int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
{
- unsigned long flags;
- int stride = runtime->frame_bits >> 3;
- int processed = urb->transfer_buffer_length / stride;
- int est_delay;
+ struct usb_device *dev;
+ struct usb_interface *iface;
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ int i, altno, err, stream;
+ int format = 0, num_channels = 0;
+ struct audioformat *fp = NULL;
+ int num, protocol, clock = 0;
+ struct uac_format_type_i_continuous_descriptor *fmt;
- spin_lock_irqsave(&subs->lock, flags);
+ dev = chip->dev;
- est_delay = snd_usb_pcm_delay(subs, runtime->rate);
- /* update delay with exact number of samples played */
- if (processed > subs->last_delay)
- subs->last_delay = 0;
- else
- subs->last_delay -= processed;
- runtime->delay = subs->last_delay;
+ /* parse the interface's altsettings */
+ iface = usb_ifnum_to_if(dev, iface_no);
+
+ num = iface->num_altsetting;
/*
- * Report when delay estimate is off by more than 2ms.
- * The error should be lower than 2ms since the estimate relies
- * on two reads of a counter updated every ms.
+ * Dallas DS4201 workaround: It presents 5 altsettings, but the last
+ * one misses syncpipe, and does not produce any sound.
*/
- if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
- snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
- est_delay, subs->last_delay);
-
- spin_unlock_irqrestore(&subs->lock, flags);
- return 0;
-}
+ if (chip->usb_id == USB_ID(0x04fa, 0x4201))
+ num = 4;
+
+ for (i = 0; i < num; i++) {
+ alts = &iface->altsetting[i];
+ altsd = get_iface_desc(alts);
+ protocol = altsd->bInterfaceProtocol;
+ /* skip invalid one */
+ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
+ altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
+ (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING &&
+ altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) ||
+ altsd->bNumEndpoints < 1 ||
+ le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0)
+ continue;
+ /* must be isochronous */
+ if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) !=
+ USB_ENDPOINT_XFER_ISOC)
+ continue;
+ /* check direction */
+ stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ altno = altsd->bAlternateSetting;
+
+ if (snd_usb_apply_interface_quirk(chip, iface_no, altno))
+ continue;
+
+ /* get audio formats */
+ switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n",
+ dev->devnum, iface_no, altno, protocol);
+ protocol = UAC_VERSION_1;
+ /* fall through */
+
+ case UAC_VERSION_1: {
+ struct uac1_as_header_descriptor *as =
+ snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
+
+ if (!as) {
+ snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
-static const char *usb_error_string(int err)
-{
- switch (err) {
- case -ENODEV:
- return "no device";
- case -ENOENT:
- return "endpoint not enabled";
- case -EPIPE:
- return "endpoint stalled";
- case -ENOSPC:
- return "not enough bandwidth";
- case -ESHUTDOWN:
- return "device disabled";
- case -EHOSTUNREACH:
- return "device suspended";
- case -EINVAL:
- case -EAGAIN:
- case -EFBIG:
- case -EMSGSIZE:
- return "internal error";
- default:
- return "unknown error";
- }
-}
+ if (as->bLength < sizeof(*as)) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
-/*
- * set up and start data/sync urbs
- */
-static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime)
-{
- unsigned int i;
- int err;
+ format = le16_to_cpu(as->wFormatTag); /* remember the format value */
+ break;
+ }
- if (subs->stream->chip->shutdown)
- return -EBADFD;
+ case UAC_VERSION_2: {
+ struct uac2_input_terminal_descriptor *input_term;
+ struct uac2_output_terminal_descriptor *output_term;
+ struct uac2_as_header_descriptor *as =
+ snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
- for (i = 0; i < subs->nurbs; i++) {
- if (snd_BUG_ON(!subs->dataurb[i].urb))
- return -EINVAL;
- if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) {
- snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i);
- goto __error;
- }
- }
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- if (snd_BUG_ON(!subs->syncurb[i].urb))
- return -EINVAL;
- if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) {
- snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i);
- goto __error;
+ if (!as) {
+ snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
+ dev->devnum, iface_no, altno);
+ continue;
}
- }
- }
- subs->active_mask = 0;
- subs->unlink_mask = 0;
- subs->running = 1;
- for (i = 0; i < subs->nurbs; i++) {
- err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC);
- if (err < 0) {
- snd_printk(KERN_ERR "cannot submit datapipe "
- "for urb %d, error %d: %s\n",
- i, err, usb_error_string(err));
- goto __error;
- }
- set_bit(i, &subs->active_mask);
- }
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC);
- if (err < 0) {
- snd_printk(KERN_ERR "cannot submit syncpipe "
- "for urb %d, error %d: %s\n",
- i, err, usb_error_string(err));
- goto __error;
+ if (as->bLength < sizeof(*as)) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
}
- set_bit(i + 16, &subs->active_mask);
- }
- }
- return 0;
-
- __error:
- // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
- deactivate_urbs(subs, 0, 0);
- return -EPIPE;
-}
+ num_channels = as->bNrChannels;
+ format = le32_to_cpu(as->bmFormats);
-/*
- */
-static struct snd_urb_ops audio_urb_ops[2] = {
- {
- .prepare = prepare_nodata_playback_urb,
- .retire = retire_playback_urb,
- .prepare_sync = prepare_playback_sync_urb,
- .retire_sync = retire_playback_sync_urb,
- },
- {
- .prepare = prepare_capture_urb,
- .retire = retire_capture_urb,
- .prepare_sync = prepare_capture_sync_urb,
- .retire_sync = retire_capture_sync_urb,
- },
-};
-
-/*
- * initialize the substream instance.
- */
+ /* lookup the terminal associated to this interface
+ * to extract the clock */
+ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+ as->bTerminalLink);
+ if (input_term) {
+ clock = input_term->bCSourceID;
+ break;
+ }
-void snd_usb_init_substream(struct snd_usb_stream *as,
- int stream, struct audioformat *fp)
-{
- struct snd_usb_substream *subs = &as->substream[stream];
-
- INIT_LIST_HEAD(&subs->fmt_list);
- spin_lock_init(&subs->lock);
-
- subs->stream = as;
- subs->direction = stream;
- subs->dev = as->chip->dev;
- subs->txfr_quirk = as->chip->txfr_quirk;
- subs->ops = audio_urb_ops[stream];
- if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH)
- subs->ops.prepare_sync = prepare_capture_sync_urb_hs;
-
- snd_usb_set_pcm_ops(as->pcm, stream);
-
- list_add_tail(&fp->list, &subs->fmt_list);
- subs->formats |= fp->formats;
- subs->endpoint = fp->endpoint;
- subs->num_formats++;
- subs->fmt_type = fp->fmt_type;
-}
+ output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+ as->bTerminalLink);
+ if (output_term) {
+ clock = output_term->bCSourceID;
+ break;
+ }
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_usb_substream *subs = substream->runtime->private_data;
+ snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n",
+ dev->devnum, iface_no, altno, as->bTerminalLink);
+ continue;
+ }
+ }
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- subs->ops.prepare = prepare_playback_urb;
- return 0;
- case SNDRV_PCM_TRIGGER_STOP:
- return deactivate_urbs(subs, 0, 0);
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- subs->ops.prepare = prepare_nodata_playback_urb;
- return 0;
- }
+ /* get format type */
+ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE);
+ if (!fmt) {
+ snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+ if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
+ ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
- return -EINVAL;
-}
+ /*
+ * Blue Microphones workaround: The last altsetting is identical
+ * with the previous one, except for a larger packet size, but
+ * is actually a mislabeled two-channel setting; ignore it.
+ */
+ if (fmt->bNrChannels == 1 &&
+ fmt->bSubframeSize == 2 &&
+ altno == 2 && num == 3 &&
+ fp && fp->altsetting == 1 && fp->channels == 1 &&
+ fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
+ protocol == UAC_VERSION_1 &&
+ le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
+ fp->maxpacksize * 2)
+ continue;
+
+ fp = kzalloc(sizeof(*fp), GFP_KERNEL);
+ if (! fp) {
+ snd_printk(KERN_ERR "cannot malloc\n");
+ return -ENOMEM;
+ }
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_usb_substream *subs = substream->runtime->private_data;
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- subs->ops.retire = retire_capture_urb;
- return start_urbs(subs, substream->runtime);
- case SNDRV_PCM_TRIGGER_STOP:
- return deactivate_urbs(subs, 0, 0);
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- subs->ops.retire = retire_paused_capture_urb;
- return 0;
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- subs->ops.retire = retire_capture_urb;
- return 0;
- }
+ fp->iface = iface_no;
+ fp->altsetting = altno;
+ fp->altset_idx = i;
+ fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+ fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->datainterval = snd_usb_parse_datainterval(chip, alts);
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ /* num_channels is only set for v2 interfaces */
+ fp->channels = num_channels;
+ if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
+ fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
+ * (fp->maxpacksize & 0x7ff);
+ fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
+ fp->clock = clock;
+
+ /* some quirks for attributes here */
+
+ switch (chip->usb_id) {
+ case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */
+ /* Optoplay sets the sample rate attribute although
+ * it seems not supporting it in fact.
+ */
+ fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE;
+ break;
+ case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
+ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+ /* doesn't set the sample rate attribute, but supports it */
+ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE;
+ break;
+ case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */
+ case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */
+ case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
+ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
+ an older model 77d:223) */
+ /*
+ * plantronics headset and Griffin iMic have set adaptive-in
+ * although it's really not...
+ */
+ fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE;
+ else
+ fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
+ break;
+ }
- return -EINVAL;
-}
+ /* ok, let's parse further... */
+ if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
+ kfree(fp->rate_table);
+ kfree(fp);
+ fp = NULL;
+ continue;
+ }
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime)
-{
- /* clear urbs (to be sure) */
- deactivate_urbs(subs, 0, 1);
- wait_clear_urbs(subs);
-
- /* for playback, submit the URBs now; otherwise, the first hwptr_done
- * updates for all URBs would happen at the same time when starting */
- if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) {
- subs->ops.prepare = prepare_nodata_playback_urb;
- return start_urbs(subs, runtime);
+ snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint);
+ err = snd_usb_add_audio_endpoint(chip, stream, fp);
+ if (err < 0) {
+ kfree(fp->rate_table);
+ kfree(fp);
+ return err;
+ }
+ /* try to set the interface... */
+ usb_set_interface(chip->dev, iface_no, altno);
+ snd_usb_init_pitch(chip, iface_no, alts, fp);
+ snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max);
}
-
return 0;
}
diff --git a/trunk/sound/usb/endpoint.h b/trunk/sound/usb/endpoint.h
index 88eb63a636eb..64dd0db023b2 100644
--- a/trunk/sound/usb/endpoint.h
+++ b/trunk/sound/usb/endpoint.h
@@ -1,21 +1,11 @@
#ifndef __USBAUDIO_ENDPOINT_H
#define __USBAUDIO_ENDPOINT_H
-void snd_usb_init_substream(struct snd_usb_stream *as,
- int stream,
- struct audioformat *fp);
+int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip,
+ int iface_no);
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
- unsigned int period_bytes,
- unsigned int rate,
- unsigned int frame_bits);
-
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force);
-
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime);
-
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd);
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd);
+int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip,
+ int stream,
+ struct audioformat *fp);
#endif /* __USBAUDIO_ENDPOINT_H */
diff --git a/trunk/sound/usb/format.c b/trunk/sound/usb/format.c
index 89421d176570..8d042dce0d16 100644
--- a/trunk/sound/usb/format.c
+++ b/trunk/sound/usb/format.c
@@ -286,7 +286,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
- tmp, sizeof(tmp));
+ tmp, sizeof(tmp), 1000);
if (ret < 0) {
snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n",
@@ -307,7 +307,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
- data, data_size);
+ data, data_size, 1000);
if (ret < 0) {
snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n",
diff --git a/trunk/sound/usb/helper.c b/trunk/sound/usb/helper.c
index 9eed8f40b179..f280c1903c25 100644
--- a/trunk/sound/usb/helper.c
+++ b/trunk/sound/usb/helper.c
@@ -81,7 +81,7 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype
*/
int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request,
__u8 requesttype, __u16 value, __u16 index, void *data,
- __u16 size)
+ __u16 size, int timeout)
{
int err;
void *buf = NULL;
@@ -92,7 +92,7 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request,
return -ENOMEM;
}
err = usb_control_msg(dev, pipe, request, requesttype,
- value, index, buf, size, 1000);
+ value, index, buf, size, timeout);
if (size > 0) {
memcpy(data, buf, size);
kfree(buf);
diff --git a/trunk/sound/usb/helper.h b/trunk/sound/usb/helper.h
index 805c300dd004..09bd943c43bf 100644
--- a/trunk/sound/usb/helper.h
+++ b/trunk/sound/usb/helper.h
@@ -8,7 +8,7 @@ void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsub
int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe,
__u8 request, __u8 requesttype, __u16 value, __u16 index,
- void *data, __u16 size);
+ void *data, __u16 size, int timeout);
unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
struct usb_host_interface *alts);
diff --git a/trunk/sound/usb/midi.c b/trunk/sound/usb/midi.c
index e21f026d9577..f9289102886a 100644
--- a/trunk/sound/usb/midi.c
+++ b/trunk/sound/usb/midi.c
@@ -816,22 +816,6 @@ static struct usb_protocol_ops snd_usbmidi_raw_ops = {
.output = snd_usbmidi_raw_output,
};
-/*
- * FTDI protocol: raw MIDI bytes, but input packets have two modem status bytes.
- */
-
-static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint* ep,
- uint8_t* buffer, int buffer_length)
-{
- if (buffer_length > 2)
- snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2);
-}
-
-static struct usb_protocol_ops snd_usbmidi_ftdi_ops = {
- .input = snd_usbmidi_ftdi_input,
- .output = snd_usbmidi_raw_output,
-};
-
static void snd_usbmidi_us122l_input(struct snd_usb_midi_in_endpoint *ep,
uint8_t *buffer, int buffer_length)
{
@@ -2179,17 +2163,6 @@ int snd_usbmidi_create(struct snd_card *card,
/* endpoint 1 is input-only */
endpoints[1].out_cables = 0;
break;
- case QUIRK_MIDI_FTDI:
- umidi->usb_protocol_ops = &snd_usbmidi_ftdi_ops;
-
- /* set baud rate to 31250 (48 MHz / 16 / 96) */
- err = usb_control_msg(umidi->dev, usb_sndctrlpipe(umidi->dev, 0),
- 3, 0x40, 0x60, 0, NULL, 0, 1000);
- if (err < 0)
- break;
-
- err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
- break;
default:
snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
err = -ENXIO;
diff --git a/trunk/sound/usb/mixer.c b/trunk/sound/usb/mixer.c
index 60f65ace7474..c22fa76e363a 100644
--- a/trunk/sound/usb/mixer.c
+++ b/trunk/sound/usb/mixer.c
@@ -152,7 +152,6 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p,
if (p && p->dB) {
cval->dBmin = p->dB->min;
cval->dBmax = p->dB->max;
- cval->initialized = 1;
}
}
@@ -296,7 +295,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v
if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
validx, snd_usb_ctrl_intf(chip) | (cval->id << 8),
- buf, val_len) >= val_len) {
+ buf, val_len, 100) >= val_len) {
*value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len));
snd_usb_autosuspend(cval->mixer->chip);
return 0;
@@ -333,7 +332,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v
ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
validx, snd_usb_ctrl_intf(chip) | (cval->id << 8),
- buf, size);
+ buf, size, 1000);
snd_usb_autosuspend(chip);
if (ret < 0) {
@@ -445,7 +444,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
usb_sndctrlpipe(chip->dev, 0), request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
validx, snd_usb_ctrl_intf(chip) | (cval->id << 8),
- buf, val_len) >= 0) {
+ buf, val_len, 100) >= 0) {
snd_usb_autosuspend(chip);
return 0;
}
@@ -881,17 +880,8 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
} else {
- if (!cval->initialized) {
- get_min_max(cval, 0);
- if (cval->initialized && cval->dBmin >= cval->dBmax) {
- kcontrol->vd[0].access &=
- ~(SNDRV_CTL_ELEM_ACCESS_TLV_READ |
- SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK);
- snd_ctl_notify(cval->mixer->chip->card,
- SNDRV_CTL_EVENT_MASK_INFO,
- &kcontrol->id);
- }
- }
+ if (! cval->initialized)
+ get_min_max(cval, 0);
uinfo->value.integer.min = 0;
uinfo->value.integer.max =
(cval->max - cval->min + cval->res - 1) / cval->res;
@@ -1102,7 +1092,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
" Switch" : " Volume");
if (control == UAC_FU_VOLUME) {
check_mapped_dB(map, cval);
- if (cval->dBmin < cval->dBmax || !cval->initialized) {
+ if (cval->dBmin < cval->dBmax) {
kctl->tlv.c = mixer_vol_tlv;
kctl->vd[0].access |=
SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1201,11 +1191,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
if (state->mixer->protocol == UAC_VERSION_1) {
csize = hdr->bControlSize;
- if (!csize) {
- snd_printdd(KERN_ERR "usbaudio: unit %u: "
- "invalid bControlSize == 0\n", unitid);
- return -EINVAL;
- }
channels = (hdr->bLength - 7) / csize - 1;
bmaControls = hdr->bmaControls;
} else {
@@ -1259,7 +1244,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0);
}
} else { /* UAC_VERSION_2 */
- for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) {
+ for (i = 0; i < 30/2; i++) {
unsigned int ch_bits = 0;
unsigned int ch_read_only = 0;
@@ -1949,13 +1934,15 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
struct mixer_build state;
int err;
const struct usbmix_ctl_map *map;
+ struct usb_host_interface *hostif;
void *p;
+ hostif = mixer->chip->ctrl_intf;
memset(&state, 0, sizeof(state));
state.chip = mixer->chip;
state.mixer = mixer;
- state.buffer = mixer->hostif->extra;
- state.buflen = mixer->hostif->extralen;
+ state.buffer = hostif->extra;
+ state.buflen = hostif->extralen;
/* check the mapping table */
for (map = usbmix_ctl_maps; map->id; map++) {
@@ -1968,8 +1955,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
}
p = NULL;
- while ((p = snd_usb_find_csint_desc(mixer->hostif->extra, mixer->hostif->extralen,
- p, UAC_OUTPUT_TERMINAL)) != NULL) {
+ while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) {
if (mixer->protocol == UAC_VERSION_1) {
struct uac1_output_terminal_descriptor *desc = p;
@@ -2176,15 +2162,17 @@ int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
/* create the handler for the optional status interrupt endpoint */
static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
{
+ struct usb_host_interface *hostif;
struct usb_endpoint_descriptor *ep;
void *transfer_buffer;
int buffer_length;
unsigned int epnum;
+ hostif = mixer->chip->ctrl_intf;
/* we need one interrupt input endpoint */
- if (get_iface_desc(mixer->hostif)->bNumEndpoints < 1)
+ if (get_iface_desc(hostif)->bNumEndpoints < 1)
return 0;
- ep = get_endpoint(mixer->hostif, 0);
+ ep = get_endpoint(hostif, 0);
if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
return 0;
@@ -2214,6 +2202,7 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
};
struct usb_mixer_interface *mixer;
struct snd_info_entry *entry;
+ struct usb_host_interface *host_iface;
int err;
strcpy(chip->card->mixername, "USB Mixer");
@@ -2230,8 +2219,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
return -ENOMEM;
}
- mixer->hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
- switch (get_iface_desc(mixer->hostif)->bInterfaceProtocol) {
+ host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+ switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
case UAC_VERSION_1:
default:
mixer->protocol = UAC_VERSION_1;
diff --git a/trunk/sound/usb/mixer.h b/trunk/sound/usb/mixer.h
index 81b2d8a32fb0..ae1a14dcfe82 100644
--- a/trunk/sound/usb/mixer.h
+++ b/trunk/sound/usb/mixer.h
@@ -3,7 +3,6 @@
struct usb_mixer_interface {
struct snd_usb_audio *chip;
- struct usb_host_interface *hostif;
struct list_head list;
unsigned int ignore_ctl_error;
struct urb *urb;
diff --git a/trunk/sound/usb/mixer_quirks.c b/trunk/sound/usb/mixer_quirks.c
index ab125ee0b0f0..3d0f4873112b 100644
--- a/trunk/sound/usb/mixer_quirks.c
+++ b/trunk/sound/usb/mixer_quirks.c
@@ -190,18 +190,18 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
err = snd_usb_ctl_msg(mixer->chip->dev,
usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
- !value, 0, NULL, 0);
+ !value, 0, NULL, 0, 100);
/* USB X-Fi S51 Pro */
if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df))
err = snd_usb_ctl_msg(mixer->chip->dev,
usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
- !value, 0, NULL, 0);
+ !value, 0, NULL, 0, 100);
else
err = snd_usb_ctl_msg(mixer->chip->dev,
usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
- value, index + 2, NULL, 0);
+ value, index + 2, NULL, 0, 100);
if (err < 0)
return err;
mixer->audigy2nx_leds[index] = value;
@@ -299,7 +299,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry,
usb_rcvctrlpipe(mixer->chip->dev, 0),
UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS |
USB_RECIP_INTERFACE, 0,
- jacks[i].unitid << 8, buf, 3);
+ jacks[i].unitid << 8, buf, 3, 100);
if (err == 3 && (buf[0] == 3 || buf[0] == 6))
snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]);
else
@@ -332,7 +332,7 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol,
err = snd_usb_ctl_msg(mixer->chip->dev,
usb_sndctrlpipe(mixer->chip->dev, 0), 0x08,
USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
- 50, 0, &new_status, 1);
+ 50, 0, &new_status, 1, 100);
if (err < 0)
return err;
mixer->xonar_u1_status = new_status;
diff --git a/trunk/sound/usb/pcm.c b/trunk/sound/usb/pcm.c
index 0220b0f335b9..b8dcbf407bbb 100644
--- a/trunk/sound/usb/pcm.c
+++ b/trunk/sound/usb/pcm.c
@@ -28,36 +28,12 @@
#include "card.h"
#include "quirks.h"
#include "debug.h"
-#include "endpoint.h"
+#include "urb.h"
#include "helper.h"
#include "pcm.h"
#include "clock.h"
#include "power.h"
-/* return the estimated delay based on USB frame counters */
-snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
- unsigned int rate)
-{
- int current_frame_number;
- int frame_diff;
- int est_delay;
-
- current_frame_number = usb_get_current_frame_number(subs->dev);
- /*
- * HCD implementations use different widths, use lower 8 bits.
- * The delay will be managed up to 256ms, which is more than
- * enough
- */
- frame_diff = (current_frame_number - subs->last_frame_number) & 0xff;
-
- /* Approximation based on number of samples per USB frame (ms),
- some truncation for 44.1 but the estimate is good enough */
- est_delay = subs->last_delay - (frame_diff * rate / 1000);
- if (est_delay < 0)
- est_delay = 0;
- return est_delay;
-}
-
/*
* return the current pcm pointer. just based on the hwptr_done value.
*/
@@ -69,8 +45,6 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream
subs = (struct snd_usb_substream *)substream->runtime->private_data;
spin_lock(&subs->lock);
hwptr_done = subs->hwptr_done;
- substream->runtime->delay = snd_usb_pcm_delay(subs,
- substream->runtime->rate);
spin_unlock(&subs->lock);
return hwptr_done / (substream->runtime->frame_bits >> 3);
}
@@ -152,7 +126,7 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep,
- data, sizeof(data))) < 0) {
+ data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n",
dev->devnum, iface, ep);
return err;
@@ -176,7 +150,7 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface,
if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
UAC2_EP_CS_PITCH << 8, 0,
- data, sizeof(data))) < 0) {
+ data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n",
dev->devnum, iface, fmt->altsetting);
return err;
@@ -443,8 +417,6 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
subs->hwptr_done = 0;
subs->transfer_done = 0;
subs->phase = 0;
- subs->last_delay = 0;
- subs->last_frame_number = 0;
runtime->delay = 0;
return snd_usb_substream_prepare(subs, runtime);
diff --git a/trunk/sound/usb/pcm.h b/trunk/sound/usb/pcm.h
index df7a003682ad..ed3e283f618d 100644
--- a/trunk/sound/usb/pcm.h
+++ b/trunk/sound/usb/pcm.h
@@ -1,9 +1,6 @@
#ifndef __USBAUDIO_PCM_H
#define __USBAUDIO_PCM_H
-snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
- unsigned int rate);
-
void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream);
int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
diff --git a/trunk/sound/usb/quirks-table.h b/trunk/sound/usb/quirks-table.h
index b61945f3af9e..dba0b7f11c54 100644
--- a/trunk/sound/usb/quirks-table.h
+++ b/trunk/sound/usb/quirks-table.h
@@ -39,17 +39,6 @@
.idProduct = prod, \
.bInterfaceClass = USB_CLASS_VENDOR_SPEC
-/* FTDI devices */
-{
- USB_DEVICE(0x0403, 0xb8d8),
- .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
- /* .vendor_name = "STARR LABS", */
- /* .product_name = "Starr Labs MIDI USB device", */
- .ifnum = 0,
- .type = QUIRK_MIDI_FTDI
- }
-},
-
/* Creative/Toshiba Multimedia Center SB-0500 */
{
USB_DEVICE(0x041e, 0x3048),
@@ -1688,20 +1677,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
-{
- /* Added support for Roland UM-ONE which differs from UM-1 */
- USB_DEVICE(0x0582, 0x012a),
- .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
- /* .vendor_name = "ROLAND", */
- /* .product_name = "UM-ONE", */
- .ifnum = 0,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const struct snd_usb_midi_endpoint_info) {
- .out_cables = 0x0001,
- .in_cables = 0x0003
- }
- }
-},
{
USB_DEVICE(0x0582, 0x011e),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
@@ -1732,40 +1707,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
-{
- USB_DEVICE(0x0582, 0x0130),
- .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
- /* .vendor_name = "BOSS", */
- /* .product_name = "MICRO BR-80", */
- .ifnum = QUIRK_ANY_INTERFACE,
- .type = QUIRK_COMPOSITE,
- .data = (const struct snd_usb_audio_quirk[]) {
- {
- .ifnum = 0,
- .type = QUIRK_IGNORE_INTERFACE
- },
- {
- .ifnum = 1,
- .type = QUIRK_AUDIO_STANDARD_INTERFACE
- },
- {
- .ifnum = 2,
- .type = QUIRK_AUDIO_STANDARD_INTERFACE
- },
- {
- .ifnum = 3,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const struct snd_usb_midi_endpoint_info) {
- .out_cables = 0x0001,
- .in_cables = 0x0001
- }
- },
- {
- .ifnum = -1
- }
- }
- }
-},
/* Guillemot devices */
{
@@ -2476,12 +2417,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.idProduct = 0x1020,
},
-/* KeithMcMillen Stringport */
-{
- USB_DEVICE(0x1f38, 0x0001),
- .bInterfaceClass = USB_CLASS_AUDIO,
-},
-
/* Miditech devices */
{
USB_DEVICE(0x4752, 0x0011),
diff --git a/trunk/sound/usb/quirks.c b/trunk/sound/usb/quirks.c
index 2e5bc7344026..77762c99afbe 100644
--- a/trunk/sound/usb/quirks.c
+++ b/trunk/sound/usb/quirks.c
@@ -34,7 +34,6 @@
#include "endpoint.h"
#include "pcm.h"
#include "clock.h"
-#include "stream.h"
/*
* handle the quirks for the contained interfaces
@@ -107,7 +106,7 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip,
alts = &iface->altsetting[0];
altsd = get_iface_desc(alts);
- err = snd_usb_parse_audio_interface(chip, altsd->bInterfaceNumber);
+ err = snd_usb_parse_audio_endpoints(chip, altsd->bInterfaceNumber);
if (err < 0) {
snd_printk(KERN_ERR "cannot setup if %d: error %d\n",
altsd->bInterfaceNumber, err);
@@ -148,7 +147,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
stream = (fp->endpoint & USB_DIR_IN)
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
- err = snd_usb_add_audio_stream(chip, stream, fp);
+ err = snd_usb_add_audio_endpoint(chip, stream, fp);
if (err < 0) {
kfree(fp);
kfree(rate_table);
@@ -255,7 +254,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
stream = (fp->endpoint & USB_DIR_IN)
? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
- err = snd_usb_add_audio_stream(chip, stream, fp);
+ err = snd_usb_add_audio_endpoint(chip, stream, fp);
if (err < 0) {
kfree(fp);
return err;
@@ -307,7 +306,6 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
[QUIRK_MIDI_CME] = create_any_midi_quirk,
[QUIRK_MIDI_AKAI] = create_any_midi_quirk,
- [QUIRK_MIDI_FTDI] = create_any_midi_quirk,
[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
[QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
@@ -340,7 +338,7 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac
snd_printdd("sending Extigy boot sequence...\n");
/* Send message to force it to reconnect with full interface. */
err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0),
- 0x10, 0x43, 0x0001, 0x000a, NULL, 0);
+ 0x10, 0x43, 0x0001, 0x000a, NULL, 0, 1000);
if (err < 0) snd_printdd("error sending boot message: %d\n", err);
err = usb_get_descriptor(dev, USB_DT_DEVICE, 0,
&dev->descriptor, sizeof(dev->descriptor));
@@ -361,11 +359,11 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a,
USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER,
- 0, 0, &buf, 1);
+ 0, 0, &buf, 1, 1000);
if (buf == 0) {
snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29,
USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
- 1, 2000, NULL, 0);
+ 1, 2000, NULL, 0, 1000);
return -ENODEV;
}
return 0;
@@ -408,7 +406,7 @@ static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 valu
buf[3] = reg;
return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION,
USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT,
- 0, 0, &buf, 4);
+ 0, 0, &buf, 4, 1000);
}
static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
@@ -428,7 +426,7 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
*/
static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
{
- int err = 0, reg;
+ int err, reg;
int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
for (reg = 0; reg < ARRAY_SIZE(val); reg++) {
diff --git a/trunk/sound/usb/stream.c b/trunk/sound/usb/stream.c
deleted file mode 100644
index 5ff8010b2d6f..000000000000
--- a/trunk/sound/usb/stream.c
+++ /dev/null
@@ -1,452 +0,0 @@
-/*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- */
-
-
-#include
-#include
-#include
-#include
-#include
-
-#include
-#include
-
-#include "usbaudio.h"
-#include "card.h"
-#include "proc.h"
-#include "quirks.h"
-#include "endpoint.h"
-#include "pcm.h"
-#include "helper.h"
-#include "format.h"
-#include "clock.h"
-#include "stream.h"
-
-/*
- * free a substream
- */
-static void free_substream(struct snd_usb_substream *subs)
-{
- struct list_head *p, *n;
-
- if (!subs->num_formats)
- return; /* not initialized */
- list_for_each_safe(p, n, &subs->fmt_list) {
- struct audioformat *fp = list_entry(p, struct audioformat, list);
- kfree(fp->rate_table);
- kfree(fp);
- }
- kfree(subs->rate_list.list);
-}
-
-
-/*
- * free a usb stream instance
- */
-static void snd_usb_audio_stream_free(struct snd_usb_stream *stream)
-{
- free_substream(&stream->substream[0]);
- free_substream(&stream->substream[1]);
- list_del(&stream->list);
- kfree(stream);
-}
-
-static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
-{
- struct snd_usb_stream *stream = pcm->private_data;
- if (stream) {
- stream->pcm = NULL;
- snd_usb_audio_stream_free(stream);
- }
-}
-
-
-/*
- * add this endpoint to the chip instance.
- * if a stream with the same endpoint already exists, append to it.
- * if not, create a new pcm stream.
- */
-int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
- int stream,
- struct audioformat *fp)
-{
- struct list_head *p;
- struct snd_usb_stream *as;
- struct snd_usb_substream *subs;
- struct snd_pcm *pcm;
- int err;
-
- list_for_each(p, &chip->pcm_list) {
- as = list_entry(p, struct snd_usb_stream, list);
- if (as->fmt_type != fp->fmt_type)
- continue;
- subs = &as->substream[stream];
- if (!subs->endpoint)
- continue;
- if (subs->endpoint == fp->endpoint) {
- list_add_tail(&fp->list, &subs->fmt_list);
- subs->num_formats++;
- subs->formats |= fp->formats;
- return 0;
- }
- }
- /* look for an empty stream */
- list_for_each(p, &chip->pcm_list) {
- as = list_entry(p, struct snd_usb_stream, list);
- if (as->fmt_type != fp->fmt_type)
- continue;
- subs = &as->substream[stream];
- if (subs->endpoint)
- continue;
- err = snd_pcm_new_stream(as->pcm, stream, 1);
- if (err < 0)
- return err;
- snd_usb_init_substream(as, stream, fp);
- return 0;
- }
-
- /* create a new pcm */
- as = kzalloc(sizeof(*as), GFP_KERNEL);
- if (!as)
- return -ENOMEM;
- as->pcm_index = chip->pcm_devs;
- as->chip = chip;
- as->fmt_type = fp->fmt_type;
- err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs,
- stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0,
- stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1,
- &pcm);
- if (err < 0) {
- kfree(as);
- return err;
- }
- as->pcm = pcm;
- pcm->private_data = as;
- pcm->private_free = snd_usb_audio_pcm_free;
- pcm->info_flags = 0;
- if (chip->pcm_devs > 0)
- sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs);
- else
- strcpy(pcm->name, "USB Audio");
-
- snd_usb_init_substream(as, stream, fp);
-
- list_add(&as->list, &chip->pcm_list);
- chip->pcm_devs++;
-
- snd_usb_proc_pcm_format_add(as);
-
- return 0;
-}
-
-static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
- struct usb_host_interface *alts,
- int protocol, int iface_no)
-{
- /* parsed with a v1 header here. that's ok as we only look at the
- * header first which is the same for both versions */
- struct uac_iso_endpoint_descriptor *csep;
- struct usb_interface_descriptor *altsd = get_iface_desc(alts);
- int attributes = 0;
-
- csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
-
- /* Creamware Noah has this descriptor after the 2nd endpoint */
- if (!csep && altsd->bNumEndpoints >= 2)
- csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
-
- if (!csep || csep->bLength < 7 ||
- csep->bDescriptorSubtype != UAC_EP_GENERAL) {
- snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
- " class specific endpoint descriptor\n",
- chip->dev->devnum, iface_no,
- altsd->bAlternateSetting);
- return 0;
- }
-
- if (protocol == UAC_VERSION_1) {
- attributes = csep->bmAttributes;
- } else {
- struct uac2_iso_endpoint_descriptor *csep2 =
- (struct uac2_iso_endpoint_descriptor *) csep;
-
- attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX;
-
- /* emulate the endpoint attributes of a v1 device */
- if (csep2->bmControls & UAC2_CONTROL_PITCH)
- attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL;
- }
-
- return attributes;
-}
-
-static struct uac2_input_terminal_descriptor *
- snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
- int terminal_id)
-{
- struct uac2_input_terminal_descriptor *term = NULL;
-
- while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
- ctrl_iface->extralen,
- term, UAC_INPUT_TERMINAL))) {
- if (term->bTerminalID == terminal_id)
- return term;
- }
-
- return NULL;
-}
-
-static struct uac2_output_terminal_descriptor *
- snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
- int terminal_id)
-{
- struct uac2_output_terminal_descriptor *term = NULL;
-
- while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
- ctrl_iface->extralen,
- term, UAC_OUTPUT_TERMINAL))) {
- if (term->bTerminalID == terminal_id)
- return term;
- }
-
- return NULL;
-}
-
-int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
-{
- struct usb_device *dev;
- struct usb_interface *iface;
- struct usb_host_interface *alts;
- struct usb_interface_descriptor *altsd;
- int i, altno, err, stream;
- int format = 0, num_channels = 0;
- struct audioformat *fp = NULL;
- int num, protocol, clock = 0;
- struct uac_format_type_i_continuous_descriptor *fmt;
-
- dev = chip->dev;
-
- /* parse the interface's altsettings */
- iface = usb_ifnum_to_if(dev, iface_no);
-
- num = iface->num_altsetting;
-
- /*
- * Dallas DS4201 workaround: It presents 5 altsettings, but the last
- * one misses syncpipe, and does not produce any sound.
- */
- if (chip->usb_id == USB_ID(0x04fa, 0x4201))
- num = 4;
-
- for (i = 0; i < num; i++) {
- alts = &iface->altsetting[i];
- altsd = get_iface_desc(alts);
- protocol = altsd->bInterfaceProtocol;
- /* skip invalid one */
- if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
- altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
- (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING &&
- altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) ||
- altsd->bNumEndpoints < 1 ||
- le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0)
- continue;
- /* must be isochronous */
- if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) !=
- USB_ENDPOINT_XFER_ISOC)
- continue;
- /* check direction */
- stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
- SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
- altno = altsd->bAlternateSetting;
-
- if (snd_usb_apply_interface_quirk(chip, iface_no, altno))
- continue;
-
- /* get audio formats */
- switch (protocol) {
- default:
- snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n",
- dev->devnum, iface_no, altno, protocol);
- protocol = UAC_VERSION_1;
- /* fall through */
-
- case UAC_VERSION_1: {
- struct uac1_as_header_descriptor *as =
- snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
-
- if (!as) {
- snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
- dev->devnum, iface_no, altno);
- continue;
- }
-
- if (as->bLength < sizeof(*as)) {
- snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
- dev->devnum, iface_no, altno);
- continue;
- }
-
- format = le16_to_cpu(as->wFormatTag); /* remember the format value */
- break;
- }
-
- case UAC_VERSION_2: {
- struct uac2_input_terminal_descriptor *input_term;
- struct uac2_output_terminal_descriptor *output_term;
- struct uac2_as_header_descriptor *as =
- snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
-
- if (!as) {
- snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n",
- dev->devnum, iface_no, altno);
- continue;
- }
-
- if (as->bLength < sizeof(*as)) {
- snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n",
- dev->devnum, iface_no, altno);
- continue;
- }
-
- num_channels = as->bNrChannels;
- format = le32_to_cpu(as->bmFormats);
-
- /* lookup the terminal associated to this interface
- * to extract the clock */
- input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
- as->bTerminalLink);
- if (input_term) {
- clock = input_term->bCSourceID;
- break;
- }
-
- output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
- as->bTerminalLink);
- if (output_term) {
- clock = output_term->bCSourceID;
- break;
- }
-
- snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n",
- dev->devnum, iface_no, altno, as->bTerminalLink);
- continue;
- }
- }
-
- /* get format type */
- fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE);
- if (!fmt) {
- snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n",
- dev->devnum, iface_no, altno);
- continue;
- }
- if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
- ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) {
- snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
- dev->devnum, iface_no, altno);
- continue;
- }
-
- /*
- * Blue Microphones workaround: The last altsetting is identical
- * with the previous one, except for a larger packet size, but
- * is actually a mislabeled two-channel setting; ignore it.
- */
- if (fmt->bNrChannels == 1 &&
- fmt->bSubframeSize == 2 &&
- altno == 2 && num == 3 &&
- fp && fp->altsetting == 1 && fp->channels == 1 &&
- fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
- protocol == UAC_VERSION_1 &&
- le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
- fp->maxpacksize * 2)
- continue;
-
- fp = kzalloc(sizeof(*fp), GFP_KERNEL);
- if (! fp) {
- snd_printk(KERN_ERR "cannot malloc\n");
- return -ENOMEM;
- }
-
- fp->iface = iface_no;
- fp->altsetting = altno;
- fp->altset_idx = i;
- fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
- fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
- fp->datainterval = snd_usb_parse_datainterval(chip, alts);
- fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
- /* num_channels is only set for v2 interfaces */
- fp->channels = num_channels;
- if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
- fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
- * (fp->maxpacksize & 0x7ff);
- fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
- fp->clock = clock;
-
- /* some quirks for attributes here */
-
- switch (chip->usb_id) {
- case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */
- /* Optoplay sets the sample rate attribute although
- * it seems not supporting it in fact.
- */
- fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE;
- break;
- case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
- case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
- /* doesn't set the sample rate attribute, but supports it */
- fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE;
- break;
- case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */
- case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */
- case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
- case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
- an older model 77d:223) */
- /*
- * plantronics headset and Griffin iMic have set adaptive-in
- * although it's really not...
- */
- fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE;
- else
- fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
- break;
- }
-
- /* ok, let's parse further... */
- if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
- kfree(fp->rate_table);
- kfree(fp);
- fp = NULL;
- continue;
- }
-
- snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint);
- err = snd_usb_add_audio_stream(chip, stream, fp);
- if (err < 0) {
- kfree(fp->rate_table);
- kfree(fp);
- return err;
- }
- /* try to set the interface... */
- usb_set_interface(chip->dev, iface_no, altno);
- snd_usb_init_pitch(chip, iface_no, alts, fp);
- snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max);
- }
- return 0;
-}
-
diff --git a/trunk/sound/usb/stream.h b/trunk/sound/usb/stream.h
deleted file mode 100644
index c97f679fc84f..000000000000
--- a/trunk/sound/usb/stream.h
+++ /dev/null
@@ -1,12 +0,0 @@
-#ifndef __USBAUDIO_STREAM_H
-#define __USBAUDIO_STREAM_H
-
-int snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
- int iface_no);
-
-int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
- int stream,
- struct audioformat *fp);
-
-#endif /* __USBAUDIO_STREAM_H */
-
diff --git a/trunk/sound/usb/urb.c b/trunk/sound/usb/urb.c
new file mode 100644
index 000000000000..e184349aee83
--- /dev/null
+++ b/trunk/sound/usb/urb.c
@@ -0,0 +1,941 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include
+#include
+#include
+#include
+
+#include
+#include
+
+#include "usbaudio.h"
+#include "helper.h"
+#include "card.h"
+#include "urb.h"
+#include "pcm.h"
+
+/*
+ * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
+ * this will overflow at approx 524 kHz
+ */
+static inline unsigned get_usb_full_speed_rate(unsigned int rate)
+{
+ return ((rate << 13) + 62) / 125;
+}
+
+/*
+ * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
+ * this will overflow at approx 4 MHz
+ */
+static inline unsigned get_usb_high_speed_rate(unsigned int rate)
+{
+ return ((rate << 10) + 62) / 125;
+}
+
+/*
+ * unlink active urbs.
+ */
+static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
+{
+ struct snd_usb_audio *chip = subs->stream->chip;
+ unsigned int i;
+ int async;
+
+ subs->running = 0;
+
+ if (!force && subs->stream->chip->shutdown) /* to be sure... */
+ return -EBADFD;
+
+ async = !can_sleep && chip->async_unlink;
+
+ if (!async && in_interrupt())
+ return 0;
+
+ for (i = 0; i < subs->nurbs; i++) {
+ if (test_bit(i, &subs->active_mask)) {
+ if (!test_and_set_bit(i, &subs->unlink_mask)) {
+ struct urb *u = subs->dataurb[i].urb;
+ if (async)
+ usb_unlink_urb(u);
+ else
+ usb_kill_urb(u);
+ }
+ }
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ if (test_bit(i+16, &subs->active_mask)) {
+ if (!test_and_set_bit(i+16, &subs->unlink_mask)) {
+ struct urb *u = subs->syncurb[i].urb;
+ if (async)
+ usb_unlink_urb(u);
+ else
+ usb_kill_urb(u);
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+
+/*
+ * release a urb data
+ */
+static void release_urb_ctx(struct snd_urb_ctx *u)
+{
+ if (u->urb) {
+ if (u->buffer_size)
+ usb_free_coherent(u->subs->dev, u->buffer_size,
+ u->urb->transfer_buffer,
+ u->urb->transfer_dma);
+ usb_free_urb(u->urb);
+ u->urb = NULL;
+ }
+}
+
+/*
+ * wait until all urbs are processed.
+ */
+static int wait_clear_urbs(struct snd_usb_substream *subs)
+{
+ unsigned long end_time = jiffies + msecs_to_jiffies(1000);
+ unsigned int i;
+ int alive;
+
+ do {
+ alive = 0;
+ for (i = 0; i < subs->nurbs; i++) {
+ if (test_bit(i, &subs->active_mask))
+ alive++;
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ if (test_bit(i + 16, &subs->active_mask))
+ alive++;
+ }
+ }
+ if (! alive)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+ if (alive)
+ snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive);
+ return 0;
+}
+
+/*
+ * release a substream
+ */
+void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force)
+{
+ int i;
+
+ /* stop urbs (to be sure) */
+ deactivate_urbs(subs, force, 1);
+ wait_clear_urbs(subs);
+
+ for (i = 0; i < MAX_URBS; i++)
+ release_urb_ctx(&subs->dataurb[i]);
+ for (i = 0; i < SYNC_URBS; i++)
+ release_urb_ctx(&subs->syncurb[i]);
+ usb_free_coherent(subs->dev, SYNC_URBS * 4,
+ subs->syncbuf, subs->sync_dma);
+ subs->syncbuf = NULL;
+ subs->nurbs = 0;
+}
+
+/*
+ * complete callback from data urb
+ */
+static void snd_complete_urb(struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+ struct snd_usb_substream *subs = ctx->subs;
+ struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
+ int err = 0;
+
+ if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) ||
+ !subs->running || /* can be stopped during retire callback */
+ (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 ||
+ (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+ clear_bit(ctx->index, &subs->active_mask);
+ if (err < 0) {
+ snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err);
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ }
+ }
+}
+
+
+/*
+ * complete callback from sync urb
+ */
+static void snd_complete_sync_urb(struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+ struct snd_usb_substream *subs = ctx->subs;
+ struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
+ int err = 0;
+
+ if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) ||
+ !subs->running || /* can be stopped during retire callback */
+ (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 ||
+ (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+ clear_bit(ctx->index + 16, &subs->active_mask);
+ if (err < 0) {
+ snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err);
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ }
+ }
+}
+
+
+/*
+ * initialize a substream for plaback/capture
+ */
+int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
+ unsigned int period_bytes,
+ unsigned int rate,
+ unsigned int frame_bits)
+{
+ unsigned int maxsize, i;
+ int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int urb_packs, total_packs, packs_per_ms;
+ struct snd_usb_audio *chip = subs->stream->chip;
+
+ /* calculate the frequency in 16.16 format */
+ if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
+ subs->freqn = get_usb_full_speed_rate(rate);
+ else
+ subs->freqn = get_usb_high_speed_rate(rate);
+ subs->freqm = subs->freqn;
+ subs->freqshift = INT_MIN;
+ /* calculate max. frequency */
+ if (subs->maxpacksize) {
+ /* whatever fits into a max. size packet */
+ maxsize = subs->maxpacksize;
+ subs->freqmax = (maxsize / (frame_bits >> 3))
+ << (16 - subs->datainterval);
+ } else {
+ /* no max. packet size: just take 25% higher than nominal */
+ subs->freqmax = subs->freqn + (subs->freqn >> 2);
+ maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3))
+ >> (16 - subs->datainterval);
+ }
+ subs->phase = 0;
+
+ if (subs->fill_max)
+ subs->curpacksize = subs->maxpacksize;
+ else
+ subs->curpacksize = maxsize;
+
+ if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL)
+ packs_per_ms = 8 >> subs->datainterval;
+ else
+ packs_per_ms = 1;
+
+ if (is_playback) {
+ urb_packs = max(chip->nrpacks, 1);
+ urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
+ } else
+ urb_packs = 1;
+ urb_packs *= packs_per_ms;
+ if (subs->syncpipe)
+ urb_packs = min(urb_packs, 1U << subs->syncinterval);
+
+ /* decide how many packets to be used */
+ if (is_playback) {
+ unsigned int minsize, maxpacks;
+ /* determine how small a packet can be */
+ minsize = (subs->freqn >> (16 - subs->datainterval))
+ * (frame_bits >> 3);
+ /* with sync from device, assume it can be 12% lower */
+ if (subs->syncpipe)
+ minsize -= minsize >> 3;
+ minsize = max(minsize, 1u);
+ total_packs = (period_bytes + minsize - 1) / minsize;
+ /* we need at least two URBs for queueing */
+ if (total_packs < 2) {
+ total_packs = 2;
+ } else {
+ /* and we don't want too long a queue either */
+ maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
+ total_packs = min(total_packs, maxpacks);
+ }
+ } else {
+ while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+ urb_packs >>= 1;
+ total_packs = MAX_URBS * urb_packs;
+ }
+ subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
+ if (subs->nurbs > MAX_URBS) {
+ /* too much... */
+ subs->nurbs = MAX_URBS;
+ total_packs = MAX_URBS * urb_packs;
+ } else if (subs->nurbs < 2) {
+ /* too little - we need at least two packets
+ * to ensure contiguous playback/capture
+ */
+ subs->nurbs = 2;
+ }
+
+ /* allocate and initialize data urbs */
+ for (i = 0; i < subs->nurbs; i++) {
+ struct snd_urb_ctx *u = &subs->dataurb[i];
+ u->index = i;
+ u->subs = subs;
+ u->packets = (i + 1) * total_packs / subs->nurbs
+ - i * total_packs / subs->nurbs;
+ u->buffer_size = maxsize * u->packets;
+ if (subs->fmt_type == UAC_FORMAT_TYPE_II)
+ u->packets++; /* for transfer delimiter */
+ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
+ if (!u->urb)
+ goto out_of_memory;
+ u->urb->transfer_buffer =
+ usb_alloc_coherent(subs->dev, u->buffer_size,
+ GFP_KERNEL, &u->urb->transfer_dma);
+ if (!u->urb->transfer_buffer)
+ goto out_of_memory;
+ u->urb->pipe = subs->datapipe;
+ u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
+ u->urb->interval = 1 << subs->datainterval;
+ u->urb->context = u;
+ u->urb->complete = snd_complete_urb;
+ }
+
+ if (subs->syncpipe) {
+ /* allocate and initialize sync urbs */
+ subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4,
+ GFP_KERNEL, &subs->sync_dma);
+ if (!subs->syncbuf)
+ goto out_of_memory;
+ for (i = 0; i < SYNC_URBS; i++) {
+ struct snd_urb_ctx *u = &subs->syncurb[i];
+ u->index = i;
+ u->subs = subs;
+ u->packets = 1;
+ u->urb = usb_alloc_urb(1, GFP_KERNEL);
+ if (!u->urb)
+ goto out_of_memory;
+ u->urb->transfer_buffer = subs->syncbuf + i * 4;
+ u->urb->transfer_dma = subs->sync_dma + i * 4;
+ u->urb->transfer_buffer_length = 4;
+ u->urb->pipe = subs->syncpipe;
+ u->urb->transfer_flags = URB_ISO_ASAP |
+ URB_NO_TRANSFER_DMA_MAP;
+ u->urb->number_of_packets = 1;
+ u->urb->interval = 1 << subs->syncinterval;
+ u->urb->context = u;
+ u->urb->complete = snd_complete_sync_urb;
+ }
+ }
+ return 0;
+
+out_of_memory:
+ snd_usb_release_substream_urbs(subs, 0);
+ return -ENOMEM;
+}
+
+/*
+ * prepare urb for full speed capture sync pipe
+ *
+ * fill the length and offset of each urb descriptor.
+ * the fixed 10.14 frequency is passed through the pipe.
+ */
+static int prepare_capture_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned char *cp = urb->transfer_buffer;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->iso_frame_desc[0].length = 3;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = subs->freqn >> 2;
+ cp[1] = subs->freqn >> 10;
+ cp[2] = subs->freqn >> 18;
+ return 0;
+}
+
+/*
+ * prepare urb for high speed capture sync pipe
+ *
+ * fill the length and offset of each urb descriptor.
+ * the fixed 12.13 frequency is passed as 16.16 through the pipe.
+ */
+static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned char *cp = urb->transfer_buffer;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->iso_frame_desc[0].length = 4;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = subs->freqn;
+ cp[1] = subs->freqn >> 8;
+ cp[2] = subs->freqn >> 16;
+ cp[3] = subs->freqn >> 24;
+ return 0;
+}
+
+/*
+ * process after capture sync complete
+ * - nothing to do
+ */
+static int retire_capture_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ return 0;
+}
+
+/*
+ * prepare urb for capture data pipe
+ *
+ * fill the offset and length of each descriptor.
+ *
+ * we use a temporary buffer to write the captured data.
+ * since the length of written data is determined by host, we cannot
+ * write onto the pcm buffer directly... the data is thus copied
+ * later at complete callback to the global buffer.
+ */
+static int prepare_capture_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ int i, offs;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ offs = 0;
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ for (i = 0; i < ctx->packets; i++) {
+ urb->iso_frame_desc[i].offset = offs;
+ urb->iso_frame_desc[i].length = subs->curpacksize;
+ offs += subs->curpacksize;
+ }
+ urb->transfer_buffer_length = offs;
+ urb->number_of_packets = ctx->packets;
+ return 0;
+}
+
+/*
+ * process after capture complete
+ *
+ * copy the data from each desctiptor to the pcm buffer, and
+ * update the current position.
+ */
+static int retire_capture_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned long flags;
+ unsigned char *cp;
+ int i;
+ unsigned int stride, frames, bytes, oldptr;
+ int period_elapsed = 0;
+
+ stride = runtime->frame_bits >> 3;
+
+ for (i = 0; i < urb->number_of_packets; i++) {
+ cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
+ if (urb->iso_frame_desc[i].status) {
+ snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
+ // continue;
+ }
+ bytes = urb->iso_frame_desc[i].actual_length;
+ frames = bytes / stride;
+ if (!subs->txfr_quirk)
+ bytes = frames * stride;
+ if (bytes % (runtime->sample_bits >> 3) != 0) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ int oldbytes = bytes;
+#endif
+ bytes = frames * stride;
+ snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
+ oldbytes, bytes);
+ }
+ /* update the current pointer */
+ spin_lock_irqsave(&subs->lock, flags);
+ oldptr = subs->hwptr_done;
+ subs->hwptr_done += bytes;
+ if (subs->hwptr_done >= runtime->buffer_size * stride)
+ subs->hwptr_done -= runtime->buffer_size * stride;
+ frames = (bytes + (oldptr % stride)) / stride;
+ subs->transfer_done += frames;
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ period_elapsed = 1;
+ }
+ spin_unlock_irqrestore(&subs->lock, flags);
+ /* copy a data chunk */
+ if (oldptr + bytes > runtime->buffer_size * stride) {
+ unsigned int bytes1 =
+ runtime->buffer_size * stride - oldptr;
+ memcpy(runtime->dma_area + oldptr, cp, bytes1);
+ memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
+ } else {
+ memcpy(runtime->dma_area + oldptr, cp, bytes);
+ }
+ }
+ if (period_elapsed)
+ snd_pcm_period_elapsed(subs->pcm_substream);
+ return 0;
+}
+
+/*
+ * Process after capture complete when paused. Nothing to do.
+ */
+static int retire_paused_capture_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ return 0;
+}
+
+
+/*
+ * prepare urb for playback sync pipe
+ *
+ * set up the offset and length to receive the current frequency.
+ */
+static int prepare_playback_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize);
+ urb->iso_frame_desc[0].offset = 0;
+ return 0;
+}
+
+/*
+ * process after playback sync complete
+ *
+ * Full speed devices report feedback values in 10.14 format as samples per
+ * frame, high speed devices in 16.16 format as samples per microframe.
+ * Because the Audio Class 1 spec was written before USB 2.0, many high speed
+ * devices use a wrong interpretation, some others use an entirely different
+ * format. Therefore, we cannot predict what format any particular device uses
+ * and must detect it automatically.
+ */
+static int retire_playback_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned int f;
+ int shift;
+ unsigned long flags;
+
+ if (urb->iso_frame_desc[0].status != 0 ||
+ urb->iso_frame_desc[0].actual_length < 3)
+ return 0;
+
+ f = le32_to_cpup(urb->transfer_buffer);
+ if (urb->iso_frame_desc[0].actual_length == 3)
+ f &= 0x00ffffff;
+ else
+ f &= 0x0fffffff;
+ if (f == 0)
+ return 0;
+
+ if (unlikely(subs->freqshift == INT_MIN)) {
+ /*
+ * The first time we see a feedback value, determine its format
+ * by shifting it left or right until it matches the nominal
+ * frequency value. This assumes that the feedback does not
+ * differ from the nominal value more than +50% or -25%.
+ */
+ shift = 0;
+ while (f < subs->freqn - subs->freqn / 4) {
+ f <<= 1;
+ shift++;
+ }
+ while (f > subs->freqn + subs->freqn / 2) {
+ f >>= 1;
+ shift--;
+ }
+ subs->freqshift = shift;
+ }
+ else if (subs->freqshift >= 0)
+ f <<= subs->freqshift;
+ else
+ f >>= -subs->freqshift;
+
+ if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) {
+ /*
+ * If the frequency looks valid, set it.
+ * This value is referred to in prepare_playback_urb().
+ */
+ spin_lock_irqsave(&subs->lock, flags);
+ subs->freqm = f;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ } else {
+ /*
+ * Out of range; maybe the shift value is wrong.
+ * Reset it so that we autodetect again the next time.
+ */
+ subs->freqshift = INT_MIN;
+ }
+
+ return 0;
+}
+
+/* determine the number of frames in the next packet */
+static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
+{
+ if (subs->fill_max)
+ return subs->maxframesize;
+ else {
+ subs->phase = (subs->phase & 0xffff)
+ + (subs->freqm << subs->datainterval);
+ return min(subs->phase >> 16, subs->maxframesize);
+ }
+}
+
+/*
+ * Prepare urb for streaming before playback starts or when paused.
+ *
+ * We don't have any data, so we send silence.
+ */
+static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned int i, offs, counts;
+ struct snd_urb_ctx *ctx = urb->context;
+ int stride = runtime->frame_bits >> 3;
+
+ offs = 0;
+ urb->dev = ctx->subs->dev;
+ for (i = 0; i < ctx->packets; ++i) {
+ counts = snd_usb_audio_next_packet_size(subs);
+ urb->iso_frame_desc[i].offset = offs * stride;
+ urb->iso_frame_desc[i].length = counts * stride;
+ offs += counts;
+ }
+ urb->number_of_packets = ctx->packets;
+ urb->transfer_buffer_length = offs * stride;
+ memset(urb->transfer_buffer,
+ runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
+ offs * stride);
+ return 0;
+}
+
+/*
+ * prepare urb for playback data pipe
+ *
+ * Since a URB can handle only a single linear buffer, we must use double
+ * buffering when the data to be transferred overflows the buffer boundary.
+ * To avoid inconsistencies when updating hwptr_done, we use double buffering
+ * for all URBs.
+ */
+static int prepare_playback_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ int i, stride;
+ unsigned int counts, frames, bytes;
+ unsigned long flags;
+ int period_elapsed = 0;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ stride = runtime->frame_bits >> 3;
+
+ frames = 0;
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->number_of_packets = 0;
+ spin_lock_irqsave(&subs->lock, flags);
+ for (i = 0; i < ctx->packets; i++) {
+ counts = snd_usb_audio_next_packet_size(subs);
+ /* set up descriptor */
+ urb->iso_frame_desc[i].offset = frames * stride;
+ urb->iso_frame_desc[i].length = counts * stride;
+ frames += counts;
+ urb->number_of_packets++;
+ subs->transfer_done += counts;
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ period_elapsed = 1;
+ if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
+ if (subs->transfer_done > 0) {
+ /* FIXME: fill-max mode is not
+ * supported yet */
+ frames -= subs->transfer_done;
+ counts -= subs->transfer_done;
+ urb->iso_frame_desc[i].length =
+ counts * stride;
+ subs->transfer_done = 0;
+ }
+ i++;
+ if (i < ctx->packets) {
+ /* add a transfer delimiter */
+ urb->iso_frame_desc[i].offset =
+ frames * stride;
+ urb->iso_frame_desc[i].length = 0;
+ urb->number_of_packets++;
+ }
+ break;
+ }
+ }
+ if (period_elapsed) /* finish at the period boundary */
+ break;
+ }
+ bytes = frames * stride;
+ if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
+ /* err, the transferred area goes over buffer boundary. */
+ unsigned int bytes1 =
+ runtime->buffer_size * stride - subs->hwptr_done;
+ memcpy(urb->transfer_buffer,
+ runtime->dma_area + subs->hwptr_done, bytes1);
+ memcpy(urb->transfer_buffer + bytes1,
+ runtime->dma_area, bytes - bytes1);
+ } else {
+ memcpy(urb->transfer_buffer,
+ runtime->dma_area + subs->hwptr_done, bytes);
+ }
+ subs->hwptr_done += bytes;
+ if (subs->hwptr_done >= runtime->buffer_size * stride)
+ subs->hwptr_done -= runtime->buffer_size * stride;
+ runtime->delay += frames;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ urb->transfer_buffer_length = bytes;
+ if (period_elapsed)
+ snd_pcm_period_elapsed(subs->pcm_substream);
+ return 0;
+}
+
+/*
+ * process after playback data complete
+ * - decrease the delay count again
+ */
+static int retire_playback_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned long flags;
+ int stride = runtime->frame_bits >> 3;
+ int processed = urb->transfer_buffer_length / stride;
+
+ spin_lock_irqsave(&subs->lock, flags);
+ if (processed > runtime->delay)
+ runtime->delay = 0;
+ else
+ runtime->delay -= processed;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ return 0;
+}
+
+static const char *usb_error_string(int err)
+{
+ switch (err) {
+ case -ENODEV:
+ return "no device";
+ case -ENOENT:
+ return "endpoint not enabled";
+ case -EPIPE:
+ return "endpoint stalled";
+ case -ENOSPC:
+ return "not enough bandwidth";
+ case -ESHUTDOWN:
+ return "device disabled";
+ case -EHOSTUNREACH:
+ return "device suspended";
+ case -EINVAL:
+ case -EAGAIN:
+ case -EFBIG:
+ case -EMSGSIZE:
+ return "internal error";
+ default:
+ return "unknown error";
+ }
+}
+
+/*
+ * set up and start data/sync urbs
+ */
+static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime)
+{
+ unsigned int i;
+ int err;
+
+ if (subs->stream->chip->shutdown)
+ return -EBADFD;
+
+ for (i = 0; i < subs->nurbs; i++) {
+ if (snd_BUG_ON(!subs->dataurb[i].urb))
+ return -EINVAL;
+ if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) {
+ snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i);
+ goto __error;
+ }
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ if (snd_BUG_ON(!subs->syncurb[i].urb))
+ return -EINVAL;
+ if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) {
+ snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i);
+ goto __error;
+ }
+ }
+ }
+
+ subs->active_mask = 0;
+ subs->unlink_mask = 0;
+ subs->running = 1;
+ for (i = 0; i < subs->nurbs; i++) {
+ err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot submit datapipe "
+ "for urb %d, error %d: %s\n",
+ i, err, usb_error_string(err));
+ goto __error;
+ }
+ set_bit(i, &subs->active_mask);
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot submit syncpipe "
+ "for urb %d, error %d: %s\n",
+ i, err, usb_error_string(err));
+ goto __error;
+ }
+ set_bit(i + 16, &subs->active_mask);
+ }
+ }
+ return 0;
+
+ __error:
+ // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
+ deactivate_urbs(subs, 0, 0);
+ return -EPIPE;
+}
+
+
+/*
+ */
+static struct snd_urb_ops audio_urb_ops[2] = {
+ {
+ .prepare = prepare_nodata_playback_urb,
+ .retire = retire_playback_urb,
+ .prepare_sync = prepare_playback_sync_urb,
+ .retire_sync = retire_playback_sync_urb,
+ },
+ {
+ .prepare = prepare_capture_urb,
+ .retire = retire_capture_urb,
+ .prepare_sync = prepare_capture_sync_urb,
+ .retire_sync = retire_capture_sync_urb,
+ },
+};
+
+/*
+ * initialize the substream instance.
+ */
+
+void snd_usb_init_substream(struct snd_usb_stream *as,
+ int stream, struct audioformat *fp)
+{
+ struct snd_usb_substream *subs = &as->substream[stream];
+
+ INIT_LIST_HEAD(&subs->fmt_list);
+ spin_lock_init(&subs->lock);
+
+ subs->stream = as;
+ subs->direction = stream;
+ subs->dev = as->chip->dev;
+ subs->txfr_quirk = as->chip->txfr_quirk;
+ subs->ops = audio_urb_ops[stream];
+ if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH)
+ subs->ops.prepare_sync = prepare_capture_sync_urb_hs;
+
+ snd_usb_set_pcm_ops(as->pcm, stream);
+
+ list_add_tail(&fp->list, &subs->fmt_list);
+ subs->formats |= fp->formats;
+ subs->endpoint = fp->endpoint;
+ subs->num_formats++;
+ subs->fmt_type = fp->fmt_type;
+}
+
+int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ subs->ops.prepare = prepare_playback_urb;
+ return 0;
+ case SNDRV_PCM_TRIGGER_STOP:
+ return deactivate_urbs(subs, 0, 0);
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ subs->ops.prepare = prepare_nodata_playback_urb;
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
+int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ subs->ops.retire = retire_capture_urb;
+ return start_urbs(subs, substream->runtime);
+ case SNDRV_PCM_TRIGGER_STOP:
+ return deactivate_urbs(subs, 0, 0);
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ subs->ops.retire = retire_paused_capture_urb;
+ return 0;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ subs->ops.retire = retire_capture_urb;
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
+int snd_usb_substream_prepare(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime)
+{
+ /* clear urbs (to be sure) */
+ deactivate_urbs(subs, 0, 1);
+ wait_clear_urbs(subs);
+
+ /* for playback, submit the URBs now; otherwise, the first hwptr_done
+ * updates for all URBs would happen at the same time when starting */
+ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+ subs->ops.prepare = prepare_nodata_playback_urb;
+ return start_urbs(subs, runtime);
+ }
+
+ return 0;
+}
+
diff --git a/trunk/sound/usb/urb.h b/trunk/sound/usb/urb.h
new file mode 100644
index 000000000000..888da38079cf
--- /dev/null
+++ b/trunk/sound/usb/urb.h
@@ -0,0 +1,21 @@
+#ifndef __USBAUDIO_URB_H
+#define __USBAUDIO_URB_H
+
+void snd_usb_init_substream(struct snd_usb_stream *as,
+ int stream,
+ struct audioformat *fp);
+
+int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
+ unsigned int period_bytes,
+ unsigned int rate,
+ unsigned int frame_bits);
+
+void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force);
+
+int snd_usb_substream_prepare(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime);
+
+int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd);
+int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd);
+
+#endif /* __USBAUDIO_URB_H */
diff --git a/trunk/sound/usb/usbaudio.h b/trunk/sound/usb/usbaudio.h
index 3e2b03577936..1e79986b5777 100644
--- a/trunk/sound/usb/usbaudio.h
+++ b/trunk/sound/usb/usbaudio.h
@@ -80,7 +80,6 @@ enum quirk_type {
QUIRK_MIDI_CME,
QUIRK_MIDI_AKAI,
QUIRK_MIDI_US122L,
- QUIRK_MIDI_FTDI,
QUIRK_AUDIO_STANDARD_INTERFACE,
QUIRK_AUDIO_FIXED_ENDPOINT,
QUIRK_AUDIO_EDIROL_UAXX,